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It is safest ++ to attach them to the start of each source file to most effectively ++ convey the exclusion of warranty; and each file should have at least ++ the "copyright" line and a pointer to where the full notice is found. ++ ++ ++ Copyright (C) 19yy ++ ++ This program is free software; you can redistribute it and/or modify ++ it under the terms of the GNU General Public License as published by ++ the Free Software Foundation; either version 2 of the License, or ++ (at your option) any later version. ++ ++ This program is distributed in the hope that it will be useful, ++ but WITHOUT ANY WARRANTY; without even the implied warranty of ++ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the ++ GNU General Public License for more details. ++ ++ You should have received a copy of the GNU General Public License ++ along with this program; if not, write to the Free Software ++ Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA ++ ++ ++ Also add information on how to contact you by electronic and paper mail. ++ ++ If the program is interactive, make it output a short notice like this ++ when it starts in an interactive mode: ++ ++ Gnomovision version 69, Copyright (C) 19yy name of author ++ Gnomovision comes with ABSOLUTELY NO WARRANTY; for details type `show w'. ++ This is free software, and you are welcome to redistribute it ++ under certain conditions; type `show c' for details. ++ ++ The hypothetical commands `show w' and `show c' should show the appropriate ++ parts of the General Public License. Of course, the commands you use may ++ be called something other than `show w' and `show c'; they could even be ++ mouse-clicks or menu items--whatever suits your program. ++ ++ You should also get your employer (if you work as a programmer) or your ++ school, if any, to sign a "copyright disclaimer" for the program, if ++ necessary. Here is a sample; alter the names: ++ ++ Yoyodyne, Inc., hereby disclaims all copyright interest in the program ++ `Gnomovision' (which makes passes at compilers) written by James Hacker. ++ ++ , 1 April 1989 ++ Ty Coon, President of Vice ++ ++ This General Public License does not permit incorporating your program into ++ proprietary programs. If your program is a subroutine library, you may ++ consider it more useful to permit linking proprietary applications with the ++ library. If this is what you want to do, use the GNU Library General ++ Public License instead of this License. +diff -r -c -N encoder/Dll/BladeMP3EncDLL.c lame3.13/Dll/BladeMP3EncDLL.c +*** encoder/Dll/BladeMP3EncDLL.c Wed Dec 31 17:00:00 1969 +--- lame3.13/Dll/BladeMP3EncDLL.c Sat Jun 19 00:17:40 1999 +*************** +*** 0 **** +--- 1,491 ---- ++ /* ++ * Blade DLL Interface for LAME. ++ * ++ * Copyright (c) 1999 A.L. Faber ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License as published by ++ * the Free Software Foundation; either version 2, or (at your option) ++ * any later version. ++ * ++ * This program is distributed in the hope that it will be useful, ++ * but WITHOUT ANY WARRANTY; without even the implied warranty of ++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the ++ * GNU General Public License for more details. ++ * ++ * You should have received a copy of the GNU General Public License ++ * along with this program; see the file COPYING. If not, write to ++ * the Free Software Foundation, 675 Mass Ave, Cambridge, MA 02139, USA. ++ */ ++ ++ #include "BladeMP3EncDLL.h" ++ #include ++ #include ++ #include ++ #include ++ #include ++ #include "common.h" ++ #include "version.h" ++ #include "VbrTag.h" ++ ++ ++ const int MAJORVERSION=0; ++ const int MINORVERSION=21; ++ ++ static short int InputBuffer[2][1152]; ++ ++ int nBladeBufferSize; ++ ++ extern encodeframe(); ++ extern void InitFormatBitStream(void); ++ extern void InitReservoir(void); ++ extern int sfb21; ++ extern int fast_mode; ++ extern int gpsycho; ++ extern int force_ms; ++ extern int VBR_q; ++ extern int g_bWriteVbrTag; ++ ++ // Local variables ++ static int nPsychoModel=2; ++ static BOOL bFirstFrame=TRUE; ++ static DWORD dwSampleBufferSize=0; ++ ++ ++ #ifdef _DEBUG ++ void dump_config( frame_params *fr_ps, int *psy, char *inPath, char *outPath); ++ #endif ++ ++ // Taken from main.c ++ static char original_file_name[MAX_NAME_SIZE]; ++ static char encoded_file_name[MAX_NAME_SIZE]; ++ static int stereo, error_protection; ++ static layer info; ++ ++ ++ // Variables that we "borrow" from Musicin.c ++ extern frame_params fr_ps; ++ extern unsigned long num_samples; ++ extern int gtkflag; ++ extern Bit_stream_struc bs; ++ extern int firstcall; ++ extern III_side_info_t l3_side; ++ ++ ++ extern void parse_args(int argc,char** argv,frame_params* fr_ps,int* psy, ++ unsigned long* num_samples,char* inPath,char* outPath); ++ ++ extern void III_FlushBitstream(); ++ ++ static void InitParams() ++ { ++ // clear buffers ++ memset(InputBuffer, 0,sizeof(InputBuffer)); ++ ++ memset(&bs, 0, sizeof(Bit_stream_struc)); ++ memset(&fr_ps, 0, sizeof(frame_params)); ++ memset(&l3_side,0x00,sizeof(III_side_info_t)); ++ ++ fr_ps.header = &info; ++ fr_ps.tab_num = -1; /* no table loaded */ ++ fr_ps.alloc = NULL; ++ info.version = MPEG_AUDIO_ID; /* =1 Default: MPEG-1 */ ++ ++ bFirstFrame=TRUE; ++ firstcall = 1; ++ ++ InitFormatBitStream(); ++ InitReservoir(); ++ ++ } ++ ++ #define NORMAL_QUALITY 0 ++ #define LOW_QUALITY 1 ++ #define HIGH_QUALITY 2 ++ ++ #define GPL_PSYCHOMODEL 0 ++ #define ISO_PSYCHOMODEL 1 ++ ++ __declspec(dllexport) BE_ERR beInitStream(PBE_CONFIG pbeConfig, PDWORD dwSamples, PDWORD dwBufferSize, PHBE_STREAM phbeStream) ++ { ++ #define MAX_ARGV 20 ++ ++ char strTmp[255]; ++ int nDllArgC=0; ++ char DllArgV[20][80]; ++ char* argv[MAX_ARGV]; ++ int i; ++ int nCRC=pbeConfig->format.mp3.bCRC; ++ int nVBR; ++ int nQuality; ++ ++ // Get VBR setting from fourth nibble ++ nVBR=(nCRC>>12)&0x0F; ++ // Get Quality from third nibble ++ nQuality=(nCRC>>8)&0x0F; ++ // Get model from lower nibble (to be compatible with standard definition) ++ nCRC=(nCRC&0x01); ++ ++ for (i=0;iformat.mp3.byMode&0x0F)== BE_MP3_MODE_MONO) ++ { ++ // Mono channel, thus MPEGFRAMSIZE samples needed ++ *dwSamples=1152; ++ } ++ else ++ { ++ // For Stereo.Joint stereo and dual channel, need 2*MPEGFRAMESIZE samples ++ *dwSamples=1152*2; ++ } ++ ++ // Set the input sample buffer size, so we know what we can expect ++ dwSampleBufferSize=*dwSamples; ++ ++ // --------------- Set arguments for ParseArg function ------------------------- ++ ++ // Set zero argument, the filename ++ strcpy(DllArgV[nDllArgC++],"LameDLLEncoder"); ++ ++ switch (pbeConfig->format.mp3.byMode) ++ { ++ case BE_MP3_MODE_STEREO: ++ strcpy(DllArgV[nDllArgC++],"-ms"); ++ break; ++ case BE_MP3_MODE_JSTEREO: ++ strcpy(DllArgV[nDllArgC++],"-mj"); ++ break; ++ case BE_MP3_MODE_MONO: ++ strcpy(DllArgV[nDllArgC++],"-mm"); ++ break; ++ case BE_MP3_MODE_DUALCHANNEL: ++ strcpy(DllArgV[nDllArgC++],"-mf"); ++ break; ++ default: ++ return BE_ERR_INVALID_FORMAT_PARAMETERS; ++ } ++ ++ switch (nQuality) ++ { ++ case NORMAL_QUALITY: // Nothing special ++ break; ++ case LOW_QUALITY: // -f flag ++ strcpy(DllArgV[nDllArgC++],"-f"); ++ break; ++ case HIGH_QUALITY: // -k flag for high qualtiy ++ strcpy(DllArgV[nDllArgC++],"-k"); ++ break; ++ } ++ ++ if (nVBR) ++ { ++ // 0=no vbr 1..10 is VBR quality setting -1 ++ sprintf(DllArgV[nDllArgC++],"-v"); ++ sprintf(DllArgV[nDllArgC++],"-V%d",nVBR-1); ++ } ++ ++ // Set frequency ++ sprintf(strTmp,"-s %f",pbeConfig->format.mp3.dwSampleRate/1000.0); ++ strcpy(DllArgV[nDllArgC++],strTmp); ++ ++ // Set bitrate ++ sprintf(strTmp,"-b %d",pbeConfig->format.mp3.wBitrate); ++ strcpy(DllArgV[nDllArgC++],strTmp); ++ ++ // Set copyright flag? ++ if (pbeConfig->format.mp3.bCopyright) ++ strcpy(DllArgV[nDllArgC++],"-c"); ++ ++ // Do we have to tag it as non original ++ if (!pbeConfig->format.mp3.bOriginal) ++ strcpy(DllArgV[nDllArgC++],"-o"); ++ ++ // Add CRC? ++ if (nCRC) ++ strcpy(DllArgV[nDllArgC++],"-p"); ++ ++ // Add input filename ++ strcpy(DllArgV[nDllArgC++],"NULL.wav"); ++ ++ // Add output filename ++ strcpy(DllArgV[nDllArgC++],"NULL.mp3"); ++ ++ // Set the encoder variables ++ parse_args(nDllArgC,argv,&fr_ps,&nPsychoModel,&num_samples, original_file_name,encoded_file_name); ++ ++ // Set private bit? ++ if (pbeConfig->bPrivate) ++ { ++ info.extension = 0; ++ } ++ else ++ { ++ info.extension = 1; ++ } ++ ++ ++ hdr_to_frps(&fr_ps); ++ ++ stereo = fr_ps.stereo; ++ ++ error_protection = info.error_protection; ++ ++ if (info.lay != 3) ++ { ++ printf("Only Layer 3 supported\n"); ++ return BE_ERR_INVALID_FORMAT_PARAMETERS; ++ } ++ if ( nPsychoModel != 2 ) ++ { ++ fprintf( stderr, "Sorry, psycho model 1 not available for layer3\n" ); ++ return BE_ERR_INVALID_FORMAT_PARAMETERS; ++ } ++ ++ #ifdef _DEBUG ++ dump_config(&fr_ps,&nPsychoModel,original_file_name,encoded_file_name); ++ #endif ++ ++ // Everything went OK, thus return SUCCESSFUL ++ return BE_ERR_SUCCESSFUL; ++ } ++ ++ ++ ++ __declspec(dllexport) BE_ERR beDeinitStream(HBE_STREAM hbeStream, PBYTE pOutput, PDWORD pdwOutput) ++ { ++ // Set output buffer ++ bs.pbtOutBuf=pOutput; ++ bs.nOutBufPos=0; ++ ++ // Crank out the last part of the buffer ++ if (fr_ps.header->lay == 3 ) ++ III_FlushBitstream(); ++ ++ // close the stream ++ close_bit_stream_w( &bs ); ++ ++ // Number of bytes in output buffer ++ *pdwOutput=bs.nOutBufPos; ++ ++ // Everything went OK, thus return SUCCESSFUL ++ return BE_ERR_SUCCESSFUL; ++ } ++ ++ ++ __declspec(dllexport) BE_ERR beCloseStream(HBE_STREAM hbeStream) ++ { ++ // DeInit encoder ++ // return DeInitEncoder(); ++ return BE_ERR_SUCCESSFUL; ++ } ++ ++ ++ ++ __declspec(dllexport) VOID beVersion(PBE_VERSION pbeVersion) ++ { ++ // DLL Release date ++ char lpszDate[20]; ++ char lpszTemp[5]; ++ ++ ++ // Set DLL interface version ++ pbeVersion->byDLLMajorVersion=MAJORVERSION; ++ pbeVersion->byDLLMinorVersion=MINORVERSION; ++ ++ // Set Engine version number (Same as Lame version) ++ pbeVersion->byMajorVersion=LAME_MAJOR_VERSION; ++ pbeVersion->byMinorVersion=LAME_MINOR_VERSION; ++ ++ // Get compilation date ++ strcpy(lpszDate,__DATE__); ++ ++ // Get the first three character, which is the month ++ strncpy(lpszTemp,lpszDate,3); ++ ++ // Set month ++ if (strcmp(lpszTemp,"Jan")==0) pbeVersion->byMonth=1; ++ if (strcmp(lpszTemp,"Feb")==0) pbeVersion->byMonth=2; ++ if (strcmp(lpszTemp,"Mar")==0) pbeVersion->byMonth=3; ++ if (strcmp(lpszTemp,"Apr")==0) pbeVersion->byMonth=4; ++ if (strcmp(lpszTemp,"May")==0) pbeVersion->byMonth=5; ++ if (strcmp(lpszTemp,"Jun")==0) pbeVersion->byMonth=6; ++ if (strcmp(lpszTemp,"Jul")==0) pbeVersion->byMonth=7; ++ if (strcmp(lpszTemp,"Aug")==0) pbeVersion->byMonth=8; ++ if (strcmp(lpszTemp,"Sep")==0) pbeVersion->byMonth=9; ++ if (strcmp(lpszTemp,"Oct")==0) pbeVersion->byMonth=10; ++ if (strcmp(lpszTemp,"Nov")==0) pbeVersion->byMonth=11; ++ if (strcmp(lpszTemp,"Dec")==0) pbeVersion->byMonth=12; ++ ++ // Get day of month string (char [4..5]) ++ pbeVersion->byDay=atoi(lpszDate+4); ++ ++ // Get year of compilation date (char [7..10]) ++ pbeVersion->wYear=atoi(lpszDate+7); ++ ++ memset(pbeVersion->zHomepage,0x00,BE_MAX_HOMEPAGE); ++ ++ strcpy(pbeVersion->zHomepage,"http://www.surf.to/cdex"); ++ } ++ ++ ++ __declspec(dllexport) BE_ERR beEncodeChunk(HBE_STREAM hbeStream, DWORD nSamples, PSHORT pSamples, PBYTE pOutput, PDWORD pdwOutput) ++ { ++ int iSampleIndex; ++ ++ // Set output buffer ++ bs.pbtOutBuf=pOutput; ++ bs.nOutBufPos=0; ++ ++ // Is this the last (incomplete) frame ++ if (nSamplesheader; ++ char strTmp[255]; ++ ++ OutputDebugString("Encoding configuration:\n"); ++ ++ sprintf(strTmp,"Algorithm=%s\n", version_names[info->version]); ++ OutputDebugString(strTmp); ++ ++ if(info->mode != MPG_MD_JOINT_STEREO) ++ { ++ sprintf(strTmp,"Layer=%s mode=%s extn=%d psy model=%d\n",layer_names[info->lay-1], mode_names[info->mode],info->mode_ext, *psy); ++ OutputDebugString(strTmp); ++ } ++ else ++ { ++ sprintf(strTmp,"Layer=%s mode=%s extn=data dependant psy model=%d\n",layer_names[info->lay-1], mode_names[info->mode], *psy); ++ OutputDebugString(strTmp); ++ } ++ ++ sprintf(strTmp,"samp frq=%.1f kHz total bitrate=%d kbps\n",s_freq[info->version][info->sampling_frequency],bitrate[info->version][info->lay-1][info->bitrate_index]); ++ OutputDebugString(strTmp); ++ ++ sprintf(strTmp,"de-emph=%d c/right=%d orig=%d errprot=%s\n",info->emphasis, info->copyright, info->original,((info->error_protection) ? "on" : "off")); ++ OutputDebugString(strTmp); ++ ++ sprintf(strTmp,"16 Khz cut off is %s\n",(sfb21)?"enabled":"disabled"); ++ OutputDebugString(strTmp); ++ ++ sprintf(strTmp,"Fast mode is %s\n",(fast_mode)?"enabled":"disabled"); ++ OutputDebugString(strTmp); ++ ++ sprintf(strTmp,"Force ms %s\n",(force_ms)?"enabled":"disabled"); ++ OutputDebugString(strTmp); ++ ++ sprintf(strTmp,"GPsycho acoustic model is %s\n",(gpsycho)?"enabled":"disabled"); ++ OutputDebugString(strTmp); ++ ++ ++ sprintf(strTmp,"input file: '%s' output file: '%s'\n", inPath, outPath); ++ OutputDebugString(strTmp); ++ ++ sprintf(strTmp,"Encoding %s to %s\n",inPath,outPath); ++ OutputDebugString(strTmp); ++ ++ sprintf(strTmp,"Encoding as %.1f kHz %d kbps %s MPEG-1 LayerIII file\n",s_freq[info->version][info->sampling_frequency],bitrate[info->version][info->lay-1][info->bitrate_index],mode_names[info->mode]); ++ OutputDebugString(strTmp); ++ } ++ ++ ++ void DispErr(LPSTR strErr) ++ { ++ MessageBox(NULL,strErr,"",MB_OK); ++ } ++ ++ #endif +diff -r -c -N encoder/Dll/BladeMP3EncDLL.def lame3.13/Dll/BladeMP3EncDLL.def +*** encoder/Dll/BladeMP3EncDLL.def Wed Dec 31 17:00:00 1969 +--- lame3.13/Dll/BladeMP3EncDLL.def Sat Jun 12 23:02:03 1999 +*************** +*** 0 **** +--- 1,9 ---- ++ LIBRARY MP3ENC.DLL ++ EXPORTS ++ ++ beInitStream @1 ++ beEncodeChunk @2 ++ beDeinitStream @3 ++ beCloseStream @4 ++ beVersion @5 ++ beWriteVBRHeader @6 +\ No newline at end of file +diff -r -c -N encoder/Dll/BladeMP3EncDLL.h lame3.13/Dll/BladeMP3EncDLL.h +*** encoder/Dll/BladeMP3EncDLL.h Wed Dec 31 17:00:00 1969 +--- lame3.13/Dll/BladeMP3EncDLL.h Tue Jun 15 18:31:36 1999 +*************** +*** 0 **** +--- 1,150 ---- ++ /* ++ ++ bladedll.h ++ ++ +++++++++++++++++++++++++++ ++ + Blade's Encoder DLL + ++ +++++++++++++++++++++++++++ ++ ++ ------------------------------------------------------ ++ - Version 1.00 (7 November 1998) - Jukka Poikolainen - ++ ------------------------------------------------------ ++ ++ Initial version ++ ++ ------------------------------------------------------ ++ - Version x.xx (x xxxxxxxx xxxx) - xxxxx xxxxxxxxxxx - ++ ------------------------------------------------------ ++ ++ */ ++ ++ #ifndef ___BLADEDLL_H_INCLUDED___ ++ #define ___BLADEDLL_H_INCLUDED___ ++ ++ #pragma pack(push) ++ #pragma pack(1) ++ ++ /* encoding formats */ ++ ++ #define BE_CONFIG_MP3 0 ++ ++ /* type definitions */ ++ ++ typedef unsigned long HBE_STREAM; ++ typedef HBE_STREAM *PHBE_STREAM; ++ typedef unsigned long BE_ERR; ++ ++ /* error codes */ ++ ++ #define BE_ERR_SUCCESSFUL 0x00000000 ++ #define BE_ERR_INVALID_FORMAT 0x00000001 ++ #define BE_ERR_INVALID_FORMAT_PARAMETERS 0x00000002 ++ #define BE_ERR_NO_MORE_HANDLES 0x00000003 ++ #define BE_ERR_INVALID_HANDLE 0x00000004 ++ ++ /* other constants */ ++ ++ #define BE_MAX_HOMEPAGE 256 ++ ++ /* format specific variables */ ++ ++ #define BE_MP3_MODE_STEREO 0 ++ #define BE_MP3_MODE_JSTEREO 1 ++ #define BE_MP3_MODE_DUALCHANNEL 2 ++ #define BE_MP3_MODE_MONO 3 ++ ++ #ifdef _BLADEDLL ++ #undef FLOAT ++ #include ++ #endif ++ ++ typedef struct { ++ ++ DWORD dwConfig; // BE_CONFIG_XXXXX ++ // Currently only BE_CONFIG_MP3 is supported ++ union { ++ ++ struct { ++ ++ DWORD dwSampleRate; // 48000, 44100 and 32000 allowed ++ BYTE byMode; // BE_MP3_MODE_STEREO, BE_MP3_MODE_DUALCHANNEL, BE_MP3_MODE_MONO ++ WORD wBitrate; // 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256 and 320 allowed ++ BOOL bPrivate; ++ BOOL bCRC; ++ BOOL bCopyright; ++ BOOL bOriginal; ++ ++ } mp3; // BE_CONFIG_MP3 ++ ++ struct { ++ ++ DWORD dwSampleRate; ++ BYTE byMode; ++ WORD wBitrate; ++ BYTE byEncodingMethod; ++ ++ } aac; ++ ++ } format; ++ ++ } BE_CONFIG, *PBE_CONFIG; ++ ++ typedef struct { ++ ++ // BladeEnc DLL Version number ++ ++ BYTE byDLLMajorVersion; ++ BYTE byDLLMinorVersion; ++ ++ // BladeEnc Engine Version Number ++ ++ BYTE byMajorVersion; ++ BYTE byMinorVersion; ++ ++ // DLL Release date ++ ++ BYTE byDay; ++ BYTE byMonth; ++ WORD wYear; ++ ++ // BladeEnc Homepage URL ++ ++ CHAR zHomepage[BE_MAX_HOMEPAGE + 1]; ++ ++ } BE_VERSION, *PBE_VERSION; ++ ++ #ifndef _BLADEDLL ++ ++ typedef BE_ERR (*BEINITSTREAM) (PBE_CONFIG, PDWORD, PDWORD, PHBE_STREAM); ++ typedef BE_ERR (*BEENCODECHUNK) (HBE_STREAM, DWORD, PSHORT, PBYTE, PDWORD); ++ typedef BE_ERR (*BEDEINITSTREAM) (HBE_STREAM, PBYTE, PDWORD); ++ typedef BE_ERR (*BECLOSESTREAM) (HBE_STREAM); ++ typedef VOID (*BEVERSION) (PBE_VERSION); ++ ++ #define TEXT_BEINITSTREAM "beInitStream" ++ #define TEXT_BEENCODECHUNK "beEncodeChunk" ++ #define TEXT_BEDEINITSTREAM "beDeinitStream" ++ #define TEXT_BECLOSESTREAM "beCloseStream" ++ #define TEXT_BEVERSION "beVersion" ++ ++ /* ++ BE_ERR beInitStream(PBE_CONFIG pbeConfig, PDWORD dwSamples, PDWORD dwBufferSize, PHBE_STREAM phbeStream); ++ BE_ERR beEncodeChunk(HBE_STREAM hbeStream, DWORD nSamples, PSHORT pSamples, PBYTE pOutput, PDWORD pdwOutput); ++ BE_ERR beDeinitStream(HBE_STREAM hbeStream, PBYTE pOutput, PDWORD pdwOutput); ++ BE_ERR beCloseStream(HBE_STREAM hbeStream); ++ VOID beVersion(PBE_VERSION pbeVersion); ++ */ ++ ++ #else ++ ++ __declspec(dllexport) BE_ERR beInitStream(PBE_CONFIG pbeConfig, PDWORD dwSamples, PDWORD dwBufferSize, PHBE_STREAM phbeStream); ++ __declspec(dllexport) BE_ERR beEncodeChunk(HBE_STREAM hbeStream, DWORD nSamples, PSHORT pSamples, PBYTE pOutput, PDWORD pdwOutput); ++ __declspec(dllexport) BE_ERR beDeinitStream(HBE_STREAM hbeStream, PBYTE pOutput, PDWORD pdwOutput); ++ __declspec(dllexport) BE_ERR beCloseStream(HBE_STREAM hbeStream); ++ __declspec(dllexport) VOID beVersion(PBE_VERSION pbeVersion); ++ ++ #endif ++ ++ #pragma pack(pop) ++ ++ #endif +diff -r -c -N encoder/Dll/MP3EncDll.dsp lame3.13/Dll/MP3EncDll.dsp +*** encoder/Dll/MP3EncDll.dsp Wed Dec 31 17:00:00 1969 +--- lame3.13/Dll/MP3EncDll.dsp Sat Jun 12 23:02:20 1999 +*************** +*** 0 **** +--- 1,275 ---- ++ # Microsoft Developer Studio Project File - Name="MP3EncDll" - Package Owner=<4> ++ # Microsoft Developer Studio Generated Build File, Format Version 5.00 ++ # ** DO NOT EDIT ** ++ ++ # TARGTYPE "Win32 (x86) Dynamic-Link Library" 0x0102 ++ ++ CFG=MP3EncDll - Win32 Debug ++ !MESSAGE This is not a valid makefile. To build this project using NMAKE, ++ !MESSAGE use the Export Makefile command and run ++ !MESSAGE ++ !MESSAGE NMAKE /f "MP3EncDll.mak". ++ !MESSAGE ++ !MESSAGE You can specify a configuration when running NMAKE ++ !MESSAGE by defining the macro CFG on the command line. For example: ++ !MESSAGE ++ !MESSAGE NMAKE /f "MP3EncDll.mak" CFG="MP3EncDll - Win32 Debug" ++ !MESSAGE ++ !MESSAGE Possible choices for configuration are: ++ !MESSAGE ++ !MESSAGE "MP3EncDll - Win32 Release" (based on\ ++ "Win32 (x86) Dynamic-Link Library") ++ !MESSAGE "MP3EncDll - Win32 Debug" (based on\ ++ "Win32 (x86) Dynamic-Link Library") ++ !MESSAGE "MP3EncDll - Win32 Pentium II Release" (based on\ ++ "Win32 (x86) Dynamic-Link Library") ++ !MESSAGE ++ ++ # Begin Project ++ # PROP Scc_ProjName "" ++ # PROP Scc_LocalPath "" ++ CPP=xicl.exe ++ MTL=midl.exe ++ RSC=rc.exe ++ ++ !IF "$(CFG)" == "MP3EncDll - Win32 Release" ++ ++ # PROP BASE Use_MFC 0 ++ # PROP BASE Use_Debug_Libraries 0 ++ # PROP BASE Output_Dir "Release" ++ # PROP BASE Intermediate_Dir "Release" ++ # PROP BASE Target_Dir "" ++ # PROP Use_MFC 0 ++ # PROP Use_Debug_Libraries 0 ++ # PROP Output_Dir "Release" ++ # PROP Intermediate_Dir "Release" ++ # PROP Ignore_Export_Lib 0 ++ # PROP Target_Dir "" ++ # ADD BASE CPP /nologo /MT /W3 /GX /O2 /D "WIN32" /D "NDEBUG" /D "_WINDOWS" /YX /FD /c ++ # ADD CPP /nologo /Zp2 /MT /W3 /GX /O2 /Ob2 /I "..\\" /D "NDEBUG" /D "WIN32" /D "_WINDOWS" /D "_BLADEDLL" /YX /FD -Qmem -Qip /c ++ # ADD BASE MTL /nologo /D "NDEBUG" /mktyplib203 /o NUL /win32 ++ # ADD MTL /nologo /D "NDEBUG" /mktyplib203 /o NUL /win32 ++ # ADD BASE RSC /l 0x409 /d "NDEBUG" ++ # ADD RSC /l 0x409 /d "NDEBUG" ++ BSC32=bscmake.exe ++ # ADD BASE BSC32 /nologo ++ # ADD BSC32 /nologo ++ LINK32=xilink.exe ++ # ADD BASE LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:windows /dll /machine:I386 ++ # ADD LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:windows /dll /map /machine:I386 /out:"..\..\mp3enc.dll" ++ ++ !ELSEIF "$(CFG)" == "MP3EncDll - Win32 Debug" ++ ++ # PROP BASE Use_MFC 0 ++ # PROP BASE Use_Debug_Libraries 1 ++ # PROP BASE Output_Dir "Debug" ++ # PROP BASE Intermediate_Dir "Debug" ++ # PROP BASE Target_Dir "" ++ # PROP Use_MFC 0 ++ # PROP Use_Debug_Libraries 1 ++ # PROP Output_Dir "Debug" ++ # PROP Intermediate_Dir "Debug" ++ # PROP Ignore_Export_Lib 0 ++ # PROP Target_Dir "" ++ # ADD BASE CPP /nologo /MTd /W3 /Gm /GX /Zi /Od /D "WIN32" /D "_DEBUG" /D "_WINDOWS" /YX /FD /c ++ # ADD CPP /nologo /Zp2 /MTd /W3 /Gm /GX /Zi /Od /I "..\\" /D "_DEBUG" /D "WIN32" /D "_WINDOWS" /D "_BLADEDLL" /YX /FD /c ++ # ADD BASE MTL /nologo /D "_DEBUG" /mktyplib203 /o NUL /win32 ++ # ADD MTL /nologo /D "_DEBUG" /mktyplib203 /o NUL /win32 ++ # ADD BASE RSC /l 0x409 /d "_DEBUG" ++ # ADD RSC /l 0x409 /d "_DEBUG" ++ BSC32=bscmake.exe ++ # ADD BASE BSC32 /nologo ++ # ADD BSC32 /nologo ++ LINK32=xilink.exe ++ # ADD BASE LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:windows /dll /debug /machine:I386 /pdbtype:sept ++ # ADD LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:windows /dll /debug /machine:I386 /out:"..\..\mp3enc.dll" /pdbtype:sept ++ # SUBTRACT LINK32 /map ++ ++ !ELSEIF "$(CFG)" == "MP3EncDll - Win32 Pentium II Release" ++ ++ # PROP BASE Use_MFC 0 ++ # PROP BASE Use_Debug_Libraries 0 ++ # PROP BASE Output_Dir "MP3EncDl" ++ # PROP BASE Intermediate_Dir "MP3EncDl" ++ # PROP BASE Ignore_Export_Lib 0 ++ # PROP BASE Target_Dir "" ++ # PROP Use_MFC 0 ++ # PROP Use_Debug_Libraries 0 ++ # PROP Output_Dir "MP3EncDl" ++ # PROP Intermediate_Dir "MP3EncDl" ++ # PROP Ignore_Export_Lib 0 ++ # PROP Target_Dir "" ++ # ADD BASE CPP /nologo /Zp2 /MT /W3 /GX /O2 /Ob2 /I "..\..\lame" /D "NDEBUG" /D "WIN32" /D "_WINDOWS" /D "DLLCOMPILE" /D "LAYER3" /Fp"Release/MP3EncDll.pch" /YX /FD /Qmem /Qipo /c ++ # ADD CPP /nologo /G6 /Zp2 /MT /W3 /GX /O2 /Ob2 /I "..\\" /D "NDEBUG" /D "WIN32" /D "_WINDOWS" /D "_BLADEDLL" /D BS_FORMAT= /Fp"Release/MP3EncDll.pch" /YX /FD /Qmem /Qipo /Qxi /c ++ # ADD BASE MTL /nologo /D "NDEBUG" /mktyplib203 /o NUL /win32 ++ # ADD MTL /nologo /D "NDEBUG" /mktyplib203 /o NUL /win32 ++ # ADD BASE RSC /l 0x409 /d "NDEBUG" ++ # ADD RSC /l 0x409 /d "NDEBUG" ++ BSC32=bscmake.exe ++ # ADD BASE BSC32 /nologo ++ # ADD BSC32 /nologo ++ LINK32=xilink.exe ++ # ADD BASE LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:windows /dll /machine:I386 /out:"..\..\cdex\mp3enc.dll" ++ # ADD LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:windows /dll /map /machine:I386 /out:"..\..\cdex\mp3enc.dll" ++ ++ !ENDIF ++ ++ # Begin Target ++ ++ # Name "MP3EncDll - Win32 Release" ++ # Name "MP3EncDll - Win32 Debug" ++ # Name "MP3EncDll - Win32 Pentium II Release" ++ # Begin Source File ++ ++ SOURCE=.\BladeMP3EncDLL.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\BladeMP3EncDLL.def ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\BladeMP3EncDLL.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\common.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\common.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\encode.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\encoder.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\formatBitstream.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\formatBitstream.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\huffman.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\huffman.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\ieeefloat.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\ieeefloat.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE="..\l3bitstream-pvt.h" ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\l3bitstream.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\l3bitstream.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\l3psy.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\l3psy.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\l3side.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE="..\loop-pvt.h" ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\loop.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\loop.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\mdct.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\mdct.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\musicin.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\portableio.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\portableio.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\psy.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\reservoir.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\reservoir.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\sqrttab.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\subs.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\subs.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\tables.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\VbrTag.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\VbrTag.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\version.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\wavheader.h ++ # End Source File ++ # End Target ++ # End Project +diff -r -c -N encoder/Dll/MP3EncDll.dsw lame3.13/Dll/MP3EncDll.dsw +*** encoder/Dll/MP3EncDll.dsw Wed Dec 31 17:00:00 1969 +--- lame3.13/Dll/MP3EncDll.dsw Sun Jun 6 16:19:39 1999 +*************** +*** 0 **** +--- 1,29 ---- ++ Microsoft Developer Studio Workspace File, Format Version 5.00 ++ # WARNING: DO NOT EDIT OR DELETE THIS WORKSPACE FILE! ++ ++ ############################################################################### ++ ++ Project: "MP3EncDll"=.\MP3EncDll.dsp - Package Owner=<4> ++ ++ Package=<5> ++ {{{ ++ }}} ++ ++ Package=<4> ++ {{{ ++ }}} ++ ++ ############################################################################### ++ ++ Global: ++ ++ Package=<5> ++ {{{ ++ }}} ++ ++ Package=<3> ++ {{{ ++ }}} ++ ++ ############################################################################### ++ +diff -r -c -N encoder/INSTALL lame3.13/INSTALL +*** encoder/INSTALL Wed Jan 22 02:43:12 1997 +--- lame3.13/INSTALL Wed Jun 16 22:40:32 1999 +*************** +*** 1,16 **** +! Building the MPEG audio encoding software: + +! If you have gnu make and are on a Unix-type system, you should only have to + type: + +! ./configure +! make + +- This will create a Makefile customized to your environment, overwriting the +- previous Makefile, using Makefile.in as a template. + +! For other systems you may have to modify the Makefile. Some flags of +! interest are: + + BS_FORMAT + Set to BINARY to work with most MPEG audio decoders (and save space). You +--- 1,87 ---- +! LAME 3.1x 6/99 Mark Taylor (www.sulaco.org/mp3) + +! ======================================================================= +! Building the MPEG audio encoding software: +! ======================================================================= +! If you are reading this, you have already applied the patch to the +! dist10 ISO demonstration source. If you do not have GTK 1.1.16 or +! higher, or do not want the graphical frame analyzer (it uses a lot of +! memory), edit the Makefile and dont define GTK and GTKLIBS. Now just + type: + +! % make +! +! If it doesn't work, figure out why and send me the fix! +! +! LAME has built in support for raw pcm, aiff and wav formats. +! If you have trouble with this, or want to encode from other +! sound formats, you can now compile LAME with Erik de Castro Lopo's +! libsndfile. To use libsndfile in LAME: +! +! 1. install libsndfile (see http://www.zip.com.au/~erikd/libsndfile/). +! 2. set SNDLIB and LIBSNDFILE as shown in the LAME Makefile. +! 3. make clean ; make +! +! Thanks to Albert Faber for the libsndfile support! +! +! +! +! ======================================================================= +! To encode: +! ======================================================================= +! % lame sample.wav sample.mp3 +! +! Variable Bitrate (VBR): +! % lame -v sample.wav sample.mp3 +! +! VBR and specialized mid/side masking thresholds: +! % lame -v -m f sample.wav sample.mp3 +! +! For more options, just type: +! +! % lame +! +! Scripts are included to run lame on multiple files: +! +! bach script: mlame Run "mlame -h" for instructions. +! sh script: auenc Run auenc for instructions +! +! +! +! +! +! +! +! +! +! +! ======================================================================= +! Portability +! ======================================================================= +! LAME 3.x has been run on many platforms - see the Makefile. +! +! +! ======================================================================= +! BUGS: +! ======================================================================= +! .wav files are assumed to be 44.1kHz stereo. (i.e. from a CD) +! .wav file reading in broken on non-Linux versions of UNIX. +! +! I haven't tried .aiff files, but pcm files will work on all platforms: +! (default is XINU format) but try with and without -x. +! +! To convert a 44.1kHz .wav file to a LAME readable pcm file, use: +! % sox sample.wav -t raw -x -w -s -c 2 -r 44100 sample.pcm +! +! +! +! + + +! ======================================================================= +! Information from the ISO README +! ======================================================================= +! Some other flags of interest are: + + BS_FORMAT + Set to BINARY to work with most MPEG audio decoders (and save space). You +diff -r -c -N encoder/Makefile lame3.13/Makefile +*** encoder/Makefile Wed Jan 22 02:43:23 1997 +--- lame3.13/Makefile Thu Jun 24 18:30:41 1999 +*************** +*** 1,68 **** +! # Generated automatically from Makefile.in by configure. +! ############################################################################ +! ## ISO MPEG Audio Subgroup Software Simulation Group (1996) +! ## ISO 13818-3 MPEG-2 Audio Decoder - Lower Sampling Frequency Extension +! ## +! ## $Id: Makefile,v 1.2 1996/08/14 23:18:34 rowlands Exp $ +! ## +! ## Makefile for encoder. Requies GNU make. +! ## +! ## $Log: Makefile,v $ +! ## Revision 1.2 1996/08/14 23:18:34 rowlands +! ## Cleanups. +! ## +! ## Revision 1.2 1996/02/14 05:18:05 rowlands +! ## Cleanups. +! ## +! ## Revision 1.1 1996/02/14 04:04:23 rowlands +! ## Initial revision +! ## +! ## Received from Mike Coleman +! ############################################################################ + + CC = gcc + + c_sources = \ + common.c \ + encode.c \ + formatBitstream.c \ + huffman.c \ +- ieeefloat.c \ + l3bitstream.c \ + l3psy.c \ + loop.c \ + mdct.c \ + musicin.c \ + portableio.c \ +- psy.c \ + reservoir.c \ + subs.c \ +! tonal.c + + OBJ = $(c_sources:.c=.o) + DEP = $(c_sources:.c=.d) + +- NINT_SWITCH = +- +- CC_SWITCHES = -g -O -DUNIX -DBS_FORMAT=BINARY $(NINT_SWITCH) -DNDEBUG +- +- PGM = encode +- +- LIBS = -lm +- + + %.o: %.c + $(CC) $(CC_SWITCHES) -c $< -o $@ + + %.d: %.c +! $(SHELL) -ec '$(CC) -M $(CC_SWITCHES) $< | sed '\''s/$*.o/& $@/g'\'' > $@' + + +! $(PGM): $(OBJ) Makefile +! $(CC) -o $(PGM) $(OBJ) $(LIBS) + + clean: +! -rm $(OBJ) $(DEP) + + tags: TAGS + +--- 1,198 ---- +! # Makefile for LAME 3.x +! # +! # LAME is reported to work under: +! # Linux (i86), Sun, SGI, NetBSD 1.3.2 (StrongARM), +! # Compaq Alpha, FreeBSD (i86), OS2 Warp, BeOS, Amiga and even VC++ +! # Macintosh PPC +! # +! # +! UNAME = $(shell uname) +! ARCH = $(shell uname -m) + ++ ++ # defaults: ++ PGM = lame + CC = gcc ++ CC_OPTS = -O ++ GTK = ++ GTKLIBS = ++ SNDLIB = ++ LIBSNDFILE = ++ LIBS = -lm ++ MAKEDEP = -M ++ NINT_SWITCH = ++ ++ ++ ++ ########################################################################## ++ #define these to use Erik de Castro Lopo's libsndfile ++ # http://www.zip.com.au/~erikd/libsndfile/ ++ ########################################################################## ++ #SNDLIB = -DLIBSNDFILE ++ #LIBSNDFILE=-lsndfile ++ # if libsndfile is in a custom location, try: ++ #LIBSNDFILE=-L $(LIBSNDHOME) -lsndfile -I $(LIBSNDHOME) ++ ++ ++ ++ ++ ++ ++ ########################################################################## ++ # set this (and run make clean) to make a binary that is distributable ++ # i.e. frame analyzier only, which doesn't produce a .mp3 file ++ ########################################################################## ++ #DISTRIB = -DDISTRIB ++ #PGM = mp3x ++ ++ ++ ++ ++ ########################################################################## ++ # LINUX ++ ########################################################################## ++ ifeq ($(UNAME),Linux) ++ # remove these if you do not have GTK, or dont want the GTK frame analyzer ++ GTK = -DHAVEGTK `gtk-config --cflags` ++ GTKLIBS = `gtk-config --libs` ++ # for debugging: ++ # CC_OPTS = -UNDEBUG -O -Wall -g -DABORTFP ++ # these options work very well with egcs-990524 ++ # CC = egcs ++ # CC_OPTS = -O9 -fomit-frame-pointer -march=pentium \ ++ # -mcpu=pentium -ffast-math -funroll-loops \ ++ # -fprofile-arcs -fbranch-probabilities ++ ++ ++ ++ ########################################################################## ++ # LINUX on Digital/Compaq Alpha CPUs ++ ########################################################################## ++ ifeq ($(ARCH),alpha) ++ # double is faster than float on Alpha ++ CC_OPTS = -O4 -Wall -fomit-frame-pointer -ffast-math -funroll-loops \ ++ -mfp-regs -fschedule-insns -fschedule-insns2 \ ++ -finline-functions \ ++ -DFLOAT=double ++ # standard Linux libm ++ LIBS = -lm ++ # optimized libffm (free fast math library) ++ #LIBS = -lffm ++ # Compaq's fast math library ++ #LIBS = -lcpml ++ endif ++ endif ++ ++ ++ ++ ########################################################################## ++ # FreeBSD ++ ########################################################################## ++ ifeq ($(UNAME),FreeBSD) ++ # remove these if you do not have GTK, or dont want the GTK frame analyzer ++ GTK = -DHAVEGTK `gtk12-config --cflags` ++ GTKLIBS = `gtk12-config --libs` ++ endif ++ ++ ++ ########################################################################## ++ # SunOS ++ ########################################################################## ++ ifeq ($(UNAME),SunOS) ++ CC = cc ++ MAKEDEP = -xM ++ endif ++ ++ ++ ########################################################################## ++ # SGI ++ ########################################################################## ++ ifeq ($(UNAME),IRIX64) ++ CC = cc ++ endif ++ ++ ++ ########################################################################## ++ # Compaq Alpha running Dec Unix (OSF) ++ ########################################################################## ++ ifeq ($(UNAME),OSF1) ++ CC = cc ++ CC_OPTS = -fast -O3 -std -g3 -non_shared ++ NINT_SWITCH = -DHAVE_NINT ++ endif ++ ++ ########################################################################## ++ # BeOS ++ ########################################################################## ++ ifeq ($(UNAME),BeOS) ++ CC = $(BE_C_COMPILER) ++ LIBS = ++ ifeq ($(ARCH),BePC) ++ CC_OPTS = -O9 -fomit-frame-pointer -march=pentium \ ++ -mcpu=pentium -ffast-math -funroll-loops \ ++ -fprofile-arcs -fbranch-probabilities ++ else ++ CC_OPTS = -opt all ++ MAKEDEP = -make ++ endif ++ endif ++ ++ ########################################################################## ++ # OS/2 ++ ########################################################################## ++ # defaults work fine except for: ++ # PGM = lame.exe ++ # and remove all the above statments having to do with non OS/2 systems ++ + ++ ++ ++ CC_SWITCHES = -DNDEBUG $(CC_OPTS) $(DISTRIB) $(SNDLIB) $(GTK) -DBS_FORMAT=BINARY $(NINT_SWITCH) + c_sources = \ + common.c \ + encode.c \ + formatBitstream.c \ ++ get_audio.c \ ++ gtkanal.c \ ++ gpkplotting.c \ + huffman.c \ + l3bitstream.c \ ++ ieeefloat.c \ + l3psy.c \ + loop.c \ + mdct.c \ + musicin.c \ + portableio.c \ + reservoir.c \ + subs.c \ +! tables.c \ +! VbrTag.c \ +! version.c \ +! mpglib/common.c \ +! mpglib/dct64_i386.c \ +! mpglib/decode_i386.c \ +! mpglib/layer3.c \ +! mpglib/tabinit.c \ +! mpglib/interface.c \ +! mpglib/main.c \ +! + + OBJ = $(c_sources:.c=.o) + DEP = $(c_sources:.c=.d) + + + %.o: %.c + $(CC) $(CC_SWITCHES) -c $< -o $@ + + %.d: %.c +! $(SHELL) -ec '$(CC) $(MAKEDEP) $(CC_SWITCHES) $< | sed '\''s;$*.o;& $@;g'\'' > $@' + + +! $(PGM): $(OBJ) Makefile +! $(CC) -o $(PGM) $(OBJ) $(LIBS) $(LIBSNDFILE) $(GTKLIBS) + + clean: +! -rm $(OBJ) $(DEP) $(PGM) + + tags: TAGS + +*************** +*** 70,72 **** +--- 200,216 ---- + etags -T ${c_sources} + + -include $(DEP) ++ ++ test8: $(PGM) ++ ./lame /home/mt/mp3/test/castanets.wav ++ cmp -l /home/mt/mp3/test/castanets.wav.mp3 /home/mt/mp3/test/castanets.ref8.mp3 | head ++ test9: $(PGM) ++ ./lame /home/mt/mp3/test/castanets.wav ++ cmp -l /home/mt/mp3/test/castanets.wav.mp3 /home/mt/mp3/test/castanets.ref9.mp3 | head ++ testf: $(PGM) ++ ./lame -m f /home/mt/mp3/test/castanets.wav ++ cmp -l /home/mt/mp3/test/castanets.wav.mp3 /home/mt/mp3/test/castanets.mf.mp3 | head ++ testg: $(PGM) ++ ./lame -g /home/mt/mp3/test/castanets.wav ++ cmp -l /home/mt/mp3/test/castanets.wav.mp3 /home/mt/mp3/test/castanets.ref8.mp3 ++ +diff -r -c -N encoder/Makefile.in lame3.13/Makefile.in +*** encoder/Makefile.in Wed Jan 22 02:43:12 1997 +--- lame3.13/Makefile.in Wed Dec 31 17:00:00 1969 +*************** +*** 1,68 **** +- ############################################################################ +- ## ISO MPEG Audio Subgroup Software Simulation Group (1996) +- ## ISO 13818-3 MPEG-2 Audio Decoder - Lower Sampling Frequency Extension +- ## +- ## $Id: Makefile.in,v 1.2 1996/02/14 05:18:05 rowlands Exp $ +- ## +- ## Makefile for encoder. Requies GNU make. +- ## +- ## $Log: Makefile.in,v $ +- ## Revision 1.2 1996/02/14 05:18:05 rowlands +- ## Cleanups. +- ## +- ## Revision 1.1 1996/02/14 04:04:23 rowlands +- ## Initial revision +- ## +- ## Received from Mike Coleman +- ############################################################################ +- +- CC = gcc +- +- c_sources = \ +- common.c \ +- encode.c \ +- formatBitstream.c \ +- huffman.c \ +- ieeefloat.c \ +- l3bitstream.c \ +- l3psy.c \ +- loop.c \ +- mdct.c \ +- musicin.c \ +- portableio.c \ +- psy.c \ +- reservoir.c \ +- subs.c \ +- tonal.c +- +- OBJ = $(c_sources:.c=.o) +- DEP = $(c_sources:.c=.d) +- +- NINT_SWITCH = @NINTSW@ +- +- CC_SWITCHES = -g -O -DUNIX -DBS_FORMAT=BINARY $(NINT_SWITCH) -DNDEBUG +- +- PGM = encode +- +- LIBS = @LIBS@ @MATHLIBS@ +- +- +- %.o: %.c +- $(CC) $(CC_SWITCHES) -c $< -o $@ +- +- %.d: %.c +- $(SHELL) -ec '$(CC) -M $(CC_SWITCHES) $< | sed '\''s/$*.o/& $@/g'\'' > $@' +- +- +- $(PGM): $(OBJ) Makefile +- $(CC) -o $(PGM) $(OBJ) $(LIBS) +- +- clean: +- -rm $(OBJ) $(DEP) +- +- tags: TAGS +- +- TAGS: ${c_sources} +- etags -T ${c_sources} +- +- -include $(DEP) +--- 0 ---- +diff -r -c -N encoder/README lame3.13/README +*** encoder/README Wed Dec 31 17:00:00 1969 +--- lame3.13/README Thu May 20 11:40:00 1999 +*************** +*** 0 **** +--- 1,29 ---- ++ LAME 3.0x ++ LAME Ain't an MP3 Encoder ++ www.sulaco.org/mp3 ++ ++ Written by Mike Cheng (www.uq.net.au/~zzmcheng). Modified ++ and maintained by Mark Taylor (www.sulaco.org/mp3). ++ ++ This is a patch agains the ISO MPEG1 demonstration source. The ++ modifications are distributed under the GNU GENERAL PUBLIC LICENSE ++ (see the file COPYING for details). ++ ++ The graphical frame analyzer uses the MPGLIB decoding engine, written by: ++ Michael Hipp (email: Michael.Hipp@student.uni-tuebingen.de) ++ and released under a more restrictive agreement. ++ ++ Various components of the patch are: ++ ++ Copyrights (c) 1999 by Mark Taylor: new psy-model, quantization and bit allocation ++ Copyrights (c) 1998 by Michael Cheng: new transforms and many other improvements ++ Copyrights (c) 1995,1996,1997 by Michael Hipp: mpglib ++ ++ ++ The ISO demonstration source (distrubuted seperatly) has its own ++ patent, copyright and distrubtion issues. The code is freely ++ available, but any comercial use (including distributing free ++ encoders) requires a license agreement from FhG. ++ ++ ++ +diff -r -c -N encoder/README.WINGTK lame3.13/README.WINGTK +*** encoder/README.WINGTK Wed Dec 31 17:00:00 1969 +--- lame3.13/README.WINGTK Mon May 24 21:14:01 1999 +*************** +*** 0 **** +--- 1,55 ---- ++ ++ If you got the LAME source from http://www.sulaco.org/mp3, the ++ libraries and binaries mentioned below are not included. You will ++ need to get the GLIB and GTK developer libraries, available from: ++ http://user.sgic.fi/~tml/gimp/win32/ ++ (or see http://www.gtk.org, and click on the GTK under win32 link). ++ ++ -Mark Taylor ++ ++ ++ *************************************************************************** ++ ++ ++ ++ May 24,1999 ++ ++ Release notes MP3X for WIN2 ++ ++ Installation instructions: ++ ++ Copy the DLL files: (in GtkWin32Dll.zip) ++ ++ gdk-1.3.dll ++ glib-1.2.dll ++ gtk-1.3.dll ++ gmodule-1.2.dll ++ ++ ++ either into the current working direcory you are running LAME, or ++ copy them in the windows system directory. ++ ++ ++ This is basically it, you can run LAME with the Graphic Psyco Analysis as follows: ++ ++ Lame -g my.wav ++ ++ (just enter Lame at the dos prompt to get all the options) Of course you can ommit the ++ -g optiont to use lame without GPA. ++ ++ ++ ++ How to compile Lame with GPA for Windows 32 platforms ++ ++ In order to compile Lame with the Graphic Psyco Analysis (GPA) with MSVC, you have to have the ++ proper GTK and GLIB include files and lib files. I've put these files together in a ++ zip file (named GtkWin32Lib.zip). Unzip these files into the Lame source directory, make ++ sure you unpack it with the directory options. The files are created in the WinGTK subdirecory ++ Furthermore, you should have at least Lame version 3.05 beta 2. ++ Open the project files in MSVC 5, and just hit the build button. There will be a few warnings, ++ but you can ignore them. ++ ++ ++ ++ Albert L. Faber ++ +diff -r -c -N encoder/TODO lame3.13/TODO +*** encoder/TODO Wed Dec 31 17:00:00 1969 +--- lame3.13/TODO Thu Jun 24 18:31:48 1999 +*************** +*** 0 **** +--- 1,17 ---- ++ Does stdin work when LAME is compiled to use libsndfile? ++ ++ make use of subblock_gain ++ slow, but best way: loop over all possible subblock_gains and ++ see which combination gives best distortion? But I dont know ++ if I trust calc_noise for short blocks. ++ ++ true mid/side masking thresholds in jstereo mode. requires computing ++ left,right, mid & side thresholds for all frames. ++ ++ 10Hz high-pass filter ++ ++ ++ ++ ++ ++ +diff -r -c -N encoder/VbrTag.c lame3.13/VbrTag.c +*** encoder/VbrTag.c Wed Dec 31 17:00:00 1969 +--- lame3.13/VbrTag.c Mon Jun 21 12:19:04 1999 +*************** +*** 0 **** +--- 1,479 ---- ++ /* ++ * Version numbering for LAME. ++ * ++ * Copyright (c) 1999 A.L. Faber ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License as published by ++ * the Free Software Foundation; either version 2, or (at your option) ++ * any later version. ++ * ++ * This program is distributed in the hope that it will be useful, ++ * but WITHOUT ANY WARRANTY; without even the implied warranty of ++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the ++ * GNU General Public License for more details. ++ * ++ * You should have received a copy of the GNU General Public License ++ * along with this program; see the file COPYING. If not, write to ++ * the Free Software Foundation, 675 Mass Ave, Cambridge, MA 02139, USA. ++ */ ++ ++ #include ++ #include ++ #include ++ #include ++ #include "VbrTag.h" ++ #include "version.h" ++ ++ #ifdef _DEBUG ++ // #define DEBUG_VBRTAG ++ #endif ++ ++ ++ int SizeOfEmptyFrame[2][2]= ++ { ++ {32,17}, ++ {17,9}, ++ }; ++ ++ static BYTE pbtStreamBuffer[216]; ++ static long g_Position[NUMTOCENTRIES]; ++ static int nZeroStreamSize=0; ++ static int TotalFrameSize=0; ++ static char VBRTag[]={"Xing"}; ++ //static char VBRTag[]={"Lame"}; ++ ++ ++ int* pVbrFrames=NULL; ++ int nVbrNumFrames=0; ++ int nVbrFrameBufferSize=0; ++ ++ /**************************************************************************** ++ * AddVbrFrame: Add VBR entry, used to fill the VBR the TOC entries ++ * Paramters: ++ * nStreamPos: how many bytes did we write to the bitstream so far ++ * (in Bytes NOT Bits) ++ **************************************************************************** ++ */ ++ void AddVbrFrame(int nStreamPos) ++ { ++ // Simple exponential growing buffer ++ if (pVbrFrames==NULL || nVbrFrameBufferSize==0) ++ { ++ // Start with 100 frames ++ nVbrFrameBufferSize=100; ++ ++ // Allocate them ++ pVbrFrames=(int*)malloc(nVbrFrameBufferSize*sizeof(int)); ++ } ++ ++ // Is buffer big enough to store this new frame ++ if (nVbrNumFrames==nVbrFrameBufferSize) ++ { ++ // Guess not, double th e buffer size ++ nVbrFrameBufferSize*=2; ++ ++ // Allocate new buffer ++ pVbrFrames=(int*)realloc(pVbrFrames,nVbrFrameBufferSize*sizeof(int)); ++ } ++ ++ // Store values ++ pVbrFrames[nVbrNumFrames++]=nStreamPos; ++ } ++ ++ ++ /*-------------------------------------------------------------*/ ++ static int ExtractI4(unsigned char *buf) ++ { ++ int x; ++ // big endian extract ++ x = buf[0]; ++ x <<= 8; ++ x |= buf[1]; ++ x <<= 8; ++ x |= buf[2]; ++ x <<= 8; ++ x |= buf[3]; ++ return x; ++ } ++ ++ void CreateI4(unsigned char *buf, int nValue) ++ { ++ // big endian create ++ buf[0]=(nValue>>24)&0xff; ++ buf[1]=(nValue>>16)&0xff; ++ buf[2]=(nValue>> 8)&0xff; ++ buf[3]=(nValue )&0xff; ++ } ++ ++ ++ /*-------------------------------------------------------------*/ ++ int GetVbrTag(VBRTAGDATA *pTagData, unsigned char *buf) ++ { ++ int i, head_flags; ++ int h_id, h_mode, h_sr_index; ++ static int sr_table[4] = { 44100, 48000, 32000, 99999 }; ++ ++ // get Vbr header data ++ pTagData->flags = 0; // clear to null incase fail ++ ++ // get selected MPEG header data ++ h_id = (buf[1] >> 3) & 1; ++ h_sr_index = (buf[2] >> 2) & 3; ++ h_mode = (buf[3] >> 6) & 3; ++ ++ // determine offset of header ++ if( h_id ) ++ { ++ // mpeg1 ++ if( h_mode != 3 ) buf+=(32+4); ++ else buf+=(17+4); ++ } ++ else ++ { ++ // mpeg2 ++ if( h_mode != 3 ) buf+=(17+4); ++ else buf+=(9+4); ++ } ++ ++ if( buf[0] != VBRTag[0] ) return 0; // fail ++ if( buf[1] != VBRTag[1] ) return 0; // header not found ++ if( buf[2] != VBRTag[2] ) return 0; ++ if( buf[3] != VBRTag[3] ) return 0; ++ ++ buf+=4; ++ ++ pTagData->h_id = h_id; ++ ++ pTagData->samprate = sr_table[h_sr_index]; ++ ++ if( h_id == 0 ) ++ pTagData->samprate >>= 1; ++ ++ head_flags = pTagData->flags = ExtractI4(buf); buf+=4; // get flags ++ ++ if( head_flags & FRAMES_FLAG ) ++ { ++ pTagData->frames = ExtractI4(buf); buf+=4; ++ } ++ ++ if( head_flags & BYTES_FLAG ) ++ { ++ pTagData->bytes = ExtractI4(buf); buf+=4; ++ } ++ ++ if( head_flags & TOC_FLAG ) ++ { ++ if( pTagData->toc != NULL ) ++ { ++ for(i=0;itoc[i] = buf[i]; ++ } ++ buf+=NUMTOCENTRIES; ++ } ++ ++ pTagData->vbr_scale = -1; ++ ++ if( head_flags & VBR_SCALE_FLAG ) ++ { ++ pTagData->vbr_scale = ExtractI4(buf); buf+=4; ++ } ++ ++ #ifdef DEBUG_VBRTAG ++ printf("\n\n********************* VBR TAG INFO *****************\n"); ++ printf("tag :%s\n",VBRTag); ++ printf("head_flags :%d\n",head_flags); ++ printf("bytes :%d\n",pTagData->bytes); ++ printf("frames :%d\n",pTagData->frames); ++ printf("VBR Scale :%d\n",pTagData->vbr_scale); ++ printf("toc:\n"); ++ if( pTagData->toc != NULL ) ++ { ++ for(i=0;itoc[i])); ++ } ++ } ++ printf("\n***************** END OF VBR TAG INFO ***************\n"); ++ #endif ++ return 1; // success ++ } ++ ++ ++ /**************************************************************************** ++ * InitVbrTag: Initializes the header, and write empty frame to stream ++ * Paramters: ++ * fpStream: pointer to output file stream ++ * nVersion: 0= MPEG1 1=MPEG2 ++ * nMode : Channel Mode: 0=STEREO 1=JS 2=DS 3=MONO ++ **************************************************************************** ++ */ ++ int InitVbrTag(Bit_stream_struc* pBs,int nVersion, int nMode, int SampIndex) ++ { ++ int i; ++ ++ // Clear Frame position array variables ++ pVbrFrames=NULL; ++ nVbrNumFrames=0; ++ nVbrFrameBufferSize=0; ++ ++ // Clear struct ++ memset(g_Position,0x00,sizeof(g_Position)); ++ ++ // Clear stream buffer ++ memset(pbtStreamBuffer,0x00,sizeof(pbtStreamBuffer)); ++ ++ // Set TOC values to 255 ++ for (i=0;ibuf_byte_idx); ++ pBs->buf[pBs->buf_byte_idx] = 0; ++ #endif ++ ++ ++ // Success ++ return 0; ++ } ++ ++ ++ ++ /**************************************************************************** ++ * PutVbrTag: Write final VBR tag to the file ++ * Paramters: ++ * lpszFileName: filename of MP3 bit stream ++ * nVbrScale : encoder quality indicator (0..100) ++ **************************************************************************** ++ */ ++ int PutVbrTag(char* lpszFileName,int nVbrScale) ++ { ++ int i; ++ long lFileSize=0; ++ int nStreamIndex=0; ++ char abyte; ++ BYTE btToc[NUMTOCENTRIES]; ++ FILE* fpStream=NULL; ++ char str1[80]; ++ ++ ++ if (nVbrNumFrames==0 || pVbrFrames==NULL) ++ return -1; ++ ++ // Open the bitstream again ++ fpStream=fopen(lpszFileName,"rb+"); ++ ++ // Assert stream is valid ++ if (fpStream==NULL) ++ return -1; ++ ++ // Clear stream buffer ++ memset(pbtStreamBuffer,0x00,sizeof(pbtStreamBuffer)); ++ ++ // Seek to end of file ++ fseek(fpStream,0,SEEK_END); ++ ++ // Get file size ++ lFileSize=ftell(fpStream); ++ ++ // Seek to first real frame ++ fseek(fpStream,TotalFrameSize,SEEK_SET); ++ ++ // Read the header (first valid frame) ++ fread(pbtStreamBuffer,4,1,fpStream); ++ ++ /* the default VBR header. 48kbs layer III, no padding, no crc */ ++ /* but sampling freq, mode andy copyright/copy protection taken */ ++ /* from first valid frame */ ++ pbtStreamBuffer[0]=(char) 0xff; ++ pbtStreamBuffer[1]=(char) 0xfb; ++ abyte = pbtStreamBuffer[2] & (char) 0x0c; ++ #ifdef XING64 ++ pbtStreamBuffer[2]=(char) 0x50 | abyte; /* 64kbs frame */ ++ #else ++ pbtStreamBuffer[2]=(char) 0x30 | abyte; /* 48kbs frame */ ++ #endif ++ ++ // Seek to the beginning of the stream ++ fseek(fpStream,0,SEEK_SET); ++ ++ // Clear all TOC entries ++ memset(btToc,0,sizeof(btToc)); ++ ++ // Set unassigned toc value with its neighbor value ++ for (i=0;i=NUMTOCENTRIES ) nPercent=NUMTOCENTRIES-1; ++ if (fRelStreamPos>255) fRelStreamPos=255; ++ ++ // Assign toc entry value ++ btToc[nPercent]=(BYTE)fRelStreamPos; ++ } ++ ++ // Fix up empty fields (if we have less than 100 frames) ++ for (i=1;i 100.0f ) percent = 100.0f; ++ ++ a = (int)percent; ++ if( a > 99 ) a = 99; ++ fa = TOC[a]; ++ if( a < 99 ) { ++ fb = TOC[a+1]; ++ } ++ else { ++ fb = 256.0f; ++ } ++ ++ ++ fx = fa + (fb-fa)*(percent-a); ++ ++ seekpoint = (int)((1.0f/256.0f)*fx*file_bytes); ++ ++ ++ return seekpoint; ++ } ++ /*-------------------------------------------------------------*/ +diff -r -c -N encoder/VbrTag.c.rej lame3.13/VbrTag.c.rej +*** encoder/VbrTag.c.rej Wed Dec 31 17:00:00 1969 +--- lame3.13/VbrTag.c.rej Mon Jun 21 12:13:55 1999 +*************** +*** 0 **** +--- 1,5 ---- ++ *************** ++ *** 422 ++ - strncpy(&pbtStreamBuffer[nStreamIndex],str1,(size_t) 20); ++ --- 422 ----- ++ + strncpy((char *)&pbtStreamBuffer[nStreamIndex],str1,(size_t) 20); +diff -r -c -N encoder/VbrTag.h lame3.13/VbrTag.h +*** encoder/VbrTag.h Wed Dec 31 17:00:00 1969 +--- lame3.13/VbrTag.h Tue Jun 15 16:46:52 1999 +*************** +*** 0 **** +--- 1,86 ---- ++ /* ++ * Version numbering for LAME. ++ * ++ * Copyright (c) 1999 A.L. Faber ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License as published by ++ * the Free Software Foundation; either version 2, or (at your option) ++ * any later version. ++ * ++ * This program is distributed in the hope that it will be useful, ++ * but WITHOUT ANY WARRANTY; without even the implied warranty of ++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the ++ * GNU General Public License for more details. ++ * ++ * You should have received a copy of the GNU General Public License ++ * along with this program; see the file COPYING. If not, write to ++ * the Free Software Foundation, 675 Mass Ave, Cambridge, MA 02139, USA. ++ */ ++ ++ #ifndef VRBTAG_H_INCLUDED ++ #define VRBTAG_H_INCLUDED ++ ++ /* ----------------------------------------------------------- ++ * A Vbr header may be present in the ancillary ++ * data field of the first frame of an mp3 bitstream ++ * The Vbr header (optionally) contains ++ * frames total number of audio frames in the bitstream ++ * bytes total number of bytes in the bitstream ++ * toc table of contents ++ ++ * toc (table of contents) gives seek points ++ * for random access ++ * the ith entry determines the seek point for ++ * i-percent duration ++ * seek point in bytes = (toc[i]/256.0) * total_bitstream_bytes ++ * e.g. half duration seek point = (toc[50]/256.0) * total_bitstream_bytes ++ */ ++ ++ #include "l3bitstream.h" ++ ++ #define FRAMES_FLAG 0x0001 ++ #define BYTES_FLAG 0x0002 ++ #define TOC_FLAG 0x0004 ++ #define VBR_SCALE_FLAG 0x0008 ++ ++ #define NUMTOCENTRIES 100 ++ ++ #define FRAMES_AND_BYTES (FRAMES_FLAG | BYTES_FLAG) ++ #ifndef BYTE ++ typedef unsigned char BYTE; ++ #endif ++ ++ // structure to receive extracted header ++ // toc may be NULL ++ typedef struct ++ { ++ int h_id; // from MPEG header, 0=MPEG2, 1=MPEG1 ++ int samprate; // determined from MPEG header ++ int flags; // from Vbr header data ++ int frames; // total bit stream frames from Vbr header data ++ int bytes; // total bit stream bytes from Vbr header data ++ int vbr_scale; // encoded vbr scale from Vbr header data ++ BYTE toc[NUMTOCENTRIES]; // may be NULL if toc not desired ++ } VBRTAGDATA; ++ ++ // 4 bytes for Header Tag ++ // 4 bytes for Header Flags ++ // 100 bytes for entry (NUMTOCENTRIES) ++ // 4 bytes for FRAME SIZE ++ // 4 bytes for STREAM_SIZE ++ // 4 bytes for VBR SCALE. a VBR quality indicator: 0=best 100=worst ++ // 20 bytes for LAME tag. for example, "LAME3.12 (beta 6)" ++ // ___________ ++ // 140 bytes ++ ++ #define VBRHEADERSIZE (NUMTOCENTRIES+4+4+4+4+4) ++ ++ int GetVbrTag(VBRTAGDATA *pTagData, unsigned char *buf); ++ int SeekPoint(unsigned char TOC[NUMTOCENTRIES], int file_bytes, float percent); ++ int InitVbrTag(Bit_stream_struc* pBs,int nVersion,int nMode, int SampIndex); ++ int PutVbrTag(char* lpszFileName,int nVbrScale); ++ void AddVbrFrame(int nStreamPos); ++ ++ ++ #endif +diff -r -c -N encoder/auenc lame3.13/auenc +*** encoder/auenc Wed Dec 31 17:00:00 1969 +--- lame3.13/auenc Mon Jun 14 23:24:36 1999 +*************** +*** 0 **** +--- 1,39 ---- ++ #!/bin/sh ++ # ++ # auenc -- version 0.1 ++ # ++ # A wrapper for lame to encode multiple files. By default, a .wav ++ # extension is removed and replaced by .mp3 . ++ # ++ # (C) 1999 Gerhard Wesp under the GPL. ++ ++ # set the variables below according to your taste ++ LAME=/usr/bin/lame ++ LAME_OPTS="-S -h -v -V 0 -b 256" # high quality, silent operation ++ ++ if [ $# -lt 1 ] ; then ++ exec 1>&2 ++ cat << _EOF_ ++ usage: $0 [options] file... ++ options: ++ -d --delete: delete original file after successful encoding ++ _EOF_ ++ exit 1 ++ fi ++ ++ unset DELETE ++ case "$1" in ( -d | --delete ) ++ DELETE=1 ++ shift ;; ++ esac ++ ++ for f in $* ; do ++ if ! $LAME $LAME_OPTS "$f" `basename "$f" .wav`.mp3 ; then ++ exec 1>&2 ++ echo "encoding of $f failed, aborting..." ++ exit 1 ++ fi ++ if [ -n "$DELETE" ] ; then ++ rm -f "$f" ++ fi ++ done +diff -r -c -N encoder/common.c lame3.13/common.c +*** encoder/common.c Wed Jan 22 02:43:13 1997 +--- lame3.13/common.c Wed Jun 23 21:49:39 1999 +*************** +*** 13,94 **** + * + * Received from Mike Coleman + **********************************************************************/ +! /********************************************************************** +! * date programmers comment * +! * 2/25/91 Doulas Wong, start of version 1.0 records * +! * Davis Pan * +! * 5/10/91 W. Joseph Carter Created this file for all common * +! * functions and global variables. * +! * Ported to Macintosh and Unix. * +! * Added Jean-Georges Fritsch's * +! * "bitstream.c" package. * +! * Added routines to handle AIFF PCM * +! * sound files. * +! * Added "mem_alloc()" and "mem_free()" * +! * routines for memory allocation * +! * portability. * +! * Added routines to convert between * +! * Apple SANE extended floating point * +! * format and IEEE double precision * +! * floating point format. For AIFF. * +! * 02jul91 dpwe (Aware Inc) Moved allocation table input here; * +! * Tables read from subdir TABLES_PATH. * +! * Added some debug printout fns (Write*)* +! * 7/10/91 Earle Jennings replacement of the one float by FLOAT * +! * port to MsDos from MacIntosh version * +! * 8/ 5/91 Jean-Georges Fritsch fixed bug in open_bit_stream_r() * +! *10/ 1/91 S.I. Sudharsanan, Ported to IBM AIX platform. * +! * Don H. Lee, * +! * Peter W. Farrett * +! *10/3/91 Don H. Lee implemented CRC-16 error protection * +! * newly introduced functions are * +! * I_CRC_calc, II_CRC_calc and * +! * update_CRC. Additions and revisions * +! * are marked with dhl for clarity * +! *10/18/91 Jean-Georges Fritsch fixed bug in update_CRC(), * +! * II_CRC_calc() and I_CRC_calc() * +! * 2/11/92 W. Joseph Carter Ported new code to Macintosh. Most * +! * important fixes involved changing * +! * 16-bit ints to long or unsigned in * +! * bit alloc routines for quant of 65535 * +! * and passing proper function args. * +! * Removed "Other Joint Stereo" option * +! * and made bitrate be total channel * +! * bitrate, irrespective of the mode. * +! * Fixed many small bugs & reorganized. * +! * 3/20/92 Jean-Georges Fritsch fixed bug in start-of-frame search * +! * 6/15/92 Juan Pineda added refill_buffer(bs) "n" * +! * initialization * +! * 7/08/92 Susanne Ritscher MS-DOS, MSC6.0 port fixes * +! * 7/27/92 Mike Li (re-)Port to MS-DOS * +! * 8/19/92 Soren H. Nielsen Fixed bug in I_CRC_calc and in * +! * II_CRC_calc. Added function: new_ext * +! * for better MS-DOS compatability * +! * 3/10/93 Kevin Peterson changed aiff_read_headers to handle * +! * chunks in any order. now returns * +! * position of sound data in file. * +! * 3/31/93 Jens Spille changed IFF_* string compares to use * +! * strcmp() * +! * 5/30/93 Masahiro Iwadare removed the previous modification * +! * for UNIX. * +! * 8/27/93 Seymour Shlien, Fixes in Unix and MSDOS ports, * +! * Daniel Lauzon, and * +! * Bill Truerniet * +! *--------------------------------------------------------------------* +! * 8/24/93 Masahiro Iwadare Included IS modification in Layer III.* +! * Changed for 1 pass decoding. * +! * 9/07/93 Toshiyuki Ishino Integrated Layer III with Ver 3.9. * +! *--------------------------------------------------------------------* +! * 11/20/93 Masahiro Iwadare Integrated Layer III with Ver 4.0. * +! *--------------------------------------------------------------------* +! * 7/14/94 Juergen Koller rewind of bitbuffer added * +! * 6/12/95 Soeren H. Nielsen Bug fix in new_ext(). * +! * 7/11/95 Soeren H. Nielsen Changes for MPEG-2 LSF Layer I and II * +! *--------------------------------------------------------------------* +! * 8/02/95 mc@fivebats.com Added code to determine byte-order, * +! * fixes to AIFF routines. Modified Mac * +! * code to work with new Apple headers * +! **********************************************************************/ + + /*********************************************************************** + * +--- 13,19 ---- + * + * Received from Mike Coleman + **********************************************************************/ +! + + /*********************************************************************** + * +*************** +*** 99,105 **** +--- 24,32 ---- + #include "common.h" + #include /* 1995-07-11 shn */ + #include ++ #include + #include ++ #include + + /*********************************************************************** + * +*************** +*** 152,292 **** + * + ***********************************************************************/ + +- /* The system uses a variety of data files. By opening them via this +- function, we can accommodate various locations. */ +- +- FILE *OpenTableFile(name) +- char *name; +- { +- char fulname[80]; +- char *envdir; +- FILE *f; +- +- fulname[0] = '\0'; +- +- #ifdef TABLES_PATH +- strcpy(fulname, TABLES_PATH); /* default relative path for tables */ +- #endif /* TABLES_PATH */ /* (includes terminal path seperator */ +- +- #ifdef UNIX /* envir. variables for UNIX only */ +- { +- char *getenv(); +- +- envdir = getenv(MPEGTABENV); /* check for environment */ +- if(envdir != NULL) +- strcpy(fulname, envdir); +- strcat(fulname, PATH_SEPARATOR); /* add a "/" on the end */ +- } +- #endif /* UNIX */ +- +- strcat(fulname, name); +- if( (f=fopen(fulname,"r"))==NULL ) { +- fprintf(stderr,"OpenTable: could not find %s\n", fulname); +- +- #ifdef UNIX +- if(envdir != NULL) +- fprintf(stderr,"Check %s directory '%s'\n",MPEGTABENV, envdir); +- else +- fprintf(stderr,"Check local directory './%s' or setenv %s\n", +- TABLES_PATH, MPEGTABENV); +- #else /* not unix : no environment variables */ +- +- #ifdef TABLES_PATH +- fprintf(stderr,"Check local directory './%s'\n",TABLES_PATH); +- #endif /* TABLES_PATH */ +- +- #endif /* UNIX */ +- +- } +- return f; +- } + + /*********************************************************************** +! * +! * Read one of the data files ("alloc_*") specifying the bit allocation/ +! * quatization parameters for each subband in layer II encoding +! * +! **********************************************************************/ +! +! int read_bit_alloc(table, alloc) /* read in table, return # subbands */ +! int table; +! al_table *alloc; +! { +! unsigned int a, b, c, d, i, j; +! FILE *fp; +! char name[16], t[80]; +! int sblim; +! +! strcpy(name, "alloc_0"); +! +! switch (table) { +! case 0 : name[6] = '0'; break; +! case 1 : name[6] = '1'; break; +! case 2 : name[6] = '2'; break; +! case 3 : name[6] = '3'; break; +! case 4 : name[6] = '4'; break; /* LSF, added 1995-07-11 shn */ +! default : name[6] = '0'; +! } +! +! if (!(fp = OpenTableFile(name))) { +! printf("Please check bit allocation table %s\n", name); +! exit(1); +! } +! +! printf("using bit allocation table %s\n", name); +! +! fgets(t, 80, fp); +! sscanf(t, "%d\n", &sblim); +! while (!feof(fp)) { +! fgets(t, 80, fp); +! sscanf(t, "%d %d %d %d %d %d\n", &i, &j, &a, &b, &c, &d); +! (*alloc)[i][j].steps = a; +! (*alloc)[i][j].bits = b; +! (*alloc)[i][j].group = c; +! (*alloc)[i][j].quant = d; +! } +! fclose(fp); +! return sblim; + } + +- /*********************************************************************** +- * +- * Using the decoded info the appropriate possible quantization per +- * subband table is loaded +- * +- **********************************************************************/ + +- int pick_table(fr_ps) /* choose table, load if necess, return # sb's */ +- frame_params *fr_ps; +- { +- int table, lay, ws, bsp, br_per_ch, sfrq; +- int sblim = fr_ps->sblimit; /* return current value if no load */ + +- lay = fr_ps->header->lay - 1; +- bsp = fr_ps->header->bitrate_index; +- br_per_ch = bitrate[fr_ps->header->version][lay][bsp] / fr_ps->stereo; +- ws = fr_ps->header->sampling_frequency; +- sfrq = s_freq[fr_ps->header->version][ws]; +- /* decision rules refer to per-channel bitrates (kbits/sec/chan) */ +- if (fr_ps->header->version == MPEG_AUDIO_ID) { /* MPEG-1 */ +- if ((sfrq == 48 && br_per_ch >= 56) || +- (br_per_ch >= 56 && br_per_ch <= 80)) table = 0; +- else if (sfrq != 48 && br_per_ch >= 96) table = 1; +- else if (sfrq != 32 && br_per_ch <= 48) table = 2; +- else table = 3; +- } +- else { /* MPEG-2 LSF */ +- table = 4; +- } +- if (fr_ps->tab_num != table) { +- if (fr_ps->tab_num >= 0) +- mem_free((void **)&(fr_ps->alloc)); +- fr_ps->alloc = (al_table FAR *) mem_alloc(sizeof(al_table), +- "alloc"); +- sblim = read_bit_alloc(fr_ps->tab_num = table, fr_ps->alloc); +- } +- return sblim; +- } + + int js_bound(lay, m_ext) + int lay, m_ext; +--- 79,128 ---- + * + ***********************************************************************/ + + + /*********************************************************************** +! * compute bitsperframe and mean_bits for a layer III frame +! **********************************************************************/ +! void getframebits(layer *info, int stereo, int *bitsPerFrame, int *mean_bits) { +! int whole_SpF; +! double bit_rate,samp; +! int samplesPerFrame,bitsPerSlot; +! int sideinfo_len; +! int mode_gr; +! +! samp = s_freq[info->version][info->sampling_frequency]; +! bit_rate = bitrate[info->version][info->lay-1][info->bitrate_index]; +! samplesPerFrame = info->version == 1 ? 1152 : 576; +! bitsPerSlot = 8; +! mode_gr = (info->version == 1) ? 2 : 1; /* mode_gr = 2 */ +! +! /* determine the mean bitrate for main data */ +! sideinfo_len = 32; +! if ( info->version == 1 ) +! { /* MPEG 1 */ +! if ( stereo == 1 ) +! sideinfo_len += 136; +! else +! sideinfo_len += 256; +! } +! else +! { /* MPEG 2 */ +! if ( stereo == 1 ) +! sideinfo_len += 72; +! else +! sideinfo_len += 136; +! } +! +! if (info->error_protection) sideinfo_len += 16; +! +! +! whole_SpF = (samplesPerFrame /samp)*(bit_rate / (double)bitsPerSlot); +! *bitsPerFrame = 8 * whole_SpF + (info->padding * 8); +! *mean_bits = (*bitsPerFrame - sideinfo_len) / mode_gr; + } + + + + + int js_bound(lay, m_ext) + int lay, m_ext; +*************** +*** 308,314 **** + + fr_ps->actual_mode = hdr->mode; + fr_ps->stereo = (hdr->mode == MPG_MD_MONO) ? 1 : 2; +! if (hdr->lay == 2) fr_ps->sblimit = pick_table(fr_ps); + else fr_ps->sblimit = SBLIMIT; + if(hdr->mode == MPG_MD_JOINT_STEREO) + fr_ps->jsbound = js_bound(hdr->lay, hdr->mode_ext); +--- 144,152 ---- + + fr_ps->actual_mode = hdr->mode; + fr_ps->stereo = (hdr->mode == MPG_MD_MONO) ? 1 : 2; +! if (hdr->lay == 2){ +! printf("Only Layer 3 supported\n"); +! exit(-1);} + else fr_ps->sblimit = SBLIMIT; + if(hdr->mode == MPG_MD_JOINT_STEREO) + fr_ps->jsbound = js_bound(hdr->lay, hdr->mode_ext); +*************** +*** 317,322 **** +--- 155,161 ---- + /* alloc, tab_num set in pick_table */ + } + ++ + void WriteHdr(fr_ps, s) + frame_params *fr_ps; + FILE *s; +*************** +*** 338,432 **** + fflush(s); + } + +- void WriteBitAlloc(bit_alloc, f_p, s) +- unsigned int bit_alloc[2][SBLIMIT]; +- frame_params *f_p; +- FILE *s; +- { +- int i,j; +- int st = f_p->stereo; +- int sbl = f_p->sblimit; +- int jsb = f_p->jsbound; +- +- fprintf(s, "BITA "); +- for(i=0; istereo; +- int sblimit = fr_ps->sblimit; +- int lay = fr_ps->header->lay; +- int i,j,k; +- +- if(lay == 2) { +- fprintf(s, "SFSI "); +- for (i=0;istereo; +- int sblimit = fr_ps->sblimit; +- +- fprintf(s, "SMPL "); +- for (i=0;isampleSize / 8) + (aiff_ptr->sampleSize % 8 ? 1 : 0); +! +! if ( fseek(file_ptr, 0L, SEEK_SET) != 0 ) +! return -1; +! +! /* write FORM chunk */ +! chunkSize = 8 + 18 + 8 + aiff_ptr->numChannels * aiff_ptr->numSampleFrames * sampleBytes; +! Write32BitsHighLow( file_ptr, IFF_ID_FORM ); +! Write32BitsHighLow( file_ptr, chunkSize ); +! Write32BitsHighLow( file_ptr, IFF_ID_AIFF ); +! /* write COMM chunk */ +! Write32BitsHighLow( file_ptr, IFF_ID_COMM ); +! Write32BitsHighLow( file_ptr, 18 ); /* chunk size */ +! Write16BitsHighLow( file_ptr, aiff_ptr->numChannels ); +! Write32BitsHighLow( file_ptr, aiff_ptr->numSampleFrames ); +! Write16BitsHighLow( file_ptr, aiff_ptr->sampleSize ); +! WriteIeeeExtendedHighLow( file_ptr, aiff_ptr->sampleRate ); +! /* write SSND chunk header */ +! chunkSize = 8 + aiff_ptr->numChannels * aiff_ptr->numSampleFrames * sampleBytes; +! Write32BitsHighLow( file_ptr, IFF_ID_SSND ); +! Write32BitsHighLow( file_ptr, chunkSize ); +! Write32BitsHighLow( file_ptr, 0 ); /* offset */ +! Write32BitsHighLow( file_ptr, 0 ); /* block size */ +! return 0; +! } + + /***************************************************************************** + * +--- 393,428 ---- + return sound_position; + } + +! #include "wavheader.h" +! #define ENDSWAP_SHORT(x) ((((x)>>8)&0xFF)|(((x)&0xFF)<<8)) +! #define ENDSWAP_INT(x) ((((x)>>24)&0xFF)|(((x)>>8)&0xFF00)|(((x)&0xFF00)<<8)|(((x)&0xFF)<<24)) +! +! void endswap_wave_fmt (WaveHeader *wavfmt) { +! wavfmt->main_chunk = ENDSWAP_INT (wavfmt->main_chunk); +! wavfmt->length = ENDSWAP_INT (wavfmt->length); +! wavfmt->chunk_type = ENDSWAP_INT (wavfmt->chunk_type); +! +! wavfmt->sub_chunk = ENDSWAP_INT (wavfmt->sub_chunk); +! wavfmt->sc_len = ENDSWAP_INT (wavfmt->sc_len); +! wavfmt->format = ENDSWAP_SHORT (wavfmt->format); +! wavfmt->modus = ENDSWAP_SHORT (wavfmt->modus); +! wavfmt->sample_fq = ENDSWAP_INT (wavfmt->sample_fq); +! wavfmt->byte_p_sec = ENDSWAP_INT (wavfmt->byte_p_sec); +! wavfmt->byte_p_spl = ENDSWAP_SHORT (wavfmt->byte_p_spl); +! wavfmt->bit_p_spl = ENDSWAP_SHORT (wavfmt->bit_p_spl); + +! wavfmt->data_chunk = ENDSWAP_INT (wavfmt->data_chunk); +! wavfmt->data_length = ENDSWAP_INT (wavfmt->data_length); + } + + +! +! +! #endif +! +! +! +! + + /***************************************************************************** + * +*************** +*** 742,756 **** + /*desalloc_buffer(); empty and close the buffer */ + /*back_track_buffer(); goes back N bits in the buffer */ + /*unsigned int get1bit(); read 1 bit from the bit stream */ +- /*unsigned long getbits(); read N bits from the bit stream */ +- /*unsigned long byte_ali_getbits(); read the next byte aligned N bits from*/ +- /* the bit stream */ + /*unsigned long look_ahead(); grep the next N bits in the bit stream without*/ + /* changing the buffer pointer */ + /*put1bit(); write 1 bit from the bit stream */ + /*put1bit(); write 1 bit from the bit stream */ + /*putbits(); write N bits from the bit stream */ +- /*byte_ali_putbits(); write byte aligned the next N bits into the bit stream*/ + /*unsigned long sstell(); return the current bit stream length (in bits) */ + /*int end_bs(); return 1 if the end of bit stream reached otherwise 0 */ + /*int seek_sync(); return 1 if a sync word was found in the bit stream */ +--- 447,457 ---- +*************** +*** 765,770 **** +--- 466,472 ---- + register int index=0; + char val[2]; + ++ #ifndef _BLADEDLL + while ((i>=0) && (!bs->eob)) { + + if (bs->format == BINARY) +*************** +*** 808,855 **** + } + + } + return 0; + } + +- static char *he = "0123456789ABCDEF"; + + /* empty the buffer to the output device when the buffer becomes full */ + void empty_buffer(bs, minimum) + Bit_stream_struc *bs; /* bit stream structure */ + int minimum; /* end of the buffer to empty */ + { +! register int i; + + #if BS_FORMAT == BINARY +! for (i=bs->buf_size-1;i>=minimum;i--) +! fwrite(&bs->buf[i], sizeof(unsigned char), 1, bs->pt); + #else + for (i=bs->buf_size-1;i>=minimum;i--) { + char val[2]; + val[0] = he[((bs->buf[i] >> 4) & 0x0F)]; + val[1] = he[(bs->buf[i] & 0x0F)]; + fwrite(val, sizeof(char), 2, bs->pt); +! } + #endif +- fflush(bs->pt); /* NEW SS to assist in debugging*/ + +! for (i=minimum-1; i>=0; i--) + bs->buf[bs->buf_size - minimum + i] = bs->buf[i]; + +! bs->buf_byte_idx = bs->buf_size -1 - minimum; + bs->buf_bit_idx = 8; + } + + /* open the device to write the bit stream into it */ +! void open_bit_stream_w(bs, bs_filenam, size) +! Bit_stream_struc *bs; /* bit stream structure */ +! char *bs_filenam; /* name of the bit stream file */ +! int size; /* size of the buffer */ +! { +! if ((bs->pt = fopen(bs_filenam, "wb")) == NULL) { + printf("Could not create \"%s\".\n", bs_filenam); + exit(1); +! } + alloc_buffer(bs, size); + bs->buf_byte_idx = size-1; + bs->buf_bit_idx=8; +--- 510,621 ---- + } + + } ++ #endif + return 0; + } + + + /* empty the buffer to the output device when the buffer becomes full */ ++ /* mt 3/99: modified so we can call empty_buffer even if buffer is not full */ ++ /* add error checking -- 1999/06 Alvaro Martinez Echevarria */ ++ /* someone gimme a good reason for writing bytes one by ++ * one; changed -- 1999/06 Alvaro Martinez Echevarria */ + void empty_buffer(bs, minimum) + Bit_stream_struc *bs; /* bit stream structure */ + int minimum; /* end of the buffer to empty */ + { +! register int i,j; +! unsigned char *tmpbuf; + ++ #ifndef DISTRIB + #if BS_FORMAT == BINARY +! +! #ifndef _BLADEDLL +! tmpbuf=(unsigned char *)malloc(bs->buf_size-minimum); +! if (tmpbuf==NULL) { +! //fprintf(stderr,"error in malloc(): %s\n",sys_errlist[errno]); +! perror("error in malloc()\n"); +! exit(1); +! } +! for (i=bs->buf_size-1,j=0;i>=minimum;i--,j++) { +! tmpbuf[j]=bs->buf[i]; +! } +! fwrite(tmpbuf,1,bs->buf_size-minimum,bs->pt); +! if (ferror(bs->pt)) { +! fprintf(stderr,"error en fwrite()\n"); +! exit(1); +! } +! free(tmpbuf); +! #else +! for (i=bs->buf_size-1;i>=minimum;i--) +! { +! if (bs->nOutBufPospbtOutBuf[bs->nOutBufPos++]=bs->buf[i]; +! } +! else +! { +! assert(0); +! } +! } +! #endif +! + #else ++ static char *he = "0123456789ABCDEF"; ++ + for (i=bs->buf_size-1;i>=minimum;i--) { + char val[2]; + val[0] = he[((bs->buf[i] >> 4) & 0x0F)]; + val[1] = he[(bs->buf[i] & 0x0F)]; + fwrite(val, sizeof(char), 2, bs->pt); +! if (ferror(bs->pt)) { +! fprintf(stderr,"error in fwrite()\n"); +! exit(1); +! } +! } +! #endif // BS_FORMAT==BINARY +! +! #ifndef _BLADEDLL +! // NEW SS to assist in debugging +! // Only file for files, not for the output buffer +! if (fflush(bs->pt)!=0) { +! //fprintf(stderr,"error in fflush(): %s\n",sys_errlist[errno]); +! perror("error in fflush(): \n"); +! exit(1); +! } +! #endif + #endif + +! /* for (i=minimum-1; i>=0; i--) */ +! for (i=minimum-1; i>=bs->buf_byte_idx+1; i--) + bs->buf[bs->buf_size - minimum + i] = bs->buf[i]; + +! /* bs->buf_byte_idx = bs->buf_size -1 - minimum; */ +! bs->buf_byte_idx = bs->buf_size -1 - minimum + bs->buf_byte_idx + 1; + bs->buf_bit_idx = 8; + } + + /* open the device to write the bit stream into it */ +! void open_bit_stream_w( Bit_stream_struc* bs, /* bit stream structure */ +! char* bs_filenam, /* name of the bit stream file */ +! int size /* size of the buffer */ +! ) +! { +! #ifdef _BLADEDLL +! // Initialize output buffer +! bs->pbtOutBuf=NULL; +! bs->nOutBufPos=0; +! #else +! if (!strcmp(bs_filenam, "-")) { +! /* Write to standard output. */ +! bs->pt = stdout; +! } else { +! if ((bs->pt = fopen(bs_filenam, "wb")) == NULL) { + printf("Could not create \"%s\".\n", bs_filenam); + exit(1); +! } +! } +! #endif + alloc_buffer(bs, size); + bs->buf_byte_idx = size-1; + bs->buf_bit_idx=8; +*************** +*** 859,949 **** + bs->eobs = FALSE; + } + +- /* open the device to read the bit stream from it */ +- void open_bit_stream_r(bs, bs_filenam, size) +- Bit_stream_struc *bs; /* bit stream structure */ +- char *bs_filenam; /* name of the bit stream file */ +- int size; /* size of the buffer */ +- { +- register unsigned long n; +- register unsigned char flag = 1; +- unsigned char val; +- +- if ((bs->pt = fopen(bs_filenam, "rb")) == NULL) { +- printf("Could not find \"%s\".\n", bs_filenam); +- exit(1); +- } +- +- do { +- n = fread(&val, sizeof(unsigned char), 1, bs->pt); +- switch (val) { +- case 0x30: +- case 0x31: +- case 0x32: +- case 0x33: +- case 0x34: +- case 0x35: +- case 0x36: +- case 0x37: +- case 0x38: +- case 0x39: +- case 0x41: +- case 0x42: +- case 0x43: +- case 0x44: +- case 0x45: +- case 0x46: +- case 0xa: /* \n */ +- case 0xd: /* cr */ +- case 0x1a: /* sub */ +- break; +- +- default: /* detection of an binary character */ +- flag--; +- break; +- } +- +- } while (flag & n); +- +- if (flag) { +- printf ("the bit stream file %s is an ASCII file\n", bs_filenam); +- bs->format = ASCII; +- } +- else { +- bs->format = BINARY; +- printf ("the bit stream file %s is a BINARY file\n", bs_filenam); +- } +- +- fclose(bs->pt); +- +- if ((bs->pt = fopen(bs_filenam, "rb")) == NULL) { +- printf("Could not find \"%s\".\n", bs_filenam); +- exit(1); +- } +- +- alloc_buffer(bs, size); +- bs->buf_byte_idx=0; +- bs->buf_bit_idx=0; +- bs->totbit=0; +- bs->mode = READ_MODE; +- bs->eob = FALSE; +- bs->eobs = FALSE; +- } +- +- /*close the device containing the bit stream after a read process*/ +- void close_bit_stream_r(bs) +- Bit_stream_struc *bs; /* bit stream structure */ +- { +- fclose(bs->pt); +- desalloc_buffer(bs); +- } + + /*close the device containing the bit stream after a write process*/ + void close_bit_stream_w(bs) + Bit_stream_struc *bs; /* bit stream structure */ + { + empty_buffer(bs, bs->buf_byte_idx); + fclose(bs->pt); + desalloc_buffer(bs); + } + +--- 625,644 ---- + bs->eobs = FALSE; + } + + + /*close the device containing the bit stream after a write process*/ + void close_bit_stream_w(bs) + Bit_stream_struc *bs; /* bit stream structure */ + { ++ // Flush last bytes + empty_buffer(bs, bs->buf_byte_idx); ++ ++ #ifndef _BLADEDLL ++ // Close the file + fclose(bs->pt); ++ #endif // _BLADEDLL ++ ++ // Deallocate all buffers + desalloc_buffer(bs); + } + +*************** +*** 966,994 **** + + int putmask[9]={0x0, 0x1, 0x3, 0x7, 0xf, 0x1f, 0x3f, 0x7f, 0xff}; + int clearmask[9]={0xff, 0xfe, 0xfc, 0xf8, 0xf0, 0xe0, 0xc0, 0x80, 0x0}; +- +- void back_track_buffer(bs, N) /* goes back N bits in the buffer */ +- Bit_stream_struc *bs; /* bit stream structure */ +- int N; +- { +- int tmp = N - (N/8)*8; +- register int i; +- +- bs->totbit -= N; +- for (i=bs->buf_byte_idx;i< bs->buf_byte_idx+N/8-1;i++) bs->buf[i] = 0; +- bs->buf_byte_idx += N/8; +- if ( (tmp + bs->buf_bit_idx) <= 8) { +- bs->buf_bit_idx += tmp; +- } +- else { +- bs->buf_byte_idx ++; +- bs->buf_bit_idx += (tmp - 8); +- } +- bs->buf[bs->buf_byte_idx] &= clearmask[bs->buf_bit_idx]; +- } +- + int mask[8]={0x1, 0x2, 0x4, 0x8, 0x10, 0x20, 0x40, 0x80}; + + /*read 1 bit from the bit stream */ + unsigned int get1bit(bs) + Bit_stream_struc *bs; /* bit stream structure */ +--- 661,669 ---- + + int putmask[9]={0x0, 0x1, 0x3, 0x7, 0xf, 0x1f, 0x3f, 0x7f, 0xff}; + int clearmask[9]={0xff, 0xfe, 0xfc, 0xf8, 0xf0, 0xe0, 0xc0, 0x80, 0x0}; + int mask[8]={0x1, 0x2, 0x4, 0x8, 0x10, 0x20, 0x40, 0x80}; + ++ #if 0 + /*read 1 bit from the bit stream */ + unsigned int get1bit(bs) + Bit_stream_struc *bs; /* bit stream structure */ +*************** +*** 1017,1022 **** +--- 692,698 ---- + bs->buf_bit_idx--; + return(bit); + } ++ #endif + + /*write 1 bit from the bit stream */ + void put1bit(bs, bit) +*************** +*** 1036,1108 **** + } + } + +- /*look ahead for the next N bits from the bit stream */ +- unsigned long look_ahead(bs, N) +- Bit_stream_struc *bs; /* bit stream structure */ +- int N; /* number of bits to read from the bit stream */ +- { +- unsigned long val=0; +- register int j = N; +- register int k, tmp; +- register int bit_idx = bs->buf_bit_idx; +- register int byte_idx = bs->buf_byte_idx; +- +- if (N > MAX_LENGTH) +- printf("Cannot read or write more than %d bits at a time.\n", MAX_LENGTH); +- +- while (j > 0) { +- if (!bit_idx) { +- bit_idx = 8; +- byte_idx--; +- } +- k = MIN (j, bit_idx); +- tmp = bs->buf[byte_idx]&putmask[bit_idx]; +- tmp = tmp >> (bit_idx-k); +- val |= tmp << (j-k); +- bit_idx -= k; +- j -= k; +- } +- return(val); +- } +- +- /*read N bit from the bit stream */ +- unsigned long getbits(bs, N) +- Bit_stream_struc *bs; /* bit stream structure */ +- int N; /* number of bits to read from the bit stream */ +- { +- unsigned long val=0; +- register int i; +- register int j = N; +- register int k, tmp; +- +- if (N > MAX_LENGTH) +- printf("Cannot read or write more than %d bits at a time.\n", MAX_LENGTH); +- +- bs->totbit += N; +- while (j > 0) { +- if (!bs->buf_bit_idx) { +- bs->buf_bit_idx = 8; +- bs->buf_byte_idx--; +- if ((bs->buf_byte_idx < MINIMUM) || (bs->buf_byte_idx < bs->eob)) { +- if (bs->eob) +- bs->eobs = TRUE; +- else { +- for (i=bs->buf_byte_idx; i>=0;i--) +- bs->buf[bs->buf_size-1-bs->buf_byte_idx+i] = bs->buf[i]; +- refill_buffer(bs); +- bs->buf_byte_idx = bs->buf_size-1; +- } +- } +- } +- k = MIN (j, bs->buf_bit_idx); +- tmp = bs->buf[bs->buf_byte_idx]&putmask[bs->buf_bit_idx]; +- tmp = tmp >> (bs->buf_bit_idx-k); +- val |= tmp << (j-k); +- bs->buf_bit_idx -= k; +- j -= k; +- } +- return(val); +- } + + /*write N bits into the bit stream */ + void putbits(bs, val, N) +--- 712,717 ---- +*************** +*** 1118,1124 **** + + bs->totbit += N; + while (j > 0) { +! k = MIN(j, bs->buf_bit_idx); + tmp = val >> (j-k); + bs->buf[bs->buf_byte_idx] |= (tmp&putmask[k]) << (bs->buf_bit_idx-k); + bs->buf_bit_idx -= k; +--- 727,733 ---- + + bs->totbit += N; + while (j > 0) { +! k = Min(j, bs->buf_bit_idx); + tmp = val >> (j-k); + bs->buf[bs->buf_byte_idx] |= (tmp&putmask[k]) << (bs->buf_bit_idx-k); + bs->buf_bit_idx -= k; +*************** +*** 1133,1170 **** + } + } + +- /*write N bits byte aligned into the bit stream */ +- void byte_ali_putbits(bs, val, N) +- Bit_stream_struc *bs; /* bit stream structure */ +- unsigned int val; /* val to write into the buffer */ +- int N; /* number of bits of val */ +- { +- unsigned long aligning, sstell(); +- +- if (N > MAX_LENGTH) +- printf("Cannot read or write more than %d bits at a time.\n", MAX_LENGTH); +- aligning = sstell(bs)%8; +- if (aligning) +- putbits(bs, (unsigned int)0, (int)(8-aligning)); +- +- putbits(bs, val, N); +- } +- +- /*read the next bute aligned N bits from the bit stream */ +- unsigned long byte_ali_getbits(bs, N) +- Bit_stream_struc *bs; /* bit stream structure */ +- int N; /* number of bits of val */ +- { +- unsigned long aligning, sstell(); +- +- if (N > MAX_LENGTH) +- printf("Cannot read or write more than %d bits at a time.\n", MAX_LENGTH); +- aligning = sstell(bs)%8; +- if (aligning) +- getbits(bs, (int)(8-aligning)); +- +- return(getbits(bs, N)); +- } + + /*return the current bit stream length (in bits)*/ + unsigned long sstell(bs) +--- 742,747 ---- +*************** +*** 1182,1439 **** + return(bs->eobs); + } + +- /*this function seeks for a byte aligned sync word in the bit stream and +- places the bit stream pointer right after the sync. +- This function returns 1 if the sync was found otherwise it returns 0 */ +- int seek_sync(bs, sync, N) +- Bit_stream_struc *bs; /* bit stream structure */ +- long sync; /* sync word maximum 32 bits */ +- int N; /* sync word length */ +- { +- #if defined(MACINTOSH) && !defined(__powerc) +- double pow(); +- #endif +- unsigned long aligning, stell(); +- unsigned long val; +- long maxi = (int)pow(2.0, (FLOAT)N) - 1; +- +- aligning = sstell(bs)%ALIGNING; +- if (aligning) +- getbits(bs, (int)(ALIGNING-aligning)); +- +- val = getbits(bs, N); +- while (((val&maxi) != sync) && (!end_bs(bs))) { +- val <<= ALIGNING; +- val |= getbits(bs, ALIGNING); +- } +- +- if (end_bs(bs)) return(0); +- else return(1); +- } + /***************************************************************************** + * + * End of bit_stream.c package + * + *****************************************************************************/ + +- /***************************************************************************** +- * +- * CRC error protection package +- * +- *****************************************************************************/ +- +- void I_CRC_calc(fr_ps, bit_alloc, crc) +- frame_params *fr_ps; +- unsigned int bit_alloc[2][SBLIMIT]; +- unsigned int *crc; +- { +- int i, k; +- layer *info = fr_ps->header; +- int stereo = fr_ps->stereo; +- int jsbound = fr_ps->jsbound; +- +- *crc = 0xffff; /* changed from '0' 92-08-11 shn */ +- update_CRC(info->bitrate_index, 4, crc); +- update_CRC(info->sampling_frequency, 2, crc); +- update_CRC(info->padding, 1, crc); +- update_CRC(info->extension, 1, crc); +- update_CRC(info->mode, 2, crc); +- update_CRC(info->mode_ext, 2, crc); +- update_CRC(info->copyright, 1, crc); +- update_CRC(info->original, 1, crc); +- update_CRC(info->emphasis, 2, crc); +- +- for (i=0;iheader; +- int stereo = fr_ps->stereo; +- int sblimit = fr_ps->sblimit; +- int jsbound = fr_ps->jsbound; +- al_table *alloc = fr_ps->alloc; +- +- *crc = 0xffff; /* changed from '0' 92-08-11 shn */ +- update_CRC(info->bitrate_index, 4, crc); +- update_CRC(info->sampling_frequency, 2, crc); +- update_CRC(info->padding, 1, crc); +- update_CRC(info->extension, 1, crc); +- update_CRC(info->mode, 2, crc); +- update_CRC(info->mode_ext, 2, crc); +- update_CRC(info->copyright, 1, crc); +- update_CRC(info->original, 1, crc); +- update_CRC(info->emphasis, 2, crc); +- +- for (i=0;i>= 1)){ +- carry = *crc & 0x8000; +- *crc <<= 1; +- if (!carry ^ !(data & masking)) +- *crc ^= CRC16_POLYNOMIAL; +- } +- *crc &= 0xffff; +- } +- +- /***************************************************************************** +- * +- * End of CRC error protection package +- * +- *****************************************************************************/ +- +- #ifdef MACINTOSH +- /***************************************************************************** +- * +- * Set Macintosh file attributes. +- * +- *****************************************************************************/ +- void set_mac_file_attr( char fileName[MAX_NAME_SIZE], short vRefNum, OSType creator, OSType fileType) +- { +- char pascal_fileName[MAX_NAME_SIZE]; +- FInfo fndrInfo; +- OSErr anErr; +- long dirID = 0L; +- +- CtoPstr(strcpy(pascal_fileName, fileName)); +- anErr = HGetFInfo(vRefNum, dirID, (ConstStr255Param) pascal_fileName, &fndrInfo); +- if ( anErr != noErr ) +- return; +- fndrInfo.fdCreator = creator; +- fndrInfo.fdType = fileType; +- anErr = HSetFInfo(vRefNum, dirID, (ConstStr255Param) pascal_fileName, &fndrInfo); +- } +- +- #endif +- +- +- #ifdef MS_DOS +- /* ------------------------------------------------------------------------ +- new_ext() +- Puts a new extension name on a file name . +- Removes the last extension name, if any. +- 1992-08-19, 1995-06-12 shn +- ------------------------------------------------------------------------ */ +- void +- new_ext(char *filename, char *extname, char *newname) +- { +- int found, dotpos; +- +- /* First, strip the extension */ +- dotpos=strlen(filename); found=0; +- do +- { +- switch (filename[dotpos]) +- { +- case '.' : found=1; break; +- case '\\': +- case '/' : +- case ':' : found=-1; break; +- default : dotpos--; if (dotpos<0) found=-1; break; +- } +- } while (found==0); +- if (found==-1) strcpy(newname,filename); +- if (found== 1) { strncpy(newname,filename,dotpos); newname[dotpos]='\0'; } +- strcat(newname,extname); +- } +- #endif +- +- +- +- #define BUFSIZE 4096 +- static unsigned long offset,totbit=0, buf_byte_idx=0; +- static unsigned int buf[BUFSIZE]; +- static unsigned int buf_bit_idx=8; +- +- /*return the current bit stream length (in bits)*/ +- unsigned long hsstell() +- { +- return(totbit); +- } +- +- /* int putmask[9]={0x0, 0x1, 0x3, 0x7, 0xf, 0x1f, 0x3f, 0x7f, 0xff}; */ +- extern int putmask[9]; +- +- /*read N bit from the bit stream */ +- unsigned long hgetbits(N) +- int N; /* number of bits to read from the bit stream */ +- { +- unsigned long val=0; +- register int j = N; +- register int k, tmp; +- +- /* +- if (N > MAX_LENGTH) +- printf("Cannot read or write more than %d bits at a time.\n", MAX_LENGTH); +- */ +- totbit += N; +- while (j > 0) { +- if (!buf_bit_idx) { +- buf_bit_idx = 8; +- buf_byte_idx++; +- if (buf_byte_idx > offset) +- { printf("Buffer overflow !!\n");exit(3); } +- } +- k = MIN (j, buf_bit_idx); +- tmp = buf[buf_byte_idx%BUFSIZE]&putmask[buf_bit_idx]; +- tmp = tmp >> (buf_bit_idx-k); +- val |= tmp << (j-k); +- buf_bit_idx -= k; +- j -= k; +- } +- return(val); +- } +- +- unsigned int hget1bit() +- { +- return(hgetbits(1)); +- } +- +- /*write N bits into the bit stream */ +- void hputbuf(val, N) +- unsigned int val; /* val to write into the buffer */ +- int N; /* number of bits of val */ +- { +- if (N != 8) { printf("Not Supported yet!!\n"); exit(-3); } +- buf[offset % BUFSIZE] = val; +- offset++; +- } +- +- void rewindNbits( N ) +- int N; +- { +- totbit -= N; +- buf_bit_idx += N; +- while( buf_bit_idx >= 8 ) +- { buf_bit_idx -= 8; +- buf_byte_idx--; +- } +- } +- +- void rewindNbytes( N ) +- int N; +- { +- totbit -= N*8; +- buf_byte_idx -= N; +- } +--- 759,767 ---- +diff -r -c -N encoder/common.h lame3.13/common.h +*** encoder/common.h Wed Jan 22 02:43:13 1997 +--- lame3.13/common.h Fri Jun 18 00:44:37 1999 +*************** +*** 10,71 **** + * + * Received from Mike Coleman + **********************************************************************/ +! /********************************************************************** +! * date programmers comment * +! * 2/25/91 Doulas Wong, start of version 1.0 records * +! * Davis Pan * +! * 5/10/91 W. Joseph Carter Reorganized & renamed all ".h" files * +! * into "common.h" and "encoder.h". * +! * Ported to Macintosh and Unix. * +! * Added additional type definitions for * +! * AIFF, double/SANE and "bitstream.c". * +! * Added function prototypes for more * +! * rigorous type checking. * +! * 27jun91 dpwe (Aware) Added "alloc_*" defs & prototypes * +! * Defined new struct 'frame_params'. * +! * Changed info.stereo to info.mode_ext * +! * #define constants for mode types * +! * Prototype arguments if PROTO_ARGS * +! * 5/28/91 Earle Jennings added MS_DOS definition * +! * MsDos function prototype declarations * +! * 7/10/91 Earle Jennings added FLOAT definition as double * +! *10/ 3/91 Don H. Lee implemented CRC-16 error protection * +! * 2/11/92 W. Joseph Carter Ported new code to Macintosh. Most * +! * important fixes involved changing * +! * 16-bit ints to long or unsigned in * +! * bit alloc routines for quant of 65535 * +! * and passing proper function args. * +! * Removed "Other Joint Stereo" option * +! * and made bitrate be total channel * +! * bitrate, irrespective of the mode. * +! * Fixed many small bugs & reorganized. * +! * Modified some function prototypes. * +! * Changed BUFFER_SIZE back to 4096. * +! * 7/27/92 Michael Li (re-)Ported to MS-DOS * +! * 7/27/92 Masahiro Iwadare Ported to Convex * +! * 8/07/92 mc@tv.tek.com * +! * 8/10/92 Amit Gulati Ported to the AIX Platform (RS6000) * +! * AIFF string constants redefined * +! * 8/27/93 Seymour Shlien, Fixes in Unix and MSDOS ports, * +! * Daniel Lauzon, and * +! * Bill Truerniet * +! *--------------------------------------------------------------------* +! * 4/23/92 J. Pineda Added code for Layer III. * +! * 11/9/92 Amit Gulati Added defines for layerIII stereo * +! * modes. * +! * 8/24/93 Masahiro Iwadare Included IS modification in Layer III.* +! * Changed for 1 pass decoding. * +! * 9/07/93 Toshiyuki Ishino Integrated Layer III with Ver 3.9. * +! *--------------------------------------------------------------------* +! * 11/20/93 Masahiro Iwadare Integrated Layer III with Ver 4.0. * +! *--------------------------------------------------------------------* +! * 7/14/94 Juergen Koller Fix for HPUX an IRIX in AIFF-Strings * +! * 6/12/95 Soeren H. Nielsen Bug fix in new_ext(). * +! * 7/11/95 Soeren H. Nielsen Added defs. for MPEG-2 LSF * +! *--------------------------------------------------------------------* +! * 8/02/95 mc@fivebats.com Changes to AIFF stuff for portability * +! * to little-endian machines * +! **********************************************************************/ + #ifndef COMMON_DOT_H + #define COMMON_DOT_H + /*********************************************************************** +--- 10,16 ---- + * + * Received from Mike Coleman + **********************************************************************/ +! + #ifndef COMMON_DOT_H + #define COMMON_DOT_H + /*********************************************************************** +*************** +*** 75,128 **** + ***********************************************************************/ + + #if 0 +- #define UNIX /* Unix conditional compile switch */ +- #define MACINTOSH /* Macintosh conditional compile switch */ +- #define MS_DOS /* IBM PC conditional compile switch */ +- #define MSC60 /* Compiled for MS_DOS with MSC v6.0 */ +- #define AIX /* AIX conditional compile switch */ + #define CONVEX /* CONVEX conditional compile switch */ + #endif + +- #if defined(MSC60) +- #ifndef MS_DOS +- #define MS_DOS +- #endif +- #ifndef PROTO_ARGS +- #define PROTO_ARGS +- #endif +- #endif +- +- #ifdef UNIX +- #define TABLES_PATH "tables" /* to find data files */ +- /* name of environment variable holding path of table files */ +- #define MPEGTABENV "MPEGTABLES" +- #define PATH_SEPARATOR "/" /* how to build paths */ +- #endif /* UNIX */ +- +- #ifdef MACINTOSH +- #define TABLES_PATH ":tables:" /* where to find data files */ +- #endif /* MACINTOSH */ +- + /* + * Don't define FAR to far unless you're willing to clean up the + * prototypes + */ + #define FAR /*far*/ + +- #ifdef __STDC__ +- #ifndef PROTO_ARGS +- #define PROTO_ARGS +- #endif +- #endif +- + #ifdef CONVEX + #define SEEK_SET 0 + #define SEEK_CUR 1 + #define SEEK_END 2 + #endif + +- /* MS_DOS and VMS do not define TABLES_PATH, so OpenTableFile will default +- to finding the data files in the default directory */ + + /*********************************************************************** + * +--- 20,40 ---- +*************** +*** 133,163 **** + #include + #include + #include + #include "portableio.h" + #include "ieeefloat.h" +- +- #ifdef UNIX +- #include +- #endif /* UNIX */ +- +- #ifdef __sgi +- #include + #endif + +! #ifdef MACINTOSH +! #include +! #include +! #endif /* MACINTOSH */ +! +! #ifdef MS_DOS + #include +- #ifdef MSC60 +- #include +- #else +- #include +- #include +- #endif /* MSC60 */ +- #endif /* MS_DOS */ + + #ifndef EXIT_SUCCESS + #define EXIT_SUCCESS 0 +--- 45,57 ---- + #include + #include + #include ++ #ifndef LIBSNDFILE + #include "portableio.h" + #include "ieeefloat.h" + #endif + +! /*#include */ + #include + + #ifndef EXIT_SUCCESS + #define EXIT_SUCCESS 0 +*************** +*** 175,185 **** + + /* General Definitions */ + +! #ifdef MS_DOS +! #define FLOAT double +! #else + #define FLOAT float + #endif + #ifndef FALSE + #define FALSE 0 + #endif +--- 69,78 ---- + + /* General Definitions */ + +! #ifndef FLOAT + #define FLOAT float + #endif ++ + #ifndef FALSE + #define FALSE 0 + #endif +*************** +*** 191,202 **** +--- 84,108 ---- + #define NULL_CHAR '\0' + + #define MAX_U_32_NUM 0xFFFFFFFF ++ #ifdef M_PI ++ #define PI M_PI ++ #endif + #ifndef PI + #define PI 3.14159265358979 + #endif ++ #ifdef M_PI_4 ++ #define PI4 M_PI_4 ++ #else + #define PI4 PI/4 ++ #endif + #define PI64 PI/64 ++ #ifdef M_LN10 ++ #define LN_TO_LOG10 (M_LN10/10) ++ #else + #define LN_TO_LOG10 0.2302585093 ++ #endif ++ ++ #define SQRT2 1.4142135623730951454746218587388284504414 + + #define VOL_REF_NUM 0 + #define MPEG_AUDIO_ID 1 +*************** +*** 206,213 **** + #define MONO 1 + #define STEREO 2 + #define BITS_IN_A_BYTE 8 +! #define WORD 16 +! #define MAX_NAME_SIZE 81 + #define SBLIMIT 32 + #define SSLIMIT 18 + #define FFT_SIZE 1024 +--- 112,118 ---- + #define MONO 1 + #define STEREO 2 + #define BITS_IN_A_BYTE 8 +! #define MAX_NAME_SIZE 300 + #define SBLIMIT 32 + #define SSLIMIT 18 + #define FFT_SIZE 1024 +*************** +*** 244,256 **** + #define ASCII 1 + + #ifndef BS_FORMAT +! #define BS_FORMAT ASCII /* BINARY or ASCII = 2x bytes */ + #endif + +! #define BUFFER_SIZE 4096 + +! #define MIN(A, B) ((A) < (B) ? (A) : (B)) +! #define MAX(A, B) ((A) > (B) ? (A) : (B)) + + /*********************************************************************** + * +--- 149,162 ---- + #define ASCII 1 + + #ifndef BS_FORMAT +! /* #define BS_FORMAT ASCII */ /* BINARY or ASCII = 2x bytes */ +! #define BS_FORMAT BINARY + #endif + +! #define BUFFER_SIZE 16384 + +! #define Min(A, B) ((A) < (B) ? (A) : (B)) +! #define Max(A, B) ((A) > (B) ? (A) : (B)) + + /*********************************************************************** + * +*************** +*** 316,321 **** +--- 222,229 ---- + unsigned long blockSize; + } blockAlign; + ++ ++ #ifndef LIBSNDFILE + typedef struct IFF_AIFF_struct { + short numChannels; + unsigned long numSampleFrames; +*************** +*** 324,329 **** +--- 232,239 ---- + unsigned long sampleType; + blockAlign blkAlgn; + } IFF_AIFF; ++ #endif ++ + + enum byte_order { order_unknown, order_bigEndian, order_littleEndian }; + extern enum byte_order NativeByteOrder; +*************** +*** 331,336 **** +--- 241,248 ---- + /* "bit_stream.h" Type Definitions */ + + typedef struct bit_stream_struc { ++ unsigned char* pbtOutBuf; /* for .DLL code */ ++ int nOutBufPos; /* for .DLL code */ + FILE *pt; /* pointer to bit stream device */ + unsigned char *buf; /* bit stream buffer */ + int buf_size; /* size of buffer (in number of bytes) */ +*************** +*** 368,491 **** + + /* The following functions are in the file "common.c" */ + +- #ifdef PROTO_ARGS +- extern FILE *OpenTableFile(char*); +- extern int read_bit_alloc(int, al_table*); +- extern int pick_table(frame_params*); + extern int js_bound(int, int); + extern void hdr_to_frps(frame_params*); +- extern void WriteHdr(frame_params*, FILE*); +- extern void WriteBitAlloc(unsigned int[2][SBLIMIT], frame_params*, +- FILE*); +- extern void WriteScale(unsigned int[2][SBLIMIT], +- unsigned int[2][SBLIMIT], unsigned int[2][3][SBLIMIT], +- frame_params*, FILE*); +- extern void WriteSamples(int, unsigned int FAR [SBLIMIT], +- unsigned int[SBLIMIT], frame_params*, FILE*); +- extern int NumericQ(char*); + extern int BitrateIndex(int, int, int); + extern int SmpFrqIndex(long, int*); +- extern int memcheck(char*, int, int); + extern void FAR *mem_alloc(unsigned long, char*); + extern void mem_free(void**); + extern void double_to_extended(double*, char[10]); + extern void extended_to_double(char[10], double*); + extern int aiff_read_headers(FILE*, IFF_AIFF*); + extern int aiff_seek_to_sound_data(FILE*); + extern int aiff_write_headers(FILE*, IFF_AIFF*); + extern int refill_buffer(Bit_stream_struc*); + extern void empty_buffer(Bit_stream_struc*, int); + extern void open_bit_stream_w(Bit_stream_struc*, char*, int); +- extern void open_bit_stream_r(Bit_stream_struc*, char*, int); +- extern void close_bit_stream_r(Bit_stream_struc*); + extern void close_bit_stream_w(Bit_stream_struc*); + extern void alloc_buffer(Bit_stream_struc*, int); + extern void desalloc_buffer(Bit_stream_struc*); +- extern void back_track_buffer(Bit_stream_struc*, int); +- extern unsigned int get1bit(Bit_stream_struc*); + extern void put1bit(Bit_stream_struc*, int); +- extern unsigned long look_ahead(Bit_stream_struc*, int); +- extern unsigned long getbits(Bit_stream_struc*, int); + extern void putbits(Bit_stream_struc*, unsigned int, int); +- extern void byte_ali_putbits(Bit_stream_struc*, unsigned int, int); +- extern unsigned long byte_ali_getbits(Bit_stream_struc*, int); + extern unsigned long sstell(Bit_stream_struc*); + extern int end_bs(Bit_stream_struc*); +- extern int seek_sync(Bit_stream_struc*, long, int); +- extern void I_CRC_calc(frame_params*, unsigned int[2][SBLIMIT], +- unsigned int*); +- extern void II_CRC_calc(frame_params*, unsigned int[2][SBLIMIT], +- unsigned int[2][SBLIMIT], unsigned int*); +- extern void update_CRC(unsigned int, unsigned int, unsigned int*); + extern void read_absthr(FLOAT*, int); +! extern unsigned int hget1bit(); /* MI */ +! extern unsigned long hgetbits(int); +! extern unsigned long hsstell(); +! extern void hputbuf(unsigned int,int); + extern enum byte_order DetermineByteOrder(); + extern void SwapBytesInWords( short *loc, int words ); + + +- #ifdef MACINTOSH +- extern void set_mac_file_attr(char[MAX_NAME_SIZE], short, OSType, +- OSType); +- #endif +- #ifdef MS_DOS +- extern void new_ext(char *filename, char *extname, char *newname); +- #endif +- +- #else +- extern FILE *OpenTableFile(); +- extern int read_bit_alloc(); +- extern int pick_table(); +- extern int js_bound(); +- extern void hdr_to_frps(); +- extern void WriteHdr(); +- extern void WriteBitAlloc(); +- extern void WriteScale(); +- extern void WriteSamples(); +- extern int NumericQ(); +- extern int BitrateIndex(); +- extern int SmpFrqIndex(); +- extern int memcheck(); +- extern void FAR *mem_alloc(); +- extern void mem_free(); +- extern void double_to_extended(); +- extern void extended_to_double(); +- extern int aiff_read_headers(); +- extern int aiff_seek_to_sound_data(); +- extern int aiff_write_headers(); +- extern int refill_buffer(); +- extern void empty_buffer(); +- extern void open_bit_stream_w(); +- extern void open_bit_stream_r(); +- extern void close_bit_stream_r(); +- extern void close_bit_stream_w(); +- extern void alloc_buffer(); +- extern void desalloc_buffer(); +- extern void back_track_buffer(); +- extern unsigned int get1bit(); +- extern void put1bit(); +- extern unsigned long look_ahead(); +- extern unsigned long getbits(); +- extern void putbits(); +- extern void byte_ali_putbits(); +- extern unsigned long byte_ali_getbits(); +- extern unsigned long sstell(); +- extern int end_bs(); +- extern int seek_sync(); +- extern void I_CRC_calc(); +- extern void II_CRC_calc(); +- extern void update_CRC(); +- extern void read_absthr(); +- +- extern unsigned int hget1bit(); +- extern unsigned long hgetbits(); +- extern unsigned long hsstell(); +- extern void hputbuf(); + +- #ifdef MS_DOS +- extern void new_ext(); +- #endif +- #endif + #endif +--- 280,317 ---- + + /* The following functions are in the file "common.c" */ + + extern int js_bound(int, int); + extern void hdr_to_frps(frame_params*); + extern int BitrateIndex(int, int, int); + extern int SmpFrqIndex(long, int*); + extern void FAR *mem_alloc(unsigned long, char*); + extern void mem_free(void**); + extern void double_to_extended(double*, char[10]); + extern void extended_to_double(char[10], double*); ++ #ifndef LIBSNDFILE + extern int aiff_read_headers(FILE*, IFF_AIFF*); + extern int aiff_seek_to_sound_data(FILE*); + extern int aiff_write_headers(FILE*, IFF_AIFF*); ++ #endif + extern int refill_buffer(Bit_stream_struc*); + extern void empty_buffer(Bit_stream_struc*, int); + extern void open_bit_stream_w(Bit_stream_struc*, char*, int); + extern void close_bit_stream_w(Bit_stream_struc*); + extern void alloc_buffer(Bit_stream_struc*, int); + extern void desalloc_buffer(Bit_stream_struc*); + extern void put1bit(Bit_stream_struc*, int); + extern void putbits(Bit_stream_struc*, unsigned int, int); + extern unsigned long sstell(Bit_stream_struc*); + extern int end_bs(Bit_stream_struc*); + extern void read_absthr(FLOAT*, int); +! + extern enum byte_order DetermineByteOrder(); + extern void SwapBytesInWords( short *loc, int words ); + ++ extern int readframe(); ++ extern int makeframe(); ++ extern void ++ getframebits(layer *info, int stereo, int *bitsPerFrame, int *mean_bits); + + + #endif +diff -r -c -N encoder/configure lame3.13/configure +*** encoder/configure Wed Jan 22 02:43:13 1997 +--- lame3.13/configure Wed Dec 31 17:00:00 1969 +*************** +*** 1,742 **** +- #!/bin/sh +- ############################################################################ +- ## ISO MPEG Audio Subgroup Software Simulation Group (1996) +- ## ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- ## +- ## $Id: configure,v 1.1 1996/02/14 04:06:24 rowlands Exp $ +- ## +- ## $Log: configure,v $ +- ## Revision 1.1 1996/02/14 04:06:24 rowlands +- ## Initial revision +- ## +- ## Received from Mike Coleman +- ## +- ############################################################################ +- +- # Guess values for system-dependent variables and create Makefiles. +- # Generated automatically using autoconf version 2.3 +- # Copyright (C) 1992, 1993, 1994 Free Software Foundation, Inc. +- # +- # This configure script is free software; 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"$ac_site_file" +- fi +- done +- +- if test -r "$cache_file"; then +- echo "loading cache $cache_file" +- . $cache_file +- else +- echo "creating cache $cache_file" +- > $cache_file +- fi +- +- ac_ext=c +- # CFLAGS is not in ac_cpp because -g, -O, etc. are not valid cpp options. +- ac_cpp='$CPP $CPPFLAGS' +- ac_compile='${CC-cc} -c $CFLAGS $CPPFLAGS conftest.$ac_ext 1>&5 2>&5' +- ac_link='${CC-cc} -o conftest $CFLAGS $CPPFLAGS $LDFLAGS conftest.$ac_ext $LIBS 1>&5 2>&5' +- +- if (echo "testing\c"; echo 1,2,3) | grep c >/dev/null; then +- # Stardent Vistra SVR4 grep lacks -e, says ghazi@caip.rutgers.edu. +- if (echo -n testing; echo 1,2,3) | sed s/-n/xn/ | grep xn >/dev/null; then +- ac_n= ac_c=' +- ' ac_t=' ' +- else +- ac_n=-n ac_c= ac_t= +- fi +- else +- ac_n= ac_c='\c' ac_t= +- fi +- +- +- +- # Extract the first word of "gcc", so it can be a program name with args. +- set dummy gcc; ac_word=$2 +- echo $ac_n "checking for $ac_word""... $ac_c" 1>&6 +- if eval "test \"`echo '$''{'ac_cv_prog_CC'+set}'`\" = set"; then +- echo $ac_n "(cached) $ac_c" 1>&6 +- else +- if test -n "$CC"; then +- ac_cv_prog_CC="$CC" # Let the user override the test. +- else +- IFS="${IFS= }"; ac_save_ifs="$IFS"; IFS="${IFS}:" +- for ac_dir in $PATH; do +- test -z "$ac_dir" && ac_dir=. +- if test -f $ac_dir/$ac_word; then +- ac_cv_prog_CC="gcc" +- break +- fi +- done +- IFS="$ac_save_ifs" +- test -z "$ac_cv_prog_CC" && ac_cv_prog_CC="cc" +- fi +- fi +- CC="$ac_cv_prog_CC" +- if test -n "$CC"; then +- echo "$ac_t""$CC" 1>&6 +- else +- echo "$ac_t""no" 1>&6 +- fi +- +- +- echo $ac_n "checking whether we are using GNU C""... $ac_c" 1>&6 +- if eval "test \"`echo '$''{'ac_cv_prog_gcc'+set}'`\" = set"; then +- echo $ac_n "(cached) $ac_c" 1>&6 +- else +- cat > conftest.c <&5 | egrep yes >/dev/null 2>&1; then +- ac_cv_prog_gcc=yes +- else +- ac_cv_prog_gcc=no +- fi +- fi +- echo "$ac_t""$ac_cv_prog_gcc" 1>&6 +- if test $ac_cv_prog_gcc = yes; then +- GCC=yes +- if test "${CFLAGS+set}" != set; then +- echo $ac_n "checking whether ${CC-cc} accepts -g""... $ac_c" 1>&6 +- if eval "test \"`echo '$''{'ac_cv_prog_gcc_g'+set}'`\" = set"; then +- echo $ac_n "(cached) $ac_c" 1>&6 +- else +- echo 'void f(){}' > conftest.c +- if test -z "`${CC-cc} -g -c conftest.c 2>&1`"; then +- ac_cv_prog_gcc_g=yes +- else +- ac_cv_prog_gcc_g=no +- fi +- rm -f conftest* +- +- fi +- echo "$ac_t""$ac_cv_prog_gcc_g" 1>&6 +- if test $ac_cv_prog_gcc_g = yes; then +- CFLAGS="-g -O" +- else +- CFLAGS="-O" +- fi +- fi +- else +- GCC= +- test "${CFLAGS+set}" = set || CFLAGS="-g" +- fi +- +- +- MATHLIBS="-lm" +- echo $ac_n "checking for -lieee""... $ac_c" 1>&6 +- if eval "test \"`echo '$''{'ac_cv_lib_ieee'+set}'`\" = set"; then +- echo $ac_n "(cached) $ac_c" 1>&6 +- else +- ac_save_LIBS="$LIBS" +- LIBS="-lieee $LIBS" +- cat > conftest.$ac_ext <&6 +- MATHLIBS="$MATHLIBS -lieee" +- else +- echo "$ac_t""no" 1>&6 +- fi +- +- +- +- NINTSW="" +- echo $ac_n "checking for nint() in -lm""... $ac_c" 1>&6 +- if eval "test \"`echo '$''{'ac_cv_lib_m'+set}'`\" = set"; then +- echo $ac_n "(cached) $ac_c" 1>&6 +- else +- ac_save_LIBS="$LIBS" +- LIBS="-lm $LIBS" +- cat > conftest.$ac_ext <&6 +- NINTSW="-DHAVE_NINT" +- else +- echo "$ac_t""no" 1>&6 +- fi +- +- +- +- +- trap '' 1 2 15 +- cat > confcache <<\EOF +- # This file is a shell script that caches the results of configure +- # tests run on this system so they can be shared between configure +- # scripts and configure runs. It is not useful on other systems. +- # If it contains results you don't want to keep, you may remove or edit it. +- # +- # By default, configure uses ./config.cache as the cache file, +- # creating it if it does not exist already. You can give configure +- # the --cache-file=FILE option to use a different cache file; that is +- # what configure does when it calls configure scripts in +- # subdirectories, so they share the cache. +- # Giving --cache-file=/dev/null disables caching, for debugging configure. +- # config.status only pays attention to the cache file if you give it the +- # --recheck option to rerun configure. +- # +- EOF +- # Ultrix sh set writes to stderr and can't be redirected directly, +- # and sets the high bit in the cache file unless we assign to the vars. +- (set) 2>&1 | +- sed -n "s/^\([a-zA-Z0-9_]*_cv_[a-zA-Z0-9_]*\)=\(.*\)/\1=\${\1='\2'}/p" \ +- >> confcache +- if cmp -s $cache_file confcache; then +- : +- else +- if test -w $cache_file; then +- echo "updating cache $cache_file" +- cat confcache > $cache_file +- else +- echo "not updating unwritable cache $cache_file" +- fi +- fi +- rm -f confcache +- +- trap 'rm -fr conftest* confdefs* core core.* *.core $ac_clean_files; exit 1' 1 2 15 +- +- test "x$prefix" = xNONE && prefix=$ac_default_prefix +- # Let make expand exec_prefix. +- test "x$exec_prefix" = xNONE && exec_prefix='${prefix}' +- +- # Any assignment to VPATH causes Sun make to only execute +- # the first set of double-colon rules, so remove it if not needed. +- # If there is a colon in the path, we need to keep it. +- if test "x$srcdir" = x.; then +- ac_vpsub='/^[ ]*VPATH[ ]*=[^:]*$/d' +- fi +- +- trap 'rm -f $CONFIG_STATUS conftest*; exit 1' 1 2 15 +- +- # Transform confdefs.h into DEFS. +- # Protect against shell expansion while executing Makefile rules. +- # Protect against Makefile macro expansion. +- cat > conftest.defs <<\EOF +- s%#define \([A-Za-z_][A-Za-z0-9_]*\) \(.*\)%-D\1=\2%g +- s%[ `~#$^&*(){}\\|;'"<>?]%\\&%g +- s%\[%\\&%g +- s%\]%\\&%g +- s%\$%$$%g +- EOF +- DEFS=`sed -f conftest.defs confdefs.h | tr '\012' ' '` +- rm -f conftest.defs +- +- +- # Without the "./", some shells look in PATH for config.status. +- : ${CONFIG_STATUS=./config.status} +- +- echo creating $CONFIG_STATUS +- rm -f $CONFIG_STATUS +- cat > $CONFIG_STATUS </dev/null | sed 1q`: +- # +- # $0 $ac_configure_args +- # +- # Compiler output produced by configure, useful for debugging +- # configure, is in ./config.log if it exists. +- +- ac_cs_usage="Usage: $CONFIG_STATUS [--recheck] [--version] [--help]" +- for ac_option +- do +- case "\$ac_option" in +- -recheck | --recheck | --rechec | --reche | --rech | --rec | --re | --r) +- echo "running \${CONFIG_SHELL-/bin/sh} $0 $ac_configure_args --no-create --no-recursion" +- exec \${CONFIG_SHELL-/bin/sh} $0 $ac_configure_args --no-create --no-recursion ;; +- -version | --version | --versio | --versi | --vers | --ver | --ve | --v) +- echo "$CONFIG_STATUS generated by autoconf version 2.3" +- exit 0 ;; +- -help | --help | --hel | --he | --h) +- echo "\$ac_cs_usage"; exit 0 ;; +- *) echo "\$ac_cs_usage"; exit 1 ;; +- esac +- done +- +- ac_given_srcdir=$srcdir +- +- trap 'rm -fr `echo "Makefile" | sed "s/:[^ ]*//g"` conftest*; exit 1' 1 2 15 +- +- # Protect against being on the right side of a sed subst in config.status. +- sed 's/%@/@@/; s/@%/@@/; s/%g$/@g/; /@g$/s/[\\\\&%]/\\\\&/g; +- s/@@/%@/; s/@@/@%/; s/@g$/%g/' > conftest.subs <<\CEOF +- $ac_vpsub +- $extrasub +- s%@CFLAGS@%$CFLAGS%g +- s%@CPPFLAGS@%$CPPFLAGS%g +- s%@CXXFLAGS@%$CXXFLAGS%g +- s%@DEFS@%$DEFS%g +- s%@LDFLAGS@%$LDFLAGS%g +- s%@LIBS@%$LIBS%g +- s%@exec_prefix@%$exec_prefix%g +- s%@prefix@%$prefix%g +- s%@program_transform_name@%$program_transform_name%g +- s%@CC@%$CC%g +- s%@MATHLIBS@%$MATHLIBS%g +- s%@NINTSW@%$NINTSW%g +- +- CEOF +- EOF +- cat >> $CONFIG_STATUS <> $CONFIG_STATUS <<\EOF +- for ac_file in .. $CONFIG_FILES; do if test "x$ac_file" != x..; then +- # Support "outfile[:infile]", defaulting infile="outfile.in". +- case "$ac_file" in +- *:*) ac_file_in=`echo "$ac_file"|sed 's%.*:%%'` +- ac_file=`echo "$ac_file"|sed 's%:.*%%'` ;; +- *) ac_file_in="${ac_file}.in" ;; +- esac +- +- # Adjust relative srcdir, etc. for subdirectories. +- +- # Remove last slash and all that follows it. Not all systems have dirname. +- ac_dir=`echo $ac_file|sed 's%/[^/][^/]*$%%'` +- if test "$ac_dir" != "$ac_file" && test "$ac_dir" != .; then +- # The file is in a subdirectory. +- test ! -d "$ac_dir" && mkdir "$ac_dir" +- ac_dir_suffix="/`echo $ac_dir|sed 's%^\./%%'`" +- # A "../" for each directory in $ac_dir_suffix. +- ac_dots=`echo $ac_dir_suffix|sed 's%/[^/]*%../%g'` +- else +- ac_dir_suffix= ac_dots= +- fi +- +- case "$ac_given_srcdir" in +- .) srcdir=. +- if test -z "$ac_dots"; then top_srcdir=. +- else top_srcdir=`echo $ac_dots|sed 's%/$%%'`; fi ;; +- /*) srcdir="$ac_given_srcdir$ac_dir_suffix"; top_srcdir="$ac_given_srcdir" ;; +- *) # Relative path. +- srcdir="$ac_dots$ac_given_srcdir$ac_dir_suffix" +- top_srcdir="$ac_dots$ac_given_srcdir" ;; +- esac +- +- echo creating "$ac_file" +- rm -f "$ac_file" +- configure_input="Generated automatically from `echo $ac_file_in|sed 's%.*/%%'` by configure." +- case "$ac_file" in +- *Makefile*) ac_comsub="1i\\ +- # $configure_input" ;; +- *) ac_comsub= ;; +- esac +- sed -e "$ac_comsub +- s%@configure_input@%$configure_input%g +- s%@srcdir@%$srcdir%g +- s%@top_srcdir@%$top_srcdir%g +- " -f conftest.subs $ac_given_srcdir/$ac_file_in > $ac_file +- fi; done +- rm -f conftest.subs +- +- +- +- exit 0 +- EOF +- chmod +x $CONFIG_STATUS +- rm -fr confdefs* $ac_clean_files +- test "$no_create" = yes || ${CONFIG_SHELL-/bin/sh} $CONFIG_STATUS || exit 1 +- +--- 0 ---- +diff -r -c -N encoder/doc/html/advanced.html lame3.13/doc/html/advanced.html +*** encoder/doc/html/advanced.html Wed Dec 31 17:00:00 1969 +--- lame3.13/doc/html/advanced.html Wed May 26 18:44:50 1999 +*************** +*** 0 **** +--- 1,54 ---- ++ ++ ++ ++ Advanced features ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++

Overriding default settings

++  Many of the following features override the encoder's ++ idea of best-quality settings. You should be aware that overriding the encoder ++ default settings is something for experts. You might wreack the encoding quality ++ in a number of ways without first noticing it. Also, the encoder is not guaranteed ++ to run at all parameter combinations. Proceed at your own risk! ++

++

++
-k  ++
Tells the encoder to use full bandwidth. By default, the encoder uses a ++ 16kHz bandwidth up to 64kbps per channel, and full bandwidth at higher bitrates. ++ Increasing the bandwidth from the default setting might produces ringing artefacts ++ at low bitrates. Use with care! ++
++
++
-O  ++
Uses the original ISO psycho-acoustic model instead of GPsycho. Used with ++ -m s, it will produce the same output as the ISO code or the encoders ++ based on it, like 8Hz, Blade or Lame 2.x
++ It will reduce a lot the quality and should only be used for comparison tests. ++
 
++
++

Tids & bits

++
++
-a  ++
This switch tells the encoder to autoconvert a stereo input signal into ++ one channel, producing mono output. The downmix is calculated as the sum of ++ the left and right channel, attenuated by 6 dB. ++
-f  ++
This switch forces the encoder to use the "extra fast" encoding ++ mode. It will reduce a lot the quality of the output. This might be useful ++ for real-time processing or streaming on slowest computers. ++
++ ++ ++ +diff -r -c -N encoder/doc/html/basics.html lame3.13/doc/html/basics.html +*** encoder/doc/html/basics.html Wed Dec 31 17:00:00 1969 +--- lame3.13/doc/html/basics.html Wed May 26 18:16:32 1999 +*************** +*** 0 **** +--- 1,134 ---- ++ ++ ++ ++ Basics ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++

Basics

++

Samplerate

++ Sample rate is the rate at which the samples are read from your sound card when ++ you sample. Sample rate is directly linked to audio bandwidth achievable: A sound ++ file with a sample rate of 8 kHz does not contain frequencies beyond 4 kHz. ++ This means that you should always use the highest sample rate that your sound ++ card supports when you sample a signal. ++

Bitrate

++ The main parameter controlling the sound quality is the bitrate that the ++ encoder runs at. In a nutshell, the higher the bitrate, the better the quality. ++

The bitrate of the bitstream output is selected via the -b switch. ++ The bitrate is specified in bits/second. The bitrate is the total bitrate for ++ all encoded channels, i.e. if you select -b 128000 and encode ++ in stereo, both channels will be stuffed into one bitstream of 128000 bits/second. ++

The encoder supports bitrates of 32, 40, 48, 56, 64, 96, 112, 128, 160, 192, ++ 224, 256 and 320 kBit/s. ++

++

Stereo mode

++

++

++
  ++
 stereo ++
In this mode, the encoder makes no use of potentially existing correlations ++ between the two input channels. It can, however, negotiate the bit demand ++ between both channel, i.e. give one channel more bits if the other contains ++ silence. ++

++

 joint stereo ++
In this mode, the encoder will make use of a correlation between both channels. ++ The signal will be matrixed into a sum ("mid") and difference ("side") ++ signal. For quasi-mono signals, this will give a significant gain in encoding ++ quality. ++

This mode does not destroy phase information like IS stereo that may be ++ used by other encoders and thus can be used to encode DOLBY ProLogic ++ surround signals. ++


++
 force joint stereo ++
This mode will force MS joint stereo on all frames. It's faster and it uses ++ some special mid & side masking threshold. ++
 
++
 mono
++
The input will be encoded as a mono signal. If it ++ was a stereo signal, it will be downsampled to mono.
++
++
++

Encoding speed

++ Several factors influence the speed of the encoder. They include: ++

++

    ++
  • Number of channels in the output signal. If your output signal has only ++ one channel, the encoder will run at twice the speed compared to stereo encoding. ++
  • Output sample rate. If the encoder produces a file at 22.050 kHz (that ++ is, a file that contains 22050 samples per second), it runs at twice the speed ++ compared to one that produces twice the number of samples per second (i.e. ++ produces a 44.1 kHz output). ++
  • Quality of the psychoaccoustic model. You can tell the encoder to use a ++ dummy psychoaccoustic model, but the quality will be lower. ++
++

Version V3.04 of the encoder reaches realtime speed on a Pentium 166 when ++ encoding at 64 kBit/s, 22,050 kHz, stereo. On a SUN Sparc Ultra-1 ++ (143 MHz) the performance is similar. ++

++

Input file specification

++ The encoder can read AIFF, AIFF-C, WAV/RIFF, and raw PCM data files. While the ++ first three only work from a file, plain PCM data can be fed into the encoder ++ via a pipe. This is useful for live encoding (also known as streaming). ++

++

Input from file: filename

++
++
filename ++
will tell the encoder the filename it reads it input from. If the file is ++ a RIFF/WAVE, or an AIFF/AIFC file, the encoder will automatically adapt to ++ the sound file format. For other formats or plain PCM data, see below. ++
++

Piping data into the encoder

++
++
- ++
tells the encoder to get its input from stdin rather than from a file. This ++ only works when the input is plain pcm data (see below). ++
++

plain PCM data input

++ If the encoder gets its input as plain pcm data (or if it does not recognize the ++ sound format by itself), you need to tell it all about the structure of the PCM ++ stream, i.e. the samplerate. ++
++
-s  samplerate ++
The sample rate in the input file. The default is 44.1 kHz. ++
-x  ++
The input file is endian inversed compared to your platform. ++
++ For stereo files, the encoder assumes that the PCM data is interleaved and that ++ the sample for the right channel follows that for the left channel. ++

As an example, -s 22050 would be used to read a 22.05 kHz. ++

Remember that this feature is only needed for input from files other than ++ RIFF/WAV, AIFF and AIFC. ++

++

Output file specification

++ The output is a plain Layer-3 stream wich can be piped into other applications. ++ This is useful for live streaming. ++

++

++
filename ++
tells the encoder the filename of the file that the encoder will write the ++ bitstream to. If the file does not exist, it is created; if it does exist, ++ it will be overwritten. ++
++

Streaming data out of the encoder

++
++
-  ++
tells the encoder to write its output into stdout rather than in a file. ++
++ ++

 

++ ++ +diff -r -c -N encoder/doc/html/contributors.html lame3.13/doc/html/contributors.html +*** encoder/doc/html/contributors.html Wed Dec 31 17:00:00 1969 +--- lame3.13/doc/html/contributors.html Sat May 29 12:10:14 1999 +*************** +*** 0 **** +--- 1,81 ---- ++ ++ ++ Contributors ++ ++ ++ ++
++

Contributors

++
++
++

The following people contributed to the LAME development:

++

Lauri Ahonen
++ Alex Ballegooy
++ Gabriel Bouvigne
++ Mike Cheng
++ Goran Dokic
++ James Droppo
++ Joakim Enerstam
++ Albert Faber
++ Joseph Fourier
++ Jani Frilander
++ Steffan Haeuser
++ Rafael W Luebbert
++ Lars Magne Ingebrigtsen
++ Scott Manley
++ Chris Matrakidis
++ Mike Oliphant
++ André Osterhues
++ Jan Peman
++ Jan Rafaj
++ Nathan Slingerland
++ Stephane Tavenard
++ Mark Taylor
++ Brad Threatt
++ Warren Toomey
++ Martin Weghofer

++

 

++

Original ISO contributors:

++

Bill Aspromonte
++ Shaun Astarabadi
++ R. Bittner
++ Karlheinz Brandenburg
++ W. Joseph Carter
++ Jack Chang
++ Mike Coleman
++ Johnathan Devine
++ Ernst Eberlein
++ Dan Ellis
++ Peter Farrett
++ Jean-Georges Fritsch
++ Vlad Fruchter
++ Hendrik Fuchs
++ Bernhard Grill
++ Amit Gulati
++ Munsi Haque
++ Chuck Hsiao
++ Toshiyuki Ishino
++ Masahiro Iwadare
++ Earl Jennings
++ James Johnston
++ Leon v.d. Kerkhof
++ Don Lee
++ Mike Li
++ Yu-Tang Lin
++ Soren Neilsen
++ Simao F. Campos Neto
++ Mark Paley
++ Davis Pan
++ Tan Ah Peng
++ Kevin Peterson
++ Juan Pineda
++ Ernst F. Schroeder
++ Peter Siebert
++ Jens Spille
++ Sam Stewart
++ Al Tabayoyon
++ Kathy Wang
++ Franz-Otto Witte
++ Douglas Wong

++ ++ +diff -r -c -N encoder/doc/html/history.html lame3.13/doc/html/history.html +*** encoder/doc/html/history.html Wed Dec 31 17:00:00 1969 +--- lame3.13/doc/html/history.html Wed May 26 12:33:16 1999 +*************** +*** 0 **** +--- 1,150 ---- ++ ++ ++ History ++ ++ ++ ++
++

History

++
++

++
++

++

3.05

++
    ++
  • Fast mode (-f)  disables psycho-acoustic model for real time encoding ++ on older machines. ++
  • New bit reservoir usage scheme to accommodate the new pre-echo detection ++ formulas. ++
  • Tuning of AWS and ENER_AWS pre-echo formulas by Gabriel Bouvigne and Mark ++ Taylor. They work great! Now on by default. ++
  • In jstereo, force blocktypes for left & right channels to be identical.  ++ FhG does this, so we will too.
    • ++
  • Patches to compile MP3x under win32 (Thanks to Albert Faber).
    • ++
  • bin_serach_stepsize limited to -210 through 45.
    • ++
  • Mid/side quantization now varies scalefactors independently. Can lead to ++ better quantizations, but it is slower (twice as many quantizations to look ++ at). Running with "-m f" does not need this and will at the old speed
    • ++
++

3.04

++
    ++
  • Preliminary documentation from Gabriel Bouvigne. ++
  • Even more speed improvements from Chris Matrakidis!  Removed one FFT ++ when using joint stereo, and many improvements in loop.c. ++
  • "Fake" ms_stereo mode renamed "Force" ms_stereo since it forces mid/side ++ stereo on all frames.  For some music this is said to be a problem, but ++ for most music mode is probably better than the default jstereo because it ++ uses specialized mid/side channel masking thresholds. ++
  • Small bugs in Force ms_stereo mode fixed. ++
  • Compaq Alpha fixes from Nathan Slingerland. ++
  • Some new experimental pre-echo detection formulas in l3psy.c (#ifdef AWS ++ and #ifdef ENER_AWS, both off by default.  Thanks to Gabriel Bouvigne ++ and Andre Osterhues) ++
  • Several bugs in the syncing of data displayed by mp3x (the frame analyzer) ++ were fixed. ++
  • Highq (-h) option added.  This turns on things (just one so far) that ++ should sound better but slow down LAME.
    • ++
++ 3.03 ++
    ++
  • Faster (20%) & cleaner FFT (Thanks to Chris Matrakidis  http://www.geocities.com/ResearchTriangle/8869/fft_summary.html) ++
  • Mods so it works with VC++ (Thanks to Gabriel Bouvigne, www.mp3tech.org) ++
  • MP3s marked "original" by default  (Thanks to Gabriel Bouvigne, www.mp3tech.org) ++
  • Can now be compiled into a BladeEnc compatible .DLL   (Thanks ++ to  Albert Faber, CDex author) ++
  • Patches for "silent mode" and stdin/stdout  (Thanks to Lars Magne Ingebrigtsen) ++
  • Fixed rare bug: if a long_block is sandwiched between two short_blocks, ++ it must be changed to a short_block, but the short_block ratios have not been ++ computed in l3psy.c.  Now always compute short_block ratios just in case. ++
  • Fixed bug with initial quantize step size when many coefficients are zero.  ++ (Thanks to Martin Weghofer). ++
  • Bug fixed in MP3x display of audible distortion. ++
  • Iimproved status display (Thanks to Lauri Ahonen).
    • ++
++

3.02

++
    ++
  • Encoder could use ms_stereo even if channel 0 and 1 block types were different.  ++ (Thanks to Jan Rafaj) ++
  • Added -k option to disable the 16kHz cutoff at 128kbs.  This cutoff ++ is never used at higher bitrates. (Thanks to Jan Rafaj) ++
  • Modified pe bit allocation formula to make sense at bit rates other than ++ 128kbs. ++
  • Fixed l3_xmin initialization problem which showed up under FreeBSD.  ++ (Thanks to Warren Toomey)
    • ++
++ 3.01 ++
    ++
  • max_name_size increased to 300  (Thanks to Mike Oliphant) ++
  • Patch to allow seeks on input file (Thanks to Scott Manley) ++
  • Fixes for mono modes (Thanks to everyone who pointed this out) ++
  • Overflow in calc_noise2 fixed ++
  • Bit reservoir overflow when encoding lots of frames with all zeros  ++ (Thanks to Jani Frilander)
    • ++
++


++ 3.0 ++

    ++
  • Added GPSYCHO (developed by Mark Taylor) ++
  • Added MP3x (developed by Mark Taylor)
    • ++
++ 2.1f ++
    ++
  • 50% faster filter_subband() routine in encode.c contributed by James Droppo ++
++ 2.1e ++
    ++
  • New command line switch -a auto-resamples a stereo input file to ++ mono. ++
  • New command line switch -r resamples from 44.1khz to 32khz [this ++ switch doesn't work really well. Very tinny sounding output files. Has to ++ do with the way I do the resampling probably] ++
  • Both of these were put into the ISO code in the encode.c file, and are simply ++ different ways of filling the input buffers from a file.
    • ++
++ 2.1d ++
    ++
  • Fixed memory alloc in musicin.c (for l3_sb_sample) ++
  • Added new command line switch (-x) to force swapping of byte order ++
  • Cleaned up memory routines in l3psy.c. All the mem_alloc() and free() routines ++ where changed so that it was only done once and not every single time ++ the routine was called. ++
  • Added a compile time switch -DTIMER that includes all timing info. It's ++ a switch for the time being until some other people have tested on their system. ++ Timing code has a tendency to do different things on different platforms. ++
    • ++
++ 2.1b ++
    ++
  • Fixed up bug: all PCM files were being read as WAV. ++
  • Played with the mem_alloc routine to fix crash under amigaos (just allocating ++ twice as much memory as needed). Might see if we can totally do without this ++ routine. Individual malloc()s where they are needed instead ++
  • Put Jan Peman's quality switch back in. This reduces quality via the '-q ' ++ switch. Fun speedup which is mostly harmless if you're not concerned with ++ quality. ++
  • Compiling with amiga-gcc works fine
    • ++
++ 2.1a ++
    ++
  • Version 2.1a released. User input/output has been cleaned up a bit. WAV ++ file reading is there in a very rudimentary sense ie the program will recognize ++ the header and skip it, but not read it. The WAV file is assumed to be 16bit ++ stereo 44.1khz.
    • ++
++ 2.1 ++
    ++
  • All tables now incorporated into the exe. Thanks to Lars Magne Ingebrigtseni(larsi@ifi.uio.no) ++
    • ++
++ 2.0 ++
    ++
  • LAME is built from the ISO source code (dist10), and incorporates modifications ++ from Mike Cheng and the 8Hz effort. The output file from LAME v2.0 is identical ++ to the output of the ISO encoder.
    • ++
++ 1.0 ++
    ++
  • LAME v1.0 is based on work from the 8Hz effort.
    • ++
++ ++ +diff -r -c -N encoder/doc/html/index.html lame3.13/doc/html/index.html +*** encoder/doc/html/index.html Wed Dec 31 17:00:00 1969 +--- lame3.13/doc/html/index.html Wed May 26 18:54:44 1999 +*************** +*** 0 **** +--- 1,81 ---- ++ ++ ++ ++ L.A.M.E. V3 Next Generation High-End MPEG Layer III Encoding ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++

++

LAME V3.05
++ ++


++ Next Generation High-End MPEG Layer III Encoding
++ ++

++

Open Source Project
++ http://www.sulaco.org/mp3

++

++

++

++

++

++
++
++   ++ ++ ++ ++ +diff -r -c -N encoder/doc/html/lame.css lame3.13/doc/html/lame.css +*** encoder/doc/html/lame.css Wed Dec 31 17:00:00 1969 +--- lame3.13/doc/html/lame.css Mon Mar 23 20:59:20 1998 +*************** +*** 0 **** +--- 1,24 ---- ++ BODY { ++ font-family: Arial, Helvetica, sans-serif; ++ position: absolute; ++ padding-left: 1cm; ++ margin-left: 1cm; ++ } ++ ++ h1, h2, h3 { ++ margin-left: -1cm; text-align: left ++ } ++ ++ TABLE { text-align: left; font-family: Arial, Helvetica, sans-serif;} ++ TH {background-color: #C0C0C0} ++ TD {font-family: Arial, Helvetica, sans-serif;} ++ ++ SMALL.TINY { font-size : xx-small } ++ SMALL.SCRIPTSIZE { font-size : xx-small } ++ SMALL.FOOTNOTESIZE { font-size : x-small } ++ SMALL.SMALL { font-size : small } ++ BIG.LARGE { font-size : large } ++ BIG.XLARGE { font-size : x-large } ++ BIG.XXLARGE { font-size : xx-large } ++ BIG.HUGE { font-size : xx-large } ++ BIG.XHUGE { font-size : xx-large } +diff -r -c -N encoder/doc/html/node10.html lame3.13/doc/html/node10.html +*** encoder/doc/html/node10.html Wed Dec 31 17:00:00 1969 +--- lame3.13/doc/html/node10.html Fri May 21 14:40:14 1999 +*************** +*** 0 **** +--- 1,72 ---- ++ ++ ++ ++ MP3Enc Features ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++
++

| Next: Basics | Up: LAME V3 Next Generation High-End | Previous: Bugfixes and changes between | Table ++ of Contents |
++
++

++

LAME Features

++
++
++   ++ ++
++ ++ ++ +diff -r -c -N encoder/doc/html/node11.html lame3.13/doc/html/node11.html +*** encoder/doc/html/node11.html Wed Dec 31 17:00:00 1969 +--- lame3.13/doc/html/node11.html Fri May 21 14:31:46 1999 +*************** +*** 0 **** +--- 1,184 ---- ++ ++ ++ ++ Basics ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++
++

| Next: Advanced features | Up: LAME Features | Previous: LAME Features | Table of Contents ++ |
++
++

Subsections ++

++ ++
++

Basics

++

Samplerate

++ Sample rate  is the rate at which the samples are read ++ from your sound card when you sample. Sample rate is directly linked to audio ++  bandwidth achievable: A sound file with a sample rate ++ of 8 kHz does not contain frequencies beyond 4 kHz. This means that ++ you should always use the highest sample rate that your sound card supports when ++ you sample a signal. ++

Bitrate

++ The main parameter controlling the sound quality is the  bitrate ++ that the encoder runs at. In a nutshell, the higher the bitrate, the better the ++ quality. ++

The bitrate of the encoder is linked to the samplerate that the encoded file ++ will have. ++

 The bitrate of the bitstream output is selected ++ via the -b  switch. The bitrate is specified ++ in bits/second. The bitrate is the total bitrate for all encoded channels, i.e. ++ if you select -br  128000 and encode ++ in stereo, both channels will be stuffed into one bitstream of 128000 bits/second. ++

The encoder supports bitrates of 32, 40, 48, 56, 64, 96, 112, 128, 160, 192, ++ 224, 256 and 320 kBit/s. ++

++

Stereo mode

++ If encoding stereo, the bitrate of the encoder is linked to a stereo mode. MPEG ++ Layer-3 knows four modes for stereo encoding. ++

++

++
 dual channel ++
(also known as  dual mono) In this mode, the ++ encoder treats the two input channels as separate entities, assuming there ++ is no similarity between the channels. This would be appropriate if you e.g. ++ have a bilingual signal where one channel contains a german speaker and one ++ contains an english speaker. The current version of the encoder does not ++ support dual channel mode. ++

++

 stereo ++
In this mode, like in dual channel above, the encoder makes no use of potentially ++ existing correlations between the two input channels. It can, however, negotiate ++ the bit demand between both channel, i.e. give one channel more bits if the ++ other contains silence. ++

++

  MS stereo ++
In this mode, the encoder will make use of a correlation between both channels. ++ The signal will be matrixed into a sum (»mid«) and difference (»side«) ++ signal. For quasi-mono signals, this will give a significant gain in encoding ++ quality. ++

This mode does not destroy phase information like IS stereo (see below) ++ and thus can be used to encode DOLBY ProLogic$^{\mbox{tm}}$  surround signals. ++

++

 MS/IS stereo ++
In this mode, high-frequency parts of the signal will be downmixed to mono ++ and transmitted with a direction information (which is basically a pan). This ++ mode (called »intensity stereo«  will ++ loose phase information and should not be used for high-quality encoding. ++
++


++

Encoding speed

++  Several factors influence the speed of the encoder. ++ They include: ++

++

    ++
  • Number of channels in the output signal. If your output signal has only ++ one channel, the encoder will run at twice the speed compared to stereo encoding. ++
  • Output sample rate. If the encoder produces a file at 22.050 kHz (that ++ is, a file that contains 22050 samples per second), it runs at twice the speed ++ compared to one that produces twice the number of samples per second (i.e. ++ produces a 44.1 kHz output). ++
  • Quality of the psychoaccoustic model. You can tell the encoder to use a ++ dummy psychoaccoustic model, but the quality will be lower. ++
++

Version V3.04 of the encoder reaches realtime speed on a Pentium 166 when ++ encoding at 64 kBit/s, 22,050 kHz, stereo. On a SUN Sparc Ultra-1 ++ (143 MHz) the performance is similar. ++

++

Input file specification

++ The encoder can read AIFF, AIFF-C, WAV/RIFF, and raw PCM data files. While the ++ first three only work from a file, plain PCM data can be fed into the encoder ++ via a pipe. This is useful for live encoding (also known as  streaming). ++

++

Input from file: filename

++
++
  filename ++
will tell the encoder the filename it reads it input from. If the file is ++ a RIFF/WAVE, or an AIFF/AIFC file, the encoder will automatically adapt to ++ the sound file format. For other formats or plain PCM data, see below. ++
++

Piping data into the encoder

++
++
-  ++
tells the encoder to get its input from stdin rather than from a file. This ++ only works when the input is plain pcm data (see below). ++
++

plain PCM data input

++  If the encoder gets its input as plain pcm data (or ++ if it does not recognize the sound format by itself), you need to tell it all ++ about the structure of the PCM stream, i.e. the samplerate. ++
++
-s  samplerate ++
The sample rate in the input file. The default is 44.1 kHz. ++
-x  ++
The input file is endian inversed compared to your platform. ++
++ For stereo files, the encoder assumes that the PCM data is interleaved and that ++ the sample for the right channel follows that for the left channel. ++

As an example, -s 22050 would be used to read a 22.05 kHz. ++

Remember that this feature is only needed for input from files other than ++ RIFF/WAV, AIFF and AIFC. ++

++

Output file specification

++ The output is a plain Layer-3 stream wich can be piped into other applications. ++ This is useful for live streaming. ++

++

++
  filename ++
tells the encoder the filename of the file that the encoder will write the ++ bitstream to. If the file does not exist, it is created; if it does exist, ++ it will be overwritten. ++
++

Streaming data out of the encoder ++  

++
++
-  ++
tells the encoder to write its output into stdout rather than in a file. ++
++ ++

 

++ ++ +diff -r -c -N encoder/doc/html/node12.html lame3.13/doc/html/node12.html +*** encoder/doc/html/node12.html Wed Dec 31 17:00:00 1969 +--- lame3.13/doc/html/node12.html Fri May 21 14:31:58 1999 +*************** +*** 0 **** +--- 1,67 ---- ++ ++ ++ ++ Advanced features ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++
++

| Next: Troubleshooting | Up: LAME Features | Previous: Basics | Table of Contents | ++
++
++

Subsections ++

++ ++
++

Advanced features

++

Overriding default settings

++  Many of the following features override the encoder's ++ idea of best-quality settings. You should be aware that overriding the encoder ++ default settings is something for experts. You might wreck the encoding quality ++ in a number of ways without first noticing it. Also, the encoder is not guaranteed ++ to run at all parameter combinations. Proceed at your own risk! ++

++

++
-k  ++
Tell the encoder to use full  bandwidth. By default, ++ the encoder uses a 16kHz bandwidth up to 64kbps per channel, and full bandwidth ++ at higher bitrates Increasing the bandwidth from the default setting might ++ produces ringing artefacts at low bitrates. Use with care! ++
++

Tids & bits

++
++
-e  ++
For transmission over serial lines with bit errors, parts of the bitstream ++ can be protected by calculating a CRC checksum. If you are just producing ++ for harddisk storage, there is no need to set this switch. This feature is ++ not yet working. ++
-a  ++
This switch tells the encoder to autoconvert a stereo input signal into ++ one channel, producing mono output. The downmix is calculated as the sum of ++ the left and right channel, attenuated by 6 dB. ++
-f  ++
  This switch forces the encoder to use the ++ "extra fast" encoding mode. It will reduce the quality ++ of the output. This might be useful for real-time processing on slowest computers. ++
++ ++ ++ +diff -r -c -N encoder/doc/html/node13.html lame3.13/doc/html/node13.html +*** encoder/doc/html/node13.html Wed Dec 31 17:00:00 1969 +--- lame3.13/doc/html/node13.html Fri May 21 14:41:16 1999 +*************** +*** 0 **** +--- 1,45 ---- ++ ++ ++ ++ Troubleshooting ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++
++

| Next: Is it really a | Up: LAME V3 Next Generation High-End | Previous: Advanced features | Table of Contents ++ |
++
++

++

Troubleshooting

++

 No software is free of errors. If you believe you ++ have found an error in the operation of LAME, and you have checked the ++ list below, please report the error to our bugtracking address. ++


++

++   ++ ++
++ ++ ++ +diff -r -c -N encoder/doc/html/node14.html lame3.13/doc/html/node14.html +*** encoder/doc/html/node14.html Wed Dec 31 17:00:00 1969 +--- lame3.13/doc/html/node14.html Fri May 21 14:32:18 1999 +*************** +*** 0 **** +--- 1,87 ---- ++ ++ ++ ++ Is it really a bug? ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++
++

| Next: Reporting the bug | Up: Troubleshooting | Previous: Troubleshooting | Table of Contents ++ |
++
++

++

Is it really a bug?

++

Before you report a bug to our programers, please verify that the bug is really ++ in the software and not in your configuration. Table 3.1 ++ helps you track down the bug yourself and see if it can be fixed. ++


++

    ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++
Table 3.1: Bug symptoms and possible causes
SymptomCheck this:
When I open the encoder application from the desktop, a ++ window opens, error messages flash by, and the window closes before I ++ can read anything.lame is a commandline application. You need to start it ++ from a shell.
could not open input file ++
Have you given an input file to the encoder (see section ++ 2.1.5)? Does the input file exist? Is it readable? ++
++
could not open output file ++
Have you given an output file to the encoder (see section ++ 2.1.6)? Does the output directory exist and ++ is it writeable? Does a file of the same name exist and is it deleteable?
++
No parameters for this bitrate/samplerateDid you override any of the encoders parameters (stereo ++ mode, samplerate)? If so, try another samplerate.
bitrate too low/highLAME only allows bitrates ranging from 32 kBit/s to ++ 320 kBit/s.
The Layer-3 file sounds muffledTry using a higher bitrate. Try using a higher bandwidth ++ (see section 2.2.1). Try changing the stereo ++ mode.
The stereo image is destroyed.Try using the -m s  switch
++
++
++ ++ ++ +diff -r -c -N encoder/doc/html/node15.html lame3.13/doc/html/node15.html +*** encoder/doc/html/node15.html Wed Dec 31 17:00:00 1969 +--- lame3.13/doc/html/node15.html Fri May 21 14:32:28 1999 +*************** +*** 0 **** +--- 1,46 ---- ++ ++ ++ ++ Reporting the bug ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++
++

| Next: Sample bug report | Up: Troubleshooting | Previous: Is it really a | Table of Contents ++ |
++
++

++

Reporting the bug

++ To assist our programmers in the processing of your bug report, we ask you to ++ include in your mail ++
    ++
  • The version of the encoder you are using. ++
  • The operating system (name and version) you are running the software with. ++ If you are using a sort of UNIX, please cite the output of uname -a. ++ If you are using Windows, please right-click on the »My Computer« ++ icon that usually resides in the top left-hand corner of your screen and report ++ the lines following »System« and »Computer«. ++
  • The exact command line that you entered before you encountered the error. ++
  • The output of the encoder when appending the -S  ++ switch to the command line. ++
++

If you have gathered this information, please write an email to mp3encoder@geek.rcc.se . ++

  ++ ++ ++ +diff -r -c -N encoder/doc/html/node16.html lame3.13/doc/html/node16.html +*** encoder/doc/html/node16.html Wed Dec 31 17:00:00 1969 +--- lame3.13/doc/html/node16.html Fri May 21 14:32:38 1999 +*************** +*** 0 **** +--- 1,61 ---- ++ ++ ++ ++ Sample bug report ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++
++

| Next: References | Up: Troubleshooting | Previous: Reporting the bug | Table of Contents ++ |
++
++

++

Sample bug report

++ This is a sample bug report that you may use as a template for your own. ++

++ 

++ To: mp3encoder@geek.rcc.se 
++ Subject: LAME bug
++ 
++ Hello,
++ 
++ I am using LAME Version V3.x on a PC (according to
++ the System Properties dialog, it is running Microsoft Windows NT
++ 4.00.1381; the computer contains a x86 Family 5 Model 2 Stepping 12
++ AT/AT compatible and 64,951 KB RAM).
++ 
++ When I run the program as
++ 
++ lame -b 127957 -S myfile.wav foobar.mp3
++ 
++ I get the following error message:
++ 
++ LAME 3.03  (www.sulaco.org/mp3)
++ WAVE : Found WAV header. Assuming 44.1khz 16bit stereo samples
++ Encoding track.wav to track.mp3
++ Encoding as 44.1 kHz 128 kbps stereo MPEG-1 LayerIII file
++ using GPSYCHO: GPL'd psycho-acoustic model
++ Frame: [ 336/ 996]   Encoding Time 00:07/00:20  (1.30x)   ETA 00:13
++ 
++ ** lame error: Illegal codebook encountered.
++ 
++ Regards,
++   Hantan
++
++


++ ++ ++ +diff -r -c -N encoder/doc/html/node17.html lame3.13/doc/html/node17.html +*** encoder/doc/html/node17.html Wed Dec 31 17:00:00 1969 +--- lame3.13/doc/html/node17.html Fri May 21 14:42:00 1999 +*************** +*** 0 **** +--- 1,103 ---- ++ ++ ++ ++ References ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++
++

| Next: Index | Up: LAME V3 Next Generation High-End | Previous: Sample bug report | Table of Contents ++ |
++
++

++

References

++
++

++
1 ++
M. Bosi, K. Brandenburg, et al.
++ ISO/IEC MPEG-2 advanced audio coding.
++ In 101st AES conference, Los Angeles, November 1996.
++ preprint 4382. ++

++

++
2 ++
K. Brandenburg and M. Bosi.
++ Overview of MPEG-audio: Current and future standards for low bit-rate audio ++ coding.
++ In 99th AES conference, New York, October 1995.
++ preprint 4130. ++

++

++
3 ++
K. Brandenburg, G. Stoll, et al.
++ The ISO/MPEG-audio codec: A generic standard for coding of high quality digital ++ audio.
++ In Neil Gilchrist and Christer Grewin, editors, Collected Papers On Digital ++ Audio Bit-Rate Reduction, pages 31-42. AES, 1996. ++

++

++
4 ++
R. Buchta, S. Meltzer, et al.
++ The Worldstar sound format.
++ In 101st AES conference, Los Angeles, November 1996.
++ preprint 4385. ++

++

++
5 ++
S. Church, B. Grill, H. Popp, et al.
++ ISDN and ISO/MPEG layer-3 audio coding: Powerful new tools for broadcast and ++ audio production.
++ In 95th AES conference, Amsterdam, October 1993.
++ preprint 3743. ++

++

++
6 ++
M. Dietz, H. Popp, et al.
++ Audio compression for network transmission.
++ In 99th AES conference, New York, October 1995.
++ preprint 4129. ++

++

++
7 ++
E. Eberlein, H. Popp, et al.
++ Layer-3, a flexible coding standard.
++ In 94th AES conference, Berlin, March 1993.
++ preprint 3493. ++

++

++
8 ++
B. Grill, J. Herre, et al.
++ Improved MPEG-2 audio multi-channel encoding.
++ In 96th AES conference, Amsterdam, February 1994.
++ preprint 3865. ++

++

++
9 ++
J. Herre, K. Brandenburg, et al.
++ Second generation ISO/MPEG audio layer-3 coding.
++ In 98th AES conference, Paris, February 1995. ++

++

++
10 ++
Witte, M. Dietz, et al.
++ Single chip implementation of an ISO/MPEG layer-3 decoder.
++ In 96th AES conference, Amsterdam, February 1994.
++ preprint 3805. ++
++ ++ ++ +diff -r -c -N encoder/doc/html/node18.html lame3.13/doc/html/node18.html +*** encoder/doc/html/node18.html Wed Dec 31 17:00:00 1969 +--- lame3.13/doc/html/node18.html Fri May 21 14:42:18 1999 +*************** +*** 0 **** +--- 1,111 ---- ++ ++ ++ ++ Index ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++
++

| Up: LAME V3 Next Generation High-End | Previous: References | Table of Contents | ++
++
++


++

Index

++
++
audio compression ++
Introduction ++
audio decoding ++
Introduction ++
audio encoding ++
Introduction ++
bandwidth ++
Samplerate | Encoding ++ speed | Overriding default settings ++
bitrate ++
Compression ratios, bitrate and | Bitrate ++
bitstream ++
The two parts of ++
command-line switch ++
++
++
-b ++
Command line switch reference | bb | Bitrate ++
-k ++
Overriding default settings ++
-e ++
Command line switch reference | Tids ++ & bits | Is it really a ++
-a ++
Command line switch reference | Tids ++ & bits ++
-f ++
Command line switch reference | Tids ++ & bits | Tids & bits | Tids ++ & bits | Tids & bits ++
-s ++
Command line switch reference | plain ++ PCM data input ++
- ++
Command line switch reference | Piping ++ data into the ++
- ++
Command line switch reference | Streaming ++ data out of ++
-S ++
Command line switch reference | Reporting ++ the bug ++
++
decoding ++
The two parts of ++
downsampling ++
Samplerate ++
dual channel ++
Stereo mode ++
dual mono ++
Stereo mode ++
encoding ++
The two parts of ++
license ++
Why should you buy ++
low-pass ++
see bandwidth ++
sample rate ++
Samplerate | Overriding ++ default settings ++
++
effective ++
Overriding default settings ++
++
stereo ++
Stereo mode ++
++
intensity ++
Command line switch reference | Stereo ++ mode | Overriding default settings ++
mid-side ++
Stereo mode ++
MS ++
see stereo, mid-side ++
++
streaming ++
Input file specification | no ++ title ++
surround sound ++
Stereo mode ++

++

++
++ ++ ++ +diff -r -c -N encoder/doc/html/node5.html lame3.13/doc/html/node5.html +*** encoder/doc/html/node5.html Wed Dec 31 17:00:00 1969 +--- lame3.13/doc/html/node5.html Mon May 24 18:15:00 1999 +*************** +*** 0 **** +--- 1,54 ---- ++ ++ ++ ++ For the impatient ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++

For the impatient

++ If you are new to audio compression, you should read section 1.1 ++ for an introduction about audio compression and MPEG Layer-3. ++

If, however, you want to jump right into the business of sound compression, ++ then Section 1.2 will show you some prefabricated ++ command lines that will give you compressed audio streams right away. ++

If you are an expert in audio coding already, the command line switch reference ++ page might come in handy. ++


++

++   ++ ++
++ ++ ++ +diff -r -c -N encoder/doc/html/node6.html lame3.13/doc/html/node6.html +*** encoder/doc/html/node6.html Wed Dec 31 17:00:00 1969 +--- lame3.13/doc/html/node6.html Wed May 26 12:06:16 1999 +*************** +*** 0 **** +--- 1,133 ---- ++ ++ ++ ++ Introduction ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ Subsections ++ ++ ++
++

Introduction

++   There is a lot of confusion surrounding the terms ++  audio compression,  audio ++ encoding, and  audio decoding. This section ++ will give you an overview what audio coding (another one of these terms...) is ++ all about. ++

++

The purpose of audio compression ++

++

Up to the advent of audio compression, high-quality digital audio data took ++ a lot of hard disk space to store. Let us go through a short example. ++

You want to, say, sample your favorite 1-minute song and store it on your ++ harddisk. Because you want CD quality, you sample at 44.1 kHz, stereo, ++ with 16 bits per sample. ++

44100 Hz means that you have 44100 values per second coming in from your sound ++ card (or input file). Multiply that by two because you have two channels. Multiply ++ by another factor of two because you have two bytes per value (that's what 16 ++ bit means). The song will take up 44100 samples/s · 2 channels ++ · 2 bytes/sample · 60 s/min ~ 10 MBytes of storage space ++ on your harddisk. ++

If you wanted to download that over the internet, given an average 28.8 modem, ++ it would take you (at least) 10000000 bytes · 8 bits/byte / (28800 ++ bits/s) · / (60 s/min) ~ 45 minutes ++

Just to download one minute of music!
++

Digital audio coding, which - in this context - is synonymously called digital ++ audio compression as well, is the art of minimizing storage space (or channel ++ bandwidth) requirements for audio data. Modern perceptual audio coding techniques ++ (like MPEG Layer-3) exploit the properties of the human ear (the perception ++ of sound) to achieve a size reduction by a factor of 12 with little or no perceptible ++ loss of quality. ++

Therefore, such schemes are the key technology for high quality low bit-rate ++ applications, like soundtracks for CD-ROM games, solid-state sound memories, ++ Internet audio, digital audio broadcasting systems, and the like. ++

++

The two parts of audio compression ++

++

Audio compression really consists of two parts. The first part, called  encoding, ++ transforms the digital audio data that resides, say, in a WAVE file, into a ++ highly compressed form called  bitstream. To play ++ the bitstream on your soundcard, you need the second part, called  decoding. ++ Decoding takes the bitstream and re-expands it to a WAVE file. ++

The program that effects the first part is called an audio encoder. ++ LAME is such an encoder . The program that does the second part is called ++ an audio decoder. One well-known MPEG Layer-3 decoder is X11Amp, ++ another mpg123. Both can be found on http://www.mp3tech.org . ++

Compression ratios, bitrate and quality ++

++

It has not been explicitly mentioned up to now: What you end up with after ++ encoding and decoding is not the same sound file anymore: All superflous information ++ has been squeezed out, so to say. It is not the same file, but it will ++ sound the same - more or less, depending on how much compression had ++ been performed on it. ++

Generally speaking, the lower the compression ratio achieved, the better the ++ sound quality will be in the end - and vice versa. Table 1.1 ++ gives you an overview about quality achievable. ++

Because compression ratio is a somewhat unwieldy measure, experts use the ++ term  bitrate when speaking of the strength of ++ compression. Bitrate denotes the average number of bits that one second of audio ++ data will take up in your compressed bitstream. Usually the units used will ++ be kbps, which is kBits/s, or 1000 bits/s. ++ To calculate the number of bytes per second of audio data, simply divide the ++ number of bits per second by eight. ++


++

    ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++
Table 1.1: Bitrate versus sound quality
BitrateBandwidthQuality comparable to or better than
8 kBps2.5 kHzPOTS (telephone sound)
16 kBps4.5 kHzshortwave radio
32 kBps7.5 kHzAM radio
64 kBps11 kHzFM radio
128 kBps16 kHzCD
++
++
++ ++ ++ +diff -r -c -N encoder/doc/html/node7.html lame3.13/doc/html/node7.html +*** encoder/doc/html/node7.html Wed Dec 31 17:00:00 1969 +--- lame3.13/doc/html/node7.html Mon May 24 18:15:22 1999 +*************** +*** 0 **** +--- 1,32 ---- ++ ++ ++ ++ Some examples ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++

Some examples

++   ++
    ++
  • Encode a WAVE-file myfile.wav to a bitrate of 128000 bits/s, writing ++ to a plain bitstream myfile.mp3
    ++ lame -b 128000 myfile.wav myfile.mp3 ++
  • Encode a plain PCM stream (2-channel, 44.1 kHz) to a plain 56 kBit/s ++ Layer-3 stream, using the encoder as filter
    ++ lame -s 44100 -b 56000 -f - - ++
++ ++ ++ +diff -r -c -N encoder/doc/html/node8.html lame3.13/doc/html/node8.html +*** encoder/doc/html/node8.html Wed Dec 31 17:00:00 1969 +--- lame3.13/doc/html/node8.html Fri May 21 14:33:56 1999 +*************** +*** 0 **** +--- 1,100 ---- ++ ++ ++ ++ Command line switch reference ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++
++

| Next: Bugfixes and changes between | Up: For the impatient | Previous: Some examples | Table of Contents ++ |
++
++

++

Command line switch reference

++

  ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++
switchparametersee section
-b bitrate2.1.2
infile input file name 
outfile output file name 
-s sample rate in input file (only needed for raw file ++ input, 44.1 kHz is default)2.1.5
-x force byte-swapping of input2.1.5
- take input from stdin 
- write output into stdout 
-f extra fast2.2.2
-e CRC checksum error protection (not working)2.2.2
-a downmix stereo file to mono2.2.2
-S be verbose 
-no-is do not use  intensity stereo2.1.3
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++ ++ ++ +diff -r -c -N encoder/doc/html/node9.html lame3.13/doc/html/node9.html +*** encoder/doc/html/node9.html Wed Dec 31 17:00:00 1969 +--- lame3.13/doc/html/node9.html Fri May 21 14:34:08 1999 +*************** +*** 0 **** +--- 1,68 ---- ++ ++ ++ ++ Bugfixes and changes between V3.0 and V3.1 ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++
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Bugfixes and changes between V3.0 and ++ V3.04

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Bug descriptionStatus
V3.0 could not read little-endian raw PCM files.Fixed
V3.0 did not use a consistent syntax for raw PCM file processing.Fixed
V3.0 did not accept input/output file paths of more than ++ 100 characters.Fixed
V3.0 did not produce more than 13..16 kHz of bandwidth.Fixed
V3.0 documentation erroneously mentioned a -dual ++ switch. The switch did not, and does not, exist.Fixed
V3.0 sometimes deleted part of non-mono compatible signals ++ were left and right channel had a 180o phase shift.Fixed
Support for the IRIX mips-2 ABI has been dicontinued.Permanent
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++ ++ ++ +diff -r -c -N encoder/doc/html/switchs.html lame3.13/doc/html/switchs.html +*** encoder/doc/html/switchs.html Wed Dec 31 17:00:00 1969 +--- lame3.13/doc/html/switchs.html Wed May 26 18:43:02 1999 +*************** +*** 0 **** +--- 1,90 ---- ++ ++ ++ ++ Command line switch reference ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++

Command line switch reference

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switchparameter
-adownmix stereo file to mono
-bbitrate (32, 40, 48, 56, 64, 96, 112, 128, ++ 160, 192, 224, 256, 320)
-cmark track as copyrighted
-ede-emphasis (n, 5, c)
-fextra fast
-gturn on MP3x graphical frame analyser (may not be ++ available on your platform)
-huse experimental improvements (for internal testing)
-kuse full bandwidth
-mstereo mode (s, j, f, m)
-omark track as non-original
-Ouse original low-quality ISO psycho-acoustic model
-ssample rate in input file (only needed for raw file ++ input, 44.1 kHz is default)
-Sdon't print progress report
-xforce byte-swapping of input
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++ ++ ++ +diff -r -c -N encoder/doc/man/lame.1 lame3.13/doc/man/lame.1 +*** encoder/doc/man/lame.1 Wed Dec 31 17:00:00 1969 +--- lame3.13/doc/man/lame.1 Sat Jun 5 12:13:31 1999 +*************** +*** 0 **** +--- 1,115 ---- ++ .TH lame 1 "June 4, 1999" " " "LAME audio compressorMP3 " ++ .SH NAME ++ lame \- create mp3 audio files ++ .SH SYNOPSIS ++ .nf ++ lame ++ .fi ++ .SH DESCRIPTION ++ .PP ++ Lame is a program which can be used to create compressed audio files. ++ (Lame aint MP3 encoder). These audio files can be played back by popular ++ mp3 players such as mpg123 ++ ++ .SH OPTIONS ++ .TP ++ .BI -m " mode" ++ (s)tereo, (j)oint, (f)orce or (m)ono (default j) ++ force = force ms_stereo on all frames. Faster and ++ uses special Mid & Side masking thresholds ++ .TP ++ .BI -s " sample" " freq" ++ frequency of input file (khz) - default 44.1 ++ .TP ++ .BI -b " bitrate" ++ set the bitrate, default 128kbps ++ .TP ++ .BI -x ++ force byte-swapping of input ++ ++ .TP ++ .BI -a ++ autoconvert from stereo to mono file for mono encoding ++ ++ .TP ++ .BI -e " emp " ++ de-emphasis n/5/c (default n) ++ .TP ++ .BI -c ++ mark as copyright ++ .TP ++ .BI -o ++ mark as non-original (i.e. do not set the original bit) ++ .TP ++ .BI -f ++ fast mode (low quality) ++ .TP ++ .BI -k ++ disable sfb=21 cutoff ++ .TP ++ .BI -d ++ allow channels to have different blocktypes ++ .TP ++ .BI -v ++ use variable bitrate ++ .TP ++ .BI -V ++ quality setting for VBR. Default n=4, 0=high quality,bigger files. ++ 9=smaller files ++ .TP ++ .BI -h ++ slower, but (maybe) higher quality ++ .TP ++ .BI -g ++ run graphical analysis on . can also be a .mp3 file. ++ .TP ++ .BI -S ++ don't print progress report ++ .TP ++ .BI -O ++ use original (buggy) ISO psycho-acoustic model ++ ++ .SH EXAMPLES ++ .LP ++ Create an mp3 file from a wav file using the defaults. ++ .IP ++ .B lame ++ .I foo.wav foo.mp3 ++ .LP ++ Copy track 7 from a regular audio CD, using cdparanoia ++ ++ .IP ++ .B cdparanoia 7 foo.wav ; ++ .B lame foo.wav foo.mp3 ++ ++ .LP ++ Copy track 7 from a regular audio CD with cdda2wav, where the cdrom device is ++ device /dev/cdrom to an mp3 file foo.mp3 ++ ++ .IP ++ .B cdda2wav \-O ++ .I wav ++ .B \-t 7 \-D ++ .I /dev/cdrom ++ .I \- ++ .B | lame ++ .I \- foo.mp3 ++ ++ .IP ++ ++ ++ ++ .SH BUGS ++ .PP ++ Quality is very good, much better than the iso mp3 compressor, but still can improve. ++ .SH SEE ALSO ++ mpg123(1) ++ .SH AUTHORS ++ .nf ++ LAME originally developed by Mike Cheng ++ (www.uq.net.au/~zzmcheng) ++ GPSYCHO psycho-acoustic model by Mark Taylor ++ (www.sulaco.org/mp3). ++ mpglib by Michael Hipp ++ Manual page by William Schelter ++ .f +diff -r -c -N encoder/encode.c lame3.13/encode.c +*** encoder/encode.c Wed Jan 22 02:43:14 1997 +--- lame3.13/encode.c Thu Jun 17 02:00:59 1999 +*************** +*** 2,113 **** + * ISO MPEG Audio Subgroup Software Simulation Group (1996) + * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension + * +! * $Id: encode.c,v 1.1 1996/02/14 04:04:23 rowlands Exp $ + * + * $Log: encode.c,v $ + * Revision 1.1 1996/02/14 04:04:23 rowlands + * Initial revision + * + * Received from Mike Coleman + **********************************************************************/ +! /********************************************************************** +! * date programmers comment * +! * 3/01/91 Douglas Wong, start of version 1.1 records * +! * Davis Pan * +! * 3/06/91 Douglas Wong rename: setup.h to endef.h * +! * efilter to enfilter * +! * ewindow to enwindow * +! * integrated "quantizer", "scalefactor",* +! * and "transmission" files * +! * update routine "window_subband" * +! * 3/31/91 Bill Aspromonte replaced read_filter by * +! * create_an_filter * +! * 5/10/91 W. Joseph Carter Ported to Macintosh and Unix. * +! * Incorporated Jean-Georges Fritsch's * +! * "bitstream.c" package. * +! * Incorporated Bill Aspromonte's * +! * filterbank coefficient matrix * +! * calculation routines and added * +! * roundoff to coincide with specs. * +! * Modified to strictly adhere to * +! * encoded bitstream specs, including * +! * "Berlin changes". * +! * Modified PCM sound file handling to * +! * process all incoming samples and fill * +! * out last encoded frame with zeros * +! * (silence) if needed. * +! * Located and fixed numerous software * +! * bugs and table data errors. * +! * 19jun91 dpwe (Aware) moved "alloc_*" reader to common.c * +! * Globals sblimit, alloc replaced by new* +! * struct 'frame_params' passed as arg. * +! * Added JOINT STEREO coding, layers I,II* +! * Affects: *_bit_allocation, * +! * subband_quantization, encode_bit_alloc* +! * sample_encoding * +! * 6/10/91 Earle Jennings modified II_subband_quantization to * +! * resolve type cast problem for MS_DOS * +! * 6/11/91 Earle Jennings modified to avoid overflow on MS_DOS * +! * in routine filter_subband * +! * 7/10/91 Earle Jennings port to MsDos from MacIntosh version * +! * 8/ 8/91 Jens Spille Change for MS-C6.00 * +! *10/ 1/91 S.I. Sudharsanan, Ported to IBM AIX platform. * +! * Don H. Lee, * +! * Peter W. Farrett * +! *10/ 3/91 Don H. Lee implemented CRC-16 error protection * +! * newly introduced function encode_CRC * +! *11/ 8/91 Kathy Wang Documentation of code * +! * All variablenames are referred to * +! * with surrounding pound (#) signs * +! * 2/11/92 W. Joseph Carter Ported new code to Macintosh. Most * +! * important fixes involved changing * +! * 16-bit ints to long or unsigned in * +! * bit alloc routines for quant of 65535 * +! * and passing proper function args. * +! * Removed "Other Joint Stereo" option * +! * and made bitrate be total channel * +! * bitrate, irrespective of the mode. * +! * Fixed many small bugs & reorganized. * +! * 6/16/92 Shaun Astarabadi Changed I_scale_factor_calc() and * +! * II_scale_factor_calc() to use scale * +! * factor 0 thru 62 only and not to * +! * encode index 63 into the bit stream. * +! * 7/27/92 Mike Li (re-)Port to MS-DOS * +! * 9/22/92 jddevine@aware.com Fixed _scale_factor_calc() defs * +! * 3/31/93 Giogio Dimino changed II_a_bit_allocation() from: * +! * if( ad > ...) to if(ad >= ...) * +! * 8/05/93 TEST changed I_a_bit_allocation() from: * +! * if( ad > ...) to if(ad >= ...) * +! * 8/02/95 mc@fivebats.com Changed audio file reading code to * +! * read samples big-endian * +! *10/15/95 mc@fivebats.com Modified get_audio() for layer3-LSF * +! **********************************************************************/ + + #include "common.h" + #include "encoder.h" + +! #ifdef MS_DOS +! extern unsigned _stklen = 16384; +! #endif +! +! +! /*=======================================================================\ +! | | +! | This segment contains all the core routines of the encoder, | +! | except for the psychoacoustic models. | +! | | +! | The user can select either one of the two psychoacoustic | +! | models. Model I is a simple tonal and noise masking threshold | +! | generator, and Model II is a more sophisticated cochlear masking | +! | threshold generator. Model I is recommended for lower complexity | +! | applications whereas Model II gives better subjective quality at low | +! | bit rates. | +! | | +! | Layers I and II of mono, stereo, and joint stereo modes are supported.| +! | Routines associated with a given layer are prefixed by "I_" for layer | +! | 1 and "II_" for layer 2. | +! \=======================================================================*/ +! + /************************************************************************ + * + * read_samples() +--- 2,26 ---- + * ISO MPEG Audio Subgroup Software Simulation Group (1996) + * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension + * +! * $Id: encode.c,v 1.2 1998/10/05 17:06:47 larsi Exp $ + * + * $Log: encode.c,v $ ++ * Revision 1.2 1998/10/05 17:06:47 larsi ++ * *** empty log message *** ++ * ++ * Revision 1.1.1.1 1998/10/05 14:47:17 larsi ++ * + * Revision 1.1 1996/02/14 04:04:23 rowlands + * Initial revision + * + * Received from Mike Coleman + **********************************************************************/ +! + + #include "common.h" + #include "encoder.h" + +! #ifndef LIBSNDFILE + /************************************************************************ + * + * read_samples() +*************** +*** 128,133 **** +--- 41,47 ---- + unsigned long samples_read; + static unsigned long samples_to_read; + static char init = TRUE; ++ extern int swapbytes; /* if TRUE then force swapping of byte order */ + + if (init) { + samples_to_read = num_samples; +*************** +*** 139,145 **** + samples_read = samples_to_read; + if ((samples_read = + fread(sample_buffer, sizeof(short), (int)samples_read, musicin)) == 0) +! printf("Hit end of audio data\n"); + /* + Samples are big-endian. If this is a little-endian machine + we must swap +--- 53,59 ---- + samples_read = samples_to_read; + if ((samples_read = + fread(sample_buffer, sizeof(short), (int)samples_read, musicin)) == 0) +! { /* printf("Hit end of audio data\n"); */ } + /* + Samples are big-endian. If this is a little-endian machine + we must swap +*************** +*** 153,164 **** + exit( 1 ); + } + } +! if ( NativeByteOrder == order_littleEndian ) + SwapBytesInWords( sample_buffer, samples_read ); + + samples_to_read -= samples_read; + if (samples_read < frame_size && samples_read > 0) { +! printf("Insufficient PCM input for one frame - fillout with zeros\n"); + for (; samples_read < frame_size; sample_buffer[samples_read++] = 0); + samples_to_read = 0; + } +--- 67,84 ---- + exit( 1 ); + } + } +! if (!iswav && ( NativeByteOrder == order_littleEndian )) +! SwapBytesInWords( sample_buffer, samples_read ); +! +! if (iswav && ( NativeByteOrder == order_bigEndian )) +! SwapBytesInWords( sample_buffer, samples_read ); +! +! if (swapbytes==TRUE) + SwapBytesInWords( sample_buffer, samples_read ); + + samples_to_read -= samples_read; + if (samples_read < frame_size && samples_read > 0) { +! /* printf("Insufficient PCM input for one frame - fillout with zeros\n"); */ + for (; samples_read < frame_size; sample_buffer[samples_read++] = 0); + samples_to_read = 0; + } +*************** +*** 176,195 **** + * + ************************************************************************/ + +! unsigned long get_audio( musicin, buffer, num_samples, stereo, info ) + FILE *musicin; + short FAR buffer[2][1152]; + unsigned long num_samples; + int stereo; + layer *info; + { + int j; +! short insamp[2304]; + unsigned long samples_read; + int lay; + lay = info->lay; + +! if ( (lay == 3) && (info->version == 0) ) + { + if ( stereo == 2 ) + { +--- 96,123 ---- + * + ************************************************************************/ + +! unsigned long get_audio( musicin, bufferL,bufferR, num_samples, stereo, info ) + FILE *musicin; ++ /* + short FAR buffer[2][1152]; ++ */ ++ short FAR bufferL[1152],bufferR[1152]; + unsigned long num_samples; + int stereo; + layer *info; + { + int j; +! short insamp[3176]; + unsigned long samples_read; + int lay; ++ extern int autoconvert; ++ + lay = info->lay; + +! /* pad with zeros in case we hit EOF */ +! memset((char *) insamp, 0, sizeof(insamp)); +! +! if ( (lay == 3) && (info->version == 0) ) /* ie MPEG-2 LSF */ + { + if ( stereo == 2 ) + { +*************** +*** 197,204 **** + (unsigned long) 1152 ); + for ( j = 0; j < 576; j++ ) + { +! buffer[0][j] = insamp[2 * j]; +! buffer[1][j] = insamp[2 * j + 1]; + } + } + else +--- 125,132 ---- + (unsigned long) 1152 ); + for ( j = 0; j < 576; j++ ) + { +! bufferL[j] = insamp[2 * j]; +! bufferR[j] = insamp[2 * j + 1]; + } + } + else +*************** +*** 207,310 **** + (unsigned long) 576 ); + for ( j = 0; j < 576; j++ ) + { +! buffer[0][j] = insamp[j]; +! buffer[1][j] = 0; + } + } + } + else +! { +! if (lay == 1){ +! if(stereo == 2){ /* layer 1, stereo */ +! samples_read = read_samples(musicin, insamp, num_samples, +! (unsigned long) 768); +! for(j=0;j<448;j++) { +! if(j<64) { +! buffer[0][j] = buffer[0][j+384]; +! buffer[1][j] = buffer[1][j+384]; +! } +! else { +! buffer[0][j] = insamp[2*j-128]; +! buffer[1][j] = insamp[2*j-127]; +! } +! } +! } +! else { /* layer 1, mono */ +! samples_read = read_samples(musicin, insamp, num_samples, +! (unsigned long) 384); +! for(j=0;j<448;j++){ +! if(j<64) { +! buffer[0][j] = buffer[0][j+384]; +! buffer[1][j] = 0; +! } +! else { +! buffer[0][j] = insamp[j-64]; +! buffer[1][j] = 0; +! } +! } +! } +! } +! else { + if(stereo == 2){ /* layer 2 (or 3), stereo */ +! samples_read = read_samples(musicin, insamp, num_samples, +! (unsigned long) 2304); +! for(j=0;j<1152;j++) { +! buffer[0][j] = insamp[2*j]; +! buffer[1][j] = insamp[2*j+1]; +! } + } + else { /* layer 2 (or 3), mono */ + samples_read = read_samples(musicin, insamp, num_samples, + (unsigned long) 1152); + for(j=0;j<1152;j++){ +! buffer[0][j] = insamp[j]; +! buffer[1][j] = 0; + } + } + } +- } + return(samples_read); + } +- +- /************************************************************************ +- * +- * read_ana_window() +- * +- * PURPOSE: Reads encoder window file "enwindow" into array #ana_win# +- * +- ************************************************************************/ +- +- void read_ana_window(ana_win) +- double FAR ana_win[HAN_SIZE]; +- { +- int i,j[4]; +- FILE *fp; +- double f[4]; +- char t[150]; +- +- if (!(fp = OpenTableFile("enwindow") ) ) { +- printf("Please check analysis window table 'enwindow'\n"); +- exit(1); +- } +- for (i=0;i<512;i+=4) { +- fgets(t, 150, fp); +- sscanf(t,"C[%d] = %lf C[%d] = %lf C[%d] = %lf C[%d] = %lf\n", +- j, f,j+1,f+1,j+2,f+2,j+3,f+3); +- if (i==j[0]) { +- ana_win[i] = f[0]; +- ana_win[i+1] = f[1]; +- ana_win[i+2] = f[2]; +- ana_win[i+3] = f[3]; +- } +- else { +- printf("Check index in analysis window table\n"); +- exit(1); +- } +- fgets(t,150,fp); +- } +- fclose(fp); +- } + + /************************************************************************ + * + * window_subband() +--- 135,181 ---- + (unsigned long) 576 ); + for ( j = 0; j < 576; j++ ) + { +! bufferL[j] = insamp[j]; +! bufferR[j] = 0; + } + } + } + else +! /* MPEG 1 */ +! { + if(stereo == 2){ /* layer 2 (or 3), stereo */ +! samples_read = read_samples(musicin, insamp, num_samples, +! (unsigned long) 2304); +! for(j=0;j<1152;j++) { +! bufferL[j] = insamp[2*j]; +! bufferR[j] = insamp[2*j+1]; +! +! } + } + else { /* layer 2 (or 3), mono */ ++ if (autoconvert==TRUE) { /* downconvert from a stereo file into a mono buffer */ ++ samples_read = read_samples(musicin, insamp, num_samples, ++ (unsigned long) 2304); ++ for(j=0;j<1152;j++){ ++ bufferL[j] = insamp[2*j]; ++ bufferR[j] = 0; ++ } ++ } ++ else { + samples_read = read_samples(musicin, insamp, num_samples, + (unsigned long) 1152); + for(j=0;j<1152;j++){ +! bufferL[j] = insamp[j]; +! bufferR[j] = 0; + } ++ } + } + } + return(samples_read); + } + ++ #endif ++ + /************************************************************************ + * + * window_subband() +*************** +*** 318,349 **** + * windowed sample #z# + * + ************************************************************************/ +! + void window_subband(buffer, z, k) +! short FAR **buffer; +! double FAR z[HAN_SIZE]; + int k; + { + typedef double FAR XX[2][HAN_SIZE]; + static XX FAR *x; +! int i, j; +! static off[2] = {0,0}; + static char init = 0; +- static double FAR *c; + if (!init) { +- c = (double FAR *) mem_alloc(sizeof(double) * HAN_SIZE, "window"); +- read_ana_window(c); + x = (XX FAR *) mem_alloc(sizeof(XX),"x"); +! for (i=0;i<2;i++) +! for (j=0;j= 0) +- modf(filter[i][k]+0.5, &filter[i][k]); +- else +- modf(filter[i][k]-0.5, &filter[i][k]); +- filter[i][k] *= 1e-9; +- } +- } +- +- /************************************************************************ +- * + * filter_subband() + * + * PURPOSE: Calculates the analysis filter bank coefficients +--- 221,226 ---- +*************** +*** 390,425 **** + * them by the filter matrix, producing 32 subband samples. + * + ************************************************************************/ +! + void filter_subband(z,s) + double FAR z[HAN_SIZE], s[SBLIMIT]; + { +! double y[64]; +! int i,j; +! static char init = 0; +! typedef double MM[SBLIMIT][64]; +! static MM FAR *m; +! #ifdef MS_DOS +! long SIZE_OF_MM; +! SIZE_OF_MM = SBLIMIT*64; +! SIZE_OF_MM *= 8; +! if (!init) { +! m = (MM FAR *) mem_alloc(SIZE_OF_MM, "filter"); +! create_ana_filter(*m); +! init = 1; +! } +! #else +! if (!init) { +! m = (MM FAR *) mem_alloc(sizeof(MM), "filter"); +! create_ana_filter(*m); +! init = 1; + } +- #endif +- for (i=0;i<64;i++) for (j=0, y[i] = 0;j<8;j++) y[i] += z[i+64*j]; +- for (i=0;i 0) ? a : -a; + } + +- /************************************************************************ +- * +- * I_combine_LR (Layer I) +- * II_combine_LR (Layer II) +- * +- * PURPOSE:Combines left and right channels into a mono channel +- * +- * SEMANTICS: The average of left and right subband samples is put into +- * #joint_sample# +- * +- * Layer I and II differ in frame length and # subbands used +- * +- ************************************************************************/ +- +- void I_combine_LR(sb_sample, joint_sample) +- double FAR sb_sample[2][3][SCALE_BLOCK][SBLIMIT]; +- double FAR joint_sample[3][SCALE_BLOCK][SBLIMIT]; +- { /* make a filtered mono for joint stereo */ +- int sb, smp; +- +- for(sb = 0; sb s[i]) +- s[i] = mod(sb_sample[k][0][j][i]); +- +- for (i=0;i=0;j--) /* $A 6/16/92 */ +- if (s[i] <= multiple[j]) { +- scalar[k][0][i] = j; +- break; +- } +- } +- } +- +- /******************************** Layer II ******************************/ +- +- void II_scale_factor_calc(sb_sample,scalar,stereo,sblimit) +- double FAR sb_sample[][3][SCALE_BLOCK][SBLIMIT]; +- unsigned int scalar[][3][SBLIMIT]; +- int stereo,sblimit; +- { +- int i,j, k,t; +- double s[SBLIMIT]; +- +- for (k=0;k s[i]) +- s[i] = mod(sb_sample[k][t][j][i]); +- +- for (i=0;i=0;j--) /* $A 6/16/92 */ +- if (s[i] <= multiple[j]) { +- scalar[k][t][i] = j; +- break; +- } +- for (i=sblimit;istereo; +- int sblimit = fr_ps->sblimit; +- +- for (k=0;k scalar[k][j][i]) max = scalar[k][j][i]; +- for (i=sblimit;istereo; +- int sblimit = fr_ps->sblimit; +- +- for (k=0;kstereo; +- int sblimit = fr_ps->sblimit; +- int dscf[2]; +- int class[2],i,j,k; +- static int pattern[5][5] = {0x123, 0x122, 0x122, 0x133, 0x123, +- 0x113, 0x111, 0x111, 0x444, 0x113, +- 0x111, 0x111, 0x111, 0x333, 0x113, +- 0x222, 0x222, 0x222, 0x333, 0x123, +- 0x123, 0x122, 0x122, 0x133, 0x123}; +- +- for (k=0;k -3 && dscf[j] <0) class[j] = 1; +- else if (dscf[j] == 0) class[j] = 2; +- else if (dscf[j] > 0 && dscf[j] < 3) class[j] = 3; +- else class[j] = 4; +- } +- switch (pattern[class[0]][class[1]]) { +- case 0x123 : scfsi[k][i] = 0; +- break; +- case 0x122 : scfsi[k][i] = 3; +- scalar[k][2][i] = scalar[k][1][i]; +- break; +- case 0x133 : scfsi[k][i] = 3; +- scalar[k][1][i] = scalar[k][2][i]; +- break; +- case 0x113 : scfsi[k][i] = 1; +- scalar[k][1][i] = scalar[k][0][i]; +- break; +- case 0x111 : scfsi[k][i] = 2; +- scalar[k][1][i] = scalar[k][2][i] = scalar[k][0][i]; +- break; +- case 0x222 : scfsi[k][i] = 2; +- scalar[k][0][i] = scalar[k][2][i] = scalar[k][1][i]; +- break; +- case 0x333 : scfsi[k][i] = 2; +- scalar[k][0][i] = scalar[k][1][i] = scalar[k][2][i]; +- break; +- case 0x444 : scfsi[k][i] = 2; +- if (scalar[k][0][i] > scalar[k][2][i]) +- scalar[k][0][i] = scalar[k][2][i]; +- scalar[k][1][i] = scalar[k][2][i] = scalar[k][0][i]; +- } +- } +- } +- +- /************************************************************************ +- * +- * I_encode_scale (Layer I) +- * II_encode_scale (Layer II) +- * +- * PURPOSE:The encoded scalar factor information is arranged and +- * queued into the output fifo to be transmitted. +- * +- * For Layer II, the three scale factors associated with +- * a given subband and channel are transmitted in accordance +- * with the scfsi, which is transmitted first. +- * +- ************************************************************************/ +- +- void I_encode_scale(scalar, bit_alloc, fr_ps, bs) +- unsigned int scalar[2][3][SBLIMIT]; +- unsigned int bit_alloc[2][SBLIMIT]; +- frame_params *fr_ps; +- Bit_stream_struc *bs; +- { +- int stereo = fr_ps->stereo; +- int sblimit = fr_ps->sblimit; +- int i,j; +- +- for (i=0;istereo; +- int sblimit = fr_ps->sblimit; +- int jsbound = fr_ps->jsbound; +- int i,j,k; +- +- for (i=0;istereo; +- int sblimit = fr_ps->sblimit; +- int jsbound = fr_ps->jsbound; +- int req_bits = 0; +- +- /* initial b_anc (header) allocation bits */ +- req_bits = 32 + 4 * ( (jsbound * stereo) + (SBLIMIT-jsbound) ); +- +- for(i=0; i= NOISY_MIN_MNR) +- break; /* we found enough bits */ +- if(stereo == 2 && i >= jsbound) /* check other JS channel */ +- for(;k<14; ++k) +- if( (-perm_smr[1-j][i] + snr[k]) >= NOISY_MIN_MNR) break; +- if(k>0) req_bits += (k+1)*SCALE_BLOCK + 6*((i>=jsbound)?stereo:1); +- } +- return req_bits; +- } +- +- /***************************** Layer II ********************************/ +- +- int II_bits_for_nonoise(perm_smr, scfsi, fr_ps) +- double FAR perm_smr[2][SBLIMIT]; +- unsigned int scfsi[2][SBLIMIT]; +- frame_params *fr_ps; +- { +- int sb,ch,ba; +- int stereo = fr_ps->stereo; +- int sblimit = fr_ps->sblimit; +- int jsbound = fr_ps->jsbound; +- al_table *alloc = fr_ps->alloc; +- int req_bits = 0, bbal = 0, berr = 0, banc = 32; +- int maxAlloc, sel_bits, sc_bits, smp_bits; +- static int sfsPerScfsi[] = { 3,2,1,2 }; /* lookup # sfs per scfsi */ +- +- /* added 92-08-11 shn */ +- if (fr_ps->header->error_protection) berr=16; else berr=0; +- +- for (sb=0; sb0)?1:0)]) +- >= NOISY_MIN_MNR) +- break; /* we found enough bits */ +- if(stereo == 2 && sb >= jsbound) /* check other JS channel */ +- for(;ba0)?1:0)]) +- >= NOISY_MIN_MNR) +- break; +- if(ba>0) { +- smp_bits = SCALE_BLOCK * ((*alloc)[sb][ba].group * (*alloc)[sb][ba].bits); +- /* scale factor bits required for subband */ +- sel_bits = 2; +- sc_bits = 6 * sfsPerScfsi[scfsi[ch][sb]]; +- if(stereo == 2 && sb >= jsbound) { +- /* each new js sb has L+R scfsis */ +- sel_bits += 2; +- sc_bits += 6 * sfsPerScfsi[scfsi[1-ch][sb]]; +- } +- req_bits += smp_bits+sel_bits+sc_bits; +- } +- } +- return req_bits; +- } +- +- /************************************************************************ +- * +- * I_main_bit_allocation (Layer I) +- * II_main_bit_allocation (Layer II) +- * +- * PURPOSE:For joint stereo mode, determines which of the 4 joint +- * stereo modes is needed. Then calls *_a_bit_allocation(), which +- * allocates bits for each of the subbands until there are no more bits +- * left, or the MNR is at the noise/no_noise threshold. +- * +- * SEMANTICS: +- * +- * For joint stereo mode, joint stereo is changed to stereo if +- * there are enough bits to encode stereo at or better than the +- * no-noise threshold (NOISY_MIN_MNR). Otherwise, the system +- * iteratively allocates less bits by using joint stereo until one +- * of the following occurs: +- * - there are no more noisy subbands (MNR >= NOISY_MIN_MNR) +- * - mode_ext has been reduced to 0, which means that all but the +- * lowest 4 subbands have been converted from stereo to joint +- * stereo, and no more subbands may be converted +- * +- * This function calls *_bits_for_nonoise() and *_a_bit_allocation(). +- * +- ************************************************************************/ +- +- void I_main_bit_allocation(perm_smr, bit_alloc, adb, fr_ps) +- double FAR perm_smr[2][SBLIMIT]; +- unsigned int bit_alloc[2][SBLIMIT]; +- int *adb; +- frame_params *fr_ps; +- { +- int noisy_sbs; +- int mode, mode_ext, lay, i; +- int rq_db, av_db = *adb; +- static int init = 0; +- +- if(init == 0) { +- /* rearrange snr for layer I */ +- snr[2] = snr[3]; +- for (i=3;i<16;i++) snr[i] = snr[i+2]; +- init = 1; +- } +- +- if((mode = fr_ps->actual_mode) == MPG_MD_JOINT_STEREO) { +- fr_ps->header->mode = MPG_MD_STEREO; +- fr_ps->header->mode_ext = 0; +- fr_ps->jsbound = fr_ps->sblimit; +- if(rq_db = I_bits_for_nonoise(perm_smr, fr_ps) > *adb) { +- fr_ps->header->mode = MPG_MD_JOINT_STEREO; +- mode_ext = 4; /* 3 is least severe reduction */ +- lay = fr_ps->header->lay; +- do { +- --mode_ext; +- fr_ps->jsbound = js_bound(lay, mode_ext); +- rq_db = I_bits_for_nonoise(perm_smr, fr_ps); +- } while( (rq_db > *adb) && (mode_ext > 0)); +- fr_ps->header->mode_ext = mode_ext; +- } /* well we either eliminated noisy sbs or mode_ext == 0 */ +- } +- noisy_sbs = I_a_bit_allocation(perm_smr, bit_alloc, adb, fr_ps); +- } +- +- /***************************** Layer II ********************************/ +- +- void II_main_bit_allocation(perm_smr, scfsi, bit_alloc, adb, fr_ps) +- double FAR perm_smr[2][SBLIMIT]; +- unsigned int scfsi[2][SBLIMIT]; +- unsigned int bit_alloc[2][SBLIMIT]; +- int *adb; +- frame_params *fr_ps; +- { +- int noisy_sbs, nn; +- int mode, mode_ext, lay; +- int rq_db, av_db = *adb; +- +- if((mode = fr_ps->actual_mode) == MPG_MD_JOINT_STEREO) { +- fr_ps->header->mode = MPG_MD_STEREO; +- fr_ps->header->mode_ext = 0; +- fr_ps->jsbound = fr_ps->sblimit; +- if((rq_db=II_bits_for_nonoise(perm_smr, scfsi, fr_ps)) > *adb) { +- fr_ps->header->mode = MPG_MD_JOINT_STEREO; +- mode_ext = 4; /* 3 is least severe reduction */ +- lay = fr_ps->header->lay; +- do { +- --mode_ext; +- fr_ps->jsbound = js_bound(lay, mode_ext); +- rq_db = II_bits_for_nonoise(perm_smr, scfsi, fr_ps); +- } while( (rq_db > *adb) && (mode_ext > 0)); +- fr_ps->header->mode_ext = mode_ext; +- } /* well we either eliminated noisy sbs or mode_ext == 0 */ +- } +- noisy_sbs = II_a_bit_allocation(perm_smr, scfsi, bit_alloc, adb, fr_ps); +- } +- +- /************************************************************************ +- * +- * I_a_bit_allocation (Layer I) +- * II_a_bit_allocation (Layer II) +- * +- * PURPOSE:Adds bits to the subbands with the lowest mask-to-noise +- * ratios, until the maximum number of bits for the subband has +- * been allocated. +- * +- * SEMANTICS: +- * 1. Find the subband and channel with the smallest MNR (#min_sb#, +- * and #min_ch#) +- * 2. Calculate the increase in bits needed if we increase the bit +- * allocation to the next higher level +- * 3. If there are enough bits available for increasing the resolution +- * in #min_sb#, #min_ch#, and the subband has not yet reached its +- * maximum allocation, update the bit allocation, MNR, and bits +- available accordingly +- * 4. Repeat until there are no more bits left, or no more available +- * subbands. (A subband is still available until the maximum +- * number of bits for the subband has been allocated, or there +- * aren't enough bits to go to the next higher resolution in the +- subband.) +- * +- ************************************************************************/ +- +- int I_a_bit_allocation(perm_smr, bit_alloc, adb, fr_ps) /* return noisy sbs */ +- double FAR perm_smr[2][SBLIMIT]; +- unsigned int bit_alloc[2][SBLIMIT]; +- int *adb; +- frame_params *fr_ps; +- { +- int i, k, smpl_bits, scale_bits, min_sb, min_ch, oth_ch; +- int bspl, bscf, ad, noisy_sbs, done = 0, bbal ; +- double mnr[2][SBLIMIT], small; +- char used[2][SBLIMIT]; +- int stereo = fr_ps->stereo; +- int sblimit = fr_ps->sblimit; +- int jsbound = fr_ps->jsbound; +- al_table *alloc = fr_ps->alloc; +- static char init= 0; +- static int banc=32, berr=0; +- +- if (!init) { +- init = 1; +- if (fr_ps->header->error_protection) berr = 16; /* added 92-08-11 shn */ +- } +- bbal = 4 * ( (jsbound * stereo) + (SBLIMIT-jsbound) ); +- *adb -= bbal + berr + banc; +- ad= *adb; +- +- for (i=0;i mnr[k][i]) { +- small = mnr[k][i]; +- min_sb = i; min_ch = k; +- } +- if(min_sb > -1) { /* there was something to find */ +- /* first step of bit allocation is biggest */ +- if (used[min_ch][min_sb]) { smpl_bits = SCALE_BLOCK; scale_bits = 0; } +- else { smpl_bits = 24; scale_bits = 6; } +- if(min_sb >= jsbound) scale_bits *= stereo; +- +- /* check to see enough bits were available for */ +- /* increasing resolution in the minimum band */ +- +- if (ad >= bspl + bscf + scale_bits + smpl_bits) { +- bspl += smpl_bits; /* bit for subband sample */ +- bscf += scale_bits; /* bit for scale factor */ +- bit_alloc[min_ch][min_sb]++; +- used[min_ch][min_sb] = 1; /* subband has bits */ +- mnr[min_ch][min_sb] = -perm_smr[min_ch][min_sb] +- + snr[bit_alloc[min_ch][min_sb]]; +- /* Check if subband has been fully allocated max bits */ +- if (bit_alloc[min_ch][min_sb] == 14 ) used[min_ch][min_sb] = 2; +- } +- else /* no room to improve this band */ +- used[min_ch][min_sb] = 2; /* for allocation anymore */ +- if(stereo == 2 && min_sb >= jsbound) { +- oth_ch = 1-min_ch; /* joint-st : fix other ch */ +- bit_alloc[oth_ch][min_sb] = bit_alloc[min_ch][min_sb]; +- used[oth_ch][min_sb] = used[min_ch][min_sb]; +- mnr[oth_ch][min_sb] = -perm_smr[oth_ch][min_sb] +- + snr[bit_alloc[oth_ch][min_sb]]; +- } +- } +- } while(min_sb>-1); /* i.e. still some sub-bands to find */ +- +- /* Calculate the number of bits left, add on to pointed var */ +- ad -= bspl+bscf; +- *adb = ad; +- +- /* see how many channels are noisy */ +- noisy_sbs = 0; small = mnr[0][0]; +- for(k=0; k mnr[k][i]) small = mnr[k][i]; +- } +- } +- return noisy_sbs; +- } +- +- /***************************** Layer II ********************************/ +- +- int II_a_bit_allocation(perm_smr, scfsi, bit_alloc, adb, fr_ps) +- double FAR perm_smr[2][SBLIMIT]; +- unsigned int scfsi[2][SBLIMIT]; +- unsigned int bit_alloc[2][SBLIMIT]; +- int *adb; +- frame_params *fr_ps; +- { +- int i, min_ch, min_sb, oth_ch, k, increment, scale, seli, ba; +- int bspl, bscf, bsel, ad, noisy_sbs, bbal=0; +- double mnr[2][SBLIMIT], small; +- char used[2][SBLIMIT]; +- int stereo = fr_ps->stereo; +- int sblimit = fr_ps->sblimit; +- int jsbound = fr_ps->jsbound; +- al_table *alloc = fr_ps->alloc; +- static char init= 0; +- static int banc=32, berr=0; +- static int sfsPerScfsi[] = { 3,2,1,2 }; /* lookup # sfs per scfsi */ +- +- if (!init) { +- init = 1; +- if (fr_ps->header->error_protection) berr=16; /* added 92-08-11 shn */ +- } +- for (i=0; i mnr[k][i]) { +- small = mnr[k][i]; +- min_sb = i; min_ch = k; +- } +- if(min_sb > -1) { /* there was something to find */ +- /* find increase in bit allocation in subband [min] */ +- increment = SCALE_BLOCK * ((*alloc)[min_sb][bit_alloc[min_ch][min_sb]+1].group * +- (*alloc)[min_sb][bit_alloc[min_ch][min_sb]+1].bits); +- if (used[min_ch][min_sb]) +- increment -= SCALE_BLOCK * ((*alloc)[min_sb][bit_alloc[min_ch][min_sb]].group* +- (*alloc)[min_sb][bit_alloc[min_ch][min_sb]].bits); +- +- /* scale factor bits required for subband [min] */ +- oth_ch = 1 - min_ch; /* above js bound, need both chans */ +- if (used[min_ch][min_sb]) scale = seli = 0; +- else { /* this channel had no bits or scfs before */ +- seli = 2; +- scale = 6 * sfsPerScfsi[scfsi[min_ch][min_sb]]; +- if(stereo == 2 && min_sb >= jsbound) { +- /* each new js sb has L+R scfsis */ +- seli += 2; +- scale += 6 * sfsPerScfsi[scfsi[oth_ch][min_sb]]; +- } +- } +- /* check to see enough bits were available for */ +- /* increasing resolution in the minimum band */ +- if (ad >= bspl + bscf + bsel + seli + scale + increment) { +- ba = ++bit_alloc[min_ch][min_sb]; /* next up alloc */ +- bspl += increment; /* bits for subband sample */ +- bscf += scale; /* bits for scale factor */ +- bsel += seli; /* bits for scfsi code */ +- used[min_ch][min_sb] = 1; /* subband has bits */ +- mnr[min_ch][min_sb] = -perm_smr[min_ch][min_sb] + +- snr[(*alloc)[min_sb][ba].quant+1]; +- /* Check if subband has been fully allocated max bits */ +- if (ba >= (1<<(*alloc)[min_sb][0].bits)-1) used[min_ch][min_sb] = 2; +- } +- else used[min_ch][min_sb] = 2; /* can't increase this alloc */ +- if(min_sb >= jsbound && stereo == 2) { +- /* above jsbound, alloc applies L+R */ +- ba = bit_alloc[oth_ch][min_sb] = bit_alloc[min_ch][min_sb]; +- used[oth_ch][min_sb] = used[min_ch][min_sb]; +- mnr[oth_ch][min_sb] = -perm_smr[oth_ch][min_sb] + +- snr[(*alloc)[min_sb][ba].quant+1]; +- } +- } +- } while(min_sb > -1); /* until could find no channel */ +- /* Calculate the number of bits left */ +- ad -= bspl+bscf+bsel; *adb = ad; +- for (i=sblimit;i mnr[k][i]) small = mnr[k][i]; +- if(mnr[k][i] < NOISY_MIN_MNR) ++noisy_sbs; /* noise is not masked */ +- +- } +- } +- return noisy_sbs; +- } +- +- /************************************************************************ +- * +- * I_subband_quantization (Layer I) +- * II_subband_quantization (Layer II) +- * +- * PURPOSE:Quantizes subband samples to appropriate number of bits +- * +- * SEMANTICS: Subband samples are divided by their scalefactors, which +- makes the quantization more efficient. The scaled samples are +- * quantized by the function a*x+b, where a and b are functions of +- * the number of quantization levels. The result is then truncated +- * to the appropriate number of bits and the MSB is inverted. +- * +- * Note that for fractional 2's complement, inverting the MSB for a +- negative number x is equivalent to adding 1 to it. +- * +- ************************************************************************/ +- +- static double a[17] = { +- 0.750000000, 0.625000000, 0.875000000, 0.562500000, 0.937500000, +- 0.968750000, 0.984375000, 0.992187500, 0.996093750, 0.998046875, +- 0.999023438, 0.999511719, 0.999755859, 0.999877930, 0.999938965, +- 0.999969482, 0.999984741 }; +- +- static double b[17] = { +- -0.250000000, -0.375000000, -0.125000000, -0.437500000, -0.062500000, +- -0.031250000, -0.015625000, -0.007812500, -0.003906250, -0.001953125, +- -0.000976563, -0.000488281, -0.000244141, -0.000122070, -0.000061035, +- -0.000030518, -0.000015259 }; +- +- void I_subband_quantization(scalar, sb_samples, j_scale, j_samps, +- bit_alloc, sbband, fr_ps) +- unsigned int scalar[2][3][SBLIMIT]; +- double FAR sb_samples[2][3][SCALE_BLOCK][SBLIMIT]; +- unsigned int j_scale[3][SBLIMIT]; +- double FAR j_samps[3][SCALE_BLOCK][SBLIMIT]; /* L+R for j-stereo if necess */ +- unsigned int bit_alloc[2][SBLIMIT]; +- unsigned int FAR sbband[2][3][SCALE_BLOCK][SBLIMIT]; +- frame_params *fr_ps; +- { +- int i, j, k, n, sig; +- int stereo = fr_ps->stereo; +- int sblimit = fr_ps->sblimit; +- int jsbound = fr_ps->jsbound; +- double d; +- static char init = 0; +- +- if (!init) { +- init = 1; +- /* rearrange quantization coef to correspond to layer I table */ +- a[1] = a[2]; b[1] = b[2]; +- for (i=2;i<15;i++) { a[i] = a[i+2]; b[i] = b[i+2]; } +- } +- for (j=0;j=jsbound) +- /* use the joint data passed in */ +- d = j_samps[0][j][i] / multiple[j_scale[0][i]]; +- else +- d = sb_samples[k][0][j][i] / multiple[scalar[k][0][i]]; +- /* scale and quantize floating point sample */ +- n = bit_alloc[k][i]; +- d = d * a[n-1] + b[n-1]; +- /* extract MSB N-1 bits from the floating point sample */ +- if (d >= 0) sig = 1; +- else { sig = 0; d += 1.0; } +- sbband[k][0][j][i] = (unsigned int) (d * (double) (1L<stereo; +- int sblimit = fr_ps->sblimit; +- int jsbound = fr_ps->jsbound; +- unsigned int stps; +- double d; +- al_table *alloc = fr_ps->alloc; +- +- for (s=0;s<3;s++) +- for (j=0;j=jsbound) /* use j-stereo samples */ +- d = j_samps[s][j][i] / multiple[j_scale[s][i]]; +- else +- d = sb_samples[k][s][j][i] / multiple[scalar[k][s][i]]; +- if (mod(d) > 1.0) +- printf("Not scaled properly %d %d %d %d\n",k,s,j,i); +- qnt = (*alloc)[i][bit_alloc[k][i]].quant; +- d = d * a[qnt] + b[qnt]; +- /* extract MSB N-1 bits from the floating point sample */ +- if (d >= 0) sig = 1; +- else { sig = 0; d += 1.0; } +- n = 0; +- #ifndef MS_DOS +- stps = (*alloc)[i][bit_alloc[k][i]].steps; +- while ((1L<stereo; +- int sblimit = fr_ps->sblimit; +- int jsbound = fr_ps->jsbound; +- +- for (i=0;istereo; +- int sblimit = fr_ps->sblimit; +- int jsbound = fr_ps->jsbound; +- al_table *alloc = fr_ps->alloc; +- +- for (i=0;istereo; +- int sblimit = fr_ps->sblimit; +- int jsbound = fr_ps->jsbound; +- +- for(j=0;jstereo; +- int sblimit = fr_ps->sblimit; +- int jsbound = fr_ps->jsbound; +- al_table *alloc = fr_ps->alloc; +- +- for (s=0;s<3;s++) +- for (j=0;jnGranules <= MAX_GRANULES ); + assert( frameInfo->nChannels <= MAX_CHANNELS ); +--- 57,67 ---- + See formatBitstream.h for more information about the data + structures and the bitstream syntax. + */ ++ static int elements, forwardFrameLength, forwardSILength; + void + BF_BitstreamFrame( BF_FrameData *frameInfo, BF_FrameResults *results ) + { +! /* int elements, forwardFrameLength, forwardSILength; */ + + assert( frameInfo->nGranules <= MAX_GRANULES ); + assert( frameInfo->nChannels <= MAX_CHANNELS ); +*************** +*** 87,108 **** + void + BF_FlushBitstream( BF_FrameData *frameInfo, BF_FrameResults *results ) + { +! int elements, forwardFrameLength, forwardSILength; + + /* get ptr to bit writing function */ + PutBits = frameInfo->putbits; + assert( PutBits ); + +- elements = side_queue_elements( &forwardFrameLength, &forwardSILength ); +- + if ( elements ) + { +! int bitsRemaining = forwardFrameLength - forwardSILength; +! int wordsRemaining = bitsRemaining / 32; +! while ( wordsRemaining-- ) +! WriteMainDataBits( 0, 32, results ); +! WriteMainDataBits( 0, (bitsRemaining % 32), results ); + } + + results->mainDataLength = forwardFrameLength - forwardSILength; + results->SILength = forwardSILength; +--- 96,117 ---- + void + BF_FlushBitstream( BF_FrameData *frameInfo, BF_FrameResults *results ) + { +! /* int elements, forwardFrameLength, forwardSILength; */ + + /* get ptr to bit writing function */ + PutBits = frameInfo->putbits; + assert( PutBits ); + + if ( elements ) + { +! int bitsRemaining = forwardFrameLength - forwardSILength; +! int wordsRemaining = bitsRemaining / 32; +! while ( wordsRemaining-- ) { +! WriteMainDataBits( 0, 32, results ); +! } +! WriteMainDataBits( 0, (bitsRemaining % 32), results ); + } ++ + + results->mainDataLength = forwardFrameLength - forwardSILength; + results->SILength = forwardSILength; +diff -r -c -N encoder/get_audio.c lame3.13/get_audio.c +*** encoder/get_audio.c Wed Dec 31 17:00:00 1969 +--- lame3.13/get_audio.c Sat Jun 5 13:29:42 1999 +*************** +*** 0 **** +--- 1,260 ---- ++ #ifdef LIBSNDFILE ++ /* ++ ** Copyright (C) 1999 Albert Faber ++ ** ++ ** This program is free software; you can redistribute it and/or modify ++ ** it under the terms of the GNU General Public License as published by ++ ** the Free Software Foundation; either version 2 of the License, or ++ ** (at your option) any later version. ++ ** ++ ** This program is distributed in the hope that it will be useful, ++ ** but WITHOUT ANY WARRANTY; without even the implied warranty of ++ ** MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the ++ ** GNU General Public License for more details. ++ ** ++ ** You should have received a copy of the GNU General Public License ++ ** along with this program; if not, write to the Free Software ++ ** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. ++ */ ++ ++ #include "common.h" ++ #include "get_audio.h" ++ #include "globalflags.h" ++ #include ++ ++ /* External references */ ++ ++ static SNDFILE* gs_pSndFileIn=NULL; ++ static SF_INFO gs_wfInfo; ++ ++ ++ #if 0 ++ static int iswav=0; /* IS BYTE SWAPPING STILL NEEDED WITH LIBSNDFILE ? */ ++ #endif ++ ++ int GetSndSampleRate() ++ { ++ return gs_wfInfo.samplerate; ++ } ++ ++ int GetSndChannels() ++ { ++ return gs_wfInfo.channels; ++ } ++ ++ void CloseSndFile() ++ { ++ if (gs_pSndFileIn) ++ { ++ if (sf_close(gs_pSndFileIn) !=0) ++ { ++ fprintf(stderr, "Could not close sound file \n"); ++ exit(2); ++ } ++ } ++ } ++ ++ void OpenSndFile(const char* lpszFileName) ++ { ++ // Try to open the sound file ++ gs_pSndFileIn=sf_open_read(lpszFileName,&gs_wfInfo); ++ ++ // Check result ++ if (gs_pSndFileIn==NULL) ++ { ++ fprintf(stderr, "Could not open sound file \"%s\".\n", lpszFileName); ++ exit(1); ++ } ++ #ifdef _DEBUG_SND_FILE ++ printf("\n\nSF_INFO structure\n"); ++ printf("samplerate :%d\n",gs_wfInfo.samplerate); ++ printf("samples :%d\n",gs_wfInfo.samples); ++ printf("channels :%d\n",gs_wfInfo.channels); ++ printf("pcmbitwidth :%d\n",gs_wfInfo.pcmbitwidth); ++ printf("format :"); ++ ++ if (gs_wfInfo.format&SF_FORMAT_WAV) printf("Microsoft WAV format (big endian)"); ++ if (gs_wfInfo.format&SF_FORMAT_AIFF) printf("Apple/SGI AIFF format (little endian)."); ++ if (gs_wfInfo.format&SF_FORMAT_AU) printf("Sun/NeXT AU format (big endian)."); ++ if (gs_wfInfo.format&SF_FORMAT_AULE) printf("DEC AU format (little endian)."); ++ if (gs_wfInfo.format&SF_FORMAT_PCM) printf("PCM data in 8, 16, 24 or 32 bits."); ++ if (gs_wfInfo.format&SF_FORMAT_FLOAT) printf("32 bit Intel x86 floats."); ++ if (gs_wfInfo.format&SF_FORMAT_ULAW) printf("U-Law encoded."); ++ if (gs_wfInfo.format&SF_FORMAT_ALAW) printf("A-Law encoded."); ++ if (gs_wfInfo.format&SF_FORMAT_MS_ADPCM) printf("Microsoft ADPCM."); ++ if (gs_wfInfo.format&SF_FORMAT_IMA_ADPCM) printf("IMA ADPCM."); ++ ++ printf("\n"); ++ printf("pcmbitwidth :%d\n",gs_wfInfo.pcmbitwidth); ++ printf("sections :%d\n",gs_wfInfo.sections); ++ printf("seekable :\n",gs_wfInfo.seekable); ++ #endif ++ } ++ ++ ++ /************************************************************************ ++ * ++ * read_samples() ++ * ++ * PURPOSE: reads the PCM samples from a file to the buffer ++ * ++ * SEMANTICS: ++ * Reads #samples_read# number of shorts from #musicin# filepointer ++ * into #sample_buffer[]#. Returns the number of samples read. ++ * ++ ************************************************************************/ ++ ++ DWORD read_samples( ++ short sample_buffer[2304], ++ DWORD num_samples, ++ DWORD frame_size ++ ) ++ { ++ DWORD samples_read; ++ static DWORD samples_to_read; ++ static char init = TRUE; ++ ++ if (init) ++ { ++ samples_to_read = num_samples; ++ init = FALSE; ++ } ++ ++ if (samples_to_read >= frame_size) ++ { ++ samples_read = frame_size; ++ } ++ else ++ { ++ samples_read = samples_to_read; ++ } ++ if (gs_pSndFileIn) ++ samples_read=sf_read_short(gs_pSndFileIn,sample_buffer,samples_read); ++ else ++ samples_read=fread(sample_buffer, sizeof(short),samples_read, stdin); ++ ++ ++ if (samples_read!=0) ++ { ++ #if 0 ++ /* Samples are big-endian. If this is a little-endian machine we must swap */ ++ if ( NativeByteOrder == order_unknown ) ++ { ++ NativeByteOrder = DetermineByteOrder(); ++ if ( NativeByteOrder == order_unknown ) ++ { ++ fprintf( stderr, "byte order not determined\n" ); ++ exit( 1 ); ++ } ++ } ++ if (!iswav && ( NativeByteOrder == order_littleEndian )) ++ SwapBytesInWords( sample_buffer, samples_read ); ++ ++ if (swapbytes==TRUE) ++ SwapBytesInWords( sample_buffer, samples_read ); ++ #endif ++ ++ samples_to_read -= samples_read; ++ if (samples_read < frame_size && samples_read > 0) ++ { ++ /* printf("Insufficient PCM input for one frame - fillout with zeros\n"); */ ++ for (; samples_read < frame_size; sample_buffer[samples_read++] = 0); ++ samples_to_read = 0; ++ } ++ } ++ return(samples_read); ++ } ++ ++ ++ /************************************************************************ ++ * ++ * get_audio() ++ * ++ * PURPOSE: reads a frame of audio data from a file to the buffer, ++ * aligns the data for future processing, and separates the ++ * left and right channels ++ * ++ * ++ ************************************************************************/ ++ ++ DWORD get_audio( FILE *musicin, /* ignored */ ++ short bufferL[1152], ++ short bufferR[1152], ++ DWORD num_samples, ++ int stereo, ++ layer* info ++ ) ++ { ++ int j; ++ short insamp[2304]; ++ DWORD samples_read; ++ int lay; ++ ++ lay = info->lay; ++ ++ /* pad with zeros in case we hit EOF */ ++ /* THIS IS ALREADY DONE IN THE SND LIB */ ++ /* memset((char *) insamp, 0, sizeof(insamp)); */ ++ ++ if ( (lay == 3) && (info->version == 0) ) /* ie MPEG-2 LSF */ ++ { ++ if ( stereo == 2 ) ++ { ++ samples_read = read_samples(insamp, num_samples,1152 ); ++ for ( j = 0; j < 576; j++ ) ++ { ++ bufferL[j] = insamp[2 * j]; ++ bufferR[j] = insamp[2 * j + 1]; ++ } ++ } ++ else ++ { ++ samples_read = read_samples(insamp, num_samples,576 ); ++ for ( j = 0; j < 576; j++ ) ++ { ++ bufferL[j] = insamp[j]; ++ bufferR[j] = 0; ++ } ++ } ++ } ++ else ++ { ++ /* MPEG 1 */ ++ if(stereo == 2) ++ { ++ /* layer 2 (or 3), stereo */ ++ samples_read = read_samples(insamp, num_samples,2304); ++ for(j=0;j<1152;j++) ++ { ++ bufferL[j] = insamp[2*j]; ++ bufferR[j] = insamp[2*j+1]; ++ } ++ } ++ else ++ { ++ /* layer 2 (or 3), mono */ ++ if (autoconvert==TRUE) ++ { ++ /* downconvert from a stereo file into a mono buffer */ ++ samples_read = read_samples(insamp, num_samples,2304); ++ for(j=0;j<1152;j++) ++ { ++ bufferL[j] = insamp[2*j]; ++ bufferR[j] = 0; ++ } ++ } ++ else ++ { ++ samples_read = read_samples(insamp, num_samples,1152); ++ for(j=0;j<1152;j++) ++ { ++ bufferL[j] = insamp[j]; ++ bufferR[j] = 0; ++ } ++ } ++ } ++ } ++ ++ return(samples_read); ++ } ++ #endif +diff -r -c -N encoder/get_audio.h lame3.13/get_audio.h +*** encoder/get_audio.h Wed Dec 31 17:00:00 1969 +--- lame3.13/get_audio.h Wed Jun 9 20:52:47 1999 +*************** +*** 0 **** +--- 1,46 ---- ++ /* ++ ** Copyright (C) 1999 Albert Faber ++ ** ++ ** This program is free software; you can redistribute it and/or modify ++ ** it under the terms of the GNU General Public License as published by ++ ** the Free Software Foundation; either version 2 of the License, or ++ ** (at your option) any later version. ++ ** ++ ** This program is distributed in the hope that it will be useful, ++ ** but WITHOUT ANY WARRANTY; without even the implied warranty of ++ ** MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the ++ ** GNU General Public License for more details. ++ ** ++ ** You should have received a copy of the GNU General Public License ++ ** along with this program; if not, write to the Free Software ++ ** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. ++ */ ++ ++ #ifndef GET_AUDIO_H_INCLUDED ++ #define GET_AUDIO_H_INCLUDED ++ ++ /* INCLUDE the sound library header file */ ++ #ifdef _WIN32 ++ /* one byte alignment for WIN32 platforms */ ++ #pragma pack(push,1) ++ #include "./libsndfile/src/sndfile.h" ++ #pragma pack(pop,1) ++ #else ++ #include "sndfile.h" ++ #endif ++ ++ #ifndef DWORD ++ typedef unsigned long DWORD; ++ #endif ++ ++ int GetSndSampleRate(); ++ int GetSndChannels(); ++ void CloseSndFile(); ++ void OpenSndFile(const char* lpszFileName); ++ ++ DWORD read_samples( short sample_buffer[2304], ++ DWORD num_samples,DWORD frame_size); ++ DWORD get_audio(FILE *musicin, short bufferL[1152],short bufferR[1152], ++ DWORD num_samples,int stereo,layer* info); ++ ++ #endif +diff -r -c -N encoder/globalflags.h lame3.13/globalflags.h +*** encoder/globalflags.h Wed Dec 31 17:00:00 1969 +--- lame3.13/globalflags.h Thu Jun 24 18:16:16 1999 +*************** +*** 0 **** +--- 1,23 ---- ++ ++ /*********************************************************************** ++ * ++ * Global Variable External Declarations ++ * ++ ***********************************************************************/ ++ extern int force_ms; ++ extern int VBR; ++ extern int VBR_q; ++ extern int VBR_min_bitrate; ++ extern int VBR_max_bitrate; ++ extern int gpsycho; ++ extern int sfb21; ++ extern int silent; ++ extern int highq; ++ extern int fast_mode; ++ extern int allow_diff_short; ++ extern int autoconvert; ++ extern int swapbytes; ++ extern int experimentalX; ++ extern int experimentalY; ++ extern int experimentalZ; ++ +diff -r -c -N encoder/gpkplotting.c lame3.13/gpkplotting.c +*** encoder/gpkplotting.c Wed Dec 31 17:00:00 1969 +--- lame3.13/gpkplotting.c Thu Jun 3 13:13:16 1999 +*************** +*** 0 **** +--- 1,276 ---- ++ #ifdef HAVEGTK ++ #include "gpkplotting.h" ++ #include "string.h" ++ ++ static gint num_plotwindows = 0; ++ static gint max_plotwindows = 10; ++ static GdkPixmap *pixmaps[10]; ++ static GtkWidget *pixmapboxes[10]; ++ ++ ++ ++ ++ /* compute a gdkcolor */ ++ void setcolor(GtkWidget *widget, GdkColor *color, gint red,gint green,gint blue) ++ { ++ ++ /* colors in GdkColor are taken from 0 to 65535, not 0 to 255. */ ++ color->red = red * (65535/255); ++ color->green = green * (65535/255); ++ color->blue = blue * (65535/255); ++ color->pixel = (gulong)(color->red*65536 + color->green*256 + color->blue); ++ /* find closest in colormap, if needed */ ++ gdk_color_alloc(gtk_widget_get_colormap(widget),color); ++ } ++ ++ ++ ++ ++ void gpk_redraw(pixmap,pixmapbox) ++ GtkWidget *pixmapbox; ++ GdkPixmap *pixmap; ++ { ++ /* redraw the entire pixmap */ ++ gdk_draw_pixmap(pixmapbox->window, ++ pixmapbox->style->fg_gc[GTK_WIDGET_STATE (pixmapbox)], ++ pixmap,0,0,0,0, ++ pixmapbox->allocation.width, ++ pixmapbox->allocation.height); ++ } ++ ++ ++ static GdkPixmap **findpixmap(GtkWidget *widget) ++ { ++ int i; ++ for (i=0; i=num_plotwindows) { ++ g_print("findpixmap(): bad argument widget \n"); ++ return NULL; ++ } ++ return &pixmaps[i]; ++ } ++ ++ void gpk_graph_draw(GtkWidget *widget, /* plot on this widged */ ++ int n, /* number of data points */ ++ gdouble *xcord, gdouble *ycord, /* data */ ++ gdouble xmn,gdouble ymn, /* coordinates of corners */ ++ gdouble xmx,gdouble ymx, ++ int clear, /* clear old plot first */ ++ char *title, /* add a title (only if clear=1) */ ++ GdkColor *color) ++ { ++ GdkPixmap **ppixmap; ++ GdkPoint *points; ++ int i; ++ gint16 width,height; ++ GdkFont *fixed_font; ++ GdkGC *gc; ++ ++ gc = gdk_gc_new(widget->window); ++ gdk_gc_set_foreground(gc, color); ++ ++ ++ ++ if ((ppixmap=findpixmap(widget))) { ++ width = widget->allocation.width; ++ height = widget->allocation.height; ++ ++ ++ if (clear) { ++ /* white background */ ++ gdk_draw_rectangle (*ppixmap, ++ widget->style->white_gc, ++ TRUE,0, 0,width,height); ++ /* title */ ++ fixed_font = gdk_font_load ("-misc-fixed-medium-r-*-*-*-100-*-*-*-*-*-*"); ++ gdk_draw_text (*ppixmap,fixed_font, ++ widget->style->fg_gc[GTK_WIDGET_STATE (widget)], ++ 0,10,title,strlen(title)); ++ } ++ ++ ++ points = g_malloc(n*sizeof(GdkPoint)); ++ for (i=0; iwindow); ++ gdk_gc_set_foreground(gc, color); ++ ++ ++ if ((ppixmap=findpixmap(widget))) { ++ width = widget->allocation.width; ++ height = widget->allocation.height; ++ ++ ++ for (i=0; i<2 ; i++) { ++ points[i].x =.5+ ((xcord[i]-xmn)*(width-1)/(xmx-xmn)); ++ points[i].y =.5+ ((ycord[i]-ymx)*(height-1)/(ymn-ymx)); ++ } ++ width=points[1].x-points[0].x + 1; ++ height=points[1].y-points[0].y + 1; ++ gdk_draw_rectangle(*ppixmap,gc,TRUE, ++ points[0].x,points[0].y,width,height); ++ gpk_redraw(*ppixmap,widget); ++ } ++ gdk_gc_destroy(gc); ++ } ++ ++ ++ ++ void gpk_bargraph_draw(GtkWidget *widget, /* plot on this widged */ ++ int n, /* number of data points */ ++ gdouble *xcord, gdouble *ycord, /* data */ ++ gdouble xmn,gdouble ymn, /* coordinates of corners */ ++ gdouble xmx,gdouble ymx, ++ int clear, /* clear old plot first */ ++ char *title, /* add a title (only if clear=1) */ ++ int barwidth, /* bar width. 0=compute based on window size */ ++ GdkColor *color) ++ { ++ GdkPixmap **ppixmap; ++ GdkPoint points[2]; ++ int i; ++ gint16 width,height,x,y,barheight; ++ GdkFont *fixed_font; ++ GdkGC *gc; ++ ++ ++ gc = gdk_gc_new(widget->window); ++ gdk_gc_set_foreground(gc, color); ++ ++ ++ if ((ppixmap=findpixmap(widget))) { ++ width = widget->allocation.width; ++ height = widget->allocation.height; ++ ++ ++ if (clear) { ++ /* white background */ ++ gdk_draw_rectangle (*ppixmap, ++ widget->style->white_gc, ++ TRUE,0, 0,width,height); ++ /* title */ ++ fixed_font = gdk_font_load ("-misc-fixed-medium-r-*-*-*-100-*-*-*-*-*-*"); ++ gdk_draw_text (*ppixmap,fixed_font, ++ widget->style->fg_gc[GTK_WIDGET_STATE (widget)], ++ 0,10,title,strlen(title)); ++ } ++ ++ ++ for (i=0; i 5 ? 5 : barwidth; ++ barwidth = barwidth < 1 ? 1 : barwidth; ++ barheight = height-1 - y; ++ /* gdk_draw_lines(*ppixmap,gc,points,2); */ ++ gdk_draw_rectangle(*ppixmap,gc,TRUE,x,y,barwidth,barheight); ++ ++ } ++ gpk_redraw(*ppixmap,widget); ++ } ++ gdk_gc_destroy(gc); ++ } ++ ++ ++ ++ ++ ++ /* Create a new backing pixmap of the appropriate size */ ++ static gint ++ configure_event (GtkWidget *widget, GdkEventConfigure *event, gpointer data) ++ { ++ GdkPixmap **ppixmap; ++ if ((ppixmap=findpixmap(widget))){ ++ if (*ppixmap) gdk_pixmap_unref(*ppixmap); ++ *ppixmap = gdk_pixmap_new(widget->window, ++ widget->allocation.width, ++ widget->allocation.height, ++ -1); ++ gdk_draw_rectangle (*ppixmap, ++ widget->style->white_gc, ++ TRUE, ++ 0, 0, ++ widget->allocation.width, ++ widget->allocation.height); ++ } ++ return TRUE; ++ } ++ ++ ++ ++ /* Redraw the screen from the backing pixmap */ ++ static gint ++ expose_event (GtkWidget *widget, GdkEventExpose *event, gpointer data) ++ { ++ GdkPixmap **ppixmap; ++ if ((ppixmap=findpixmap(widget))){ ++ gdk_draw_pixmap(widget->window, ++ widget->style->fg_gc[GTK_WIDGET_STATE (widget)], ++ *ppixmap, ++ event->area.x, event->area.y, ++ event->area.x, event->area.y, ++ event->area.width, event->area.height); ++ } ++ ++ return FALSE; ++ } ++ ++ ++ ++ ++ ++ GtkWidget *gpk_plot_new(int width, int height) ++ { ++ GtkWidget *pixmapbox; ++ ++ pixmapbox = gtk_drawing_area_new(); ++ gtk_drawing_area_size(GTK_DRAWING_AREA(pixmapbox),width,height); ++ gtk_signal_connect (GTK_OBJECT (pixmapbox), "expose_event", ++ (GtkSignalFunc) expose_event, NULL); ++ gtk_signal_connect (GTK_OBJECT(pixmapbox),"configure_event", ++ (GtkSignalFunc) configure_event, NULL); ++ gtk_widget_set_events (pixmapbox, GDK_EXPOSURE_MASK); ++ ++ if (num_plotwindows < max_plotwindows) { ++ pixmapboxes[num_plotwindows] = pixmapbox; ++ pixmaps[num_plotwindows] = NULL; ++ num_plotwindows ++; ++ } else { ++ g_print("gtk_plotarea_new(): exceeded maximum of 10 plotarea windows\n"); ++ } ++ ++ return pixmapbox; ++ } ++ ++ ++ #endif +diff -r -c -N encoder/gpkplotting.h lame3.13/gpkplotting.h +*** encoder/gpkplotting.h Wed Dec 31 17:00:00 1969 +--- lame3.13/gpkplotting.h Wed Apr 28 20:04:35 1999 +*************** +*** 0 **** +--- 1,27 ---- ++ #include ++ ++ /* allocate a graphing widget */ ++ GtkWidget *gpk_plot_new(int width,int height); ++ ++ /* graph a function in the graphing widged */ ++ void gpk_graph_draw(GtkWidget *widget, ++ int n, gdouble *xcord, gdouble *ycord, ++ gdouble xmn, gdouble ymn,gdouble xmx,gdouble ymx, ++ int clear, char * title,GdkColor *color); ++ ++ /* draw a rectangle in the graphing widget */ ++ void gpk_rectangle_draw(GtkWidget *widget, /* plot on this widged */ ++ gdouble xcord[2], gdouble ycord[2], /* corners */ ++ gdouble xmn,gdouble ymn, /* coordinates of corners */ ++ gdouble xmx,gdouble ymx, ++ GdkColor *color); /* color to use */ ++ ++ /* make a bar graph in the graphing widged */ ++ void gpk_bargraph_draw(GtkWidget *widget, ++ int n, gdouble *xcord, gdouble *ycord, ++ gdouble xmn, gdouble ymn,gdouble xmx,gdouble ymx, ++ int clear, char * title, int bwidth,GdkColor *color); ++ ++ /* set forground color */ ++ void setcolor(GtkWidget *widget, GdkColor *color,int red,int green,int blue); ++ +diff -r -c -N encoder/gtkanal.c lame3.13/gtkanal.c +*** encoder/gtkanal.c Wed Dec 31 17:00:00 1969 +--- lame3.13/gtkanal.c Tue Jun 22 00:01:44 1999 +*************** +*** 0 **** +--- 1,1190 ---- ++ #ifdef HAVEGTK ++ #include ++ #include ++ #include "gpkplotting.h" ++ #include "common.h" ++ #include "encoder.h" ++ #include "l3psy.h" ++ #include "gtkanal.h" ++ #include "version.h" ++ ++ /* global variables for the state of the system */ ++ static gint idle_keepgoing; /* processing of frames is ON */ ++ static gint idle_count_max; /* number of frames to process before plotting */ ++ static gint idle_count; /* pause & plot when idle_count=idel_count_max */ ++ static gint idle_finish=0; /* process all frames, finish mp3 file and exit */ ++ static gint idle_back = 0; /* set when we are displaying the old data */ ++ static int mp3done = 0; /* last frame has been read */ ++ static GtkWidget *frameprogress; /* progress bar */ ++ static GtkWidget *framecounter; /* progress counter */ ++ ++ /* main window */ ++ GtkWidget *window; ++ /* Backing pixmap for drawing areas */ ++ GtkWidget *pcmbox; /* PCM data plotted here */ ++ GtkWidget *winbox; /* mpg123 synthesis data plotted here */ ++ GtkWidget *enerbox[2]; /* spectrum, gr=0,1 plotted here */ ++ GtkWidget *mdctbox[2]; /* mdct coefficients gr=0,1 plotted here */ ++ GtkWidget *headerbox; /* mpg123 header info shown here */ ++ ++ plotting_data *pinfo; ++ plotting_data Pinfo[NUMPINFO]; ++ ++ struct gtkinfostruct { ++ int totalframes; /* total number of frames in mp3 file */ ++ int filetype; /* input file type 0=WAV, 1=MP3 */ ++ int msflag; /* toggle between L&R vs M&S PCM data display */ ++ int chflag; /* toggle between L & R channels */ ++ int kbflag; /* toggle between wave # and barks */ ++ int flag123; /* show mpg123 frame info, OR ISO encoder frame info */ ++ int gpsycho; /* TRUE if using gpsycho model */ ++ double avebits; /* running average bits per frame */ ++ int approxbits; /* (approx) bits per frame */ ++ int maxbits; /* max bits per frame used so far*/ ++ int totemph; /* total of frames with de-emphasis */ ++ int totms; /* total frames with ms_stereo */ ++ int totis; /* total frames with i_stereo */ ++ int totshort; /* total granules with short blocks */ ++ int totmix; /* total granules with mixed blocks */ ++ int pupdate; /* plot while processing, or only when needed */ ++ } gtkinfo; ++ ++ ++ ++ ++ ++ void plot_frame() ++ { ++ int i,j,n,ch,gr,grtemp; ++ gdouble *xcord,*ycord; ++ gdouble xmx,xmn,ymx,ymn; ++ double *data,*data2,*data3; ++ char title2[80]; ++ char *title; ++ plotting_data *pplot; ++ plotting_data *pplot1; ++ plotting_data *pplot2; ++ plotting_data *ptemp; ++ ++ double en,samp; ++ static int firstcall=1; ++ static GdkColor *barcolor,*color,*grcolor[2]; ++ static GdkColor yellow,gray,cyan,magenta,orange,pink,red,green,blue,black,oncolor,offcolor; ++ int blocktype[2][2]; ++ int headbits; ++ int mode_gr = 2; ++ ++ /* find the delay between the frame we are plotting and the frame for ++ * which MPG123 finally produced some output. */ ++ /* ++ { int mpglag=0; ++ for (i=0; i<=(pinfo->mpglag); i++ ) { ++ if ((pinfo-i)->mpglag > mpglag ) mpglag = (pinfo-i)->mpglag; ++ } ++ pplot2 = pinfo-mpglag; ++ } ++ */ ++ { int mpglag=0; ++ for (i=0; i<(READ_AHEAD); i++ ) { ++ if ((pinfo-i)->mpglag == (pinfo-i-1)->mpglag) { ++ mpglag = (pinfo-i)->mpglag; ++ break; ++ } ++ } ++ pplot2 = pinfo-mpglag; ++ } ++ ++ ++ ++ ++ ++ /* however, the PCM data is delayed by 528 samples in the encoder filterbanks. ++ * We added another 1152-528 delay to this so the PCM data is *exactly* one ++ * frame behind the header & MDCT information */ ++ pplot1 =pplot2 +1; /* back one frame for header info, MDCT */ ++ pplot = pinfo; /* reference frame pcm input, FFT's */ ++ ++ /* allocate these GC's only once */ ++ if (firstcall) { ++ firstcall=0; ++ /* grcolor[0]=&magenta; */ ++ grcolor[0]=&blue; ++ grcolor[1]=&green; ++ barcolor = &gray; ++ ++ setcolor(headerbox,&oncolor,255,0,0); ++ setcolor(headerbox,&offcolor,175,175,175); ++ setcolor(pcmbox,&red,255,0,0); ++ setcolor(pcmbox,&pink,255,0,255); ++ setcolor(pcmbox,&magenta,255,0,100); ++ setcolor(pcmbox,&orange,255,127,0); ++ setcolor(pcmbox,&cyan,0,255,255); ++ setcolor(pcmbox,&green,0,255,0); ++ setcolor(pcmbox,&blue,0,0,255); ++ setcolor(pcmbox,&black,0,0,0); ++ setcolor(pcmbox,&gray,100,100,100); ++ setcolor(pcmbox,&yellow,255,255,0); ++ ++ } ++ ++ /******************************************************************* ++ * frame header info ++ *******************************************************************/ ++ if (pplot1->sampfreq) ++ samp=pplot1->sampfreq; ++ else samp=1; ++ ++ headbits = 32 + ((pplot1->stereo==2) ? 256 : 136); ++ gtkinfo.approxbits = (pplot1->bitrate*1000*1152.0/samp) - headbits; ++ /*font = gdk_font_load ("-misc-fixed-medium-r-*-*-*-100-*-*-*-*-*-*");*/ ++ sprintf(title2,"%3.1fkHz %ikbs ",samp/1000,pplot1->bitrate); ++ gtk_text_freeze (GTK_TEXT(headerbox)); ++ gtk_text_backward_delete(GTK_TEXT(headerbox), ++ gtk_text_get_length(GTK_TEXT(headerbox))); ++ gtk_text_set_point(GTK_TEXT(headerbox),0); ++ gtk_text_insert(GTK_TEXT(headerbox),NULL,&oncolor,NULL,title2, -1); ++ title = " mono "; ++ if (2==pplot1->stereo) title = pplot1->js ? " jstereo " : " stereo "; ++ gtk_text_insert (GTK_TEXT(headerbox), NULL, &oncolor, NULL,title, -1); ++ color = pplot1->ms_stereo ? &oncolor : &offcolor ; ++ gtk_text_insert (GTK_TEXT(headerbox), NULL, color, NULL,"ms_stereo ", -1); ++ color = pplot1->i_stereo ? &oncolor : &offcolor ; ++ gtk_text_insert (GTK_TEXT(headerbox), NULL, color, NULL,"i_stereo ", -1); ++ color = pplot1->emph ? &oncolor : &offcolor ; ++ gtk_text_insert (GTK_TEXT(headerbox), NULL, color, NULL,"emph ", -1); ++ color = pplot1->scfsi ? &oncolor : &offcolor ; ++ gtk_text_insert (GTK_TEXT(headerbox), NULL, color, NULL,"scfsi ", -1); ++ if (gtkinfo.filetype) ++ sprintf(title2," mdb=%i %i/NA",pplot1->maindata,pplot1->totbits); ++ else ++ sprintf(title2," mdb=%i %i/%i", ++ pplot1->maindata,pplot1->totbits,pplot->resvsize); ++ gtk_text_insert (GTK_TEXT(headerbox), NULL, &oncolor, NULL,title2, -1); ++ gtk_text_thaw (GTK_TEXT(headerbox)); ++ ++ ++ ch = gtkinfo.chflag; ++ /******************************************************************* ++ * block type ++ *******************************************************************/ ++ for (gr = 0 ; gr < mode_gr ; gr ++) ++ if (gtkinfo.flag123) ++ blocktype[gr][ch]=pplot1->mpg123blocktype[gr][ch]; ++ else blocktype[gr][ch]=pplot->blocktype[gr][ch]; ++ ++ ++ /******************************************************************* ++ * draw the PCM data * ++ *******************************************************************/ ++ n = 1600; /* PCM frame + FFT window: 224 + 1152 + 224 */ ++ xcord = g_malloc(n*sizeof(gdouble)); ++ ycord = g_malloc(n*sizeof(gdouble)); ++ ++ ++ if (gtkinfo.msflag) ++ title=ch ? "Side Channel" : "Mid Channel"; ++ else ++ title=ch ? "Right Channel" : "Left Channel"; ++ ++ ymn = -32767 ; ++ ymx = 32767; ++ xmn = 0; ++ xmx = 1600-1; ++ ++ /* 0 ... 224 draw in black, connecting to 224 pixel ++ * 1375 .. 1599 draw in black connecting to 1375 pixel ++ * 224 ... 1375 MP3 frame. draw in blue ++ */ ++ ++ /* draw the title */ ++ gpk_graph_draw(pcmbox,0,xcord,ycord,xmn,ymn,xmx,ymx,1,title, ++ &black); ++ ++ ++ /* draw some hash marks dividing the frames */ ++ ycord[0] = ymx*.8; ycord[1] = ymn*.8; ++ for (gr=0 ; gr<=2; gr++) { ++ xcord[0] = 223.5 + gr*576; xcord[1] = 223.5 +gr*576; ++ gpk_rectangle_draw(pcmbox,xcord,ycord,xmn,ymn,xmx,ymx,&yellow); ++ } ++ for (gr = 0 ; gr < mode_gr ; gr++) { ++ if (blocktype[gr][ch]==2) ++ for (i=1 ; i<=2; i++) { ++ xcord[0] = 223.5+gr*576 + i*192; ++ xcord[1] = 223.5+gr*576 + i*192; ++ gpk_rectangle_draw(pcmbox,xcord,ycord,xmn,ymn,xmx,ymx,&yellow); ++ } ++ } ++ /* bars representing FFT windows */ ++ xcord[0] = 0; ycord[0] = ymn+3000; ++ xcord[1] = 1024-1; ycord[1] = ymn+1000; ++ gpk_rectangle_draw(pcmbox,xcord,ycord,xmn,ymn,xmx,ymx,grcolor[0]); ++ xcord[0] = 576; ycord[0] = ymn+2000; ++ xcord[1] = 576+1024-1; ycord[1] = ymn; ++ gpk_rectangle_draw(pcmbox,xcord,ycord,xmn,ymn,xmx,ymx,grcolor[1]); ++ ++ ++ /* plot PCM data */ ++ for (i=0; ipcmdata[0][i]-pplot->pcmdata[1][i]) : ++ .5*(pplot->pcmdata[0][i]+pplot->pcmdata[1][i]); ++ else ++ ycord[i]=pplot->pcmdata[ch][i]; ++ } ++ ++ /* skip plot if we are doing an mp3 file */ ++ if (!gtkinfo.filetype) { ++ n = 224; /* number of points on end of blue part */ ++ /* data left of frame */ ++ gpk_graph_draw(pcmbox,n+1,xcord,ycord,xmn,ymn,xmx,ymx,0,title,&black); ++ /* data right of frame */ ++ gpk_graph_draw(pcmbox,n+1,&xcord[1152+n-1],&ycord[1152+n-1], ++ xmn,ymn,xmx,ymx,0,title,&black); ++ /* the actual frame */ ++ gpk_graph_draw(pcmbox,1152,&xcord[n],&ycord[n],xmn,ymn,xmx,ymx,0,title,&black); ++ } ++ ++ ++ /*******************************************************************/ ++ /* draw the PCM re-synthesis data */ ++ /*******************************************************************/ ++ n = 1152; ++ ++ sprintf(title2,"Re-synthesis ms_ratio=%3.2f %3.2f", ++ pplot->ms_ratio[0],pplot->ms_ratio[1]); ++ title=title2; ++ ++ ++ ymn = -32767 ; ++ ymx = 32767; ++ xmn = 0; ++ xmx = 1600-1; ++ gpk_graph_draw(winbox,0,xcord,ycord, ++ xmn,ymn,xmx,ymx,1,title,&black); ++ /* draw some hash marks dividing the frames */ ++ ycord[0] = ymx*.8; ycord[1] = ymn*.8; ++ for (gr=0 ; gr<=2; gr++) { ++ xcord[0] = 223.5 + gr*576; xcord[1] = 223.5 +gr*576; ++ gpk_rectangle_draw(winbox,xcord,ycord,xmn,ymn,xmx,ymx,&yellow); ++ } ++ for (gr = 0 ; gr < 2 ; gr++) { ++ if (blocktype[gr][ch]==2) ++ for (i=1 ; i<=2; i++) { ++ xcord[0] = 223.5+gr*576 + i*192; ++ xcord[1] = 223.5+gr*576 + i*192; ++ gpk_rectangle_draw(winbox,xcord,ycord,xmn,ymn,xmx,ymx,&yellow); ++ } ++ } ++ ++ ++ ++ n = 224; ++ for (j=1152-n,i=0; i<=n; i++,j++) { ++ xcord[i] = i; ++ if (gtkinfo.msflag) ++ ycord[i] = ch ? .5*(pplot1->pcmdata2[0][j]- ++ pplot1->pcmdata2[1][j]) : ++ .5*(pplot1->pcmdata2[0][j]+pplot1->pcmdata2[1][j]); ++ else ++ ycord[i]=pplot1->pcmdata2[ch][j]; ++ } ++ gpk_graph_draw(winbox,n+1,xcord,ycord, ++ xmn,ymn,xmx,ymx,0,title,&black); ++ ++ n = 1152; ++ for (i=0; ipcmdata2[0][i]-pplot2->pcmdata2[1][i]) : ++ .5*(pplot2->pcmdata2[0][i]+pplot2->pcmdata2[1][i]); ++ else ++ ycord[i]=pplot2->pcmdata2[ch][i]; ++ } ++ gpk_graph_draw(winbox,n,xcord,ycord, ++ xmn,ymn,xmx,ymx,0,title,&black); ++ ++ ++ ++ ++ ++ /*******************************************************************/ ++ /* draw the MDCT energy spectrum */ ++ /*******************************************************************/ ++ for (gr = 0 ; gr < mode_gr ; gr ++) { ++ int bits; ++ char *blockname; ++ char label[80],label2[80]; ++ switch (blocktype[gr][ch]) { ++ case 0: blockname = "normal"; break; ++ case 1: blockname = "start"; break; ++ case 2: blockname = "short"; break; ++ case 3: blockname = "end"; break; ++ } ++ strcpy(label,blockname); ++ if (pplot1->mixed[gr][ch]) strcat(label,"(mixed)"); ++ if (blocktype[gr][ch]==2) { ++ sprintf(label2, ++ "%i %i%i%i", ++ pplot1->scalefac_scale[gr][ch], ++ pplot1->sub_gain[gr][ch][0], ++ pplot1->sub_gain[gr][ch][1], ++ pplot1->sub_gain[gr][ch][2]); ++ strcat(label,label2); ++ }else{ ++ sprintf(label2,"%i",pplot1->scalefac_scale[gr][ch]); ++ strcat(label,label2); ++ } ++ ++ ++ ++ ++ n = 576; ++ if (gtkinfo.flag123) { ++ data = pplot1->mpg123xr[gr][0]; ++ data2 = pplot1->mpg123xr[gr][1]; ++ }else{ ++ data = pplot->xr[gr][0]; ++ data2 = pplot->xr[gr][1]; ++ } ++ ++ ++ ++ ymn=9e20; ++ ymx=-9e20; ++ for (i=0; i ymx) ? ycord[i] : ymx; ++ ymn=(ycord[i] < ymn) ? ycord[i] : ymn; ++ } ++ /* print the min/max ++ sprintf(title2,"MDCT%1i %5.2f %5.2f bits=%i",gr,ymn,ymx, ++ pplot1->mainbits[gr][ch]); ++ */ ++ if (gtkinfo.flag123) bits=pplot1->mainbits[gr][ch]; ++ else bits=pplot->LAMEmainbits[gr][ch]; ++ ++ ++ sprintf(title2,"MDCT%1i(%s) bits=%i q=%i n=%3.1f",gr,label,bits, ++ pplot1->qss[gr][ch]-210,pplot->noise[gr][ch]*1e6); ++ ++ xmn = 1; ++ xmx = n; ++ ymn=0; ++ ymx=11; ++ gpk_bargraph_draw(mdctbox[gr],n,xcord,ycord, ++ xmn,ymn,xmx,ymx,1,title2,0,barcolor); ++ } ++ ++ ++ ++ ++ /******************************************************************* ++ * draw the psy model energy spectrum (k space) ++ * l3psy.c computes pe, en, thm for THIS granule. ++ * To encode, ISO uses (bug): en,thm from last granule. ++ * pe from this granule ++ * gpsycho model corrects this ++ *******************************************************************/ ++ if (gtkinfo.msflag || gtkinfo.filetype) { ++ /* erase everything */ ++ gpk_bargraph_draw(enerbox[0],0,xcord,ycord, ++ xmn,ymn,xmx,ymx,1," ",0,barcolor); ++ gpk_bargraph_draw(enerbox[1],0,xcord,ycord, ++ xmn,ymn,xmx,ymx,1," ",0,barcolor); ++ } ++ else if (gtkinfo.kbflag){ ++ for (gr = 0 ; gr < mode_gr ; gr ++) { ++ n = HBLKSIZE; /* only show half the spectrum */ ++ if (gtkinfo.gpsycho) ++ {ptemp=pplot; grtemp=gr;} ++ else { ++ /* display info from one granule behind */ ++ if (gr==0) {ptemp=(pplot+1); grtemp=1;} ++ if (gr==1) {ptemp=pplot; grtemp=0;} ++ } ++ data = &ptemp->energy[grtemp][ch][0]; ++ ++ ymn=9e20; ++ ymx=-9e20; ++ for (i=0; i ymx) ? ycord[i] : ymx; ++ ymn=(ycord[i] < ymn) ? ycord[i] : ymn; ++ } ++ for (en=0 , j=0; jenergy[grtemp][ch][j]; ++ ++ sprintf(title2,"FFT%1i pe=%4.1fK en=%5.2e ",gr, ++ ptemp->pe[grtemp][ch]/1000,en); ++ ++ ymn = 3; ++ ymx = 15; ++ xmn = 1; ++ xmx = n; ++ gpk_bargraph_draw(enerbox[gr],n,xcord,ycord, ++ xmn,ymn,xmx,ymx,1,title2,0,barcolor); ++ ++ } ++ }else{ ++ /******************************************************************* ++ * draw the psy model energy spectrum (scalefactor bands) ++ * ISO model (because of bugs) uses: ++ * pe,en,thm,energy delayed 1 granule ++ *******************************************************************/ ++ for (gr = 0 ; gr < mode_gr ; gr ++) { ++ if (gtkinfo.gpsycho) ++ { ptemp=pplot; grtemp=gr;} ++ else { ++ if (gr==0) {ptemp=(pplot+1); grtemp=1;} ++ if (gr==1) {ptemp=pplot; grtemp=0;} ++ } ++ ++ if (blocktype[gr][ch]==2) { ++ n = 3*SBMAX_s; ++ data = &ptemp->en_s[grtemp][ch][0]; ++ data2 = &ptemp->thr_s[grtemp][ch][0]; ++ data3 = &pplot->xfsf_s[grtemp][ch][0]; ++ } else { ++ n = SBMAX_l; ++ data = &ptemp->en[grtemp][ch][0]; ++ data2 = &ptemp->thr[grtemp][ch][0]; ++ data3 = &pplot->xfsf[grtemp][ch][0]; ++ } ++ ymn=9e20; ++ ymx=-9e20; ++ for (i=0; i ymx) ? ycord[i] : ymx; ++ ymn=(ycord[i] < ymn) ? ycord[i] : ymn; ++ } ++ ++ ++ ++ ++ /* en = max energy difference amoung the 3 short FFTs for this granule */ ++ en = pplot->ers[gr][ch]; ++ sprintf(title2,"FFT%1i pe=%4.1fK del_en=%3.1f over=%i",gr, ++ ptemp->pe[grtemp][ch]/1000,en,pplot->over[gr][ch]); ++ ++ ++ ymn = 3; ++ ymx = 15; ++ xmn = 1; ++ xmx = n+1; /* a little extra because of the bar thickness */ ++ gpk_bargraph_draw(enerbox[gr],n,xcord,ycord, ++ xmn,ymn,xmx,ymx,1,title2,0,barcolor); ++ ++ ++ for (i=0; i ymx) ? ycord[i] : ymx; ++ ymn=(ycord[i] < ymn) ? ycord[i] : ymn; ++ } ++ gpk_bargraph_draw(enerbox[gr],n,xcord,ycord, ++ xmn,ymn,xmx,ymx,0,title2,3,&red); ++ ++ ++ for (i=0; i ymx) ? ycord[i] : ymx; ++ ymn=(ycord[i] < ymn) ? ycord[i] : ymn; ++ } ++ gpk_bargraph_draw(enerbox[gr],n,xcord,ycord, ++ xmn,ymn,xmx,ymx,0,title2,3,grcolor[gr]); ++ } ++ } ++ } ++ ++ ++ ++ static void update_progress() ++ { ++ char label[80]; ++ sprintf(label,"Frame:%4i/%4i %6.2fs", ++ pinfo->frameNum,gtkinfo.totalframes, pinfo->frametime); ++ gtk_progress_set_value (GTK_PROGRESS (frameprogress), (gdouble) pinfo->frameNum); ++ gtk_label_set_text(GTK_LABEL(framecounter),label); ++ } ++ ++ ++ ++ static void analyze() ++ { ++ if ( idle_keepgoing) { ++ idle_count = 0; ++ idle_count_max=0; ++ idle_keepgoing=0; ++ idle_finish=0; ++ } ++ plot_frame(); ++ update_progress(); ++ } ++ ++ static void plotclick( GtkWidget *widget, gpointer data ) ++ { analyze(); } ++ ++ ++ ++ ++ static int frameadv1( GtkWidget *widget, gpointer data ) ++ { ++ int i; ++ if (idle_keepgoing ){ ++ if (idle_back) { ++ /* frame displayed is the old frame. to advance, just swap in new frame */ ++ idle_back--; ++ pinfo = &Pinfo[READ_AHEAD+idle_back]; ++ }else{ ++ /* advance the frame by reading in a new frame */ ++ if (mp3done) { ++ /* dont try to read any more frames, and quit if "finish MP3" was selected */ ++ if (idle_finish) gtk_main_quit(); ++ idle_count_max=0; ++ } else { ++ /* read in the next frame */ ++ for (i=NUMPINFO-1 ; i>0 ; i--) ++ memcpy(&Pinfo[i],&Pinfo[i-1],sizeof(plotting_data)); ++ pinfo = &Pinfo[0]; ++ if (gtkinfo.filetype) { ++ /* analyzing a .mp3 file */ ++ pinfo->num_samples = readframe() ; ++ }else{ ++ pinfo->num_samples = makeframe(); ++ } ++ ++ ++ if (pinfo->sampfreq) ++ pinfo->frametime = (pinfo->frameNum-1)*1152.0/pinfo->sampfreq; ++ else pinfo->frametime=0; ++ if (!pinfo->num_samples) { ++ mp3done = 1; ++ if (idle_finish) gtk_main_quit(); ++ } ++ ++ pinfo->totbits = 0; ++ { int gr,ch; ++ for (gr = 0 ; gr < 2 ; gr ++) ++ for (ch = 0 ; ch < 2 ; ch ++) { ++ gtkinfo.totshort += (pinfo->mpg123blocktype[gr][ch]==2); ++ gtkinfo.totmix += !(pinfo->mixed[gr][ch]==0); ++ pinfo->totbits += pinfo->mainbits[gr][ch]; ++ } ++ } ++ if (pinfo->frameNum > 1) /* start averaging at frame = 2*/ ++ gtkinfo.avebits = (gtkinfo.avebits*((pinfo->frameNum)-2) ++ + pinfo->totbits ) /(pinfo->frameNum-1); ++ ++ gtkinfo.maxbits=MAX(gtkinfo.maxbits,pinfo->totbits); ++ gtkinfo.totemph += !(pinfo->emph==0); ++ gtkinfo.totms += !(pinfo->ms_stereo==0); ++ gtkinfo.totis += !(pinfo->i_stereo==0); ++ ++ pinfo = &Pinfo[READ_AHEAD]; ++ } ++ } ++ ++ idle_count++; ++ if (gtkinfo.pupdate) plot_frame(); ++ update_progress(); ++ if ((idle_count>=idle_count_max) && (! idle_finish)) analyze(); ++ } ++ return 1; ++ } ++ ++ ++ static void frameadv( GtkWidget *widget, gpointer data ) ++ { ++ int adv; ++ ++ if (!strcmp((char *) data,"-1")) { ++ /* ignore if we've already gone back as far as possible */ ++ if (pinfo->frameNum==1 || (idle_back==NUMBACK)) return; ++ idle_back++; ++ pinfo = &Pinfo[READ_AHEAD+idle_back]; ++ analyze(); ++ return; ++ } ++ ++ ++ adv = 1; ++ if (!strcmp((char *) data,"1")) adv = 1; ++ if (!strcmp((char *) data,"10")) adv = 10; ++ if (!strcmp((char *) data,"100")) adv = 100; ++ if (!strcmp((char *) data,"finish")) idle_finish = 1; ++ ++ ++ if (idle_keepgoing) { ++ /* already running - que up additional frame advance requests */ ++ idle_count_max += adv; ++ } ++ else { ++ /* turn on idleing */ ++ idle_count_max = adv; ++ idle_count = 0; ++ idle_keepgoing = 1; ++ } ++ } ++ ++ ++ ++ ++ /* another callback */ ++ static void delete_event( GtkWidget *widget, ++ GdkEvent *event, ++ gpointer data ) ++ { ++ gtk_main_quit (); ++ } ++ ++ ++ ++ ++ ++ ++ ++ static void channel_option (GtkWidget *widget, gpointer data) ++ { ++ int option; ++ option = (int) data; ++ switch (option) { ++ case 1: ++ gtkinfo.msflag=0; ++ gtkinfo.chflag=0; ++ break; ++ case 2: ++ gtkinfo.msflag=0; ++ gtkinfo.chflag=1; ++ break; ++ case 3: ++ gtkinfo.msflag=1; ++ gtkinfo.chflag=0; ++ break; ++ case 4: ++ gtkinfo.msflag=1; ++ gtkinfo.chflag=1; ++ } ++ analyze(); ++ } ++ static void spec_option (GtkWidget *widget, gpointer data) ++ { ++ int option; ++ option = (int) data; ++ switch (option) { ++ case 1: ++ gtkinfo.kbflag=0; ++ break; ++ case 2: ++ gtkinfo.kbflag=1; ++ break; ++ case 3: ++ gtkinfo.flag123=0; ++ break; ++ case 4: ++ gtkinfo.flag123=1; ++ break; ++ case 5: ++ gtkinfo.pupdate=1; ++ break; ++ case 6: ++ gtkinfo.pupdate=0; ++ break; ++ } ++ analyze(); ++ } ++ ++ ++ ++ static void text_window (GtkWidget *widget, gpointer data) ++ { ++ int option; ++ GtkWidget *hbox,*vbox,*button,*box; ++ GtkWidget *textwindow,*vscrollbar; ++ char text[80]; ++ ++ option = (int) data; ++ ++ textwindow = gtk_window_new(GTK_WINDOW_DIALOG); ++ gtk_signal_connect_object (GTK_OBJECT (window), "delete_event", ++ GTK_SIGNAL_FUNC(gtk_widget_destroy), ++ GTK_OBJECT (textwindow)); ++ ++ gtk_container_set_border_width (GTK_CONTAINER (textwindow), 0); ++ vbox = gtk_vbox_new(FALSE,0); ++ hbox = gtk_hbox_new(FALSE,0); ++ ++ button = gtk_button_new_with_label ("close"); ++ gtk_signal_connect_object (GTK_OBJECT (button), "clicked", ++ GTK_SIGNAL_FUNC(gtk_widget_destroy), ++ GTK_OBJECT (textwindow)); ++ ++ box = gtk_text_new (NULL, NULL); ++ gtk_text_set_editable (GTK_TEXT (box), FALSE); ++ vscrollbar = gtk_vscrollbar_new (GTK_TEXT(box)->vadj); ++ ++ ++ switch (option) { ++ case 0: ++ gtk_window_set_title (GTK_WINDOW (textwindow), "Documentation"); ++ gtk_widget_set_usize(box,450,500); ++ gtk_text_set_word_wrap(GTK_TEXT(box),TRUE); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL, ++ "Frame header information: "\ ++ "First the bitrate, sampling frequency and mono, stereo or jstereo "\ ++ "indicators are displayed . If the bitstream is jstereo, then mid/side "\ ++ "stereo or intensity stereo may be on (indicated in red). If "\ ++ "de-emphasis is used, this is also indicated in red. The mdb value is "\ ++ "main_data_begin. The encoded data starts this many bytes *before* the "\ ++ "frame header. A large value of mdb means the bitstream has saved some "\ ++ "bits into the reservoir, which it may allocate for some future frame. "\ ++ "The two numbers after mdb are the size (in bits) used to encode the "\ ++ "MDCT coefficients for this frame, followed byt the size of the bit "\ ++ "resevoir before encoding this frame. The maximum frame size and a "\ ++ "running average are given in the Stats pull down menu. A large "\ ++ "maximum frame size indicates the bitstream has made use of the bit "\ ++ "reservoir. \n\n",-1); ++ ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL, ++ "PCM data (top graph): "\ ++ "The PCM data is plotted in black. The layer3 frame is divided into 2 "\ ++ "granules of 576 samples (marked with yellow vertical lines). In the "\ ++ "case of normal, start and stop blocks, the MDCT coefficients for each "\ ++ "granule are computed using a 1152 sample window centered over the "\ ++ "granule. In the case of short blocks, the granule is further divided "\ ++ "into 3 blocks of 192 samples (also marked with yellow vertical lines)."\ ++ "The MDCT coefficients for these blocks are computed using 384 sample "\ ++ "windows centered over the 192 sample window. (This info not available "\ ++ "when analyzing .mp3 files.) For the psycho-acoustic model, a windowed "\ ++ "FFT is computed for each granule. The range of these windows "\ ++ "is denoted by the blue and green bars.\n\n",-1); ++ ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL, ++ "PCM re-synthesis data (second graph): "\ ++ "Same as the PCM window described above. The data displayed is the "\ ++ "result of encoding and then decoding the original sample. \n\n",-1); ++ ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL, ++ "MDCT windows: "\ ++ "Shows the energy in the MDCT spectrum for granule 0 (left window) "\ ++ "and granule 1 (right window). The text also shows the blocktype "\ ++ "used, the number of bits used to encode the coefficients and the "\ ++ "number of extra bits allocated from the reservoir. The MDCT pull down "\ ++ "window will toggle between the original unquantized MDCT coefficients "\ ++ "and the compressed (quantized) coefficients.\n\n",-1); ++ ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL, ++ "FFT window: "\ ++ "The gray bars show the energy in the FFT spectrum used by the "\ ++ "psycho-acoustic model. Granule 0 is in the left window, granule 1 in "\ ++ "the right window. The green and blue bars show how much distortion is "\ ++ "allowable, as computed by the psycho-acoustic model. The red bars show "\ ++ "the actual distortion after encoding. There is one FFT for each "\ ++ "granule, computed with a 1024 Hann window centered over the "\ ++ "appropriate granule. (the range of this 1024 sample window is shown "\ ++ "by the blue and green bars in the PCM data window). The Spectrum pull "\ ++ "down window will toggle between showing the energy in equally spaced "\ ++ "frequency domain and the scale factor bands used by layer3. Finally, "\ ++ "the perceptual entropy, total energy and number of scalefactor bands "\ ++ "with audible distortion is shown. (This info not available when "\ ++ "analyzing .mp3 files.)",-1); ++ ++ break; ++ case 1: ++ /* Set the about box information */ ++ gtk_window_set_title (GTK_WINDOW (textwindow), "About"); ++ gtk_widget_set_usize(box,350,260); ++ ++ sprintf(text,"LAME version %s \nwww.sulaco.org/mp3\n\n",get_lame_version()); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ ++ sprintf(text,"psycho-acoustic model: GPSYCHO version %s \nMark Taylor\n\n",get_psy_version()); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ ++ sprintf(text,"frame analyzer: MP3x version %s \nMark Taylor\n\n",get_mp3x_version()); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL, ++ "decoder: mpg123/mpglib .59q \nMichael Hipp \nwww-ti.informatik.uni-tuebingen.de/~hippm/mpg123.html\n\n",-1); ++ ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL, ++ "Encoder, decoder & psy-models based on ISO\ndemonstration source. ",-1); ++ break; ++ ++ case 2: ++ gtk_window_set_title (GTK_WINDOW (textwindow), "Statistics"); ++ gtk_widget_set_usize(box,350,260); ++ sprintf(text,"frames processed so far: %i \n",Pinfo[0].frameNum); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ sprintf(text,"granules processed so far: %i \n\n",4*Pinfo[0].frameNum); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ sprintf(text,"mean bits/frame (approximate): %i\n", ++ gtkinfo.approxbits); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ sprintf(text,"mean bits/frame (from LAME): %i\n", ++ 4*Pinfo[0].mean_bits); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ sprintf(text,"bitsize of largest frame: %i \n",gtkinfo.maxbits); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ sprintf(text,"average bits/frame: %3.1f \n\n",gtkinfo.avebits); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ sprintf(text,"ms_stereo frames: %i \n",gtkinfo.totms); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ sprintf(text,"i_stereo frames: %i \n",gtkinfo.totis); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ sprintf(text,"de-emphasis frames: %i \n",gtkinfo.totemph); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ sprintf(text,"short block granules: %i \n",gtkinfo.totshort); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ sprintf(text,"mixed block granules: %i \n",gtkinfo.totmix); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ break; ++ } ++ ++ ++ ++ gtk_widget_show (vscrollbar); ++ gtk_widget_show (box); ++ gtk_widget_show (vbox); ++ gtk_widget_show (hbox); ++ gtk_widget_show (button); ++ ++ gtk_box_pack_start (GTK_BOX(hbox), box, FALSE, TRUE, 0); ++ gtk_box_pack_start(GTK_BOX(hbox), vscrollbar, FALSE, FALSE, 0); ++ gtk_box_pack_start (GTK_BOX (vbox), hbox, FALSE, TRUE, 0); ++ gtk_box_pack_end (GTK_BOX (vbox), button, FALSE, TRUE, 0); ++ gtk_container_add (GTK_CONTAINER (textwindow), vbox); ++ gtk_widget_show(textwindow); ++ ++ } ++ ++ ++ ++ ++ /* #include */ ++ ++ ++ /* This is the GtkItemFactoryEntry structure used to generate new menus. ++ Item 1: The menu path. The letter after the underscore indicates an ++ accelerator key once the menu is open. ++ Item 2: The accelerator key for the entry ++ Item 3: The callback function. ++ Item 4: The callback action. This changes the parameters with ++ which the function is called. The default is 0. ++ Item 5: The item type, used to define what kind of an item it is. ++ Here are the possible values: ++ ++ NULL -> "" ++ "" -> "" ++ "" -> create a title item ++ "<Item>" -> create a simple item ++ "<CheckItem>" -> create a check item ++ "<ToggleItem>" -> create a toggle item ++ "<RadioItem>" -> create a radio item ++ <path> -> path of a radio item to link against ++ "<Separator>" -> create a separator ++ "<Branch>" -> create an item to hold sub items ++ "<LastBranch>" -> create a right justified branch ++ */ ++ ++ static GtkItemFactoryEntry menu_items[] = { ++ {"/_File", NULL, NULL, 0, "<Branch>"}, ++ /* ++ {"/File/_New", "<control>N", print_hello, 0, NULL}, ++ {"/File/_Open", "<control>O", print_hello, 0, NULL}, ++ {"/File/_Save", "<control>S", print_hello, 0, NULL}, ++ {"/File/Save _As", NULL, NULL, 0, NULL}, ++ {"/File/sep1", NULL, NULL, 0, "<Separator>"}, ++ {"/File/Quit", "<control>Q", gtk_main_quit, 0, NULL}, ++ */ ++ {"/File/_Quit", "<control>Q", delete_event, 0, NULL}, ++ ++ {"/_Plotting", NULL, NULL, 0, "<Branch>"}, ++ {"/Plotting/_While advancing" , NULL, spec_option, 5, NULL}, ++ {"/Plotting/_After advancing", NULL, spec_option, 6, NULL}, ++ ++ {"/_Channel", NULL, NULL, 0, "<Branch>"}, ++ {"/Channel/show _Left" , NULL, channel_option, 1, NULL}, ++ {"/Channel/show _Right", NULL, channel_option, 2, NULL}, ++ {"/Channel/show _Mid" , NULL, channel_option, 3, NULL}, ++ {"/Channel/show _Side", NULL, channel_option, 4, NULL}, ++ ++ {"/_Spectrum", NULL, NULL, 0, "<Branch>"}, ++ {"/Spectrum/_Scalefactor bands", NULL, spec_option, 1, NULL}, ++ {"/Spectrum/_Wave number", NULL, spec_option, 2, NULL}, ++ ++ {"/_MDCT", NULL, NULL, 0, "<Branch>"}, ++ {"/MDCT/_Original", NULL, spec_option, 3, NULL}, ++ {"/MDCT/_Compressed", NULL, spec_option, 4, NULL}, ++ ++ {"/_Stats", NULL, NULL, 0, "<Branch>"}, ++ {"/Stats/_Show", NULL, text_window, 2, NULL}, ++ ++ {"/_Help", NULL, NULL, 0, "<LastBranch>"}, ++ {"/_Help/_Documentation", NULL, text_window, 0, NULL}, ++ {"/_Help/_About", NULL, text_window, 1, NULL}, ++ }; ++ ++ ++ static void get_main_menu(GtkWidget *window, GtkWidget ** menubar) { ++ int nmenu_items = sizeof(menu_items) / sizeof(menu_items[0]); ++ GtkItemFactory *item_factory; ++ GtkAccelGroup *accel_group; ++ ++ accel_group = gtk_accel_group_new(); ++ ++ /* This function initializes the item factory. ++ Param 1: The type of menu - can be GTK_TYPE_MENU_BAR, GTK_TYPE_MENU, ++ or GTK_TYPE_OPTION_MENU. ++ Param 2: The path of the menu. ++ Param 3: A pointer to a gtk_accel_group. The item factory sets up ++ the accelerator table while generating menus. ++ */ ++ ++ item_factory = gtk_item_factory_new(GTK_TYPE_MENU_BAR, "<main>", ++ accel_group); ++ ++ /* This function generates the menu items. Pass the item factory, ++ the number of items in the array, the array itself, and any ++ callback data for the the menu items. */ ++ gtk_item_factory_create_items(item_factory, nmenu_items, menu_items, NULL); ++ ++ /* Attach the new accelerator group to the window. */ ++ gtk_accel_group_attach (accel_group, GTK_OBJECT (window)); ++ ++ if (menubar) ++ /* Finally, return the actual menu bar created by the item factory. */ ++ *menubar = gtk_item_factory_get_widget(item_factory, "<main>"); ++ } ++ ++ ++ ++ ++ int gtkcontrol(int totalframes,int filetype,char *infile, int gpsycho) ++ { ++ /* GtkWidget is the storage type for widgets */ ++ GtkWidget *button; ++ GtkAdjustment *adj; ++ GtkWidget *mbox; /* main box */ ++ GtkWidget *box1; /* frame control buttons go */ ++ GtkWidget *box2; /* frame counters */ ++ GtkWidget *box3; /* frame header info */ ++ GtkWidget *table; /* table for all the plotting areas */ ++ GtkWidget *menubar; ++ ++ gint tableops,graphx,graphy; ++ char frameinfo[80]; ++ ++ graphx = 500; /* minimum allowed size of pixmap */ ++ graphy = 100; ++ ++ ++ /* set some global defaults/variables */ ++ gtkinfo.totalframes = totalframes; ++ gtkinfo.filetype = filetype; ++ gtkinfo.msflag=0; ++ gtkinfo.chflag=0; ++ gtkinfo.kbflag=0; ++ gtkinfo.flag123=filetype; ++ gtkinfo.pupdate=0; ++ gtkinfo.avebits = 0; ++ gtkinfo.maxbits = 0; ++ gtkinfo.approxbits = 0; ++ gtkinfo.totemph = 0; ++ gtkinfo.totms = 0; ++ gtkinfo.totis = 0; ++ gtkinfo.totshort = 0; ++ gtkinfo.totmix = 0; ++ ++ gtkinfo.gpsycho=gpsycho; ++ memset((char *) Pinfo, 0, sizeof(Pinfo)); ++ pinfo = &Pinfo[READ_AHEAD]; ++ ++ strcpy(frameinfo,"MP3x: "); ++ strncat(frameinfo,infile,70); ++ ++ window = gtk_window_new (GTK_WINDOW_TOPLEVEL); ++ gtk_window_set_title (GTK_WINDOW (window), frameinfo); ++ gtk_signal_connect (GTK_OBJECT (window), "delete_event", ++ GTK_SIGNAL_FUNC (delete_event), NULL); ++ gtk_container_set_border_width (GTK_CONTAINER (window), 0); ++ ++ ++ mbox = gtk_vbox_new(FALSE, 0); ++ ++ ++ /* layout of mbox */ ++ box1 = gtk_hbox_new(FALSE, 0); ++ box2 = gtk_hbox_new(FALSE, 0); ++ box3 = gtk_hbox_new(FALSE, 0); ++ table = gtk_table_new (4, 2, FALSE); ++ tableops = GTK_FILL | GTK_EXPAND | GTK_SHRINK; ++ get_main_menu(window, &menubar); ++ ++ gtk_box_pack_start(GTK_BOX(mbox), menubar, FALSE, TRUE, 0); ++ gtk_box_pack_end (GTK_BOX (mbox), box1, FALSE, TRUE, 0); ++ gtk_box_pack_end(GTK_BOX (mbox),box2, FALSE, TRUE, 0); ++ gtk_box_pack_start(GTK_BOX (mbox),box3, FALSE, TRUE, 0); ++ gtk_box_pack_start (GTK_BOX (mbox), table, TRUE, TRUE, 0); ++ gtk_container_add (GTK_CONTAINER (window), mbox); ++ ++ ++ /*********************************************************************/ ++ /* stuff in box3 frame header info */ ++ /*********************************************************************/ ++ /* ++ headerbox = gtk_label_new(" "); ++ gtk_label_set_justify(GTK_LABEL(headerbox),GTK_JUSTIFY_LEFT); ++ */ ++ headerbox = gtk_text_new (NULL, NULL); ++ gtk_text_set_editable (GTK_TEXT (headerbox), FALSE); ++ gtk_widget_set_usize(headerbox,200,20); ++ gtk_widget_show (headerbox); ++ gtk_box_pack_start(GTK_BOX (box3),headerbox, TRUE, TRUE, 0); ++ ++ ++ ++ /*********************************************************************/ ++ /* stuff in box2 frame counters */ ++ /*********************************************************************/ ++ framecounter = gtk_label_new(""); ++ gtk_widget_show(framecounter); ++ gtk_box_pack_start(GTK_BOX (box2),framecounter, FALSE, TRUE, 0); ++ ++ adj = (GtkAdjustment *) gtk_adjustment_new (0, 1,(gint) totalframes, 0, 0, 0); ++ frameprogress = gtk_progress_bar_new_with_adjustment (adj); ++ /* Set the format of the string that can be displayed in the ++ * trough of the progress bar: ++ * %p - percentage ++ * %v - value ++ * %l - lower range value ++ * %u - upper range value */ ++ gtk_progress_set_format_string (GTK_PROGRESS (frameprogress), ++ "%p%%"); ++ gtk_progress_set_value (GTK_PROGRESS (frameprogress), (gdouble) 0); ++ gtk_progress_set_show_text (GTK_PROGRESS (frameprogress),TRUE); ++ gtk_widget_show (frameprogress); ++ gtk_box_pack_end (GTK_BOX (box2), frameprogress, FALSE, TRUE, 0); ++ ++ ++ ++ /*********************************************************************/ ++ /* stuff in box1 buttons along bottom */ ++ /*********************************************************************/ ++ button = gtk_button_new_with_label ("-1"); ++ gtk_signal_connect (GTK_OBJECT (button), "clicked", ++ GTK_SIGNAL_FUNC (frameadv), (gpointer) "-1"); ++ gtk_box_pack_start(GTK_BOX(box1), button, TRUE, TRUE, 0); ++ gtk_widget_show(button); ++ ++ button = gtk_button_new_with_label ("+1"); ++ gtk_signal_connect (GTK_OBJECT (button), "clicked", ++ GTK_SIGNAL_FUNC (frameadv), (gpointer) "1"); ++ gtk_box_pack_start(GTK_BOX(box1), button, TRUE, TRUE, 0); ++ gtk_widget_show(button); ++ ++ button = gtk_button_new_with_label ("+10"); ++ gtk_signal_connect (GTK_OBJECT (button), "clicked", ++ GTK_SIGNAL_FUNC (frameadv), (gpointer) "10"); ++ gtk_box_pack_start(GTK_BOX(box1), button, TRUE, TRUE, 0); ++ gtk_widget_show(button); ++ ++ button = gtk_button_new_with_label ("+100"); ++ gtk_signal_connect (GTK_OBJECT (button), "clicked", ++ GTK_SIGNAL_FUNC (frameadv), (gpointer) "100"); ++ gtk_box_pack_start(GTK_BOX(box1), button, TRUE, TRUE, 0); ++ gtk_widget_show(button); ++ ++ button = gtk_button_new_with_label ("finish mp3"); ++ gtk_signal_connect (GTK_OBJECT (button), "clicked", ++ GTK_SIGNAL_FUNC (frameadv), (gpointer) "finish"); ++ gtk_box_pack_start(GTK_BOX(box1), button, TRUE, TRUE, 0); ++ gtk_widget_show(button); ++ ++ button = gtk_button_new_with_label ("stop/plot"); ++ gtk_signal_connect (GTK_OBJECT (button), "clicked", ++ GTK_SIGNAL_FUNC (plotclick), NULL); ++ gtk_box_pack_start(GTK_BOX(box1), button, TRUE, TRUE, 0); ++ gtk_widget_show(button); ++ ++ ++ /*********************************************************************/ ++ /* stuff in table. all the plotting windows */ ++ /*********************************************************************/ ++ pcmbox = gpk_plot_new(graphx,graphy); ++ gtk_table_attach (GTK_TABLE(table),pcmbox,0,2,0,1,tableops,tableops,2,2 ); ++ gtk_widget_show (pcmbox); ++ ++ winbox = gpk_plot_new(graphy,graphy); ++ gtk_table_attach(GTK_TABLE(table),winbox,0,2,1,2,tableops,tableops,2,2); ++ gtk_widget_show (winbox); ++ ++ ++ mdctbox[0] = gpk_plot_new(graphy,graphy); ++ gtk_table_attach(GTK_TABLE(table),mdctbox[0],0,1,2,3,tableops,tableops,2,2); ++ gtk_widget_show (mdctbox[0]); ++ ++ mdctbox[1] = gpk_plot_new(graphy,graphy); ++ gtk_table_attach (GTK_TABLE(table),mdctbox[1],1,2,2,3,tableops,tableops,2,2); ++ gtk_widget_show (mdctbox[1]); ++ ++ enerbox[0] = gpk_plot_new(graphy,graphy); ++ gtk_table_attach(GTK_TABLE(table),enerbox[0],0,1,3,4,tableops,tableops,2,2); ++ gtk_widget_show (enerbox[0]); ++ ++ enerbox[1] = gpk_plot_new(graphy,graphy); ++ gtk_table_attach (GTK_TABLE(table),enerbox[1],1,2,3,4,tableops,tableops,2,2); ++ gtk_widget_show (enerbox[1]); ++ ++ ++ ++ ++ gtk_idle_add((GtkFunction) frameadv1, NULL); ++ gtk_widget_show(menubar); ++ gtk_widget_show(box2); ++ gtk_widget_show(box3); ++ gtk_widget_show(table); ++ gtk_widget_show(box1); ++ gtk_widget_show (mbox); ++ gtk_widget_show (window); /* show smallest allowed window */ ++ ++ /* make window bigger. */ ++ /* now the user will be able to shrink it, if desired */ ++ /* gtk_widget_set_usize(mbox,500,500); */ ++ /* gtk_widget_show (window); */ /* show smallest allowed window */ ++ ++ ++ ++ idle_keepgoing=1; /* processing of frames is ON */ ++ idle_count_max=READ_AHEAD+1; /* number of frames to process before plotting */ ++ idle_count=0; /* pause & plot when idle_count=idle_count_max */ ++ ++ ++ gtk_main (); ++ if (!mp3done) exit(2); ++ return(0); ++ } ++ ++ #endif ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ +diff -r -c -N encoder/gtkanal.h lame3.13/gtkanal.h +*** encoder/gtkanal.h Wed Dec 31 17:00:00 1969 +--- lame3.13/gtkanal.h Mon Jun 21 23:59:02 1999 +*************** +*** 0 **** +--- 1,104 ---- ++ /* ++ layerIII enc->dec delay: 1056 (observed) ++ layerII enc->dec dealy: 480 (observed) ++ ++ ++ polyphase 256-16 (dec or enc) ++ mdct 256+32 (9*32) (dec or enc) ++ total: 512+16 ++ ++ ++ SYNC_FLUSH calculation: ++ layerIII enc delay: 528 ++ extra bits to display full fft window: 224 ++ total: 752 ++ ++ layerII sync_flush: should be 464 ++ layerIII sync_flush: should be 752 ++ ++ ++ Also we would like to remove mpg123 decoder delay with ++ DECDELAY of -528 (not possible) ++ so delay by a enough to make the total delay 1 whole frame (1152-528) ++ */ ++ ++ ++ ++ #define ENCDELAY 528 ++ #define DECDELAY (1152-528) ++ ++ /* use this delay if you want to also include the data from the FFT window */ ++ #define WINDELAY (224 + ENCDELAY) ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ #ifdef HAVEGTK ++ ++ /* mpg123 ouput is delayed by 1 frame ++ * HOWEVER, sometimes mpg123 doesn't produce a frame until we send the ++ * next frame. Everytime this happens, bump up MPGLAG. ++ */ ++ #define READ_AHEAD 10 /* number of frames to read ahead */ ++ #define MAXMPGLAG READ_AHEAD /* if the mpg123 lag becomes bigger than this we have to stop */ ++ #define NUMBACK 5 /* number of frames we can back up */ ++ #define NUMPINFO (NUMBACK+READ_AHEAD+1) ++ ++ #define BLKSIZE 1024 ++ #define HBLKSIZE 513 ++ #define SBMAX_l 21 ++ #define SBMAX_s 12 ++ ++ ++ ++ typedef struct { ++ int frameNum; /* current frame number */ ++ int num_samples; /* number of pcm samples read for this frame */ ++ double frametime; /* starting time of frame, in seconds */ ++ double pcmdata[2][1152+WINDELAY]; ++ double pcmdata2[2][1152+DECDELAY]; ++ double energy[2][2][BLKSIZE]; ++ double xr[2][2][576]; ++ double mpg123xr[2][2][576]; ++ double pe[2][2]; ++ double ms_ratio[2]; ++ ++ double thr[2][2][SBMAX_l]; ++ double en[2][2][SBMAX_l]; ++ double thr_s[2][2][3*SBMAX_s]; ++ double en_s[2][2][3*SBMAX_s]; ++ double ers[2][2]; ++ ++ double xfsf[2][2][SBMAX_l]; ++ double xfsf_s[2][2][3*SBMAX_s]; ++ ++ int mpglag; ++ int qss[2][2]; ++ int sub_gain[2][2][3]; ++ int over[2][2]; ++ double noise[2][2]; ++ int blocktype[2][2]; ++ int scalefac_scale[2][2]; ++ int mpg123blocktype[2][2]; ++ int mixed[2][2]; ++ int mainbits[2][2]; ++ int LAMEmainbits[2][2]; ++ int stereo,js,ms_stereo,i_stereo,emph,bitrate,sampfreq,maindata; ++ int scfsi,mean_bits,resvsize; ++ int totbits; ++ } plotting_data; ++ ++ ++ int gtkcontrol(int,int,char *,int); ++ ++ extern int gtkflag; ++ extern plotting_data *pinfo; ++ ++ #endif +diff -r -c -N encoder/huffman.c lame3.13/huffman.c +*** encoder/huffman.c Wed Jan 22 02:43:15 1997 +--- lame3.13/huffman.c Sat Jun 19 19:52:37 1999 +*************** +*** 1,224 **** +! /********************************************************************** +! * ISO MPEG Audio Subgroup Software Simulation Group (1996) +! * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +! * +! * $Id: huffman.c,v 1.1 1996/02/14 04:04:23 rowlands Exp $ +! * +! * $Log: huffman.c,v $ +! * Revision 1.1 1996/02/14 04:04:23 rowlands +! * Initial revision +! * +! * Received from Mike Coleman +! **********************************************************************/ +! /********************************************************************** +! * changes made since last update: * +! * date programmers comment * +! *27.2.92 F.O.Witte (ITT Intermetall) * +! * email: otto.witte@itt-sc.de * +! * tel: ++49 (761)517-125 * +! * fax: ++49 (761)517-880 * +! *12.6.92 J. Pineda Added sign bit to decoder. * +! * 08/24/93 M. Iwadare Changed for 1 pass decoding. * +! *--------------------------------------------------------------------* +! * 7/14/94 Juergen Koller Bug fixes in Layer III code * +! *********************************************************************/ + #include <stdlib.h> + #include "common.h" + #include "huffman.h" +- +- HUFFBITS dmask = 1 << (sizeof(HUFFBITS)*8-1); +- unsigned int hs = sizeof(HUFFBITS)*8; +- +- struct huffcodetab ht[HTN]; /* array of all huffcodtable headers */ +- /* 0..31 Huffman code table 0..31 */ +- /* 32,33 count1-tables */ +- +- /* read the huffman encode table */ +- int read_huffcodetab(fi) +- FILE *fi; +- { +- +- char line[100],command[40],huffdata[40]; +- unsigned int t,i,j,k,nn,x,y,n=0; +- unsigned int xl, yl, len; +- HUFFBITS h; +- int hsize; +- +- hsize = sizeof(HUFFBITS)*8; +- do { +- fgets(line,99,fi); +- } while ((line[0] == '#') || (line[0] < ' ') ); +- +- do { +- while ((line[0]=='#') || (line[0] < ' ')) { +- fgets(line,99,fi); +- } +- +- sscanf(line,"%s %s %u %u %u",command,ht[n].tablename, +- &xl,&yl,&ht[n].linbits); +- if (strcmp(command,".end")==0) +- return n; +- else if (strcmp(command,".table")!=0) { +- fprintf(stderr,"huffman table %u data corrupted\n",n); +- return -1; +- } +- ht[n].linmax = (1<<ht[n].linbits)-1; +- +- sscanf(ht[n].tablename,"%u",&nn); +- if (nn != n) { +- fprintf(stderr,"wrong table number %u\n",n); +- return(-2); +- } + +- ht[n].xlen = xl; +- ht[n].ylen = yl; + +! do { +! fgets(line,99,fi); +! } while ((line[0] == '#') || (line[0] < ' ')); + +! sscanf(line,"%s %u",command,&t); +! if (strcmp(command,".reference")==0) { +! ht[n].ref = t; +! ht[n].table = ht[t].table; +! ht[n].hlen = ht[t].hlen; +! if ( (xl != ht[t].xlen) || +! (yl != ht[t].ylen) ) { +! fprintf(stderr,"wrong table %u reference\n",n); +! return (-3); +! }; +! do { +! fgets(line,99,fi); +! } while ((line[0] == '#') || (line[0] < ' ') ); +! } +! else { +! ht[n].ref = -1; +! ht[n].table=(HUFFBITS *) calloc(xl*yl,sizeof(HUFFBITS)); +! if (ht[n].table == NULL) { +! fprintf(stderr,"unsufficient heap error\n"); +! return (-4); +! } +! ht[n].hlen=(unsigned char *) calloc(xl*yl,sizeof(unsigned char)); +! if (ht[n].hlen == NULL) { +! fprintf(stderr,"unsufficient heap error\n"); +! return (-4); +! } +! for (i=0; i<xl; i++) { +! for (j=0;j<yl; j++) { +! if (xl>1) +! sscanf(line,"%u %u %u %s",&x, &y, &len,huffdata); +! else +! sscanf(line,"%u %u %s",&x,&len,huffdata); +! h=0;k=0; +! while (huffdata[k]) { +! h <<= 1; +! if (huffdata[k] == '1') +! h++; +! else if (huffdata[k] != '0'){ +! fprintf(stderr,"huffman-table %u bit error\n",n); +! return (-5); +! }; +! k++; +! }; +! if (k != len) { +! fprintf(stderr, +! "warning: wrong codelen in table %u, pos [%2u][%2u]\n", +! n,i,j); +! }; +! ht[n].table[i*xl+j] = h; +! ht[n].hlen[i*xl+j] = (unsigned char) len; +! do { +! fgets(line,99,fi); +! } while ((line[0] == '#') || (line[0] < ' ')); +! } +! } +! } +! n++; +! } while (1); +! } + +! /* read the huffman decoder table */ +! int read_decoder_table(fi) +! FILE *fi; + { +! int n,i,nn,t; +! unsigned int v0,v1; +! char command[100],line[100]; +! for (n=0;n<HTN;n++) { +! /* .table number treelen xlen ylen linbits */ +! do { +! fgets(line,99,fi); +! } while ((line[0] == '#') || (line[0] < ' ')); +! +! sscanf(line,"%s %s %u %u %u %u",command,ht[n].tablename, +! &ht[n].treelen, &ht[n].xlen, &ht[n].ylen, &ht[n].linbits); +! if (strcmp(command,".end")==0) +! return n; +! else if (strcmp(command,".table")!=0) { +! fprintf(stderr,"huffman table %u data corrupted\n",n); +! return -1; +! } +! ht[n].linmax = (1<<ht[n].linbits)-1; +! +! sscanf(ht[n].tablename,"%u",&nn); +! if (nn != n) { +! fprintf(stderr,"wrong table number %u\n",n); +! return(-2); +! } +! do { +! fgets(line,99,fi); +! } while ((line[0] == '#') || (line[0] < ' ')); +! +! sscanf(line,"%s %u",command,&t); +! if (strcmp(command,".reference")==0) { +! ht[n].ref = t; +! ht[n].val = ht[t].val; +! ht[n].treelen = ht[t].treelen; +! if ( (ht[n].xlen != ht[t].xlen) || +! (ht[n].ylen != ht[t].ylen) ) { +! fprintf(stderr,"wrong table %u reference\n",n); +! return (-3); +! }; +! while ((line[0] == '#') || (line[0] < ' ') ) { +! fgets(line,99,fi); +! } +! } +! else if (strcmp(command,".treedata")==0) { +! ht[n].ref = -1; +! if ( ht[n].treelen ) +! { +! ht[n].val = (unsigned char (*)[2]) +! calloc(2*(ht[n].treelen),sizeof(unsigned char)); +! if (ht[n].val == NULL) { +! fprintf(stderr, "heaperror at table %d\n",n); +! exit (-10); +! } +! } +! else +! ht[n].val = NULL; +! for (i=0;i<ht[n].treelen; i++) { +! fscanf(fi,"%x %x",&v0, &v1); +! ht[n].val[i][0]=(unsigned char)v0; +! ht[n].val[i][1]=(unsigned char)v1; +! } +! fgets(line,99,fi); /* read the rest of the line */ +! } +! else { +! fprintf(stderr,"huffman decodertable error at table %d\n",n); +! } +! } +! return n; +! } + + + /* do the huffman coding, */ + /* note! for counta,countb - the 4 bit value is passed in y, set x to 0 */ + /* return value: 0-no error, 1 decode error */ +! void huffman_coder( x, y, h, bs) +! unsigned int x; /* x-value */ +! unsigned int y; /* y-value */ +! struct huffcodetab *h; /* pointer to huffman code record */ +! Bit_stream_struc *bs; /* pointer to open write bitstream */ + { + HUFFBITS huffbits; /* data left aligned */ + HUFFBITS linbitsX; +--- 1,133 ---- +! #include <string.h> + #include <stdlib.h> + #include "common.h" + #include "huffman.h" + + +! +! /* HUFFBITS dmask = 1 << (sizeof(HUFFBITS)*8-1); */ +! unsigned int hs = sizeof(HUFFBITS)*8; + +! static HUFFBITS t1HB[] = {1, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, +! 1, 0}; +! static HUFFBITS t2HB[] = {1, 2, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, +! 3, 1, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 3, 2, 0}; +! static HUFFBITS t3HB[] = {3, 2, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 1, 1, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 3, 2, 0}; +! static HUFFBITS t5HB[] = {1, 2, 6, 5, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 3, 1, 4, 4, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 7, 5, 7, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 6, 1, 1, 0}; +! static HUFFBITS t6HB[] = {7, 3, 5, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 6, 2, 3, 2, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 5, 4, 4, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 3, 3, 2, 0}; +! static HUFFBITS t7HB[] = {1, 2, 10, 19, 16, 10, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 3, 3, 7, 10, 5, 3, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 11, 4, 13, 17, 8, 4, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 12, 11, 18, 15, 11, 2, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 7, 6, 9, 14, 3, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 6, 4, 5, 3, 2, 0}; +! static HUFFBITS t8HB[] = {3, 4, 6, 18, 12, 5, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 5, 1, 2, 16, 9, 3, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 7, 3, 5, 14, 7, 3, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 19, 17, 15, 13, 10, 4, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 13, 5, 8, 11, 5, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 12, 4, 4, 1, 1, 0}; +! static HUFFBITS t9HB[] = {7, 5, 9, 14, 15, 7, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 6, 4, 5, 5, 6, 7, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 7, 6, 8, 8, 8, 5, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 15, 6, 9, 10, 5, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 11, 7, 9, 6, 4, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 14, 4, 6, 2, 6, 0}; +! static HUFFBITS t10HB[] = {1, 2, 10, 23, 35, 30, 12, 17, 0, 0, 0, 0, 0, 0, 0, 0, 3, 3, 8, 12, 18, 21, 12, 7, 0, 0, 0, 0, 0, 0, 0, 0, 11, 9, 15, 21, 32, 40, 19, 6, 0, 0, 0, 0, 0, 0, 0, 0, 14, 13, 22, 34, 46, 23, 18, 7, 0, 0, 0, 0, 0, 0, 0, 0, 20, 19, 33, 47, 27, 22, 9, 3, 0, 0, 0, 0, 0, 0, 0, 0, 31, 22, 41, 26, 21, 20, 5, 3, 0, 0, 0, 0, 0, 0, 0, 0, 14, 13, 10, 11, 16, 6, 5, 1, 0, 0, 0, 0, 0, 0, 0, 0, 9, 8, 7, 8, 4 , 4, 2, 0}; +! static HUFFBITS t11HB[] = {3, 4, 10, 24, 34, 33, 21, 15, 0, 0, 0, 0, 0, 0, 0, 0, 5, 3, 4, 10, 32, 17, 11, 10, 0, 0, 0, 0, 0, 0, 0, 0, 11, 7, 13, 18, 30, 31, 20, 5, 0, 0, 0, 0, 0, 0, 0, 0, 25, 11, 19, 59, 27, 18, 12, 5, 0, 0, 0, 0, 0, 0, 0, 0, 35, 33, 31, 58, 30, 16, 7, 5, 0, 0, 0, 0, 0, 0, 0, 0, 28, 26, 32, 19, 17, 15, 8, 14, 0, 0, 0, 0, 0, 0, 0, 0, 14, 12, 9, 13, 14, 9, 4, 1, 0, 0, 0, 0, 0, 0, 0, 0, 11, 4, 6, 6, 6, 3, 2, 0}; +! static HUFFBITS t12HB[] = {9, 6, 16, 33, 41, 39, 38, 26, 0, 0, 0, 0, 0, 0, 0, 0, 7, 5, 6, 9, 23, 16, 26, 11, 0, 0, 0, 0, 0, 0, 0, 0, 17, 7, 11, 14, 21, 30, 10, 7, 0, 0, 0, 0, 0, 0, 0, 0, 17, 10, 15, 12, 18, 28, 14, 5, 0, 0, 0, 0, 0, 0, 0, 0, 32, 13, 22, 19, 18, 16, 9, 5, 0, 0, 0, 0, 0, 0, 0, 0, 40, 17, 31, 29, 17, 13, 4, 2, 0, 0, 0, 0, 0, 0, 0, 0, 27, 12, 11, 15, 10, 7, 4, 1, 0, 0, 0, 0, 0, 0, 0, 0, 27, 12, 8, 12 , 6, 3, 1, 0}; +! static HUFFBITS t13HB[] = {1, 5, 14, 21, 34, 51, 46, 71, 42, 52, 68, 52, 67, 44, 43, 19, 3, 4, 12, 19, 31, 26, 44, 33, 31, 24, 32, 24, 31, 35, 22, 14, 15, 13, 23, 36, 59, 49, 77, 65, 29, 40, 30, 40, 27, 33, 42, 16, 22, +! 20, 37, 61, 56, 79, 73, 64, 43, 76, 56, 37, 26, 31, 25, 14, 35, 16, 60, 57, 97, 75, 114, 91, 54, 73, 55, 41, 48, 53, 23, 24, 58, 27, 50, 96, 76, 70, 93, 84, 77, 58, 79, 29, 74, 49, 41, 17, 47, +! 45, 78, 74, 115, 94, 90, 79, 69, 83, 71, 50, 59, 38, 36, 15, 72, 34, 56, 95, 92, 85, 91, 90, 86, 73, 77, 65, 51, 44, 43, 42, 43, 20, 30, 44, 55, 78, 72, 87, 78, 61, 46, 54, 37, 30, 20, 16, 53, +! 25, 41, 37, 44, 59, 54, 81, 66, 76, 57, 54, 37, 18, 39, 11, 35, 33, 31, 57, 42, 82, 72, 80, 47, 58, 55, 21, 22, 26, 38, 22, 53, 25, 23, 38, 70, 60, 51, 36, 55, 26, 34, 23, 27, 14, 9, 7, 34, 32, +! 28, 39, 49, 75, 30, 52, 48, 40, 52, 28, 18, 17, 9, 5, 45, 21, 34, 64, 56, 50, 49, 45, 31, 19, 12, 15, 10, 7, 6, 3, 48, 23, 20, 39, 36, 35, 53, 21, 16, 23, 13, 10, 6, 1, 4, 2, 16, 15, 17, 27, 25, +! 20, 29, 11, 17, 12, 16, 8, 1, 1, 0, 1}; +! static HUFFBITS t15HB[] = {7, 12, 18, 53, 47, 76, 124, 108, 89, 123, 108, 119, 107, 81, 122, 63, 13, 5, 16, 27, 46, 36, 61, 51, 42, 70, 52, 83, 65, 41, 59, 36, 19, 17, 15, 24, 41, 34, 59, 48, 40, 64, 50, 78, 62, 80, 56, +! 33, 29, 28, 25, 43, 39, 63, 55, 93, 76, 59, 93, 72, 54, 75, 50, 29, 52, 22, 42, 40, 67, 57, 95, 79, 72, 57, 89, 69, 49, 66, 46, 27, 77, 37, 35, 66, 58, 52, 91, 74, 62, 48, 79, 63, 90, 62, 40, 38, +! 125, 32, 60, 56, 50, 92, 78, 65, 55, 87, 71, 51, 73, 51, 70, 30, 109, 53, 49, 94, 88, 75, 66, 122, 91, 73, 56, 42, 64, 44, 21, 25, 90, 43, 41, 77, 73, 63, 56, 92, 77, 66, 47, 67, 48, 53, 36, 20, +! 71, 34, 67, 60, 58, 49, 88, 76, 67, 106, 71, 54, 38, 39, 23, 15, 109, 53, 51, 47, 90, 82, 58, 57, 48, 72, 57, 41, 23, 27, 62, 9, 86, 42, 40, 37, 70, 64, 52, 43, 70, 55, 42, 25, 29, 18, 11, 11, +! 118, 68, 30, 55, 50, 46, 74, 65, 49, 39, 24, 16, 22, 13, 14, 7, 91, 44, 39, 38, 34, 63, 52, 45, 31, 52, 28, 19, 14, 8, 9, 3, 123, 60, 58, 53, 47, 43, 32, 22, 37, 24, 17, 12, 15, 10, 2, 1, 71, +! 37, 34, 30, 28, 20, 17, 26, 21, 16, 10, 6, 8, 6, 2, 0}; +! static HUFFBITS t16HB[] = {1, 5, 14, 44, 74, 63, 110, 93, 172, 149, 138, 242, 225, 195, 376, 17, 3, 4, 12, 20, 35, 62, 53, 47, 83, 75, 68, 119, 201, 107, 207, 9, 15, 13, 23, 38, 67, 58, 103, 90, 161, 72, 127, 117, +! 110, 209, 206, 16, 45, 21, 39, 69, 64, 114, 99, 87, 158, 140, 252, 212, 199, 387, 365, 26, 75, 36, 68, 65, 115, 101, 179, 164, 155, 264, 246, 226, 395, 382, 362, 9, 66, 30, 59, 56, 102, +! 185, 173, 265, 142, 253, 232, 400, 388, 378, 445, 16, 111, 54, 52, 100, 184, 178, 160, 133, 257, 244, 228, 217, 385, 366, 715, 10, 98, 48, 91, 88, 165, 157, 148, 261, 248, 407, 397, 372, +! 380, 889, 884, 8, 85, 84, 81, 159, 156, 143, 260, 249, 427, 401, 392, 383, 727, 713, 708, 7, 154, 76, 73, 141, 131, 256, 245, 426, 406, 394, 384, 735, 359, 710, 352, 11, 139, 129, 67, 125, +! 247, 233, 229, 219, 393, 743, 737, 720, 885, 882, 439, 4, 243, 120, 118, 115, 227, 223, 396, 746, 742, 736, 721, 712, 706, 223, 436, 6, 202, 224, 222, 218, 216, 389, 386, 381, 364, 888, +! 443, 707, 440, 437, 1728, 4, 747, 211, 210, 208, 370, 379, 734, 723, 714, 1735, 883, 877, 876, 3459, 865, 2, 377, 369, 102, 187, 726, 722, 358, 711, 709, 866, 1734, 871, 3458, 870, 434, +! 0, 12, 10, 7, 11, 10, 17, 11, 9, 13, 12, 10, 7, 5, 3, 1, 3}; +! static HUFFBITS t24HB[] = {15, 13, 46, 80, 146, 262, 248, 434, 426, 669, 653, 649, 621, 517, 1032, 88, 14, 12, 21, 38, 71, 130, 122, 216, 209, 198, 327, 345, 319, 297, 279, 42, 47, 22, 41, 74, 68, 128, 120, 221, +! 207, 194, 182, 340, 315, 295, 541, 18, 81, 39, 75, 70, 134, 125, 116, 220, 204, 190, 178, 325, 311, 293, 271, 16, 147, 72, 69, 135, 127, 118, 112, 210, 200, 188, 352, 323, 306, 285, +! 540, 14, 263, 66, 129, 126, 119, 114, 214, 202, 192, 180, 341, 317, 301, 281, 262, 12, 249, 123, 121, 117, 113, 215, 206, 195, 185, 347, 330, 308, 291, 272, 520, 10, 435, 115, 111, +! 109, 211, 203, 196, 187, 353, 332, 313, 298, 283, 531, 381, 17, 427, 212, 208, 205, 201, 193, 186, 177, 169, 320, 303, 286, 268, 514, 377, 16, 335, 199, 197, 191, 189, 181, 174, 333, +! 321, 305, 289, 275, 521, 379, 371, 11, 668, 184, 183, 179, 175, 344, 331, 314, 304, 290, 277, 530, 383, 373, 366, 10, 652, 346, 171, 168, 164, 318, 309, 299, 287, 276, 263, 513, 375, +! 368, 362, 6, 648, 322, 316, 312, 307, 302, 292, 284, 269, 261, 512, 376, 370, 364, 359, 4, 620, 300, 296, 294, 288, 282, 273, 266, 515, 380, 374, 369, 365, 361, 357, 2, 1033, 280, 278, +! 274, 267, 264, 259, 382, 378, 372, 367, 363, 360, 358, 356, 0, 43, 20, 19, 17, 15, 13, 11, 9, 7, 6, 4, 7, 5, 3, 1, 3}; +! static HUFFBITS t32HB[] = {1, 5, 4, 5, 6, 5, 4, 4, 7, 3, 6, 0, 7, 2, 3, 1}; +! static HUFFBITS t33HB[] = {15, 14, 13, 12, 11, 10, 9, 8, 7, 6, 5, 4, 3, 2, 1, 0}; +! +! static unsigned char t1l[] = {1, 3, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 2, 3}; +! static unsigned char t2l[] = {1, 3, 6, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 3, 3, 5, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 5, 5, 6}; +! static unsigned char t3l[] = {2, 2, 6, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 3, 2, 5, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 5, 5, 6}; +! static unsigned char t5l[] = {1, 3, 6, 7, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 3, 3, 6, 7, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 6, 6, 7, 8, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 7, 6, 7, 8}; +! static unsigned char t6l[] = {3, 3, 5, 7, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 3, 2, 4, 5, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 4, 4, 5, 6, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 6, 5, 6, 7}; +! static unsigned char t7l[] = {1, 3, 6, 8, 8, 9, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 3, 4, 6, 7, 7, 8, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 6, 5, 7, 8, 8, 9, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 7, 7, 8, 9, 9, 9, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 7, 7, 8, 9, 9, 10, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 8, 8, 9, 10, 10, 10}; +! static unsigned char t8l[] = {2, 3, 6, 8, 8, 9, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 3, 2, 4, 8, 8, 8, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 6, 4, 6, 8, 8, 9, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 8, 8, 8, 9, 9, 10, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 8, 7, 8, 9, 10, 10, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 9, 8, 9, 9, 11, 11}; +! static unsigned char t9l[] = {3, 3, 5, 6, 8, 9, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 3, 3, 4, 5, 6, 8, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 4, 4, 5, 6, 7, 8, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 6, 5, 6, 7, 7, 8, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 7, 6, 7, 7, 8, 9, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 8, 7, 8, 8, 9, 9}; +! static unsigned char t10l[] = {1, 3, 6, 8, 9, 9, 9, 10, 0, 0, 0, 0, 0, 0, 0, 0, 3, 4, 6, 7, 8, 9, 8, 8, 0, 0, 0, 0, 0, 0, 0, 0, 6, 6, 7, 8, 9, 10, 9, 9, 0, 0, 0, 0, 0, 0, 0, 0, 7, 7, 8, 9, 10, 10, 9, 10, 0, 0, 0, 0, 0, 0, 0, 0, 8, 8, 9, 10, 10, 10, 10, 10, 0, 0, 0, 0, 0, 0, 0, 0, 9, 9, 10, 10, 11, 11, 10, 11, 0, 0, 0, 0, 0, 0, 0, 0, 8, 8, 9, 10, 10, 10, 11, 11, 0, 0, 0, 0, 0, 0, 0, 0, 9, 8, 9, 10, 10, 11, 11, 11}; +! static unsigned char t11l[] = {2, 3, 5, 7, 8, 9, 8, 9, 0, 0, 0, 0, 0, 0, 0, 0, 3, 3, 4, 6, 8, 8, 7, 8, 0, 0, 0, 0, 0, 0, 0, 0, 5, 5, 6, 7, 8, 9, 8, 8, 0, 0, 0, 0, 0, 0, 0, 0, 7, 6, 7, 9, 8, 10, 8, 9, 0, 0, 0, 0, 0, 0, 0, 0, 8, 8, 8, 9, 9, 10, 9, 10, 0, 0, 0, 0, 0, 0, 0, 0, 8, 8, 9, 10, 10, 11, 10, 11, 0, 0, 0, 0, 0, 0, 0, 0, 8, 7, 7, 8, 9, 10, 10, 10, 0, 0, 0, 0, 0, 0, 0, 0, 8, 7, 8, 9, 10, 10, 10, 10}; +! static unsigned char t12l[] = {4, 3, 5, 7, 8, 9, 9, 9, 0, 0, 0, 0, 0, 0, 0, 0, 3, 3, 4, 5, 7, 7, 8, 8, 0, 0, 0, 0, 0, 0, 0, 0, 5, 4, 5, 6, 7, 8, 7, 8, 0, 0, 0, 0, 0, 0, 0, 0, 6, 5, 6, 6, 7, 8, 8, 8, 0, 0, 0, 0, 0, 0, 0, 0, 7, 6, 7, 7, 8, 8, 8, 9, 0, 0, 0, 0, 0, 0, 0, 0, 8, 7, 8, 8, 8, 9, 8, 9, 0, 0, 0, 0, 0, 0, 0, 0, 8, 7, 7, 8, 8, 9, 9, 10, 0, 0, 0, 0, 0, 0, 0, 0, 9, 8, 8, 9, 9, 9, 9, 10}; +! static unsigned char t13l[] = {1, 4, 6, 7, 8, 9, 9, 10, 9, 10, 11, 11, 12, 12, 13, 13, 3, 4, 6, 7, 8, 8, 9, 9, 9, 9, 10, 10, 11, 12, 12, 12, 6, 6, 7, 8, 9, 9, 10, 10, 9, 10, 10, 11, 11, 12, 13, 13, 7, 7, 8, 9, 9, 10, 10, 10, 10, 11, 11, 11, 11, 12, 13, 13, +! 8, 7, 9, 9, 10, 10, 11, 11, 10, 11, 11, 12, 12, 13, 13, 14, 9, 8, 9, 10, 10, 10, 11, 11, 11, 11, 12, 11, 13, 13, 14, 14, 9, 9, 10, 10, 11, 11, 11, 11, 11, 12, 12, 12, 13, 13, 14, 14, 10, 9, 10, 11, 11, 11, 12, 12, 12, 12, 13, 13, 13, 14, 16, 16, 9, 8, 9, 10, +! 10, 11, 11, 12, 12, 12, 12, 13, 13, 14, 15, 15, 10, 9, 10, 10, 11, 11, 11, 13, 12, 13, 13, 14, 14, 14, 16, 15, 10, 10, 10, 11, 11, 12, 12, 13, 12, 13, 14, 13, 14, 15, 16, 17, 11, 10, 10, 11, 12, 12, 12, 12, 13, 13, 13, 14, 15, 15, 15, 16, 11, 11, 11, 12, 12, +! 13, 12, 13, 14, 14, 15, 15, 15, 16, 16, 16, 12, 11, 12, 13, 13, 13, 14, 14, 14, 14, 14, 15, 16, 15, 16, 16, 13, 12, 12, 13, 13, 13, 15, 14, 14, 17, 15, 15, 15, 17, 16, 16, 12, 12, 13, 14, 14, 14, 15, 14, 15, 15, 16, 16, 19, 18, 19, 16}; +! static unsigned char t15l[] = {3, 4, 5, 7, 7, 8, 9, 9, 9, 10, 10, 11, 11, 11, 12, 13, 4, 3, 5, 6, 7, 7, 8, 8, 8, 9, 9, 10, 10, 10, 11, 11, 5, 5, 5, 6, 7, 7, 8, 8, 8, 9, 9, 10, 10, 11, 11, 11, 6, 6, 6, 7, 7, 8, 8, 9, 9, 9, 10, 10, 10, 11, 11, 11, 7, 6, 7, +! 7, 8, 8, 9, 9, 9, 9, 10, 10, 10, 11, 11, 11, 8, 7, 7, 8, 8, 8, 9, 9, 9, 9, 10, 10, 11, 11, 11, 12, 9, 7, 8, 8, 8, 9, 9, 9, 9, 10, 10, 10, 11, 11, 12, 12, 9, 8, 8, 9, 9, 9, 9, 10, 10, 10, 10, 10, 11, 11, 11, 12, 9, 8, 8, 9, 9, 9, 9, 10, 10, 10, 10, 11, 11, +! 12, 12, 12, 9, 8, 9, 9, 9, 9, 10, 10, 10, 11, 11, 11, 11, 12, 12, 12, 10, 9, 9, 9, 10, 10, 10, 10, 10, 11, 11, 11, 11, 12, 13, 12, 10, 9, 9, 9, 10, 10, 10, 10, 11, 11, 11, 11, 12, 12, 12, 13, 11, 10, 9, 10, 10, 10, 11, 11, 11, 11, 11, 11, 12, 12, 13, 13, +! 11, 10, 10, 10, 10, 11, 11, 11, 11, 12, 12, 12, 12, 12, 13, 13, 12, 11, 11, 11, 11, 11, 11, 11, 12, 12, 12, 12, 13, 13, 12, 13, 12, 11, 11, 11, 11, 11, 11, 12, 12, 12, 12, 12, 13, 13, 13, 13}; +! static unsigned char t16l[] = {1, 4, 6, 8, 9, 9, 10, 10, 11, 11, 11, 12, 12, 12, 13, 9, 3, 4, 6, 7, 8, 9, 9, 9, 10, 10, 10, 11, 12, 11, 12, 8, 6, 6, 7, 8, 9, 9, 10, 10, 11, 10, 11, 11, 11, 12, 12, 9, 8, 7, 8, 9, 9, 10, 10, 10, 11, 11, 12, 12, 12, 13, 13, +! 10, 9, 8, 9, 9, 10, 10, 11, 11, 11, 12, 12, 12, 13, 13, 13, 9, 9, 8, 9, 9, 10, 11, 11, 12, 11, 12, 12, 13, 13, 13, 14, 10, 10, 9, 9, 10, 11, 11, 11, 11, 12, 12, 12, 12, 13, 13, 14, 10, 10, 9, 10, 10, 11, 11, 11, 12, 12, 13, 13, 13, 13, 15, 15, 10, 10, 10, +! 10, 11, 11, 11, 12, 12, 13, 13, 13, 13, 14, 14, 14, 10, 11, 10, 10, 11, 11, 12, 12, 13, 13, 13, 13, 14, 13, 14, 13, 11, 11, 11, 10, 11, 12, 12, 12, 12, 13, 14, 14, 14, 15, 15, 14, 10, 12, 11, 11, 11, 12, 12, 13, 14, 14, 14, 14, 14, 14, 13, 14, 11, 12, 12, +! 12, 12, 12, 13, 13, 13, 13, 15, 14, 14, 14, 14, 16, 11, 14, 12, 12, 12, 13, 13, 14, 14, 14, 16, 15, 15, 15, 17, 15, 11, 13, 13, 11, 12, 14, 14, 13, 14, 14, 15, 16, 15, 17, 15, 14, 11, 9, 8, 8, 9, 9, 10, 10, 10, 11, 11, 11, 11, 11, 11, 11, 8}; +! static unsigned char t24l[] = {4, 4, 6, 7, 8, 9, 9, 10, 10, 11, 11, 11, 11, 11, 12, 9, 4, 4, 5, 6, 7, 8, 8, 9, 9, 9, 10, 10, 10, 10, 10, 8, 6, 5, 6, 7, 7, 8, 8, 9, 9, 9, 9, 10, 10, 10, 11, 7, 7, 6, 7, 7, 8, 8, 8, 9, 9, 9, 9, 10, 10, 10, 10, 7, 8, 7, 7, 8, +! 8, 8, 8, 9, 9, 9, 10, 10, 10, 10, 11, 7, 9, 7, 8, 8, 8, 8, 9, 9, 9, 9, 10, 10, 10, 10, 10, 7, 9, 8, 8, 8, 8, 9, 9, 9, 9, 10, 10, 10, 10, 10, 11, 7, 10, 8, 8, 8, 9, 9, 9, 9, 10, 10, 10, 10, 10, 11, 11, 8, 10, 9, 9, 9, 9, 9, 9, 9, 9, 10, 10, 10, 10, 11, 11, +! 8, 10, 9, 9, 9, 9, 9, 9, 10, 10, 10, 10, 10, 11, 11, 11, 8, 11, 9, 9, 9, 9, 10, 10, 10, 10, 10, 10, 11, 11, 11, 11, 8, 11, 10, 9, 9, 9, 10, 10, 10, 10, 10, 10, 11, 11, 11, 11, 8, 11, 10, 10, 10, 10, 10, 10, 10, 10, 10, 11, 11, 11, 11, 11, 8, 11, 10, 10, +! 10, 10, 10, 10, 10, 11, 11, 11, 11, 11, 11, 11, 8, 12, 10, 10, 10, 10, 10, 10, 11, 11, 11, 11, 11, 11, 11, 11, 8, 8, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 8, 8, 8, 8, 4}; +! static unsigned char t32l[] = {1, 4, 4, 5, 4, 6, 5, 6, 4, 5, 5, 6, 5, 6, 6, 6}; +! static unsigned char t33l[] = {4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4}; + +! #define NOREF -1 +! struct huffcodetab ht[HTN] = + { +! { 0, 0, 0, 0,NULL,NULL}, +! { 2,16, 0, 0,t1HB, t1l}, +! { 3,16, 0, 0,t2HB, t2l}, +! { 3,16, 0, 0,t3HB, t3l}, +! { 0, 0, 0, 0,NULL,NULL},/* Apparently not used */ +! { 4,16, 0, 0,t5HB, t5l}, +! { 4,16, 0, 0,t6HB, t6l}, +! { 6,16, 0, 0,t7HB, t7l}, +! { 6,16, 0, 0,t8HB, t8l}, +! { 6,16, 0, 0,t9HB, t9l}, +! { 8,16, 0, 0,t10HB, t10l}, +! { 8,16, 0, 0,t11HB, t11l}, +! { 8,16, 0, 0,t12HB, t12l}, +! {16,16, 0, 0,t13HB, t13l}, +! { 0, 0, 0, 0,NULL,NULL},/* Apparently not used */ +! {16,16, 0, 0,t15HB, t15l}, +! {16,16, 1, 1,t16HB, t16l}, +! {16,16, 2, 3,t16HB, t16l}, +! {16,16, 3, 7,t16HB, t16l}, +! {16,16, 4, 15,t16HB, t16l}, +! {16,16, 6, 63,t16HB, t16l}, +! {16,16, 8, 255,t16HB, t16l}, +! {16,16,10,1023,t16HB, t16l}, +! {16,16,13,8191,t16HB, t16l}, +! {16,16, 4, 15,t24HB, t24l}, +! {16,16, 5, 31,t24HB, t24l}, +! {16,16, 6, 63,t24HB, t24l}, +! {16,16, 7, 127,t24HB, t24l}, +! {16,16, 8, 255,t24HB, t24l}, +! {16,16, 9, 511,t24HB, t24l}, +! {16,16,11,2047,t24HB, t24l}, +! {16,16,13,8191,t24HB, t24l}, +! { 1,16, 0, 0,t32HB, t32l}, +! { 1,16, 0, 0,t33HB, t33l}, +! }; + + ++ #if 0 + /* do the huffman coding, */ + /* note! for counta,countb - the 4 bit value is passed in y, set x to 0 */ + /* return value: 0-no error, 1 decode error */ +! void huffman_coder(unsigned int x, +! unsigned int y, +! struct huffcodetab *h, +! Bit_stream_struc *bs) + { + HUFFBITS huffbits; /* data left aligned */ + HUFFBITS linbitsX; +*************** +*** 238,244 **** + else if (x >= xl1) { + linbitsX = x-xl1; + if (linbitsX > h->linmax) { +- fprintf(stderr,"warning: Huffman X table overflow\n"); + linbitsX= h->linmax; + }; + if (y >= yl1) { +--- 147,152 ---- +*************** +*** 247,253 **** + putbits(bs,huffbits,len); + linbitsY = y-yl1; + if (linbitsY > h->linmax) { +- fprintf(stderr,"warning: Huffman Y table overflow\n"); + linbitsY = h->linmax; + }; + if (h->linbits) { +--- 155,160 ---- +*************** +*** 270,276 **** + putbits(bs,huffbits,len); + linbitsY = y-yl1; + if (linbitsY > h->linmax) { +- fprintf(stderr,"warning: Huffman Y table overflow\n"); + linbitsY = h->linmax; + }; + if (h->linbits) { +--- 177,182 ---- +*************** +*** 279,380 **** + } + } + +! /* do the huffman-decoding */ +! /* note! for counta,countb -the 4 bit value is returned in y, discard x */ +! int huffman_decoder(h, x, y, v, w) +! struct huffcodetab *h; /* pointer to huffman code record */ +! /* unsigned */ int *x; /* returns decoded x value */ +! /* unsigned */ int *y; /* returns decoded y value */ +! int *v; +! int *w; +! { +! HUFFBITS level; +! int point = 0; +! int error = 1; +! level = dmask; +! if (h->val == NULL) return 2; +! +! /* table 0 needs no bits */ +! if ( h->treelen == 0) +! { *x = *y = 0; +! return 0; +! } +! +! +! /* Lookup in Huffman table. */ +! +! do { +! if (h->val[point][0]==0) { /*end of tree*/ +! *x = h->val[point][1] >> 4; +! *y = h->val[point][1] & 0xf; +! +! error = 0; +! break; +! } +! if (hget1bit()) { +! while (h->val[point][1] >= MXOFF) point += h->val[point][1]; +! point += h->val[point][1]; +! } +! else { +! while (h->val[point][0] >= MXOFF) point += h->val[point][0]; +! point += h->val[point][0]; +! } +! level >>= 1; +! } while (level || (point < ht->treelen) ); +! +! /* Check for error. */ +! +! if (error) { /* set x and y to a medium value as a simple concealment */ +! printf("Illegal Huffman code in data.\n"); +! *x = (h->xlen-1 << 1); +! *y = (h->ylen-1 << 1); +! } +! +! /* Process sign encodings for quadruples tables. */ +! +! if (h->tablename[0] == '3' +! && (h->tablename[1] == '2' || h->tablename[1] == '3')) { +! *v = (*y>>3) & 1; +! *w = (*y>>2) & 1; +! *x = (*y>>1) & 1; +! *y = *y & 1; +! +! /* v, w, x and y are reversed in the bitstream. +! switch them around to make test bistream work. */ +! +! /* {int i=*v; *v=*y; *y=i; i=*w; *w=*x; *x=i;} MI */ +! +! if (*v) +! if (hget1bit() == 1) *v = -*v; +! if (*w) +! if (hget1bit() == 1) *w = -*w; +! if (*x) +! if (hget1bit() == 1) *x = -*x; +! if (*y) +! if (hget1bit() == 1) *y = -*y; +! } +! +! /* Process sign and escape encodings for dual tables. */ +! +! else { +! +! /* x and y are reversed in the test bitstream. +! Reverse x and y here to make test bitstream work. */ +! +! /* removed 11/11/92 -ag +! {int i=*x; *x=*y; *y=i;} +! */ +! if (h->linbits) +! if ((h->xlen-1) == *x) +! *x += hgetbits(h->linbits); +! if (*x) +! if (hget1bit() == 1) *x = -*x; +! if (h->linbits) +! if ((h->ylen-1) == *y) +! *y += hgetbits(h->linbits); +! if (*y) +! if (hget1bit() == 1) *y = -*y; +! } +! +! return error; +! } +--- 185,188 ---- + } + } + +! #endif +diff -r -c -N encoder/huffman.h lame3.13/huffman.h +*** encoder/huffman.h Wed Jan 22 02:43:15 1997 +--- lame3.13/huffman.h Wed May 5 10:15:21 1999 +*************** +*** 1,60 **** +- /********************************************************************** +- * ISO MPEG Audio Subgroup Software Simulation Group (1996) +- * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- * +- * $Id: huffman.h,v 1.1 1996/02/14 04:04:23 rowlands Exp $ +- * +- * $Log: huffman.h,v $ +- * Revision 1.1 1996/02/14 04:04:23 rowlands +- * Initial revision +- * +- * Received from Mike Coleman +- **********************************************************************/ +- /********************************************************************** +- * date programmers comment * +- * 27.2.92 F.O.Witte (ITT Intermetall) * +- * 8/24/93 M. Iwadare Changed for 1 pass decoding. * +- * 7/14/94 J. Koller useless 'typedef' before huffcodetab * +- * removed * +- *********************************************************************/ +- + #define HUFFBITS unsigned long int + #define HTN 34 + #define MXOFF 250 + + struct huffcodetab { +- char tablename[3]; /*string, containing table_description */ + unsigned int xlen; /*max. x-index+ */ + unsigned int ylen; /*max. y-index+ */ + unsigned int linbits; /*number of linbits */ + unsigned int linmax; /*max number to be stored in linbits */ +- int ref; /*a positive value indicates a reference*/ + HUFFBITS *table; /*pointer to array[xlen][ylen] */ + unsigned char *hlen; /*pointer to array[xlen][ylen] */ +- unsigned char(*val)[2];/*decoder tree */ +- unsigned int treelen; /*length of decoder tree */ + }; + + extern struct huffcodetab ht[HTN];/* global memory block */ + /* array of all huffcodtable headers */ + /* 0..31 Huffman code table 0..31 */ + /* 32,33 count1-tables */ +- #ifdef PROTO_ARGS + +- extern int read_huffcodetab(FILE *); +- extern int read_decoder_table(FILE *); +- + extern void huffman_coder(unsigned int, unsigned int, +! struct huffcodetab *, Bit_stream_struc *); +! +! extern int huffman_decoder(struct huffcodetab *, +! /* unsigned */ int *, /* unsigned */ int*, int*, int*); +! +! #else +! +! extern int read_huffcodetab(); +! extern int read_decoder_table(); +! extern void huffman_coder(); +! extern int huffman_decoder(); +! +! #endif +--- 1,20 ---- + #define HUFFBITS unsigned long int + #define HTN 34 + #define MXOFF 250 + + struct huffcodetab { + unsigned int xlen; /*max. x-index+ */ + unsigned int ylen; /*max. y-index+ */ + unsigned int linbits; /*number of linbits */ + unsigned int linmax; /*max number to be stored in linbits */ + HUFFBITS *table; /*pointer to array[xlen][ylen] */ + unsigned char *hlen; /*pointer to array[xlen][ylen] */ + }; + + extern struct huffcodetab ht[HTN];/* global memory block */ + /* array of all huffcodtable headers */ + /* 0..31 Huffman code table 0..31 */ + /* 32,33 count1-tables */ + + extern void huffman_coder(unsigned int, unsigned int, +! struct huffcodetab*, Bit_stream_struc*); +diff -r -c -N encoder/ieeefloat.c lame3.13/ieeefloat.c +*** encoder/ieeefloat.c Wed Jan 22 02:43:17 1997 +--- lame3.13/ieeefloat.c Sat Jun 5 13:45:49 1999 +*************** +*** 1,3 **** +--- 1,4 ---- ++ #ifndef LIBSNDFILE + /* Copyright (C) 1988-1991 Apple Computer, Inc. + * All Rights Reserved. + * +*************** +*** 969,971 **** +--- 970,973 ---- + */ + + #endif /* TEST_FP */ ++ #endif +diff -r -c -N encoder/l3bitstream.c lame3.13/l3bitstream.c +*** encoder/l3bitstream.c Wed Jan 22 02:43:15 1997 +--- lame3.13/l3bitstream.c Sat Jun 19 17:00:40 1999 +*************** +*** 75,85 **** + char *ancillary, + int ancillary_bits ) + { +! int gr, ch, i, mode_gr; + fr_ps = in_fr_ps; + bs = in_bs; + stereo = fr_ps->stereo; + mode_gr = (fr_ps->header->version == 1) ? 2 : 1; + + if ( frameData == NULL ) + { +--- 75,88 ---- + char *ancillary, + int ancillary_bits ) + { +! int gr, ch, mode_gr; + fr_ps = in_fr_ps; + bs = in_bs; + stereo = fr_ps->stereo; + mode_gr = (fr_ps->header->version == 1) ? 2 : 1; ++ ++ ++ + + if ( frameData == NULL ) + { +*************** +*** 111,130 **** + PartHoldersInitialized = 1; + } + +! #if 1 + for ( gr = 0; gr < mode_gr; gr++ ) + for ( ch = 0; ch < stereo; ch++ ) + { + int *pi = &l3_enc[gr][ch][0]; +! double *pr = &xr[gr][ch][0]; +! for ( i = 0; i < 576; i++, pr++, pi++ ) +! { +! if ( (*pr < 0) && (*pi > 0) ) +! *pi *= -1; + } + } + #endif + + encodeSideInfo( l3_side ); + encodeMainData( l3_enc, l3_side, scalefac ); + write_ancillary_data( ancillary, ancillary_bits ); +--- 114,142 ---- + PartHoldersInitialized = 1; + } + +! #if 0 +! /* this makes more sense in loop.c */ + for ( gr = 0; gr < mode_gr; gr++ ) + for ( ch = 0; ch < stereo; ch++ ) + { + int *pi = &l3_enc[gr][ch][0]; +! int i; +! for ( i = 0; i < 576; i++) { +! double pr = xr[gr][ch][i]; +! /* ms stereo fix. if ms_stereo, l3_enc will be M/S values */ +! if (in_fr_ps->header->mode_ext == 2) { +! pr = ch ? (xr[gr][0][i]-xr[gr][1][i]) : (xr[gr][0][i]+xr[gr][1][i]); +! /* pr /= sqrt(2.0); */ +! } +! +! if ( (pr < 0) && (pi[i] > 0) ) pi[i] *= -1; + } + } + #endif + ++ ++ ++ + encodeSideInfo( l3_side ); + encodeMainData( l3_enc, l3_side, scalefac ); + write_ancillary_data( ancillary, ancillary_bits ); +*************** +*** 159,164 **** +--- 171,177 ---- + + /* we set this here -- it will be tested in the next loops iteration */ + l3_side->main_data_begin = frameResults->nextBackPtr; ++ + } + + void +*************** +*** 179,184 **** +--- 192,200 ---- + int i, gr, ch, sfb, window, mode_gr; + layer *info = fr_ps->header; + ++ ++ ++ + if ( info->version == 1 ) + mode_gr = 2; + else +*************** +*** 206,211 **** +--- 222,228 ---- + + if ( (gi->window_switching_flag == 1) && (gi->block_type == 2) ) + { ++ #ifdef ALLOW_MIXED + if ( gi->mixed_block_flag ) + { + for ( sfb = 0; sfb < 8; sfb++ ) +*************** +*** 221,226 **** +--- 238,244 ---- + + } + else ++ #endif + { + for ( sfb = 0; sfb < 6; sfb++ ) + for ( window = 0; window < 3; window++ ) +*************** +*** 262,272 **** + gr_info *gi = &(si->gr[gr].ch[ch].tt); + int *ix = &l3_enc[gr][ch][0]; + int sfb_partition; +- + assert( gi->sfb_partition_table ); + + if ( (gi->window_switching_flag == 1) && (gi->block_type == 2) ) + { + if ( gi->mixed_block_flag ) + { + sfb_partition = 0; +--- 280,290 ---- + gr_info *gi = &(si->gr[gr].ch[ch].tt); + int *ix = &l3_enc[gr][ch][0]; + int sfb_partition; + assert( gi->sfb_partition_table ); + + if ( (gi->window_switching_flag == 1) && (gi->block_type == 2) ) + { ++ #ifdef ALLOW_MIXED + if ( gi->mixed_block_flag ) + { + sfb_partition = 0; +*************** +*** 283,288 **** +--- 301,307 ---- + } + } + else ++ #endif + { + for ( sfb = 0, sfb_partition = 0; sfb_partition < 4; sfb_partition++ ) + { +*************** +*** 304,315 **** + *pph = BF_addEntry( *pph, scalefac->l[gr][ch][sfb], slen ); + } + } + Huffmancodebits( &codedDataPH[gr][ch], ix, gi ); + } /* for ch */ + } + } /* main_data */ + +! static unsigned int crc = 0; + + static int encodeSideInfo( III_side_info_t *si ) + { +--- 323,351 ---- + *pph = BF_addEntry( *pph, scalefac->l[gr][ch][sfb], slen ); + } + } ++ ++ ++ + Huffmancodebits( &codedDataPH[gr][ch], ix, gi ); + } /* for ch */ + } + } /* main_data */ + +! static unsigned int crc = 0; /* (jo) current crc */ +! +! /* (jo) this wrapper function for BF_addEntry() updates also the crc */ +! static BF_PartHolder *CRC_BF_addEntry( BF_PartHolder *thePH, uint32 value, uint16 length ) +! { +! unsigned int bit = 1 << length; +! +! while((bit >>= 1)){ +! crc <<= 1; +! if (!(crc & 0x10000) ^ !(value & bit)) +! crc ^= CRC16_POLYNOMIAL; +! } +! crc &= 0xffff; +! return BF_addEntry(thePH, value, length); +! } + + static int encodeSideInfo( III_side_info_t *si ) + { +*************** +*** 317,345 **** + layer *info = fr_ps->header; + + mode_gr = (info->version == 1) ? 2 : 1; + + headerPH->part->nrEntries = 0; + headerPH = BF_addEntry( headerPH, 0xfff, 12 ); + headerPH = BF_addEntry( headerPH, info->version, 1 ); + headerPH = BF_addEntry( headerPH, 4 - info->lay, 2 ); + headerPH = BF_addEntry( headerPH, !info->error_protection, 1 ); +! headerPH = BF_addEntry( headerPH, info->bitrate_index, 4 ); +! headerPH = BF_addEntry( headerPH, info->sampling_frequency, 2 ); +! headerPH = BF_addEntry( headerPH, info->padding, 1 ); +! headerPH = BF_addEntry( headerPH, info->extension, 1 ); +! headerPH = BF_addEntry( headerPH, info->mode, 2 ); +! headerPH = BF_addEntry( headerPH, info->mode_ext, 2 ); +! headerPH = BF_addEntry( headerPH, info->copyright, 1 ); +! headerPH = BF_addEntry( headerPH, info->original, 1 ); +! headerPH = BF_addEntry( headerPH, info->emphasis, 2 ); + + bits_sent = 32; +! +! if ( fr_ps->header->error_protection ) +! { +! headerPH = BF_addEntry( headerPH, crc, 16 ); +! bits_sent += 16; +! } + + frameSIPH->part->nrEntries = 0; + +--- 353,380 ---- + layer *info = fr_ps->header; + + mode_gr = (info->version == 1) ? 2 : 1; ++ ++ crc = 0xffff; /* (jo) init crc16 for error_protection */ + + headerPH->part->nrEntries = 0; + headerPH = BF_addEntry( headerPH, 0xfff, 12 ); + headerPH = BF_addEntry( headerPH, info->version, 1 ); + headerPH = BF_addEntry( headerPH, 4 - info->lay, 2 ); + headerPH = BF_addEntry( headerPH, !info->error_protection, 1 ); +! /* (jo) from now on call the CRC_BF_addEntry() wrapper to update crc */ +! headerPH = CRC_BF_addEntry( headerPH, info->bitrate_index, 4 ); +! headerPH = CRC_BF_addEntry( headerPH, info->sampling_frequency, 2 ); +! headerPH = CRC_BF_addEntry( headerPH, info->padding, 1 ); +! headerPH = CRC_BF_addEntry( headerPH, info->extension, 1 ); +! headerPH = CRC_BF_addEntry( headerPH, info->mode, 2 ); +! headerPH = CRC_BF_addEntry( headerPH, info->mode_ext, 2 ); +! headerPH = CRC_BF_addEntry( headerPH, info->copyright, 1 ); +! headerPH = CRC_BF_addEntry( headerPH, info->original, 1 ); +! headerPH = CRC_BF_addEntry( headerPH, info->emphasis, 2 ); + + bits_sent = 32; +! +! /* (jo) see below for BF_addEntry( headerPH, crc, 16 ); */ + + frameSIPH->part->nrEntries = 0; + +*************** +*** 352,369 **** + + if ( info->version == 1 ) + { /* MPEG1 */ +! frameSIPH = BF_addEntry( frameSIPH, si->main_data_begin, 9 ); + + if ( stereo == 2 ) +! frameSIPH = BF_addEntry( frameSIPH, si->private_bits, 3 ); + else +! frameSIPH = BF_addEntry( frameSIPH, si->private_bits, 5 ); + + for ( ch = 0; ch < stereo; ch++ ) + for ( scfsi_band = 0; scfsi_band < 4; scfsi_band++ ) + { + BF_PartHolder **pph = &channelSIPH[ch]; +! *pph = BF_addEntry( *pph, si->scfsi[ch][scfsi_band], 1 ); + } + + for ( gr = 0; gr < 2; gr++ ) +--- 387,404 ---- + + if ( info->version == 1 ) + { /* MPEG1 */ +! frameSIPH = CRC_BF_addEntry( frameSIPH, si->main_data_begin, 9 ); + + if ( stereo == 2 ) +! frameSIPH = CRC_BF_addEntry( frameSIPH, si->private_bits, 3 ); + else +! frameSIPH = CRC_BF_addEntry( frameSIPH, si->private_bits, 5 ); + + for ( ch = 0; ch < stereo; ch++ ) + for ( scfsi_band = 0; scfsi_band < 4; scfsi_band++ ) + { + BF_PartHolder **pph = &channelSIPH[ch]; +! *pph = CRC_BF_addEntry( *pph, si->scfsi[ch][scfsi_band], 1 ); + } + + for ( gr = 0; gr < 2; gr++ ) +*************** +*** 371,405 **** + { + BF_PartHolder **pph = &spectrumSIPH[gr][ch]; + gr_info *gi = &(si->gr[gr].ch[ch].tt); +! *pph = BF_addEntry( *pph, gi->part2_3_length, 12 ); +! *pph = BF_addEntry( *pph, gi->big_values, 9 ); +! *pph = BF_addEntry( *pph, gi->global_gain, 8 ); +! *pph = BF_addEntry( *pph, gi->scalefac_compress, 4 ); +! *pph = BF_addEntry( *pph, gi->window_switching_flag, 1 ); + + if ( gi->window_switching_flag ) + { +! *pph = BF_addEntry( *pph, gi->block_type, 2 ); +! *pph = BF_addEntry( *pph, gi->mixed_block_flag, 1 ); + + for ( region = 0; region < 2; region++ ) +! *pph = BF_addEntry( *pph, gi->table_select[region], 5 ); + for ( window = 0; window < 3; window++ ) +! *pph = BF_addEntry( *pph, gi->subblock_gain[window], 3 ); + } + else + { + assert( gi->block_type == 0 ); + for ( region = 0; region < 3; region++ ) +! *pph = BF_addEntry( *pph, gi->table_select[region], 5 ); + +! *pph = BF_addEntry( *pph, gi->region0_count, 4 ); +! *pph = BF_addEntry( *pph, gi->region1_count, 3 ); + } + +! *pph = BF_addEntry( *pph, gi->preflag, 1 ); +! *pph = BF_addEntry( *pph, gi->scalefac_scale, 1 ); +! *pph = BF_addEntry( *pph, gi->count1table_select, 1 ); + } + + if ( stereo == 2 ) +--- 406,440 ---- + { + BF_PartHolder **pph = &spectrumSIPH[gr][ch]; + gr_info *gi = &(si->gr[gr].ch[ch].tt); +! *pph = CRC_BF_addEntry( *pph, gi->part2_3_length, 12 ); +! *pph = CRC_BF_addEntry( *pph, gi->big_values, 9 ); +! *pph = CRC_BF_addEntry( *pph, gi->global_gain, 8 ); +! *pph = CRC_BF_addEntry( *pph, gi->scalefac_compress, 4 ); +! *pph = CRC_BF_addEntry( *pph, gi->window_switching_flag, 1 ); + + if ( gi->window_switching_flag ) + { +! *pph = CRC_BF_addEntry( *pph, gi->block_type, 2 ); +! *pph = CRC_BF_addEntry( *pph, gi->mixed_block_flag, 1 ); + + for ( region = 0; region < 2; region++ ) +! *pph = CRC_BF_addEntry( *pph, gi->table_select[region], 5 ); + for ( window = 0; window < 3; window++ ) +! *pph = CRC_BF_addEntry( *pph, gi->subblock_gain[window], 3 ); + } + else + { + assert( gi->block_type == 0 ); + for ( region = 0; region < 3; region++ ) +! *pph = CRC_BF_addEntry( *pph, gi->table_select[region], 5 ); + +! *pph = CRC_BF_addEntry( *pph, gi->region0_count, 4 ); +! *pph = CRC_BF_addEntry( *pph, gi->region1_count, 3 ); + } + +! *pph = CRC_BF_addEntry( *pph, gi->preflag, 1 ); +! *pph = CRC_BF_addEntry( *pph, gi->scalefac_scale, 1 ); +! *pph = CRC_BF_addEntry( *pph, gi->count1table_select, 1 ); + } + + if ( stereo == 2 ) +*************** +*** 409,459 **** + } + else + { /* MPEG2 */ +! frameSIPH = BF_addEntry( frameSIPH, si->main_data_begin, 8 ); + + if ( stereo == 2 ) +! frameSIPH = BF_addEntry( frameSIPH, si->private_bits, 2 ); + else +! frameSIPH = BF_addEntry( frameSIPH, si->private_bits, 1 ); + + gr = 0; + for ( ch = 0; ch < stereo; ch++ ) + { + BF_PartHolder **pph = &spectrumSIPH[gr][ch]; + gr_info *gi = &(si->gr[gr].ch[ch].tt); +! *pph = BF_addEntry( *pph, gi->part2_3_length, 12 ); +! *pph = BF_addEntry( *pph, gi->big_values, 9 ); +! *pph = BF_addEntry( *pph, gi->global_gain, 8 ); +! *pph = BF_addEntry( *pph, gi->scalefac_compress, 9 ); +! *pph = BF_addEntry( *pph, gi->window_switching_flag, 1 ); + + if ( gi->window_switching_flag ) + { +! *pph = BF_addEntry( *pph, gi->block_type, 2 ); +! *pph = BF_addEntry( *pph, gi->mixed_block_flag, 1 ); + + for ( region = 0; region < 2; region++ ) +! *pph = BF_addEntry( *pph, gi->table_select[region], 5 ); + for ( window = 0; window < 3; window++ ) +! *pph = BF_addEntry( *pph, gi->subblock_gain[window], 3 ); + } + else + { + for ( region = 0; region < 3; region++ ) +! *pph = BF_addEntry( *pph, gi->table_select[region], 5 ); + +! *pph = BF_addEntry( *pph, gi->region0_count, 4 ); +! *pph = BF_addEntry( *pph, gi->region1_count, 3 ); + } + +! *pph = BF_addEntry( *pph, gi->scalefac_scale, 1 ); +! *pph = BF_addEntry( *pph, gi->count1table_select, 1 ); + } + if ( stereo == 2 ) + bits_sent += 136; + else + bits_sent += 72; + } + return bits_sent; + } + +--- 444,501 ---- + } + else + { /* MPEG2 */ +! frameSIPH = CRC_BF_addEntry( frameSIPH, si->main_data_begin, 8 ); + + if ( stereo == 2 ) +! frameSIPH = CRC_BF_addEntry( frameSIPH, si->private_bits, 2 ); + else +! frameSIPH = CRC_BF_addEntry( frameSIPH, si->private_bits, 1 ); + + gr = 0; + for ( ch = 0; ch < stereo; ch++ ) + { + BF_PartHolder **pph = &spectrumSIPH[gr][ch]; + gr_info *gi = &(si->gr[gr].ch[ch].tt); +! *pph = CRC_BF_addEntry( *pph, gi->part2_3_length, 12 ); +! *pph = CRC_BF_addEntry( *pph, gi->big_values, 9 ); +! *pph = CRC_BF_addEntry( *pph, gi->global_gain, 8 ); +! *pph = CRC_BF_addEntry( *pph, gi->scalefac_compress, 9 ); +! *pph = CRC_BF_addEntry( *pph, gi->window_switching_flag, 1 ); + + if ( gi->window_switching_flag ) + { +! *pph = CRC_BF_addEntry( *pph, gi->block_type, 2 ); +! *pph = CRC_BF_addEntry( *pph, gi->mixed_block_flag, 1 ); + + for ( region = 0; region < 2; region++ ) +! *pph = CRC_BF_addEntry( *pph, gi->table_select[region], 5 ); + for ( window = 0; window < 3; window++ ) +! *pph = CRC_BF_addEntry( *pph, gi->subblock_gain[window], 3 ); + } + else + { + for ( region = 0; region < 3; region++ ) +! *pph = CRC_BF_addEntry( *pph, gi->table_select[region], 5 ); + +! *pph = CRC_BF_addEntry( *pph, gi->region0_count, 4 ); +! *pph = CRC_BF_addEntry( *pph, gi->region1_count, 3 ); + } + +! *pph = CRC_BF_addEntry( *pph, gi->scalefac_scale, 1 ); +! *pph = CRC_BF_addEntry( *pph, gi->count1table_select, 1 ); + } + if ( stereo == 2 ) + bits_sent += 136; + else + bits_sent += 72; + } ++ ++ if ( fr_ps->header->error_protection ) ++ { /* (jo) error_protection: add crc16 information to header */ ++ headerPH = BF_addEntry( headerPH, crc, 16 ); ++ bits_sent += 16; ++ } ++ + return bits_sent; + } + +*************** +*** 527,535 **** + struct huffcodetab *h; + int bvbits, c1bits, tablezeros, r0, r1, r2, rt, *pr; + int bitsWritten = 0; +- int idx = 0; + tablezeros = 0; + r0 = r1 = r2 = 0; + + /* 1: Write the bigvalues */ + bigvalues = gi->big_values * 2; +--- 569,577 ---- + struct huffcodetab *h; + int bvbits, c1bits, tablezeros, r0, r1, r2, rt, *pr; + int bitsWritten = 0; + tablezeros = 0; + r0 = r1 = r2 = 0; ++ + + /* 1: Write the bigvalues */ + bigvalues = gi->big_values * 2; +*************** +*** 569,576 **** + { + x = (*ix_s)[line][window]; + y = (*ix_s)[line + 1][window]; +- assert( idx < 576 ); +- assert( idx >= 0 ); + bits = HuffmanCode( tableindex, x, y, &code, &ext, &cbits, &xbits ); + *pph = BF_addEntry( *pph, code, cbits ); + *pph = BF_addEntry( *pph, ext, xbits ); +--- 611,616 ---- +*************** +*** 580,585 **** +--- 620,626 ---- + } + } + else ++ #ifdef ALLOW_MIXED + if ( gi->mixed_block_flag && gi->block_type == 2 ) + { /* Mixed blocks long, short */ + int sfb, window, line, start, end; +*************** +*** 625,630 **** +--- 666,672 ---- + + } + else ++ #endif + { /* Long blocks */ + int *scalefac = &sfBandIndex[fr_ps->header->sampling_frequency + (fr_ps->header->version * 3)].l[0]; + unsigned scalefac_index = 100; +*************** +*** 644,649 **** +--- 686,692 ---- + region2Start = scalefac[ scalefac_index ]; + assert( region1Start == gi->address1 ); + } ++ + for ( i = 0; i < bigvalues; i += 2 ) + { + unsigned tableindex = 100; +*************** +*** 671,676 **** +--- 714,720 ---- + y = ix[i + 1]; + if ( tableindex ) + { ++ + bits = HuffmanCode( tableindex, x, y, &code, &ext, &cbits, &xbits ); + *pph = BF_addEntry( *pph, code, cbits ); + *pph = BF_addEntry( *pph, ext, xbits ); +*************** +*** 701,706 **** +--- 745,751 ---- + bitsWritten += L3_huffman_coder_count1( pph, h, v, w, x, y ); + } + c1bits = bitsWritten - bvbits; ++ + if ( (stuffingBits = gi->part2_3_length - gi->part2_length - bitsWritten) ) + { + int stuffingWords = stuffingBits / 32; +*************** +*** 719,725 **** + } + assert( bitsWritten == gi->part2_3_length - gi->part2_length ); + #ifdef DEBUG +! printf( "#### %d Huffman bits written (%02d + %02d), part2_length = %d, part2_3_length = %d, %d stuffing ####\n", + bitsWritten, bvbits, c1bits, gi->part2_length, gi->part2_3_length, stuffingBits ); + #endif + } +--- 764,770 ---- + } + assert( bitsWritten == gi->part2_3_length - gi->part2_length ); + #ifdef DEBUG +! printf( "## %d Huffman bits written (%02d + %02d), part2_length = %d, part2_3_length = %d, %d stuffing ##\n", + bitsWritten, bvbits, c1bits, gi->part2_length, gi->part2_3_length, stuffingBits ); + #endif + } +diff -r -c -N encoder/l3psy.c lame3.13/l3psy.c +*** encoder/l3psy.c Wed Jan 22 02:43:15 1997 +--- lame3.13/l3psy.c Thu Jun 24 18:16:37 1999 +*************** +*** 2,10 **** + * ISO MPEG Audio Subgroup Software Simulation Group (1996) + * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension + * +! * $Id: l3psy.c,v 1.2 1997/01/19 22:28:29 rowlands Exp $ + * + * $Log: l3psy.c,v $ + * Revision 1.2 1997/01/19 22:28:29 rowlands + * Layer 3 bug fixes from Seymour Shlien + * +--- 2,15 ---- + * ISO MPEG Audio Subgroup Software Simulation Group (1996) + * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension + * +! * $Id: l3psy.c,v 1.2 1998/10/05 17:06:48 larsi Exp $ + * + * $Log: l3psy.c,v $ ++ * Revision 1.2 1998/10/05 17:06:48 larsi ++ * *** empty log message *** ++ * ++ * Revision 1.1.1.1 1998/10/05 14:47:18 larsi ++ * + * Revision 1.2 1997/01/19 22:28:29 rowlands + * Layer 3 bug fixes from Seymour Shlien + * +*************** +*** 25,38 **** +--- 30,62 ---- + **********************************************************************/ + + #include "common.h" ++ #include "globalflags.h" + #include "encoder.h" + #include "l3psy.h" + #include "l3side.h" + #include <assert.h> ++ #include "gtkanal.h" ++ #define MSFREQ 20 + + #define maximum(x,y) ( (x>y) ? x : y ) + #define minimum(x,y) ( (x<y) ? x : y ) + ++ /* some different types of adaptive window switching */ ++ ++ ++ /* This mode will turn on short_blocks if there is a localized surge in energy */ ++ #define ENER_AWS ++ ++ /* from Gabriel Bouvigne <bouvigne@multimania.com> */ ++ #define AWS ++ ++ static int switch_pe=1800; ++ ++ ++ void sprdngf1(FLOAT *, double *); ++ void sprdngf2(double *, FLOAT *); ++ static double s3_l[CBANDS][CBANDS]; /* needed global static by sprdngfs */ ++ + void L3para_read( double sfreq, int numlines[CBANDS], int partition_l[HBLKSIZE], + double minval[CBANDS], double qthr_l[CBANDS], double norm_l[CBANDS], + double s3_l[CBANDS][CBANDS], int partition_s[HBLKSIZE_s], double qthr_s[CBANDS_s], +*************** +*** 42,71 **** + int cbw_s[SBMAX_s], int bu_s[SBMAX_s], int bo_s[SBMAX_s], + double w1_s[SBMAX_s], double w2_s[SBMAX_s] ); + +! void L3psycho_anal( short int *buffer, short int savebuf[1344], int chn, int lay, FLOAT snr32[32], +! double sfreq, double ratio_d[21], double ratio_ds[12][3], +! double *pe, gr_info *cod_info ) + { +! static double ratio[2][21]; +! static double ratio_s[2][12][3]; +! int blocktype; +! unsigned int b, i, j, k; +! double r_prime, phi_prime; /* not FLOAT */ +! FLOAT freq_mult, bval_lo, min_thres, sum_energy; +! double tb, temp1,temp2,temp3; + +! /* nint(); Layer III */ +! double thr[CBANDS]; + + /* The static variables "r", "phi_sav", "new", "old" and "oldest" have */ + /* to be remembered for the unpredictability measure. For "r" and */ + /* "phi_sav", the first index from the left is the channel select and */ + /* the second index is the "age" of the data. */ +- +- +- static FLOAT window_s[BLKSIZE_s] ; + static int new = 0, old = 1, oldest = 0; +! static int init = 0, flush, sync_flush, syncsize, sfreq_idx; + static double cw[HBLKSIZE], eb[CBANDS]; + static double ctb[CBANDS]; + static double SNR_l[CBANDS], SNR_s[CBANDS_s]; +--- 66,259 ---- + int cbw_s[SBMAX_s], int bu_s[SBMAX_s], int bo_s[SBMAX_s], + double w1_s[SBMAX_s], double w2_s[SBMAX_s] ); + +! +! +! +! +! +! +! +! +! +! +! +! +! void L3psycho_energy( short int *buffer, +! FLOAT energy[HBLKSIZE], +! FLOAT ax[HBLKSIZE], FLOAT bx[HBLKSIZE], +! FLOAT energy_s[3][HBLKSIZE_s], +! FLOAT ax_s[3][HBLKSIZE_s], FLOAT bx_s[3][HBLKSIZE_s], +! int chn,int gr_out , layer * info, +! int check_ms_stereo, double *ms_ener_ratio) +! { +! static short int savebuffer[2][1344]; +! static double ms_ener[2]; +! static int sync_flush,flush,syncsize; +! static int init=0; +! static FLOAT scalefac; +! static FLOAT window_s[BLKSIZE_s]; +! static FLOAT window[BLKSIZE]; +! +! #ifdef HAVEGTK +! static FLOAT energy_save[2][HBLKSIZE]; +! #endif +! +! short int *savebuf; +! int i,j,k,sblock; +! +! static FLOAT wsamp_r_int[2][BLKSIZE]; +! FLOAT wsamp_rs[256],*wsamp_r; +! +! +! wsamp_r=wsamp_r_int[chn]; +! savebuf = &savebuffer[chn][0]; +! +! if(init==0){ +! init=1; +! memset((char *) savebuffer, 0, sizeof(savebuffer)); +! if (gpsycho) {sync_flush=WINDELAY; flush=576; syncsize=WINDELAY+576;} +! else {sync_flush=768; flush=576; syncsize=1344;} +! +! scalefac=1.0; +! if (force_ms) scalefac=SQRT2; +! +! /* calculate HANN window coefficients */ +! /* for(i=0;i<BLKSIZE;i++) window[i] =0.5*(1-cos(2.0*PI*i/(BLKSIZE-1.0)));*/ +! for(i=0;i<BLKSIZE;i++) window[i] =0.5*(1-cos(2.0*PI*(i-0.5)/BLKSIZE)); +! for(i=0;i<BLKSIZE_s;i++)window_s[i]=0.5*(1-cos(2.0*PI*(i-0.5)/BLKSIZE_s)); +! } +! +! /********************************************************************** +! * Delay signal by sync_flush samples * +! **********************************************************************/ +! for ( j = 0; j < sync_flush; j++ ) /* for long window samples */ +! savebuf[j] = savebuf[j+flush]; +! +! for ( j = sync_flush; j < syncsize; j++ ) +! savebuf[j] = buffer[j-sync_flush]; +! +! +! +! /********************************************************************** +! * compute FFTs +! **********************************************************************/ +! for ( j = 0; j < BLKSIZE; j++ ) +! wsamp_r[j] = window[j] * savebuf[j] * scalefac; +! fft( wsamp_r, energy, ax, bx, 1024 ); /**long FFT**/ +! +! #ifdef HAVEGTK +! if(gtkflag) { +! for (j=0; j<HBLKSIZE ; j++) { +! pinfo->energy[gr_out][chn][j]=energy_save[chn][j]; +! energy_save[chn][j]=energy[j]; +! } +! } +! #endif +! for ( sblock = 0; sblock < 3; sblock++ ) { +! int shlen = 128; +! int shoff = 2; +! if (gpsycho) { +! shlen = 192; +! shoff = 1; +! } +! for ( j = 0, k = shlen * (shoff + sblock); j < 256; j++, k++ ) +! wsamp_rs[j] = window_s[j]* savebuf[k]*scalefac; +! fft( wsamp_rs, &energy_s[sblock][0], &ax_s[sblock][0], &bx_s[sblock][0], 256 ); +! } +! +! /**********************************************************************/ +! /* compute side_energy / total_energy ratio */ +! /**********************************************************************/ +! *ms_ener_ratio=0; +! if (check_ms_stereo) { +! double ms_ener_side; +! +! /* compute energy in frequencies above MSFREQ */ +! for (ms_ener[chn]=0, j=MSFREQ; j<HBLKSIZE ; j++) { +! ms_ener[chn] += energy[j]; +! } +! if (chn == 1) { +! /* compute energy in side channel */ +! ms_ener_side = fft_side( wsamp_r_int, MSFREQ); +! +! /* ener_side / total_energy */ +! if (ms_ener_side <= .01*(ms_ener[0]+ms_ener[1])) +! *ms_ener_ratio = .01; +! else +! *ms_ener_ratio = ms_ener_side/(ms_ener[0]+ms_ener[1]); +! } +! } +! +! if (force_ms) { +! for (ms_ener[chn]=0, j=MSFREQ; j<HBLKSIZE ; j++) +! ms_ener[chn] += energy[j]; +! if (chn == 1) { +! if (ms_ener[1] <= .01*(ms_ener[0]+ms_ener[1])) +! *ms_ener_ratio = .01; +! else +! *ms_ener_ratio = ms_ener[1]/(ms_ener[0]+ms_ener[1]); +! } +! } +! +! +! } +! +! +! +! +! +! +! +! +! +! void L3psycho_anal( short int *buffer[2], int stereo, +! int gr_out , layer * info, +! double sfreq, +! int check_ms_stereo, double *ms_ener_ratio, +! double ratio_d[2][21], double ratio_ds[2][12][3], +! double percep_energy[2], int blocktype_d[2]) + { +! static double pe[2]={0,0}; +! static double ms_ratio_s_old=0,ms_ratio_l_old=0; +! +! static double ratio[2][SBMAX_l]; +! static double ratio_s[2][SBMAX_s][3]; +! #ifdef HAVEGTK +! static double pe_save[2]; +! static double ers_save[2]; +! #endif +! static double thm_save[2][SBMAX_l]; +! static double en_save[2][SBMAX_l]; +! static double thm_s_save[2][SBMAX_s][3]; +! static double en_s_save[2][SBMAX_s][3]; +! +! +! int blocktype[2],uselongblock[2],chn; +! unsigned int b, i, j, k; +! FLOAT freq_mult, bval_lo; +! double temp1,temp2,temp3; +! double ms_ratio_l=0,ms_ratio_s=0; +! double estot[2][3]; +! +! +! /* nint(); Layer III */ +! double thr[CBANDS]; +! FLOAT ax[HBLKSIZE], bx[HBLKSIZE]; +! FLOAT energy[HBLKSIZE]; +! FLOAT energy_s[3][HBLKSIZE_s]; +! FLOAT ax_s[3][HBLKSIZE_s], bx_s[3][HBLKSIZE_s]; /* 256 samples not 129. */ + +! static float mld_l[SBMAX_l],mld_s[SBMAX_s]; +! +! +! + + /* The static variables "r", "phi_sav", "new", "old" and "oldest" have */ + /* to be remembered for the unpredictability measure. For "r" and */ + /* "phi_sav", the first index from the left is the channel select and */ + /* the second index is the "age" of the data. */ + static int new = 0, old = 1, oldest = 0; +! static int init = 0, sfreq_idx; + static double cw[HBLKSIZE], eb[CBANDS]; + static double ctb[CBANDS]; + static double SNR_l[CBANDS], SNR_s[CBANDS_s]; +*************** +*** 73,79 **** + static double minval[CBANDS],qthr_l[CBANDS],norm_l[CBANDS]; + static double qthr_s[CBANDS_s],norm_s[CBANDS_s]; + static double nb_1[2][CBANDS], nb_2[2][CBANDS]; +- static double s3_l[CBANDS][CBANDS]; /* s3_s[CBANDS_s][CBANDS_s]; */ + + /* Scale Factor Bands */ + static int cbw_l[SBMAX_l],bu_l[SBMAX_l],bo_l[SBMAX_l] ; +--- 261,266 ---- +*************** +*** 88,192 **** + + /* The following static variables are constants. */ + +- static double nmt = 5.5; +- + static FLOAT crit_band[27] = {0, 100, 200, 300, 400, 510, 630, 770, + 920, 1080, 1270,1480,1720,2000,2320, 2700, + 3150, 3700, 4400,5300,6400,7700,9500,12000, + 15500,25000,30000}; + +! static FLOAT bmax[27] = {20.0, 20.0, 20.0, 20.0, 20.0, 17.0, 15.0, +! 10.0, 7.0, 4.4, 4.5, 4.5, 4.5, 4.5, +! 4.5, 4.5, 4.5, 4.5, 4.5, 4.5, 4.5, +! 4.5, 4.5, 4.5, 3.5, 3.5, 3.5}; +! +! /* The following pointer variables point to large areas of memory */ +! /* dynamically allocated by the mem_alloc() function. Dynamic memory */ +! /* allocation is used in order to avoid stack frame or data area */ +! /* overflow errors that otherwise would have occurred at compile time */ +! /* on the Macintosh computer. */ +! +! FLOAT *grouped_c, *grouped_e, *nb, *cb, *ecb, *bc; +! FLOAT *wsamp_r, *wsamp_i, *phi, *energy; +! static FLOAT energy_s[3][256]; +! static FLOAT phi_s[3][256] ; /* 256 samples not 129 */ +! FLOAT *c, *fthr; +! F32 *snrtmp; + + static int *numlines ; + static int *partition; + static FLOAT *cbval, *rnorm; +- static FLOAT *window; +- static FLOAT *absthr; + static double *tmn; + static FCB *s; + static FHBLK *lthr; +! static F2HBLK *r, *phi_sav; + +- /* These dynamic memory allocations simulate "automatic" variables */ +- /* placed on the stack. For each mem_alloc() call here, there must be */ +- /* a corresponding mem_free() call at the end of this function. */ + +! grouped_c = (FLOAT *) mem_alloc(sizeof(FCB), "grouped_c"); +! grouped_e = (FLOAT *) mem_alloc(sizeof(FCB), "grouped_e"); + nb = (FLOAT *) mem_alloc(sizeof(FCB), "nb"); + cb = (FLOAT *) mem_alloc(sizeof(FCB), "cb"); + ecb = (FLOAT *) mem_alloc(sizeof(FCB), "ecb"); + bc = (FLOAT *) mem_alloc(sizeof(FCB), "bc"); +! wsamp_r = (FLOAT *) mem_alloc(sizeof(FBLK), "wsamp_r"); +! wsamp_i = (FLOAT *) mem_alloc(sizeof(FBLK), "wsamp_i"); +! phi = (FLOAT *) mem_alloc(sizeof(FBLK), "phi"); +! energy = (FLOAT *) mem_alloc(sizeof(FBLK), "energy"); + c = (FLOAT *) mem_alloc(sizeof(FHBLK), "c"); + fthr = (FLOAT *) mem_alloc(sizeof(FHBLK), "fthr"); + snrtmp = (F32 *) mem_alloc(sizeof(F2_32), "snrtmp"); + +- assert( lay == 3 ); +- if(init==0){ +- +- /* These dynamic memory allocations simulate "static" variables placed */ +- /* in the data space. Each mem_alloc() call here occurs only once at */ +- /* initialization time. The mem_free() function must not be called. */ + numlines = (int *) mem_alloc(sizeof(ICB), "numlines"); + partition = (int *) mem_alloc(sizeof(IHBLK), "partition"); + cbval = (FLOAT *) mem_alloc(sizeof(FCB), "cbval"); + rnorm = (FLOAT *) mem_alloc(sizeof(FCB), "rnorm"); +! window = (FLOAT *) mem_alloc(sizeof(FBLK), "window"); +! absthr = (FLOAT *) mem_alloc(sizeof(FHBLK), "absthr"); +! tmn = (double *) mem_alloc(sizeof(DCB), "tmn"); + s = (FCB *) mem_alloc(sizeof(FCBCB), "s"); + lthr = (FHBLK *) mem_alloc(sizeof(F2HBLK), "lthr"); +! r = (F2HBLK *) mem_alloc(sizeof(F22HBLK), "r"); +! phi_sav = (F2HBLK *) mem_alloc(sizeof(F22HBLK), "phi_sav"); + +- /*#if 0 */ + i = sfreq + 0.5; + switch(i){ + case 32000: sfreq_idx = 0; break; + case 44100: sfreq_idx = 1; break; + case 48000: sfreq_idx = 2; break; + default: printf("error, invalid sampling frequency: %d Hz\n",i); + exit(-1); + } +! printf("absthr[][] sampling frequency index: %d\n",sfreq_idx); +! read_absthr(absthr, sfreq_idx); +! switch(lay){ +! case 1: sync_flush=576; flush=384; syncsize=1024; break; +! case 2: sync_flush=480; flush=576; syncsize=1056; break; +! case 3: sync_flush=768; flush=576; syncsize=1344; break; +! default: printf("Bad lay value:(%d)",lay); exit(-1); break; +! } +! /* #endif */ +! +! /* calculate HANN window coefficients */ +! /* for(i=0;i<BLKSIZE;i++) window[i] =0.5*(1-cos(2.0*PI*i/(BLKSIZE-1.0)));*/ +! for(i=0;i<BLKSIZE;i++) window[i] =0.5*(1-cos(2.0*PI*(i-0.5)/BLKSIZE)); +! for(i=0;i<BLKSIZE_s;i++)window_s[i]=0.5*(1-cos(2.0*PI*(i-0.5)/BLKSIZE_s)); + /* reset states used in unpredictability measure */ + for(i=0;i<HBLKSIZE;i++){ +- r[0][0][i]=r[1][0][i]=r[0][1][i]=r[1][1][i]=0; +- phi_sav[0][0][i]=phi_sav[1][0][i]=0; +- phi_sav[0][1][i]=phi_sav[1][1][i]=0; + lthr[0][i] = 60802371420160.0; + lthr[1][i] = 60802371420160.0; + } +--- 275,359 ---- + + /* The following static variables are constants. */ + + static FLOAT crit_band[27] = {0, 100, 200, 300, 400, 510, 630, 770, + 920, 1080, 1270,1480,1720,2000,2320, 2700, + 3150, 3700, 4400,5300,6400,7700,9500,12000, + 15500,25000,30000}; + +! static FLOAT *nb, *cb, *ecb, *bc; + ++ static FLOAT *c, *fthr; ++ static F32 *snrtmp; + static int *numlines ; ++ static int *numlines2 ; + static int *partition; + static FLOAT *cbval, *rnorm; + static double *tmn; + static FCB *s; + static FHBLK *lthr; +! static F2HBLK *ax_sav, *bx_sav, *rx_sav; +! static double decibel_lower; +! +! +! assert( info->lay == 3 ); + + +! if(init==0){ +! /* multiply an energy by decibel_lower to reduce its volume by +! * vdb Decibels. */ +! double vdb = 5; +! decibel_lower = pow(10.0,-vdb/10.0); +! +! +! +! /* These dynamic memory allocations simulate "static" variables placed */ +! /* in the data space. Each mem_alloc() call here occurs only once at */ +! /* initialization time. The mem_free() function must not be called. */ + nb = (FLOAT *) mem_alloc(sizeof(FCB), "nb"); + cb = (FLOAT *) mem_alloc(sizeof(FCB), "cb"); + ecb = (FLOAT *) mem_alloc(sizeof(FCB), "ecb"); + bc = (FLOAT *) mem_alloc(sizeof(FCB), "bc"); +! + c = (FLOAT *) mem_alloc(sizeof(FHBLK), "c"); + fthr = (FLOAT *) mem_alloc(sizeof(FHBLK), "fthr"); + snrtmp = (F32 *) mem_alloc(sizeof(F2_32), "snrtmp"); + + numlines = (int *) mem_alloc(sizeof(ICB), "numlines"); ++ if (gpsycho) ++ numlines2 = (int *) mem_alloc(sizeof(ICB), "numlines"); ++ else /* ISO model overwrites numlines with garbage */ ++ numlines2 = numlines; ++ + partition = (int *) mem_alloc(sizeof(IHBLK), "partition"); + cbval = (FLOAT *) mem_alloc(sizeof(FCB), "cbval"); + rnorm = (FLOAT *) mem_alloc(sizeof(FCB), "rnorm"); +! tmn = (double *) mem_alloc(sizeof(DCBA), "tmn"); + s = (FCB *) mem_alloc(sizeof(FCBCB), "s"); + lthr = (FHBLK *) mem_alloc(sizeof(F2HBLK), "lthr"); +! rx_sav = (F2HBLK *) mem_alloc(sizeof(F22HBLK), "rx_sav"); +! ax_sav = (F2HBLK *) mem_alloc(sizeof(F22HBLK), "ax_sav"); +! bx_sav = (F2HBLK *) mem_alloc(sizeof(F22HBLK), "bx_sav"); +! + + i = sfreq + 0.5; + switch(i){ + case 32000: sfreq_idx = 0; break; + case 44100: sfreq_idx = 1; break; + case 48000: sfreq_idx = 2; break; ++ case 16000: sfreq_idx = 3; break; ++ case 22050: sfreq_idx = 4; break; ++ case 24000: sfreq_idx = 5; break; + default: printf("error, invalid sampling frequency: %d Hz\n",i); + exit(-1); + } +! +! +! + /* reset states used in unpredictability measure */ ++ memset (rx_sav,0, 4* HBLKSIZE); ++ memset (ax_sav,0, 4* HBLKSIZE); ++ memset (bx_sav,0, 4* HBLKSIZE); + for(i=0;i<HBLKSIZE;i++){ + lthr[0][i] = 60802371420160.0; + lthr[1][i] = 60802371420160.0; + } +*************** +*** 241,247 **** + temp1 = (cbval[i] - cbval[j])*1.05; + if(temp1>=0.5 && temp1<=2.5){ + temp2 = temp1 - 0.5; +! temp2 = 8.0 * (temp2*temp2 - 2.0 * temp2); + } + else temp2 = 0; + temp1 += 0.474; +--- 408,414 ---- + temp1 = (cbval[i] - cbval[j])*1.05; + if(temp1>=0.5 && temp1<=2.5){ + temp2 = temp1 - 0.5; +! temp2 = 8.0 * (temp2*(temp2 - 2.0)); + } + else temp2 = 0; + temp1 += 0.474; +*************** +*** 264,1033 **** + rnorm[j] += s[j][i]; + } + } + init++; + } +! + /************************* End of Initialization *****************************/ +- switch(lay) { +- case 1: +- case 2: +- for ( i=0; i<lay; i++) +- { +- /***************************************************************************** +- * Net offset is 480 samples (1056-576) for layer 2; this is because one must* +- * stagger input data by 256 samples to synchronize psychoacoustic model with* +- * filter bank outputs, then stagger so that center of 1024 FFT window lines * +- * up with center of 576 "new" audio samples. * +- * * +- * For layer 1, the input data still needs to be staggered by 256 samples, * +- * then it must be staggered again so that the 384 "new" samples are centered* +- * in the 1024 FFT window. The net offset is then 576 and you need 448 "new"* +- * samples for each iteration to keep the 384 samples of interest centered * +- *****************************************************************************/ +- for (j=0; j<syncsize; j++) +- { +- if (j < (sync_flush) ) +- savebuf[j] = savebuf[j+flush]; +- else +- savebuf[j] = *buffer++; + +- /**window data with HANN window***********************************************/ +- if (j<BLKSIZE) +- { +- wsamp_r[j] = window[j]*((FLOAT) savebuf[j]); +- wsamp_i[j] = 0; +- } +- } +- /**Compute FFT****************************************************************/ +- fft(wsamp_r,wsamp_i,energy,phi,1024); +- /***************************************************************************** +- * calculate the unpredictability measure, given energy[f] and phi[f] * +- *****************************************************************************/ +- for(j=0; j<HBLKSIZE; j++){ +- r_prime = 2.0 * r[chn][old][j] - r[chn][oldest][j]; +- phi_prime = 2.0 * phi_sav[chn][old][j] - phi_sav[chn][oldest][j]; +- r[chn][new][j] = sqrt((double) energy[j]); +- phi_sav[chn][new][j] = phi[j]; +- temp1 = r[chn][new][j] * cos((double) phi[j]) +- - r_prime * cos(phi_prime); +- temp2=r[chn][new][j] * sin((double) phi[j]) +- - r_prime * sin(phi_prime); +- temp3=r[chn][new][j] + fabs(r_prime); +- if(temp3 != 0)c[j]=sqrt(temp1*temp1+temp2*temp2)/temp3; +- else c[j] = 0; +- } +- /*only update data "age" pointers after you are done with the second channel */ +- /*for layer 1 computations, for the layer 2 double computations, the pointers*/ +- /*are reset automatically on the second pass */ +- if(lay==2 || chn==1){ +- if(new==0){new = 1; oldest = 1;} +- else {new = 0; oldest = 0;} +- if(old==0)old = 1; else old = 0; +- } +- /***************************************************************************** +- * Calculate the grouped, energy-weighted, unpredictability measure, * +- * grouped_c[], and the grouped energy. grouped_e[] * +- *****************************************************************************/ +- for(j=1;j<CBANDS;j++){ +- grouped_e[j] = 0; +- grouped_c[j] = 0; +- } +- grouped_e[0] = energy[0]; +- grouped_c[0] = energy[0]*c[0]; +- for(j=1;j<HBLKSIZE;j++){ +- grouped_e[partition[j]] += energy[j]; +- grouped_c[partition[j]] += energy[j]*c[j]; +- } +- /***************************************************************************** +- * convolve the grouped energy-weighted unpredictability measure * +- * and the grouped energy with the spreading function, s[j][k] * +- *****************************************************************************/ +- for(j=0;j<CBANDS;j++){ +- ecb[j] = 0; +- cb[j] = 0; +- for(k=0;k<CBANDS;k++){ +- if(s[j][k] != 0.0){ +- ecb[j] += s[j][k]*grouped_e[k]; +- cb[j] += s[j][k]*grouped_c[k]; +- } +- } +- if(ecb[j] !=0)cb[j] = cb[j]/ecb[j]; +- else cb[j] = 0; +- } +- /***************************************************************************** +- * Calculate the required SNR for each of the frequency partitions * +- * this whole section can be accomplished by a table lookup * +- *****************************************************************************/ +- for(j=0;j<CBANDS;j++){ +- if(cb[j]<.05)cb[j]=0.05; +- else if(cb[j]>.5)cb[j]=0.5; +- /* tb = -0.434294482*log((double) cb[j])-0.301029996; */ +- tb = -0.43 *log((double) cb[j]) - 0.29 ; +- if(tb<0.0) tb=0.0; else if(tb>1.0) tb=1.0; +- bc[j] = tmn[j]*tb + nmt*(1.0-tb); +- k = cbval[j] + 0.5; +- bc[j] = (bc[j] > bmax[k]) ? bc[j] : bmax[k]; +- bc[j] = exp((double) -bc[j]*LN_TO_LOG10); +- } +- /***************************************************************************** +- * Calculate the permissible noise energy level in each of the frequency * +- * partitions. Include absolute threshold and pre-echo controls * +- * this whole section can be accomplished by a table lookup * +- *****************************************************************************/ +- for(j=0;j<CBANDS;j++) +- if(rnorm[j] && numlines[j]) +- nb[j] = ecb[j]*bc[j]/(rnorm[j]*numlines[j]); +- else nb[j] = 0; +- for(j=0;j<HBLKSIZE;j++){ +- temp1=nb[partition[j]]; /* preliminary threshold */ +- temp1=(temp1>absthr[j])?temp1:absthr[j]; +- /*do not use pre-echo control for layer 2 because it may do bad things to the*/ +- /* MUSICAM bit allocation algorithm */ +- if(lay==1){ +- fthr[j] = (temp1 < lthr[chn][j]) ? temp1 : lthr[chn][j]; +- temp2 = temp1 * 0.00316; +- fthr[j] = (temp2 > fthr[j]) ? temp2 : fthr[j]; +- } +- else fthr[j] = temp1; +- lthr[chn][j] = LXMIN*temp1; +- } +- /***************************************************************************** +- * Translate the 512 threshold values to the 32 filter bands of the coder * +- *****************************************************************************/ +- for(j=0;j<193;j += 16){ +- min_thres = fthr[j]; +- sum_energy = energy[j]; +- for(k=1;k<17;k++){ +- if(min_thres>fthr[j+k]) min_thres = fthr[j+k]; +- sum_energy += energy[j+k]; +- } +- snrtmp[i][j/16] = sum_energy/(min_thres * 17.0); +- snrtmp[i][j/16] = 4.342944819 * log((double)snrtmp[i][j/16]); +- } +- for(j=208;j<(HBLKSIZE-1);j += 16){ +- min_thres = 0.0; +- sum_energy = 0.0; +- for(k=0;k<17;k++){ +- min_thres += fthr[j+k]; +- sum_energy += energy[j+k]; +- } +- snrtmp[i][j/16] = sum_energy/min_thres; +- snrtmp[i][j/16] = 4.342944819 * log((double)snrtmp[i][j/16]); +- } +- /***************************************************************************** +- * End of Psychoacuostic calculation loop * +- *****************************************************************************/ +- } +- for(i=0; i<32; i++){ +- if(lay==2) /* if(lay==2 && chn==2) MI */ +- snr32[i]=(snrtmp[0][i]>snrtmp[1][i])?snrtmp[0][i]:snrtmp[1][i]; +- else snr32[i]=snrtmp[0][i]; +- } +- break; +- +- /*************************************************************************/ +- /** LAYER 3 */ +- /*************************************************************************/ + +- case 3: +- if ( init_L3 == 0 ) +- { +- L3para_read( sfreq,numlines,partition_l,minval,qthr_l,norm_l,s3_l, +- partition_s,qthr_s,norm_s,SNR_s, +- cbw_l,bu_l,bo_l,w1_l,w2_l, cbw_s,bu_s,bo_s,w1_s,w2_s ); +- init_L3 ++ ; +- } +- +- for ( j = 0; j < 21; j++ ) +- ratio_d[j] = ratio[chn][j]; +- for ( j = 0; j < 12; j++ ) +- for ( i = 0; i < 3; i++ ) +- ratio_ds[j][i] = ratio_s[chn][j][i]; +- +- if ( chn == 0 ) +- if ( new == 0 ) +- { +- new = 1; +- old = 0; +- oldest = 1; +- } +- else +- { +- new = 0; +- old = 1; +- oldest = 0; +- } + + + /********************************************************************** +! * Delay signal by sync_flush=768 samples * + **********************************************************************/ +! +! for ( j = 0; j < sync_flush; j++ ) /* for long window samples */ +! savebuf[j] = savebuf[j+flush]; +! +! for ( j = sync_flush; j < syncsize; j++ ) +! savebuf[j] = *buffer++; +! +! for ( j = 0; j < BLKSIZE; j++ ) +! { /**window data with HANN window**/ +! wsamp_r[j] = window[j] * savebuf[j]; +! wsamp_i[j] = 0.0; +! } +! + + /********************************************************************** + * compute unpredicatability of first six spectral lines * + **********************************************************************/ + +! fft( wsamp_r, wsamp_i, energy, phi, 1024 ); /**long FFT**/ +! for ( j = 0; j < 6; j++ ) +! { /* calculate unpredictability measure cw */ +! r_prime = 2.0 * r[chn][old][j] - r[chn][oldest][j]; +! phi_prime = 2.0 * phi_sav[chn][old][j]-phi_sav[chn][oldest][j]; +! r[chn][new][j] = sqrt((double) energy[j]); +! phi_sav[chn][new][j] = phi[j]; +! temp1 = r[chn][new][j] * cos((double) phi[j]) +! - r_prime * cos(phi_prime); +! temp2 = r[chn][new][j] * sin((double) phi[j]) +! - r_prime * sin(phi_prime); +! temp3=r[chn][new][j] + fabs(r_prime); +! +! if ( temp3 != 0.0 ) +! cw[j] = sqrt( temp1*temp1+temp2*temp2 ) / temp3; +! else +! cw[j] = 0; +! } + + + /********************************************************************** + * compute unpredicatibility of next 200 spectral lines * + **********************************************************************/ +! for ( sblock = 0; sblock < 3; sblock++ ) +! { /**window data with HANN window**/ +! for ( j = 0, k = 128 * (2 + sblock); j < 256; j++, k++ ) +! { +! wsamp_r[j] = window_s[j]* savebuf[k]; +! wsamp_i[j] = 0.0; +! } /* short FFT*/ +! +! fft( wsamp_r, wsamp_i, &energy_s[sblock][0], &phi_s[sblock][0], 256 ); +! } +! +! sblock = 1; + +! for ( j = 6; j < 206; j += 4 ) +! {/* calculate unpredictability measure cw */ +! double r2, phi2, temp1, temp2, temp3; +! +! k = (j+2) / 4; +! r_prime = 2.0 * sqrt((double) energy_s[0][k]) +! - sqrt((double) energy_s[2][k]); +! phi_prime = 2.0 * phi_s[0][k] - phi_s[2][k]; +! r2 = sqrt((double) energy_s[1][k]); +! phi2 = phi_s[1][k]; +! temp1 = r2 * cos( phi2 ) - r_prime * cos( phi_prime ); +! temp2 = r2 * sin( phi2 ) - r_prime * sin( phi_prime ); +! temp3 = r2 + fabs( r_prime ); +! if ( temp3 != 0.0 ) +! cw[j] = sqrt( temp1 * temp1 + temp2 * temp2 ) / temp3; +! else +! cw[j] = 0.0; +! cw[j+1] = cw[j+2] = cw[j+3] = cw[j]; +! } + + + /********************************************************************** +! * Set unpredicatiblility of remaining spectral lines to 0.4 * + **********************************************************************/ +! for ( j = 206; j < HBLKSIZE; j++ ) +! cw[j] = 0.4; + + + + /********************************************************************** + * Calculate the energy and the unpredictability in the threshold * + * calculation partitions * + **********************************************************************/ +! +! for ( b = 0; b < CBANDS; b++ ) +! { +! eb[b] = 0.0; +! cb[b] = 0.0; +! } +! for ( j = 0; j < HBLKSIZE; j++ ) +! { +! int tp = partition_l[j]; +! if ( tp >= 0 ) +! { +! eb[tp] += energy[j]; +! cb[tp] += cw[j] * energy[j]; +! } +! } +! + + /********************************************************************** + * convolve the partitioned energy and unpredictability * + * with the spreading function, s3_l[b][k] * + ******************************************************************** */ +! +! for ( b = 0; b < CBANDS; b++ ) +! { +! ecb[b] = 0.0; +! ctb[b] = 0.0; +! } +! for ( b = 0;b < CBANDS; b++ ) +! { +! for ( k = 0; k < CBANDS; k++ ) +! { +! ecb[b] += s3_l[b][k] * eb[k]; /* sprdngf for Layer III */ +! ctb[b] += s3_l[b][k] * cb[k]; +! } +! } + +! /* calculate the tonality of each threshold calculation partition */ +! /* calculate the SNR in each threshhold calculation partition */ + +! for ( b = 0; b < CBANDS; b++ ) +! { +! double cbb,tbb; +! if (ecb[b] != 0.0 ) +! { +! cbb = ctb[b]/ecb[b]; +! if (cbb <0.01) cbb = 0.01; +! cbb = log( cbb); +! } +! else +! cbb = 0.0 ; +! tbb = -0.299 - 0.43*cbb; /* conv1=-0.299, conv2=-0.43 */ +! tbb = minimum( 1.0, maximum( 0.0, tbb) ) ; /* 0<tbb<1 */ +! SNR_l[b] = maximum( minval[b], 29.0*tbb+6.0*(1.0-tbb) ); +! } /* TMN=29.0,NMT=6.0 for all calculation partitions */ +! +! for ( b = 0; b < CBANDS; b++ ) /* calculate the threshold for each partition */ +! nb[b] = ecb[b] * norm_l[b] * exp( -SNR_l[b] * LN_TO_LOG10 ); + +! for ( b = 0; b < CBANDS; b++ ) +! { /* pre-echo control */ +! double temp_1; /* BUG of IS */ +! temp_1 = minimum( nb[b], minimum(2.0*nb_1[chn][b],16.0*nb_2[chn][b]) ); +! thr[b] = maximum( qthr_l[b], temp_1 );/* rpelev=2.0, rpelev2=16.0 */ +! nb_2[chn][b] = nb_1[chn][b]; +! nb_1[chn][b] = nb[b]; +! } + + +- *pe = 0.0; /* calculate percetual entropy */ +- for ( b = 0; b < CBANDS; b++ ) +- { +- double tp ; +- tp = minimum( 0.0, log((thr[b]+1.0) / (eb[b]+1.0) ) ) ; /*not log*/ +- *pe -= numlines[b] * tp ; +- } /* thr[b] -> thr[b]+1.0 : for non sound portition */ +- +- #define switch_pe 1800 +- blocktype = NORM_TYPE; +- + +- if ( *pe < switch_pe ) +- { /* no attack : use long blocks */ +- switch( blocktype_old[chn] ) +- { +- case NORM_TYPE: +- case STOP_TYPE: +- blocktype = NORM_TYPE; +- break; +- +- case SHORT_TYPE: +- blocktype = STOP_TYPE; +- break; +- +- case START_TYPE: +- fprintf( stderr, "Error in block selecting\n" ); +- abort(); +- break; /* problem */ +- } + +- /* threshold calculation (part 2) */ +- for ( sb = 0; sb < SBMAX_l; sb++ ) +- { +- en[sb] = w1_l[sb] * eb[bu_l[sb]] + w2_l[sb] * eb[bo_l[sb]]; +- thm[sb] = w1_l[sb] *thr[bu_l[sb]] + w2_l[sb] * thr[bo_l[sb]]; +- for ( b = bu_l[sb]+1; b < bo_l[sb]; b++ ) +- { +- en[sb] += eb[b]; +- thm[sb] += thr[b]; +- } +- if ( en[sb] != 0.0 ) +- ratio[chn][sb] = thm[sb]/en[sb]; +- else +- ratio[chn][sb] = 0.0; +- } +- } +- else +- { +- /* attack : use short blocks */ +- blocktype = SHORT_TYPE; +- +- if ( blocktype_old[chn] == NORM_TYPE ) +- blocktype_old[chn] = START_TYPE; +- if ( blocktype_old[chn] == STOP_TYPE ) +- blocktype_old[chn] = SHORT_TYPE ; +- +- /* threshold calculation for short blocks */ +- +- for ( sblock = 0; sblock < 3; sblock++ ) +- { +- for ( b = 0; b < CBANDS_s; b++ ) +- { +- eb[b] = 0.0; +- ecb[b] = 0.0; +- } +- for ( j = 0; j < HBLKSIZE_s; j++ ) +- eb[partition_s[j]] += energy_s[sblock][j]; +- for ( b = 0; b < CBANDS_s; b++ ) +- for ( k = 0; k < CBANDS_s; k++ ) +- ecb[b] += s3_l[b][k] * eb[k]; +- for ( b = 0; b < CBANDS_s; b++ ) +- { +- nb[b] = ecb[b] * norm_l[b] * exp( (double) SNR_s[b] * LN_TO_LOG10 ); +- thr[b] = maximum (qthr_s[b],nb[b]); +- } +- for ( sb = 0; sb < SBMAX_s; sb++ ) +- { +- en[sb] = w1_s[sb] * eb[bu_s[sb]] + w2_s[sb] * eb[bo_s[sb]]; +- thm[sb] = w1_s[sb] *thr[bu_s[sb]] + w2_s[sb] * thr[bo_s[sb]]; +- for ( b = bu_s[sb]+1; b < bo_s[sb]; b++ ) +- { +- en[sb] += eb[b]; +- thm[sb] += thr[b]; +- } +- if ( en[sb] != 0.0 ) +- ratio_s[chn][sb][sblock] = thm[sb]/en[sb]; +- else +- ratio_s[chn][sb][sblock] = 0.0; +- } +- } +- } +- +- cod_info->block_type = blocktype_old[chn]; +- blocktype_old[chn] = blocktype; + +! if ( cod_info->block_type == NORM_TYPE ) +! cod_info->window_switching_flag = 0; +! else +! cod_info->window_switching_flag = 1; +! cod_info->mixed_block_flag = 0; +! break; + + +! default: +! printf("error, invalid MPEG/audio coding layer: %d\n",lay); + } + +! /* These mem_free() calls must correspond with the mem_alloc() calls */ +! /* used at the beginning of this function to simulate "automatic" */ +! /* variables placed on the stack. */ +! +! mem_free((void **) &grouped_c); +! mem_free((void **) &grouped_e); +! mem_free((void **) &nb); +! mem_free((void **) &cb); +! mem_free((void **) &ecb); +! mem_free((void **) &bc); +! mem_free((void **) &wsamp_r); +! mem_free((void **) &wsamp_i); +! mem_free((void **) &phi); +! mem_free((void **) &energy); +! mem_free((void **) &c); +! mem_free((void **) &fthr); +! mem_free((void **) &snrtmp); +! } +! #ifdef DEBUG +! #undef DEBUG + #endif + + + void L3para_read(double sfreq, int *numlines, int *partition_l, double *minval, double *qthr_l, double *norm_l, double (*s3_l)[63], int *partition_s, double *qthr_s, double *norm_s, double *SNR, int *cbw_l, int *bu_l, int *bo_l, double *w1_l, double *w2_l, int *cbw_s, int *bu_s, int *bo_s, double *w1_s, double *w2_s) + { +! double freq_tp; +! static double bval_l[CBANDS], bval_s[CBANDS]; +! int cbmax, cbmax_tp; +! static double s3_s[CBANDS][CBANDS]; +! +! FILE *fin; +! char tp[256]; +! int sbmax ; +! int i,j,k,k2,loop, part_max ; +! +! fin = OpenTableFile( "psy_data" ); +! if (fin == NULL) +! exit( 1 ); +! +! /* Read long block data */ +! +! for(loop=0;loop<6;loop++) +! { +! fscanf(fin,"freq = %lf partition = %d\n",&freq_tp,&cbmax_tp); +! cbmax_tp++; +! #ifdef DEBUG +! printf("freq = %f partition = %d\n",freq_tp,cbmax); +! #endif + +! if (sfreq == freq_tp ) +! { +! cbmax = cbmax_tp; +! for(i=0,k2=0;i<cbmax_tp;i++) +! { +! fscanf(fin, +! "No=%d #lines=%d minval=%lf qthr=%lf norm=%lf bval=%lf\n", +! &j,&numlines[i],&minval[i],&qthr_l[i],&norm_l[i],&bval_l[i]); +! if (j!=i) +! { printf("please check \"psy_data\""); +! exit(-1); +! } +! for(k=0;k<numlines[i];k++) +! partition_l[k2++] = i ; +! #ifdef DEBUG +! printf("No=%2d #lines=%2d minval=%4.1f qthr=%8.3f norm=%5.3f bval=%8.3f\n", +! i,numlines[i],minval[i],qthr_l[i],norm_l[i],bval_l[i] ); +! #endif + } +! } +! else +! { +! for(j=0;j<cbmax_tp;j++) +! { +! fgets(tp,255,fin); +! sscanf(tp,"No=%d %s\n",&i,tp); +! if (j!=i) +! { printf("please check \"psy_data.\"\n"); +! exit(-1); +! } +! } +! } +! } + +! /************************************************************************ +! * Now compute the spreading function, s[j][i], the value of the spread-* +! * ing function, centered at band j, for band i, store for later use * +! ************************************************************************/ +! #ifdef DEBUG +! printf("freq = %f\n",sfreq); +! #endif +! part_max = cbmax ; +! for(i=0;i<part_max;i++) +! { +! double tempx,x,tempy,temp; +! for(j=0;j<part_max;j++) + { +! tempx = (bval_l[i] - bval_l[j])*1.05; +! if (j>=i) tempx = (bval_l[i] - bval_l[j])*3.0; +! else tempx = (bval_l[i] - bval_l[j])*1.5; +! /* if (j>=i) tempx = (bval_l[j] - bval_l[i])*3.0; +! else tempx = (bval_l[j] - bval_l[i])*1.5; */ +! if(tempx>=0.5 && tempx<=2.5) +! { +! temp = tempx - 0.5; +! x = 8.0 * (temp*temp - 2.0 * temp); +! } +! else x = 0.0; +! tempx += 0.474; +! tempy = 15.811389 + 7.5*tempx - 17.5*sqrt(1.0+tempx*tempx); +! if (tempy <= -60.0) s3_l[i][j] = 0.0; +! else s3_l[i][j] = exp( (x + tempy)*LN_TO_LOG10 ); +! #ifdef DEBUG_S3 +! printf("s[%d][%d]=%f\n",i,j,s3_l[i][j]); +! #endif + +! #ifdef DEBUGP +! printf("j=%d i=%d tempy=%f s[i][j]=%f \n",i,j,tempy,s[i][j]); +! minval[i] = bmax[j-1]; +! printf("minval[%d] = %f, j-1=%d %f\n",i, minval[i] , j,fthr[i]) ; +! #endif +! } +! } + + +! /* Read short block data */ + +! for(loop=0;loop<6;loop++) +! { +! fscanf(fin,"freq = %lf partition = %d\n",&freq_tp,&cbmax_tp); +! cbmax_tp++; +! #ifdef DEBUG +! printf("freq = %f partition = %d\n",freq_tp,cbmax); +! #endif + +! if (sfreq == freq_tp ) +! { +! cbmax = cbmax_tp; +! for(i=0,k2=0;i<cbmax_tp;i++) +! { +! fscanf(fin, +! "No=%d #lines=%d qthr=%lf norm=%lf SNR=%lf bval=%lf\n", +! &j,&numlines[i],&qthr_s[i],&norm_s[i],&SNR[i],&bval_s[i]); +! if (j!=i) +! { printf("please check \"psy_data\""); +! exit(-1); +! } +! for(k=0;k<numlines[i];k++) +! partition_s[k2++] = i ; +! #ifdef DEBUG +! printf("No=%2d #lines=%2d qthr=%8.3f norm=%5.3f SNR=%6.3f bval=%8.3f\n", +! i,numlines[i],qthr_s[i],norm_s[i],SNR[i],bval_s[i] ); +! #endif + } +- } +- else +- { +- for(j=0;j<cbmax_tp;j++) +- { +- fgets(tp,255,fin); +- sscanf(tp,"No=%d %s\n",&i,tp); +- if (j!=i) +- { printf("please check \"psy_data.\"\n"); +- exit(-1); +- } +- } +- } +- } + +! /************************************************************************ +! * Now compute the spreading function, s[j][i], the value of the spread-* +! * ing function, centered at band j, for band i, store for later use * +! ************************************************************************/ +! #ifdef DEBUG_S3 +! fpp=fopen("s3_s","w"); +! #endif +! part_max = cbmax ; +! for(i=0;i<part_max;i++) +! { +! double tempx,x,tempy,temp; +! for(j=0;j<part_max;j++) + { +! tempx = (bval_s[i] - bval_s[j])*1.05; +! if (j>=i) tempx = (bval_s[i] - bval_s[j])*3.0; +! else tempx = (bval_s[i] - bval_s[j])*1.5; +! if(tempx>=0.5 && tempx<=2.5) +! { +! temp = tempx - 0.5; +! x = 8.0 * (temp*temp - 2.0 * temp); +! } +! else x = 0.0; +! tempx += 0.474; +! tempy = 15.811389 + 7.5*tempx - 17.5*sqrt(1.0+tempx*tempx); +! if (tempy <= -60.0) s3_s[i][j] = 0.0; +! else s3_s[i][j] = exp( (x + tempy)*LN_TO_LOG10 ); +! #ifdef DEBUG_S3 +! fprintf(fpp,"s3_s[%d][%d]=%f\n",i,j,s3_s[i][j]); +! #endif +! #ifdef DEBUGP +! printf("j=%d i=%d tempy=%f s[i][j]=%f \n",i,j,tempy,s[i][j]); +! minval[i] = bmax[j-1]; +! printf("minval[%d] = %f, j-1=%d %f\n",i, minval[i] , j,fthr[i]) ; +! #endif +! } +! } +! #ifdef DEBUG_S3 +! fclose(fpp); +! #endif +! /* Read long block data for converting threshold calculation +! partitions to scale factor bands */ + +! for(loop=0;loop<6;loop++) +! { +! fscanf(fin,"freq=%lf sb=%d\n",&freq_tp,&sbmax); +! sbmax++; +! #ifdef DEBUG +! printf("freq = %f sb = %d\n",freq_tp,sbmax); +! #endif + +! if (sfreq == freq_tp) +! { +! for(i=0;i<sbmax;i++) +! { +! fscanf(fin, +! "sb=%d cbw=%d bu=%d bo=%d w1=%lf w2=%lf\n", +! &j,&cbw_l[i],&bu_l[i],&bo_l[i],&w1_l[i],&w2_l[i]); +! if (j!=i) +! { printf("30:please check \"psy_data\"\n"); +! exit(-1); +! } +! #ifdef DEBUG +! printf( +! "sb=%2d cbw=%1d bu=%2d bo=%2d w1=%5.3f w2=%5.3f\n", +! j,cbw_l[i],bu_l[i],bo_l[i],w1_l[i],w2_l[i]); +! #endif +! if (i!=0) +! if ( (bo_l[i] != (bu_l[i]+cbw_l[i])) || +! (fabs(1.0-w1_l[i]-w2_l[i-1]) > 0.01 ) ) +! { printf("31:please check \"psy_data.\"\n"); +! exit(-1); +! } +! } +! } +! else +! { +! for(j=0;j<sbmax;j++) +! { +! fgets(tp,255,fin); +! sscanf(tp,"sb=%d %s\n",&i,tp); +! if (j!=i) +! { printf("please check \"psy_data.\"\n"); +! exit(-1); +! } +! } +! } +! } + +- /* Read short block data for converting threshold calculation +- partitions to scale factor bands */ + +! for(loop=0;loop<6;loop++) +! { +! fscanf(fin,"freq=%lf sb=%d\n",&freq_tp,&sbmax); +! sbmax++; +! #ifdef DEBUG +! printf("freq = %f sb = %d\n",freq_tp,sbmax); +! #endif + +- if (sfreq == freq_tp) +- { +- for(i=0;i<sbmax;i++) +- { +- fscanf(fin, +- "sb=%d cbw=%d bu=%d bo=%d w1=%lf w2=%lf\n", +- &j,&cbw_s[i],&bu_s[i],&bo_s[i],&w1_s[i],&w2_s[i]); +- if (j!=i) +- { printf("30:please check \"psy_data\"\n"); +- exit(-1); +- } +- #ifdef DEBUG +- printf( +- "sb=%2d cbw=%1d bu=%2d bo=%2d w1=%5.3f w2=%5.3f\n", +- j,cbw_s[i],bu_s[i],bo_s[i],w1_s[i],w2_s[i]); +- #endif +- if (i!=0) +- if ( (bo_s[i] != (bu_s[i]+cbw_s[i])) || +- (fabs(1.0-w1_s[i]-w2_s[i-1]) > 0.01 ) ) +- { printf("31:please check \"psy_data.\"\n"); +- exit(-1); +- } +- } +- } +- else +- { +- for(j=0;j<sbmax;j++) +- { +- fgets(tp,255,fin); +- sscanf(tp,"sb=%d %s\n",&i,tp); +- if (j!=i) +- { printf("please check \"psy_data.\"\n"); +- exit(-1); +- } +- } +- } +- } + + } +--- 431,1417 ---- + rnorm[j] += s[j][i]; + } + } ++ ++ ++ /* setup stereo demasking thresholds */ ++ /* formula reverse enginerred from plot in paper */ ++ for ( sb = 0; sb < SBMAX_s; sb++ ) { ++ double mld = 1.25*(1-cos(3.14159*sb/SBMAX_s))-2.5; ++ mld_s[sb] = pow(10.0,mld); ++ } ++ for ( sb = 0; sb < SBMAX_l; sb++ ) { ++ double mld = 1.25*(1-cos(3.14159*sb/SBMAX_l))-2.5; ++ mld_l[sb] = pow(10.0,mld); ++ } ++ ++ ++ ++ + init++; + } +! +! if ( init_L3 == 0 ) +! { +! L3para_read( sfreq,numlines2,partition_l,minval,qthr_l,norm_l,s3_l, +! partition_s,qthr_s,norm_s,SNR_s, +! cbw_l,bu_l,bo_l,w1_l,w2_l, cbw_s,bu_s,bo_s,w1_s,w2_s ); +! init_L3 ++ ; +! } + /************************* End of Initialization *****************************/ + + + + ++ ++ #ifdef AWS ++ /* reduce switch_pe if there where preecho events in previous granules */ ++ { ++ int prev_granule_used_shortblock=0; ++ ++ for (chn=0; chn<stereo; chn++) ++ if (blocktype_old[chn]==SHORT_TYPE) ++ prev_granule_used_shortblock=1; ++ ++ if (prev_granule_used_shortblock) ++ switch_pe=maximum(switch_pe-700,900); ++ else switch_pe = minimum(switch_pe+200,1800); ++ } ++ #endif ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ for (chn=0; chn<stereo; chn++) { ++ ++ ++ for ( j = 0; j < 21; j++ ) ++ ratio_d[chn][j] = ratio[chn][j]; ++ for ( j = 0; j < 12; j++ ) ++ for ( i = 0; i < 3; i++ ) ++ ratio_ds[chn][j][i] = ratio_s[chn][j][i]; ++ percep_energy[chn] = pe[chn]; ++ ++ ++ /*if ( chn == 0 )*/ ++ if ( new == 0 ) ++ { ++ new = 1; ++ old = 0; ++ oldest = 1; ++ } ++ else ++ { ++ new = 0; ++ old = 1; ++ oldest = 0; ++ } ++ ++ + /********************************************************************** +! * compute FFTs + **********************************************************************/ +! L3psycho_energy( buffer[chn], energy, ax, bx, energy_s, ax_s, bx_s, +! chn,gr_out,info,check_ms_stereo,&ms_ratio_l); +! ms_ratio_s=ms_ratio_l; + + /********************************************************************** + * compute unpredicatability of first six spectral lines * + **********************************************************************/ ++ for ( j = 0; j < 6; j++ ) ++ { /* calculate unpredictability measure cw */ ++ double an, a1, a2; ++ double bn, b1, b2; ++ double rn, r1, r2; ++ double numre, numim, den; ++ ++ a2 = ax_sav[chn][oldest][j]; ++ b2 = bx_sav[chn][oldest][j]; ++ r2 = rx_sav[chn][oldest][j]; ++ a1 = ax_sav[chn][oldest][j] = ax_sav[chn][old][j]; ++ b1 = bx_sav[chn][oldest][j] = bx_sav[chn][old][j]; ++ r1 = rx_sav[chn][oldest][j] = rx_sav[chn][old][j]; ++ an = ax_sav[chn][old][j] = ax[j]; ++ bn = bx_sav[chn][old][j] = bx[j]; ++ rn = rx_sav[chn][old][j] = sqrt(energy[j]); ++ ++ { /* square (x1,y1) */ ++ if( r1 != 0.0 ) { ++ numre = (a1*b1); ++ numim = (a1*a1-b1*b1)*0.5; ++ den = r1*r1; ++ } else { ++ numre = 1.0; ++ numim = 0.0; ++ den = 1.0; ++ } ++ } + +! { /* multiply by (x2,-y2) */ +! if( r2 != 0.0 ) { +! double tmp2 = (numim+numre)*(a2+b2)*0.5; +! double tmp1 = -a2*numre+tmp2; +! numre = -b2*numim+tmp2; +! numim = tmp1; +! den *= r2; +! } else { +! /* do nothing */ +! } +! } + ++ { /* r-prime factor */ ++ double tmp = (2.0*r1-r2)/den; ++ numre *= tmp; ++ numim *= tmp; ++ } ++ ++ if( (den=rn+fabs(2.0*r1-r2)) != 0.0 ) { ++ numre = (an+bn)/2.0-numre; ++ numim = (an-bn)/2.0-numim; ++ cw[j] = sqrt(numre*numre+numim*numim)/den; ++ } else { ++ cw[j] = 0.0; ++ } + ++ } ++ + /********************************************************************** + * compute unpredicatibility of next 200 spectral lines * + **********************************************************************/ +! sblock = 1; +! +! for ( j = 6; j < 206; j += 4 ) +! {/* calculate unpredictability measure cw */ +! double rn, r1, r2; +! double numre, numim, den; +! +! k = (j+2) / 4; +! +! { /* square (x1,y1) */ +! r1 = sqrt((double)energy_s[0][k]); +! if( r1 != 0.0 ) { +! double a1 = ax_s[0][k]; +! double b1 = bx_s[0][k]; +! numre = (a1*b1); +! numim = (a1*a1-b1*b1)*0.5; +! den = r1*r1; +! } else { +! numre = 1.0; +! numim = 0.0; +! den = 1.0; +! } +! } + +! { /* multiply by (x2,-y2) */ +! r2 = sqrt((double)energy_s[2][k]); +! if( r2 != 0.0 ) { +! double a2 = ax_s[2][k]; +! double b2 = bx_s[2][k]; +! +! double tmp2 = (numim+numre)*(a2+b2)*0.5; +! double tmp1 = -a2*numre+tmp2; +! numre = -b2*numim+tmp2; +! numim = tmp1; +! +! den *= r2; +! } else { +! /* do nothing */ +! } +! } +! +! { /* r-prime factor */ +! double tmp = (2.0*r1-r2)/den; +! numre *= tmp; +! numim *= tmp; +! } +! +! rn = sqrt((double)energy_s[1][k]); +! if( (den=rn+fabs(2.0*r1-r2)) != 0.0 ) { +! double an = ax_s[1][k]; +! double bn = bx_s[1][k]; +! numre = (an+bn)/2.0-numre; +! numim = (an-bn)/2.0-numim; +! cw[j] = sqrt(numre*numre+numim*numim)/den; +! } else { +! cw[j] = 0.0; +! } +! +! cw[j+1] = cw[j+2] = cw[j+3] = cw[j]; +! } + + + /********************************************************************** +! * Set unpredicatiblility of remaining spectral lines to 0.4 206..513 * + **********************************************************************/ +! for ( j = 206; j < HBLKSIZE; j++ ) +! cw[j] = 0.4; + + + ++ #if 0 ++ for ( j = 14; j < HBLKSIZE-4; j += 4 ) ++ {/* calculate energy from short ffts */ ++ double tot,ave; ++ k = (j+2) / 4; ++ for (tot=0, sblock=0; sblock < 3; sblock++) ++ tot+=energy_s[sblock][k]; ++ ave = energy[j+1]+ energy[j+2]+ energy[j+3]+ energy[j]; ++ ave /= 4.; ++ /* ++ printf("energy / tot %i %5.2f %e %e\n",j,ave/(tot*16./3.), ++ ave,tot*16./3.); ++ */ ++ energy[j+1] = energy[j+2] = energy[j+3] = energy[j]=tot; ++ } ++ #endif ++ ++ ++ ++ ++ ++ ++ ++ + /********************************************************************** + * Calculate the energy and the unpredictability in the threshold * + * calculation partitions * + **********************************************************************/ +! for ( b = 0; b < CBANDS; b++ ) +! { +! eb[b] = 0.0; +! cb[b] = 0.0; +! } +! for ( j = 0; j < HBLKSIZE; j++ ) +! { +! int tp = partition_l[j]; +! if ( tp >= 0 ) +! { +! eb[tp] += energy[j]; +! cb[tp] += cw[j] * energy[j]; +! } +! } +! + + /********************************************************************** + * convolve the partitioned energy and unpredictability * + * with the spreading function, s3_l[b][k] * + ******************************************************************** */ +! for ( b = 0; b < CBANDS; b++ ) +! { +! ecb[b] = 0.0; +! ctb[b] = 0.0; +! } +! /* s3_l is a sparse matrix for 44.1khz */ +! if (sfreq_idx==1) { +! sprdngf1(ecb,eb); +! sprdngf2(ctb,cb); +! } +! else { +! for ( b = 0;b < CBANDS; b++ ) +! { +! for ( k = 0; k < CBANDS; k++ ) +! { +! ecb[b] += s3_l[b][k] * eb[k]; /* sprdngf for Layer III */ +! ctb[b] += s3_l[b][k] * cb[k]; +! } +! } +! } +! /* calculate the tonality of each threshold calculation partition */ +! /* calculate the SNR in each threshhold calculation partition */ + +! for ( b = 0; b < CBANDS; b++ ) +! { +! double cbb,tbb; +! if (ecb[b] != 0.0 ) +! { +! cbb = ctb[b]/ecb[b]; +! if (cbb <0.01) cbb = 0.01; +! cbb = log( cbb); +! } +! else +! cbb = 0.0 ; +! tbb = -0.299 - 0.43*cbb; /* conv1=-0.299, conv2=-0.43 */ +! tbb = minimum( 1.0, maximum( 0.0, tbb) ) ; /* 0<tbb<1 */ +! SNR_l[b] = maximum( minval[b], 29.0*tbb+6.0*(1.0-tbb) ); +! } /* TMN=29.0,NMT=6.0 for all calculation partitions */ +! +! for ( b = 0; b < CBANDS; b++ ) /* calculate the threshold for each partition */ +! nb[b] = ecb[b] * norm_l[b] * exp( -SNR_l[b] * LN_TO_LOG10 ); +! +! for ( b = 0; b < CBANDS; b++ ) +! { /* pre-echo control */ +! double temp_1; /* BUG of IS */ +! int rpelev=2; int rpelev2=16; +! temp_1 = minimum( nb[b], minimum(rpelev*nb_1[chn][b],rpelev2*nb_2[chn][b]) ); +! if (experimentalX) temp_1 = temp_1*decibel_lower; +! thr[b] = maximum( qthr_l[b], temp_1 );/* rpelev=2.0, rpelev2=16.0 */ +! nb_2[chn][b] = nb_1[chn][b]; +! nb_1[chn][b] = nb[b]; +! } +! /* note: all surges in PE are because of the above pre-echo formula +! * for temp_1. it this is not used, PE is always around 600 +! */ +! +! pe[chn] = 0.0; /* calculate percetual entropy */ +! for ( b = 0; b < CBANDS; b++ ) +! { +! double tp ; +! tp = minimum( 0.0, log((thr[b]+1.0) / (eb[b]+1.0) ) ) ; /*not log*/ +! pe[chn] -= numlines[b] * tp ; +! +! } /* thr[b] -> thr[b]+1.0 : for non sound portition */ +! +! +! /*************************************************************** +! * Check to see if we also need to compute long block thresholds +! ***************************************************************/ +! uselongblock[chn] = (pe[chn] < switch_pe); +! +! { +! double mn,mx; +! for (sblock=0; sblock < 3; sblock++) +! estot[chn][sblock]=0; +! for ( j = HBLKSIZE_s/2; j < HBLKSIZE_s; j ++) +! for (sblock=0; sblock < 3; sblock++) +! estot[chn][sblock]+=energy_s[sblock][j]; +! mn = minimum(estot[chn][0],estot[chn][1]); +! mn = minimum(mn,estot[chn][2]); +! mx = maximum(estot[chn][0],estot[chn][1]); +! mx = maximum(mx,estot[chn][2]); +! #ifdef HAVEGTK +! if (gtkflag) { +! pinfo->ers[gr_out][chn]=ers_save[chn]; +! ers_save[chn]=mx/(1e-12+mn); +! pinfo->pe[gr_out][chn]=pe_save[chn]; +! pe_save[chn]=pe[chn]; +! } +! #endif +! #ifdef ENER_AWS +! uselongblock[chn] = 1; + +! /* old ISO formula. For t3.wav, add this and lower 25 to a 20. */ +! if (pe[chn] > 1800) uselongblock[chn]=0; + +! /* big surge of energy - always use short blocks */ +! if ( mx > 25*mn) uselongblock[chn] = 0; + ++ /* switch based on pe, but only if there is some energy surge */ ++ if (( mx > 2.5*mn ) && (pe[chn] > switch_pe)) uselongblock[chn]=0; ++ #endif ++ } ++ + + + + +! if ((chn == 1) && (force_ms)) { +! /* Forced ms_stereo mode. */ +! /* ch=0 (mid) blocktype determines ch=1 (side) blocktype */ +! uselongblock[1] = uselongblock[0]; +! } +! +! +! /*************************************************************** +! * compute masking thresholds +! * we need to always compute short block masking thresholds because +! * we dont know if this granule will be changed to a short block later. +! ***************************************************************/ +! if ( uselongblock[chn]) +! { /* no attack : use long blocks */ +! /* threshold calculation (part 2) */ +! for ( sb = 0; sb < SBMAX_l; sb++ ) +! { +! en[sb] = w1_l[sb] * eb[bu_l[sb]] + w2_l[sb] * eb[bo_l[sb]]; +! thm[sb] = w1_l[sb] *thr[bu_l[sb]] + w2_l[sb] * thr[bo_l[sb]]; +! for ( b = bu_l[sb]+1; b < bo_l[sb]; b++ ) +! { +! en[sb] += eb[b]; +! thm[sb] += thr[b]; +! } + ++ if ( en[sb] != 0.0 ) ++ ratio[chn][sb] = thm[sb]/en[sb]; ++ else ++ ratio[chn][sb] = 0.0; ++ } ++ #ifdef HAVEGTK ++ if (gtkflag) { ++ for (sb=0; sb< SBMAX_l; sb ++ ) { ++ pinfo->thr[gr_out][chn][sb]=thm_save[chn][sb]; ++ pinfo->en[gr_out][chn][sb]=en_save[chn][sb]; ++ thm_save[chn][sb]=thm[sb]; ++ en_save[chn][sb]=en[sb]; ++ } ++ } ++ #endif ++ for (sb=0; sb< SBMAX_l; sb ++ ) { ++ thm_save[chn][sb]=thm[sb]; ++ en_save[chn][sb]=en[sb]; ++ } + +! } +! +! /* in some cases, when computing the next granule, we may switch this +! * granule to a short block. compute short block thresholds just in case */ +! +! /* threshold calculation for short blocks */ +! for ( sblock = 0; sblock < 3; sblock++ ) { +! for ( b = 0; b < CBANDS_s; b++ ) +! { +! eb[b] = 0.0; +! ecb[b] = 0.0; +! } +! for ( j = 0; j < HBLKSIZE_s; j++ ) +! eb[partition_s[j]] += energy_s[sblock][j]; +! for ( b = 0; b < CBANDS_s; b++ ) +! for ( k = 0; k < CBANDS_s; k++ ) +! ecb[b] += s3_l[b][k] * eb[k]; +! for ( b = 0; b < CBANDS_s; b++ ) +! { +! if (gpsycho) +! nb[b] = ecb[b] * norm_s[b] * exp( (double) SNR_s[b] * LN_TO_LOG10 ); +! else +! nb[b] = ecb[b] * norm_l[b] * exp( (double) SNR_s[b] * LN_TO_LOG10 ); +! if (experimentalX) nb[b] = nb[b]*decibel_lower; +! thr[b] = maximum (qthr_s[b],nb[b]); +! } +! for ( sb = 0; sb < SBMAX_s; sb++ ) +! { +! en[sb] = w1_s[sb] * eb[bu_s[sb]] + w2_s[sb] * eb[bo_s[sb]]; +! thm[sb] = w1_s[sb] *thr[bu_s[sb]] + w2_s[sb] * thr[bo_s[sb]]; +! for ( b = bu_s[sb]+1; b < bo_s[sb]; b++ ) +! { +! en[sb] += eb[b]; +! thm[sb] += thr[b]; +! } +! +! if ( en[sb] != 0.0 ) +! ratio_s[chn][sb][sblock] = thm[sb]/en[sb]; +! else +! ratio_s[chn][sb][sblock] = 0.0; +! #ifdef HAVEGTK +! if (gtkflag) { +! pinfo->thr_s[gr_out][chn][3*sb+sblock]=thm_s_save[chn][sb][sblock]; +! pinfo->en_s[gr_out][chn][3*sb+sblock]=en_s_save[chn][sb][sblock]; +! } +! #endif +! thm_s_save[chn][sb][sblock]=thm[sb]; +! en_s_save[chn][sb][sblock]=en[sb]; +! } +! +! } +! +! +! /* compute M/S thresholds from Johnston & Ferreira 1992 ICASSP paper */ +! #define JOHNSTON +! #ifdef JOHNSTON +! if (force_ms) { +! if (chn == 1) { +! double rside,rmid,mld; +! if ( uselongblock[chn]) { +! for ( sb = 0; sb < SBMAX_l; sb++ ) { +! mld = mld_l[sb]; +! /* +! x1 = minimum(thm_save[0][sb],thm_save[1][sb]); +! x2 = maximum(thm_save[0][sb],thm_save[1][sb]); +! */ +! /* if db = 10*log10(x2/x1) < 2 */ +! /* if (x2 < 1.58*x1) { */ +! rmid = Max(ratio[0][sb],Min(ratio[1][sb],mld)); +! rside = Max(ratio[1][sb],Min(ratio[0][sb],mld)); +! ratio[0][sb]=rmid; +! ratio[1][sb]=rside; +! } +! } +! /* alwasy compute these - we may need them later */ +! for ( sblock = 0; sblock < 3; sblock++ ){ +! for ( sb = 0; sb < SBMAX_s; sb++ ) { +! mld = mld_s[sb]; +! rmid = Max(ratio_s[0][sb][sblock],Min(ratio_s[1][sb][sblock],mld)); +! rside = Max(ratio_s[1][sb][sblock],Min(ratio_s[0][sb][sblock],mld)); +! ratio_s[0][sb][sblock]=rmid; +! ratio_s[1][sb][sblock]=rside; +! } +! } +! } + } ++ #endif + +! +! +! } /* end loop over chn */ +! +! +! #if 1 +! if (check_ms_stereo) { +! /* determin ms_ratio from masking thresholds*/ +! /* use ms_stereo (ms_ratio < .35) if average thresh. diff < 5 db */ +! { double db,x1,x2,sidetot=0,tot=0; +! for (sb= SBMAX_l/4 ; sb< SBMAX_l; sb ++ ) { +! x1 = minimum(thm_save[0][sb],thm_save[1][sb]); +! x2 = maximum(thm_save[0][sb],thm_save[1][sb]); +! /* thresholds difference in db */ +! if (x2 >= 1000*x1) db=30; +! else db = 10*log10(x2/x1); +! sidetot += db; +! tot++; +! } +! ms_ratio_l= .35*(sidetot/tot)/5.0; +! +! sidetot=0; tot=0; +! for ( sblock = 0; sblock < 3; sblock++ ) +! for ( sb = SBMAX_s/4; sb < SBMAX_s; sb++ ) { +! x1 = minimum(thm_s_save[0][sb][sblock],thm_s_save[1][sb][sblock]); +! x2 = maximum(thm_s_save[0][sb][sblock],thm_s_save[1][sb][sblock]); +! /* thresholds difference in db */ +! if (x2 >= 1000*x1) db=30; +! else db = 10*log10(x2/x1); +! sidetot += db; +! tot++; +! } +! ms_ratio_s = .35*(sidetot/tot)/5.0; +! } +! } + #endif + + ++ /*************************************************************** ++ * determin final block type ++ ***************************************************************/ ++ for (chn=0; chn<stereo; chn++) { ++ blocktype[chn] = NORM_TYPE; ++ } ++ ++ if (!allow_diff_short) ++ if (info->mode==MPG_MD_JOINT_STEREO) { ++ /* force both channels to use the same block type */ ++ /* this is necessary if the frame is to be encoded in ms_stereo. */ ++ /* But even without ms_stereo, FhG does this */ ++ int bothlong= (uselongblock[0] && uselongblock[1]); ++ if (!bothlong) { ++ uselongblock[0]=0; ++ uselongblock[1]=0; ++ } ++ } ++ ++ ++ ++ ++ /* update the blocktype of the previous granule, since it depends on what ++ * happend in this granule */ ++ for (chn=0; chn<stereo; chn++) { ++ if ( uselongblock[chn]) ++ { /* no attack : use long blocks */ ++ switch( blocktype_old[chn] ) ++ { ++ case NORM_TYPE: ++ case STOP_TYPE: ++ blocktype[chn] = NORM_TYPE; ++ break; ++ case SHORT_TYPE: ++ blocktype[chn] = STOP_TYPE; ++ break; ++ case START_TYPE: ++ fprintf( stderr, "Error in block selecting\n" ); ++ abort(); ++ break; /* problem */ ++ } ++ } else { ++ /* attack : use short blocks */ ++ blocktype[chn] = SHORT_TYPE; ++ if ( blocktype_old[chn] == NORM_TYPE ) { ++ blocktype_old[chn] = START_TYPE; ++ } ++ if ( blocktype_old[chn] == STOP_TYPE ) { ++ blocktype_old[chn] = SHORT_TYPE ; ++ } ++ } ++ ++ ++ blocktype_d[chn] = blocktype_old[chn]; /* value returned to calling program */ ++ blocktype_old[chn] = blocktype[chn]; /* save for next call to l3psy_anal */ ++ } ++ ++ if (blocktype_d[0]==2) ++ *ms_ener_ratio = ms_ratio_s_old; ++ else ++ *ms_ener_ratio = ms_ratio_l_old; ++ ms_ratio_s_old = ms_ratio_s; ++ ms_ratio_l_old = ms_ratio_l; ++ ++ ++ ++ } ++ ++ ++ ++ ++ ++ ++ extern double psy_data[]; ++ + void L3para_read(double sfreq, int *numlines, int *partition_l, double *minval, double *qthr_l, double *norm_l, double (*s3_l)[63], int *partition_s, double *qthr_s, double *norm_s, double *SNR, int *cbw_l, int *bu_l, int *bo_l, double *w1_l, double *w2_l, int *cbw_s, int *bu_s, int *bo_s, double *w1_s, double *w2_s) + { +! double freq_tp; +! static double bval_l[CBANDS], bval_s[CBANDS]; +! int cbmax=0, cbmax_tp; +! static double s3_s[CBANDS][CBANDS]; +! double *p = psy_data; + +! int sbmax ; +! int i,j,k,k2,loop, part_max ; +! +! /* Read long block data */ +! +! for(loop=0;loop<6;loop++) +! { +! freq_tp = *p++; +! cbmax_tp = (int) *p++; +! cbmax_tp++; +! +! if (sfreq == freq_tp ) +! { +! cbmax = cbmax_tp; +! for(i=0,k2=0;i<cbmax_tp;i++) +! { +! j = (int) *p++; +! numlines[i] = (int) *p++; +! minval[i] = *p++; +! qthr_l[i] = *p++; +! norm_l[i] = *p++; +! bval_l[i] = *p++; +! if (j!=i) +! { +! printf("1. please check \"psy_data\""); +! exit(-1); + } +! for(k=0;k<numlines[i];k++) +! partition_l[k2++] = i ; +! } +! } +! else +! p += cbmax_tp * 6; +! } + +! /************************************************************************ +! * Now compute the spreading function, s[j][i], the value of the spread-* +! * ing function, centered at band j, for band i, store for later use * +! ************************************************************************/ +! part_max = cbmax ; +! for(i=0;i<part_max;i++) +! { +! double tempx,x,tempy,temp; +! for(j=0;j<part_max;j++) +! { +! tempx = (bval_l[i] - bval_l[j])*1.05; +! if (j>=i) tempx = (bval_l[i] - bval_l[j])*3.0; +! else tempx = (bval_l[i] - bval_l[j])*1.5; +! /* if (j>=i) tempx = (bval_l[j] - bval_l[i])*3.0; +! else tempx = (bval_l[j] - bval_l[i])*1.5; */ +! if(tempx>=0.5 && tempx<=2.5) + { +! temp = tempx - 0.5; +! x = 8.0 * (temp*temp - 2.0 * temp); +! } +! else x = 0.0; +! tempx += 0.474; +! tempy = 15.811389 + 7.5*tempx - 17.5*sqrt(1.0+tempx*tempx); +! if (tempy <= -60.0) s3_l[i][j] = 0.0; +! else s3_l[i][j] = exp( (x + tempy)*LN_TO_LOG10 ); +! } +! } + +! /* Read short block data */ + ++ for(loop=0;loop<6;loop++) ++ { ++ freq_tp = *p++; ++ cbmax_tp = (int) *p++; ++ cbmax_tp++; + +! if (sfreq == freq_tp ) +! { +! cbmax = cbmax_tp; +! for(i=0,k2=0;i<cbmax_tp;i++) +! { +! j = (int) *p++; +! numlines[i] = (int) *p++; +! qthr_s[i] = *p++; +! norm_s[i] = *p++; +! SNR[i] = *p++; +! bval_s[i] = *p++; +! if (j!=i) +! { +! printf("3. please check \"psy_data\""); +! exit(-1); +! } +! for(k=0;k<numlines[i];k++) +! partition_s[k2++] = i ; +! } +! } +! else +! p += cbmax_tp * 6; +! } + +! /************************************************************************ +! * Now compute the spreading function, s[j][i], the value of the spread-* +! * ing function, centered at band j, for band i, store for later use * +! ************************************************************************/ +! part_max = cbmax ; +! for(i=0;i<part_max;i++) +! { +! double tempx,x,tempy,temp; +! for(j=0;j<part_max;j++) +! { +! tempx = (bval_s[i] - bval_s[j])*1.05; +! if (j>=i) tempx = (bval_s[i] - bval_s[j])*3.0; +! else tempx = (bval_s[i] - bval_s[j])*1.5; +! if(tempx>=0.5 && tempx<=2.5) +! { +! temp = tempx - 0.5; +! x = 8.0 * (temp*temp - 2.0 * temp); +! } +! else x = 0.0; +! tempx += 0.474; +! tempy = 15.811389 + 7.5*tempx - 17.5*sqrt(1.0+tempx*tempx); +! if (tempy <= -60.0) s3_s[i][j] = 0.0; +! else s3_s[i][j] = exp( (x + tempy)*LN_TO_LOG10 ); +! } +! } +! /* Read long block data for converting threshold calculation +! partitions to scale factor bands */ + +! for(loop=0;loop<6;loop++) +! { +! freq_tp = *p++; +! sbmax = (int) *p++; +! sbmax++; +! +! if (sfreq == freq_tp) +! { +! for(i=0;i<sbmax;i++) +! { +! j = (int) *p++; +! cbw_l[i] = (int) *p++; +! bu_l[i] = (int) *p++; +! bo_l[i] = (int) *p++; +! w1_l[i] = (double) *p++; +! w2_l[i] = (double) *p++; +! if (j!=i) +! { printf("30:please check \"psy_data\"\n"); +! exit(-1); + } + +! if (i!=0) +! if ( (bo_l[i] != (bu_l[i]+cbw_l[i])) || +! (fabs(1.0-w1_l[i]-w2_l[i-1]) > 0.01 ) ) +! { +! printf("31l: please check \"psy_data.\"\n"); +! exit(-1); +! } +! } +! } +! else +! p += sbmax * 6; +! } +! +! /* Read short block data for converting threshold calculation +! partitions to scale factor bands */ +! +! for(loop=0;loop<6;loop++) +! { +! freq_tp = *p++; +! sbmax = (int) *p++; +! sbmax++; +! +! if (sfreq == freq_tp) +! { +! for(i=0;i<sbmax;i++) + { +! j = (int) *p++; +! cbw_s[i] = (int) *p++; +! bu_s[i] = (int) *p++; +! bo_s[i] = (int) *p++; +! w1_s[i] = *p++; +! w2_s[i] = *p++; +! if (j!=i) +! { printf("30:please check \"psy_data\"\n"); +! exit(-1); +! } + +! if (i!=0) +! if ( (bo_s[i] != (bu_s[i]+cbw_s[i])) || +! (fabs(1.0-w1_s[i]-w2_s[i-1]) > 0.01 ) ) +! { printf("31s: please check \"psy_data.\"\n"); +! exit(-1); +! } +! } +! } +! else +! p += sbmax * 6; +! } + +! } +! void sprdngf1(dest,source) +! FLOAT *dest; +! double *source; +! { +! int b,k; +! extern double s3_l[CBANDS][CBANDS]; /* s3_l is a sparse matrix. fudge with it for 44.1 khz */ +! static int s3ind[CBANDS][2] = { +! {0,2}, +! {0,3}, +! {0,4}, +! {0,5}, +! {0,6}, +! {0,7}, +! {0,8}, +! {0,9}, +! {0,10}, +! {0,11}, +! {0,12}, +! {1,14}, +! {1,14}, +! {2,15}, +! {3,15}, +! {5,16}, +! {6,17}, +! {7,19}, +! {9,20}, +! {10,21}, +! {11,22}, +! {12,23}, +! {14,24}, +! {15,25}, +! {15,27}, +! {16,28}, +! {16,28}, +! {17,29}, +! {18,30}, +! {19,31}, +! {19,32}, +! {20,34}, +! {21,35}, +! {22,36}, +! {22,36}, +! {23,37}, +! {24,38}, +! {25,39}, +! {26,41}, +! {27,42}, +! {28,43}, +! {29,44}, +! {30,45}, +! {31,46}, +! {32,47}, +! {33,48}, +! {34,49}, +! {35,50}, +! {36,51}, +! {37,52}, +! {37,53}, +! {38,54}, +! {39,55}, +! {40,56}, +! {41,57}, +! {42,58}, +! {43,59}, +! {44,60}, +! {45,61}, +! {46,62}, +! {47,62}, +! {48,62}, +! {48,62}, +! }; + + +! for ( b = 0;b < CBANDS; b++ ) +! for ( k = s3ind[b][0]; k <= s3ind[b][1]; k++ ) +! dest[b] += s3_l[b][k] * source[k]; +! } +! +! +! void sprdngf2(dest,source) +! double *dest; +! FLOAT *source; +! { +! int b,k; +! extern double s3_l[CBANDS][CBANDS]; /* s3_l is a sparse matrix. fudge with it for 44.1 khz */ +! static int s3ind[CBANDS][2] = { +! {0,2}, +! {0,3}, +! {0,4}, +! {0,5}, +! {0,6}, +! {0,7}, +! {0,8}, +! {0,9}, +! {0,10}, +! {0,11}, +! {0,12}, +! {1,14}, +! {1,14}, +! {2,15}, +! {3,15}, +! {5,16}, +! {6,17}, +! {7,19}, +! {9,20}, +! {10,21}, +! {11,22}, +! {12,23}, +! {14,24}, +! {15,25}, +! {15,27}, +! {16,28}, +! {16,28}, +! {17,29}, +! {18,30}, +! {19,31}, +! {19,32}, +! {20,34}, +! {21,35}, +! {22,36}, +! {22,36}, +! {23,37}, +! {24,38}, +! {25,39}, +! {26,41}, +! {27,42}, +! {28,43}, +! {29,44}, +! {30,45}, +! {31,46}, +! {32,47}, +! {33,48}, +! {34,49}, +! {35,50}, +! {36,51}, +! {37,52}, +! {37,53}, +! {38,54}, +! {39,55}, +! {40,56}, +! {41,57}, +! {42,58}, +! {43,59}, +! {44,60}, +! {45,61}, +! {46,62}, +! {47,62}, +! {48,62}, +! {48,62}, +! }; + + ++ for ( b = 0;b < CBANDS; b++ ) ++ for ( k = s3ind[b][0]; k <= s3ind[b][1]; k++ ) ++ dest[b] += s3_l[b][k] * source[k]; + } +diff -r -c -N encoder/l3psy.h lame3.13/l3psy.h +*** encoder/l3psy.h Wed Jan 22 02:43:16 1997 +--- lame3.13/l3psy.h Wed Jun 16 17:28:32 1999 +*************** +*** 30,37 **** + + /* l3psy.c */ + #include "l3side.h" +! void L3psycho_anal( short int *buffer, short int savebuf[1344], int chn, int lay, FLOAT snr32[32], +! double sfreq, double ratio_d[21], double ratio_ds[12][3], +! double *pe, gr_info *cod_info ); + /* void read_absthr(float *absthr, int table); */ + #endif +--- 30,41 ---- + + /* l3psy.c */ + #include "l3side.h" +! void L3psycho_anal( short int *buffer[2], int chn, +! int gr , layer *info, +! double sfreq, +! int check_ms, double *ms_ener_ratio, +! double ratio_d[2][21], double ratio_ds[2][12][3], +! double pe[2], +! int blocktype_d[2]); + /* void read_absthr(float *absthr, int table); */ + #endif +diff -r -c -N encoder/l3side.h lame3.13/l3side.h +*** encoder/l3side.h Wed Jan 22 02:43:16 1997 +--- lame3.13/l3side.h Sat Jun 5 16:44:56 1999 +*************** +*** 49,63 **** + } III_psy_xmin; + + typedef struct { +- double xr[576]; +- double xr_s[3][192]; +- double xmin[21]; +- double xmin_s[3][192]; +- int ix[576]; +- int ix_s[3][192]; +- } III_input3; /* ch */ +- +- typedef struct { + unsigned part2_3_length; + unsigned big_values; + unsigned count1; +--- 49,54 ---- +diff -r -c -N encoder/lame.dsp lame3.13/lame.dsp +*** encoder/lame.dsp Wed Dec 31 17:00:00 1969 +--- lame3.13/lame.dsp Sat Jun 12 23:01:40 1999 +*************** +*** 0 **** +--- 1,304 ---- ++ # Microsoft Developer Studio Project File - Name="lame" - Package Owner=<4> ++ # Microsoft Developer Studio Generated Build File, Format Version 5.00 ++ # ** DO NOT EDIT ** ++ ++ # TARGTYPE "Win32 (x86) Console Application" 0x0103 ++ ++ CFG=lame - Win32 Debug GTK ++ !MESSAGE This is not a valid makefile. To build this project using NMAKE, ++ !MESSAGE use the Export Makefile command and run ++ !MESSAGE ++ !MESSAGE NMAKE /f "lame.mak". ++ !MESSAGE ++ !MESSAGE You can specify a configuration when running NMAKE ++ !MESSAGE by defining the macro CFG on the command line. For example: ++ !MESSAGE ++ !MESSAGE NMAKE /f "lame.mak" CFG="lame - Win32 Debug GTK" ++ !MESSAGE ++ !MESSAGE Possible choices for configuration are: ++ !MESSAGE ++ !MESSAGE "lame - Win32 Release" (based on "Win32 (x86) Console Application") ++ !MESSAGE "lame - Win32 Debug" (based on "Win32 (x86) Console Application") ++ !MESSAGE "lame - Win32 Debug GTK" (based on "Win32 (x86) Console Application") ++ !MESSAGE "lame - Win32 Release GTK" (based on\ ++ "Win32 (x86) Console Application") ++ !MESSAGE ++ ++ # Begin Project ++ # PROP Scc_ProjName "" ++ # PROP Scc_LocalPath "" ++ CPP=xicl.exe ++ RSC=rc.exe ++ ++ !IF "$(CFG)" == "lame - Win32 Release" ++ ++ # PROP BASE Use_MFC 0 ++ # PROP BASE Use_Debug_Libraries 0 ++ # PROP BASE Output_Dir "Release" ++ # PROP BASE Intermediate_Dir "Release" ++ # PROP BASE Target_Dir "" ++ # PROP Use_MFC 0 ++ # PROP Use_Debug_Libraries 0 ++ # PROP Output_Dir "Release" ++ # PROP Intermediate_Dir "Release" ++ # PROP Ignore_Export_Lib 0 ++ # PROP Target_Dir "" ++ # ADD BASE CPP /nologo /W3 /GX /O2 /D "WIN32" /D "NDEBUG" /D "_CONSOLE" /D "_MBCS" /YX /FD /c ++ # ADD CPP /nologo /W3 /GX /O2 /I "./WinGTK/gtk-plus" /I "./WinGTK/glib-1.2" /D "NDEBUG" /D "WIN32" /D "_CONSOLE" /D "_MBCS" /D BS_FORMAT=BINARY /YX /FD /c ++ # ADD BASE RSC /l 0x409 /d "NDEBUG" ++ # ADD RSC /l 0x409 /d "NDEBUG" ++ BSC32=bscmake.exe ++ # ADD BASE BSC32 /nologo ++ # ADD BSC32 /nologo ++ LINK32=xilink.exe ++ # ADD BASE LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:console /machine:I386 ++ # ADD LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:console /profile /map /machine:I386 ++ ++ !ELSEIF "$(CFG)" == "lame - Win32 Debug" ++ ++ # PROP BASE Use_MFC 0 ++ # PROP BASE Use_Debug_Libraries 1 ++ # PROP BASE Output_Dir "Debug" ++ # PROP BASE Intermediate_Dir "Debug" ++ # PROP BASE Target_Dir "" ++ # PROP Use_MFC 0 ++ # PROP Use_Debug_Libraries 1 ++ # PROP Output_Dir "Debug" ++ # PROP Intermediate_Dir "Debug" ++ # PROP Ignore_Export_Lib 0 ++ # PROP Target_Dir "" ++ # ADD BASE CPP /nologo /W3 /Gm /GX /Zi /Od /D "WIN32" /D "_DEBUG" /D "_CONSOLE" /D "_MBCS" /YX /FD /c ++ # ADD CPP /nologo /Zp4 /W3 /Gm /GX /Zi /Od /D "_DEBUG" /D "WIN32" /D "_CONSOLE" /D "_MBCS" /D BS_FORMAT=BINARY /YX /FD /c ++ # ADD BASE RSC /l 0x409 /d "_DEBUG" ++ # ADD RSC /l 0x409 /d "_DEBUG" ++ BSC32=bscmake.exe ++ # ADD BASE BSC32 /nologo ++ # ADD BSC32 /nologo ++ LINK32=xilink.exe ++ # ADD BASE LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:console /debug /machine:I386 /pdbtype:sept ++ # ADD LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:console /debug /machine:I386 /pdbtype:sept ++ ++ !ELSEIF "$(CFG)" == "lame - Win32 Debug GTK" ++ ++ # PROP BASE Use_MFC 0 ++ # PROP BASE Use_Debug_Libraries 1 ++ # PROP BASE Output_Dir "lame___W" ++ # PROP BASE Intermediate_Dir "lame___W" ++ # PROP BASE Ignore_Export_Lib 0 ++ # PROP BASE Target_Dir "" ++ # PROP Use_MFC 0 ++ # PROP Use_Debug_Libraries 1 ++ # PROP Output_Dir "DebugGTK" ++ # PROP Intermediate_Dir "DebugGTK" ++ # PROP Ignore_Export_Lib 0 ++ # PROP Target_Dir "" ++ # ADD BASE CPP /nologo /Zp4 /W3 /Gm /GX /Zi /Od /I "../WinGTK/gtk-plus" /I "../WinGTK/glib-1.2" /D "_DEBUG" /D "WIN32" /D "_CONSOLE" /D "_MBCS" /D BS_FORMAT=BINARY /D "HAVEGTK" /YX /FD /c ++ # ADD CPP /nologo /Zp4 /W3 /Gm /GX /Zi /Od /I "./WinGTK/gtk-plus" /I "./WinGTK/glib-1.2" /D "_DEBUG" /D "WIN32" /D "_CONSOLE" /D "_MBCS" /D BS_FORMAT=BINARY /D "HAVEGTK" /YX /FD /c ++ # ADD BASE RSC /l 0x409 /d "_DEBUG" ++ # ADD RSC /l 0x409 /d "_DEBUG" ++ BSC32=bscmake.exe ++ # ADD BASE BSC32 /nologo ++ # ADD BSC32 /nologo ++ LINK32=xilink.exe ++ # ADD BASE LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:console /debug /machine:I386 /pdbtype:sept ++ # ADD LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib glib-1.2.lib gdk-1.3.lib gtk-1.3.lib /nologo /subsystem:console /debug /machine:I386 /pdbtype:sept /libpath:"./WinGTK/glib-1.2" /libpath:"./WinGTK/gtk-plus/gdk" /libpath:"./WinGTK/gtk-plus/gtk" ++ ++ !ELSEIF "$(CFG)" == "lame - Win32 Release GTK" ++ ++ # PROP BASE Use_MFC 0 ++ # PROP BASE Use_Debug_Libraries 0 ++ # PROP BASE Output_Dir "lame___0" ++ # PROP BASE Intermediate_Dir "lame___0" ++ # PROP BASE Ignore_Export_Lib 0 ++ # PROP BASE Target_Dir "" ++ # PROP Use_MFC 0 ++ # PROP Use_Debug_Libraries 0 ++ # PROP Output_Dir "ReleaseGTK" ++ # PROP Intermediate_Dir "ReleaseGTK" ++ # PROP Ignore_Export_Lib 0 ++ # PROP Target_Dir "" ++ # ADD BASE CPP /nologo /W3 /GX /O2 /I "../WinGTK/gtk-plus" /I "../WinGTK/glib-1.2" /D "NDEBUG" /D "WIN32" /D "_CONSOLE" /D "_MBCS" /D BS_FORMAT=BINARY /D "HAVEGTK" /YX /FD /c ++ # ADD CPP /nologo /W3 /GX /O2 /I "./WinGTK/gtk-plus" /I "./WinGTK/glib-1.2" /D "NDEBUG" /D "WIN32" /D "_CONSOLE" /D "_MBCS" /D BS_FORMAT=BINARY /D "HAVEGTK" /YX /FD /c ++ # ADD BASE RSC /l 0x409 /d "NDEBUG" ++ # ADD RSC /l 0x409 /d "NDEBUG" ++ BSC32=bscmake.exe ++ # ADD BASE BSC32 /nologo ++ # ADD BSC32 /nologo ++ LINK32=xilink.exe ++ # ADD BASE LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:console /profile /map /machine:I386 ++ # ADD LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib glib-1.2.lib gdk-1.3.lib gtk-1.3.lib /nologo /subsystem:console /profile /map /machine:I386 /libpath:"./WinGTK/glib-1.2" /libpath:"./WinGTK/gtk-plus/gdk" /libpath:"./WinGTK/gtk-plus/gtk" ++ ++ !ENDIF ++ ++ # Begin Target ++ ++ # Name "lame - Win32 Release" ++ # Name "lame - Win32 Debug" ++ # Name "lame - Win32 Debug GTK" ++ # Name "lame - Win32 Release GTK" ++ # Begin Source File ++ ++ SOURCE=.\common.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\common.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\encode.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\encoder.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\formatBitstream.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\formatBitstream.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\gpkplotting.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\gpkplotting.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\gtkanal.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\gtkanal.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\huffman.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\huffman.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\ieeefloat.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\ieeefloat.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=".\l3bitstream-pvt.h" ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\l3bitstream.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\l3bitstream.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\l3psy.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\l3psy.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\l3side.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=".\loop-pvt.h" ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\loop.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\loop.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\mdct.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\mdct.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\musicin.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\portableio.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\portableio.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\profile.txt ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\psy.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\reservoir.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\reservoir.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\sqrttab.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\subs.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\subs.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\tables.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\VbrTag.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\VbrTag.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\version.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\version.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\WinGTK.txt ++ # End Source File ++ # End Target ++ # End Project +diff -r -c -N encoder/lame.dsw lame3.13/lame.dsw +*** encoder/lame.dsw Wed Dec 31 17:00:00 1969 +--- lame3.13/lame.dsw Sun Jun 6 16:19:25 1999 +*************** +*** 0 **** +--- 1,44 ---- ++ Microsoft Developer Studio Workspace File, Format Version 5.00 ++ # WARNING: DO NOT EDIT OR DELETE THIS WORKSPACE FILE! ++ ++ ############################################################################### ++ ++ Project: "lame"=.\lame.dsp - Package Owner=<4> ++ ++ Package=<5> ++ {{{ ++ }}} ++ ++ Package=<4> ++ {{{ ++ Begin Project Dependency ++ Project_Dep_Name mpglib ++ End Project Dependency ++ }}} ++ ++ ############################################################################### ++ ++ Project: "mpglib"=.\mpglib\mpglib.dsp - Package Owner=<4> ++ ++ Package=<5> ++ {{{ ++ }}} ++ ++ Package=<4> ++ {{{ ++ }}} ++ ++ ############################################################################### ++ ++ Global: ++ ++ Package=<5> ++ {{{ ++ }}} ++ ++ Package=<3> ++ {{{ ++ }}} ++ ++ ############################################################################### ++ +diff -r -c -N encoder/loop-pvt.h lame3.13/loop-pvt.h +*** encoder/loop-pvt.h Wed Jan 22 02:43:16 1997 +--- lame3.13/loop-pvt.h Sat Jun 19 16:26:39 1999 +*************** +*** 25,39 **** + + */ + +! int outer_loop( double xr[2][2][576], /*vector of the magnitudees of the spectral values */ +! int max_bits, + III_psy_xmin *l3_xmin, /* the allowed distortion of the scalefactor */ + int l3_enc[2][2][576], /* vector of quantized values ix(0..575) */ + frame_params *fr_ps, + III_scalefac_t *scalefac, /* scalefactors */ + int gr, + int ch, +! III_side_info_t *l3_side ); + + int part2_length( III_scalefac_t *scalefac, + frame_params *fr_ps, +--- 25,59 ---- + + */ + +! void outer_loop( double xr[2][2][576], /*vector of the magnitudees of the spectral values */ +! double xr_org[2][2][576], +! int mean_bits, +! int VBRbits[2][2], +! int bit_rate, +! int best_over[2], +! III_psy_xmin *l3_xmin, /* the allowed distortion of the scalefactor */ +! int l3_enc[2][2][576], /* vector of quantized values ix(0..575) */ +! frame_params *fr_ps, +! III_scalefac_t *scalefac, /* scalefactors */ +! int gr, +! int ch, +! III_side_info_t *l3_side, +! III_psy_ratio *ratio, +! double pe[2][2], +! double ms_ratio[2]); +! +! void outer_loop_old( double xr[2][2][576], /*vector of the magnitudees of the spectral values */ +! double xr_org[2][2][576], +! int mean_bits, + III_psy_xmin *l3_xmin, /* the allowed distortion of the scalefactor */ + int l3_enc[2][2][576], /* vector of quantized values ix(0..575) */ + frame_params *fr_ps, + III_scalefac_t *scalefac, /* scalefactors */ + int gr, + int ch, +! III_side_info_t *l3_side, +! III_psy_ratio *ratio, double pe[2][2]); +! + + int part2_length( III_scalefac_t *scalefac, + frame_params *fr_ps, +*************** +*** 43,49 **** + + int quantanf_init( double xr[576] ); + +! int inner_loop( double xr[2][2][576], + int l3_enc[2][2][576], + int max_bits, + gr_info *cod_info, +--- 63,69 ---- + + int quantanf_init( double xr[576] ); + +! int inner_loop( double xr[2][2][576], double xrpow[2][2][576], + int l3_enc[2][2][576], + int max_bits, + gr_info *cod_info, +*************** +*** 80,89 **** + gr_info *cod_info, + int gr, + int ch ); + void calc_noise( double xr[576], + int ix[576], + gr_info *cod_info, +! double xfsf[4][CBLIMIT] ); + + + int loop_break( III_scalefac_t *scalefac, +--- 100,126 ---- + gr_info *cod_info, + int gr, + int ch ); ++ int scale_bitcount_lsf( III_scalefac_t *scalefac, gr_info *cod_info, ++ int gr, int ch ); + void calc_noise( double xr[576], + int ix[576], + gr_info *cod_info, +! double xfsf[4][CBLIMIT]); +! int calc_noise1( double xr[576], +! int ix[576], +! gr_info *cod_info, +! double xfsf[4][CBLIMIT], +! int distort[4][CBLIMIT], +! III_psy_xmin *l3_xmin, +! int gr, int ch, double noise); +! +! void calc_noise2( double xr[2][576], +! int ix[2][576], +! gr_info *cod_info[2], +! double xfsf[2][4][CBLIMIT], +! int distort[2][4][CBLIMIT], +! III_psy_xmin *l3_xmin, +! int gr, int ch, int over[2], double noise[2]); + + + int loop_break( III_scalefac_t *scalefac, +*************** +*** 96,101 **** +--- 133,144 ---- + int gr, + int ch, + III_side_info_t *l3_side ); ++ int preemphasis2( double xr[576], double xrpow[576], ++ III_psy_xmin *l3_xmin, ++ int gr, ++ int ch, ++ III_side_info_t *l3_side, ++ int distort[4][CBLIMIT] ); + int amp_scalefac_bands( double xr[576], + double xfsf[4][CBLIMIT], + III_psy_xmin *l3_xmin, +*************** +*** 103,110 **** + III_scalefac_t *scalefac, + int gr, + int ch, +! int iteration ); +! void quantize( double xr[576], + int ix[576], + gr_info *cod_info ); + int ix_max( int ix[576], +--- 146,164 ---- + III_scalefac_t *scalefac, + int gr, + int ch, +! int iteration); +! int amp_scalefac_bands2( double xr[576], double xrpow[576], +! III_psy_xmin *l3_xmin, +! III_side_info_t *l3_side, +! III_scalefac_t *scalefac, +! int gr, +! int ch, +! int iteration, +! int distort[4][CBLIMIT]); +! int quantize( double xr[576], +! int ix[576], +! gr_info *cod_info ); +! void quantize_xrpow( double xr[576], + int ix[576], + gr_info *cod_info ); + int ix_max( int ix[576], +*************** +*** 115,119 **** + int + new_choose_table( int ix[576], + unsigned int begin, +! unsigned int end ); + #endif +--- 169,182 ---- + int + new_choose_table( int ix[576], + unsigned int begin, +! unsigned int end, int * s ); +! +! /* New SS 20-12-96 */ +! int bin_search_StepSize(int desired_rate, double start, int bot, int ix[576], +! double xrs[576], double xrspow[576], gr_info * cod_info); +! int bin_search_StepSize2(int desired_rate, double start, int bot, int ix[576], +! double xrs[576], double xrspow[576], gr_info * cod_info); +! int count_bits(); +! +! + #endif +diff -r -c -N encoder/loop.c lame3.13/loop.c +*** encoder/loop.c Wed Jan 22 02:43:16 1997 +--- lame3.13/loop.c Thu Jun 24 18:28:31 1999 +*************** +*** 2,8 **** + * ISO MPEG Audio Subgroup Software Simulation Group (1996) + * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension + * +! * $Id: loop.c,v 1.2 1997/01/19 22:28:29 rowlands Exp $ + * + * $Log: loop.c,v $ + * Revision 1.2 1997/01/19 22:28:29 rowlands +--- 2,8 ---- + * ISO MPEG Audio Subgroup Software Simulation Group (1996) + * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension + * +! * $Id: loop.c,v 1.2 1997/01/19 22:28:29 rowlands Exp $ + * + * $Log: loop.c,v $ + * Revision 1.2 1997/01/19 22:28:29 rowlands +*************** +*** 13,65 **** + * + * Received from Mike Coleman + **********************************************************************/ +- /********************************************************************** +- * date programmers comment * +- *25. 6.92 Toshiyuki Ishino Ver 1.0 * +- *29.10.92 Masahiro Iwadare Ver 2.0 * +- *17. 4.93 Masahiro Iwadare Updated for IS Modification * +- *04.11.93 Seymour Shlien Speed up inner loop * +- *09.09.95 mc@fivebats.com Several changes for updated IS,* +- * and some MPEG2-LSF support * +- *20.12.96 seymour.shlien@crc.doc.ca Fixed some bugs and improved * +- * the appearance * +- *********************************************************************/ + + #include <stdio.h> + #include <stdlib.h> + #include <math.h> + #include <assert.h> + #include "l3side.h" + #include "loop.h" + #include "huffman.h" + #include "l3bitstream.h" + #include "reservoir.h" + #include "loop-pvt.h" + + /* #define DEBUG */ + /* #define DEBUGSC */ +! /* #define PERFORM 3 */ +! /* If PERFORM is defined to some number, then a file encode.log +! is preduced showing the intermediate results produced by the +! outer_loop code for the frame number = PERFORM. + */ +- #define BIN_SEARCH +- /* for speeding up the iteration_loop algorithm */ + +- #ifdef PERFORM +- FILE *log_output; +- extern int frameNum; +- float worst_xfsf_to_xmin_ratio(III_psy_xmin *l3_xmin, double xfsf[4][CBLIMIT] +- ,int block_type,int gr,int ch); +- #endif + +- /* New SS 20-12-96 */ +- #ifdef BIN_SEARCH +- int bin_search_StepSize(int desired_rate, double start, int ix[576], +- double xrs[576], gr_info * cod_info); +- int count_bits(); +- float worst_xfsf_to_xmin_ratio(); +- #endif + + + /* +--- 13,66 ---- + * + * Received from Mike Coleman + **********************************************************************/ + + #include <stdio.h> + #include <stdlib.h> + #include <math.h> + #include <assert.h> ++ #include "globalflags.h" ++ #include "common.h" + #include "l3side.h" + #include "loop.h" + #include "huffman.h" + #include "l3bitstream.h" + #include "reservoir.h" + #include "loop-pvt.h" ++ #include "gtkanal.h" ++ + + /* #define DEBUG */ + /* #define DEBUGSC */ +! +! +! static int convert_mdct, convert_psy, reduce_sidechannel; +! /* +! mt 5/99. These global flags denote 4 possibilities: +! +! 1 MDCT input L/R, quantize L/R, psy-model thresholds: L/R -m s +! 2 MDCT input L/R, quantize M/S, psy-model thresholds: L/R -m j +! 3 MDCT input M/S, quantize M/S, psy-model thresholds: M/S -m f +! 4 MDCT input L/R, quantize M/S, psy-model thresholds: M/S coming soon +! +! 1: convert_mdct = 0, convert_psy=0, reduce_sidechannel=0 +! 2: convert_mdct = 1, convert_psy=1, reduce_sidechannel=1 +! 3: convert_mdct = 0, convert_psy=0, reduce_sidechannel=1 +! 4: convert_mdct = 1, convert_psy=0, reduce_sidechannel=1 +! +! if (convert_mdct), then iteration_loop will quantize M/S data from +! the L/R input MDCT coefficients. +! +! if (convert_psy), then calc_noise will compute the noise for the L/R +! channels from M/S MDCT data and L/R psy-model threshold information. +! Distortion in ether L or R channel will be marked as distortion in +! both Mid and Side channels. +! +! if (reduce_sidechannel) then outer_loop will allocate less bits +! to the side channel and more bits to the mid channel based on relative +! energies. + */ + + + + + /* +*************** +*** 82,88 **** + {0,4,8,12,18,26,36,48,62,80,104,136,180,192} + }, + { /* Table B.2.a: 16 kHz */ +! {0,6,12,18,24,30,36,44,45,66,80,96,116,140,168,200,238,248,336,396,464,522,576}, + {0,4,8,12,18,26,36,48,62,80,104,134,174,192} + }, + { /* Table B.8.b: 44.1 kHz */ +--- 83,89 ---- + {0,4,8,12,18,26,36,48,62,80,104,136,180,192} + }, + { /* Table B.2.a: 16 kHz */ +! {0,6,12,18,24,30,36,44,54,66,80,96,116,140,168,200,238,248,336,396,464,522,576}, + {0,4,8,12,18,26,36,48,62,80,104,134,174,192} + }, + { /* Table B.8.b: 44.1 kHz */ +*************** +*** 169,665 **** + + + + /************************************************************************/ + /* iteration_loop() */ + /************************************************************************/ + void +! iteration_loop( double pe[][2], double xr_org[2][2][576], III_psy_ratio *ratio, + III_side_info_t *l3_side, int l3_enc[2][2][576], +! int mean_bits, int stereo, double xr_dec[2][2][576], +! III_scalefac_t *scalefac, frame_params *fr_ps, +! int ancillary_pad, int bitsPerFrame ) +! { +! static int firstcall = 1; +! III_psy_xmin l3_xmin; +! gr_info *cod_info; +! layer *info; +! int *main_data_begin; +! +! int max_bits; +! int ch, gr, sfb, i, mode_gr; +! static int huffman_read_flag = 0; +! +! double xr[2][2][576]; +! I576 *ix; +! +! main_data_begin = &l3_side->main_data_begin; +! l3_side->resvDrain = 0; +! +! if ( firstcall ) +! { +! *main_data_begin = 0; +! firstcall = 0; +! #ifdef PERFORM +! log_output = fopen("encode.log","w"); +! #endif +! } + +! info = fr_ps->header; +! mode_gr = (info->version == 1) ? 2 : 1; + +! scalefac_band_long = &sfBandIndex[info->sampling_frequency + (info->version * 3)].l[0]; +! scalefac_band_short = &sfBandIndex[info->sampling_frequency + (info->version * 3)].s[0]; + +! /* reading huffman code table */ +! if (huffman_read_flag == 0) { +! FILE *fi = OpenTableFile( "huffcode" ); +! if ( fi == NULL ) +! exit( EXIT_FAILURE ); +! read_huffcodetab( fi ); +! huffman_read_flag++; +! fclose( fi ); + } + + +! for ( gr = 0; gr < mode_gr; gr++ ) +! { +! for ( ch = 0; ch < stereo; ch++ ) +! { +! for ( i = 0; i < 576; i++ ) +! xr[gr][ch][i] = xr_org[gr][ch][i]; +! } +! } + + ResvFrameBegin( fr_ps, l3_side, mean_bits, bitsPerFrame ); + +- for ( gr = 0; gr < mode_gr; gr++ ) +- { +- for ( ch = 0; ch < stereo; ch++ ) +- { +- ix = (I576 *) l3_enc[gr][ch]; +- cod_info = (gr_info *) &(l3_side->gr[gr].ch[ch]); +- gr_deco(cod_info); +- calc_xmin( xr, ratio, cod_info, &l3_xmin, gr, ch ); +- +- #ifdef DEBUG +- printf( "----- start gr[%d] ch[%1d] : block_type=%1d, window_switching_flag=%1d (loop)\n", +- gr,ch, cod_info->block_type, cod_info->window_switching_flag ); +- #endif +- +- if ( info->version == 1 ) +- calc_scfsi( xr[gr][ch], l3_side, &l3_xmin, ch, gr ); +- +- /* calculation of number of available bit( per granule ) */ +- max_bits = ResvMaxBits( fr_ps, l3_side, &pe[gr][ch], mean_bits ); +- #ifdef DEBUG +- printf( " max_bits = %d, mean_bits = %d (iteration_loop)\n", max_bits, mean_bits ); +- #endif +- +- /* reset of iteration variables */ +- +- for ( sfb = 0; sfb < 21; sfb++ ) +- scalefac->l[gr][ch][sfb] = 0; +- for ( sfb = 0; sfb < 13; sfb++ ) +- for ( i = 0; i < 3; i++ ) +- scalefac->s[gr][ch][sfb][i] = 0; +- +- for ( i = 0; i < 4; i++ ) +- cod_info->slen[i] = 0; +- cod_info->sfb_partition_table = &nr_of_sfb_block[0][0][0]; +- +- cod_info->part2_3_length = 0; +- cod_info->big_values = 0; +- cod_info->count1 = 0; +- cod_info->scalefac_compress = 0; +- cod_info->table_select[0] = 0; +- cod_info->table_select[1] = 0; +- cod_info->table_select[2] = 0; +- cod_info->subblock_gain[0] = 0; +- cod_info->subblock_gain[1] = 0; +- cod_info->subblock_gain[2] = 0; +- cod_info->region0_count = 0; +- cod_info->region1_count = 0; +- cod_info->part2_length = 0; +- cod_info->preflag = 0; +- cod_info->scalefac_scale = 0; +- cod_info->quantizerStepSize = 0.0; +- cod_info->count1table_select= 0; +- +- /* all spectral values zero ? */ +- if ( fabs(xr_max(xr[gr][ch], 0, 576)) != 0.0 ) +- { +- cod_info->quantizerStepSize = +- (double) quantanf_init( xr[gr][ch] ); +- cod_info->part2_3_length = outer_loop( xr, max_bits, &l3_xmin, +- l3_enc, fr_ps, scalefac, +- gr, ch, l3_side ); +- } +- ResvAdjust( fr_ps, cod_info, l3_side, mean_bits ); + +! cod_info->global_gain = nint( cod_info->quantizerStepSize + 210.0 ); +! assert( cod_info->global_gain < 256 ); +! } /* for ch */ +! } /* for gr */ +! ResvFrameEnd( fr_ps, l3_side, mean_bits ); + } + + + + /************************************************************************/ +! /* quantanf_init */ + /************************************************************************/ +! int quantanf_init( double xr[576] ) +! /* Function: Calculate the first quantization step quantanf. */ + { +! int i, tp = 0; +! double system_const, minlimit; +! double sfm = 0.0, sum1 = 0.0, sum2 = 0.0; + +! system_const = 8.0; +! minlimit = -100.0; + +! for ( i = 0; i < 576; i++ ) +! { +! if ( xr[i] != 0 ) +! { +! double tpd = xr[i] * xr[i]; +! sum1 += log( tpd ); +! sum2 += tpd; +! } + } +! if ( sum2 != 0.0 ) +! { +! sfm = exp( sum1 / 576.0 ) / (sum2 / 576.0); +! tp = nint( system_const * log(sfm) ); +! if ( tp < minlimit ) +! tp = minlimit; +! #ifdef DEBUG +! printf(" quantanf = %d (quantanf_init)\n",tp ); + #endif +- } +- return(tp-70.0); /* SS 19-12-96. Starting value of +- global_gain or quantizerStepSize +- has to be reduced for iteration_loop +- */ + } + + + + +! + /************************************************************************/ +! /* outer_loop */ + /************************************************************************/ + /* Function: The outer iteration loop controls the masking conditions */ + /* of all scalefactorbands. It computes the best scalefac and */ +! /* global gain. This module calls the inner iteration loop */ +! /************************************************************************/ +! int outer_loop( +! double xr[2][2][576], /* magnitudes of the spectral values */ +! int max_bits, + III_psy_xmin *l3_xmin, /* the allowed distortion of the scalefactor */ + int l3_enc[2][2][576], /* vector of quantized values ix(0..575) */ + frame_params *fr_ps, + III_scalefac_t *scalefac, /* scalefactors */ +! int gr, int ch, III_side_info_t *l3_side ) + { +! int status ; +! int scalesave_l[CBLIMIT], scalesave_s[CBLIMIT][3]; +! int sfb, bits, huff_bits, save_preflag, save_compress; +! double xfsf[4][CBLIMIT]; +! int i, over, iteration; +! float max_ratio; + + +- /* D576 *xrs; */ /* to eliminate warning messages from gcc compiler */ +- /* I576 *ix; */ /* replace this code with below. S. Shlien 15-1-97 */ + +! double *xrs; +! int *ix; +! gr_info *cod_info = &l3_side->gr[gr].ch[ch].tt; + +! /* xrs = (D576 *) &xr[gr][ch][0]; */ +! /* ix = (I576 *) l3_enc[gr][ch]; */ +! + +! xrs = (double *) &(xr[gr][ch][0]); +! ix = (int *) &(l3_enc[gr][ch][0]); + + +- iteration = 0; +- #ifdef PERFORM +- if(frameNum == PERFORM) +- fprintf(log_output,"\n\nframe = %d ch = %d gr= %d\n",frameNum,ch,gr); +- #endif +- do +- { +- iteration += 1; +- cod_info->part2_length = part2_length( scalefac, fr_ps, gr, ch, l3_side ); +- huff_bits = max_bits - cod_info->part2_length; + +! #ifdef BIN_SEARCH +! if(iteration == 1) +! { +! bin_search_StepSize(max_bits,cod_info->quantizerStepSize, +! ix,xrs,cod_info); /* speeds things up a bit */ +! } +! #endif +! #ifdef PERFORM +! if(frameNum==PERFORM) +! fprintf(log_output,"\n Interim Results %d\n\n",iteration); +! bits = test_inner_loop( xr, l3_enc, huff_bits, cod_info, gr, ch, +! xfsf,l3_xmin); +! #else +! bits = inner_loop( xr, l3_enc, huff_bits, cod_info, gr, ch ); +! #endif + +! calc_noise( &xr[gr][ch][0], &l3_enc[gr][ch][0], cod_info, xfsf ); /* distortion calculation */ + +! for ( sfb = 0; sfb < CBLIMIT; sfb++ ) /* save scaling factors */ +! scalesave_l[sfb] = scalefac->l[gr][ch][sfb]; + +- for ( sfb = 0; sfb < SFB_SMAX; sfb++ ) +- for ( i = 0; i < 3; i++ ) +- scalesave_s[sfb][i] = scalefac->s[gr][ch][sfb][i]; +- +- save_preflag = cod_info->preflag; +- save_compress = cod_info->scalefac_compress; + +! preemphasis( &xr[gr][ch][0], xfsf, l3_xmin, gr, ch, l3_side ); + + + +! #ifdef PERFORM +! if(frameNum == PERFORM) +! { +! fprintf(log_output,"\nbits = %d huff_bits= %d ", bits,huff_bits); +! fprintf(log_output," max_bits = %d\n",max_bits); +! fprintf(log_output,"Stepsize = %f ",cod_info->quantizerStepSize); +! fprintf(log_output," scale_bits = %d \n", cod_info->part2_length ); +! print_scalefacs(log_output,scalefac,cod_info->block_type,gr,ch); +! /*if (gr==0 && ch==0) +! print_quantized_values(log_output,l3_enc[gr][ch] ,cod_info);*/ +! max_ratio = worst_xfsf_to_xmin_ratio(l3_xmin,xfsf,cod_info->block_type,gr,ch); +! fprintf(log_output,"max_ratio = %6.2f\n",max_ratio ); +! print_ratios(log_output,l3_xmin,xfsf,cod_info->block_type,gr,ch); +! fprintf(log_output,"\n\n"); +! fflush(log_output); + } + #endif +! over = amp_scalefac_bands( &xr[gr][ch][0], xfsf, l3_xmin, +! l3_side, scalefac, gr, ch, iteration ); + +- #if 1 +- if ( (status = loop_break(scalefac, cod_info, gr, ch)) == 0 ) +- { +- if ( fr_ps->header->version == 1 ) +- status = scale_bitcount( scalefac, cod_info, gr, ch ); +- else +- status = scale_bitcount_lsf( scalefac, cod_info, gr, ch ); + } +! #else +! status = loop_break( scalefac, cod_info, gr, ch ); +! if ( fr_ps->header->version == 1 ) +! status += scale_bitcount( scalefac, cod_info, gr, ch ); +! else +! status += scale_bitcount_lsf( scalefac, cod_info, gr, ch ); +! #endif +! + } +- while ( (status == 0) && (over > 0) ); + +! cod_info->preflag = save_preflag; +! cod_info->scalefac_compress = save_compress; + +- for ( sfb = 0; sfb < 21; sfb++ ) +- scalefac->l[gr][ch][sfb] = scalesave_l[sfb]; + +- for ( i = 0; i < 3; i++ ) +- for ( sfb = 0; sfb < 12; sfb++ ) +- scalefac->s[gr][ch][sfb][i] = scalesave_s[sfb][i]; + +! cod_info->part2_length = part2_length( scalefac, fr_ps, gr, ch, l3_side ); +! cod_info->part2_3_length = cod_info->part2_length + bits; + +! #ifdef PERFORM +! if(frameNum == PERFORM) +! { +! fprintf(log_output,"\n Final Results\n"); +! fprintf(log_output,"bits = %d huff_bits= %d", bits,huff_bits); +! fprintf(log_output," max_bits = %d\n",max_bits); +! fprintf(log_output," Stepsize = %f ",cod_info->quantizerStepSize); +! fprintf(log_output, " scale_bits = %d \n", cod_info->part2_length ); +! max_ratio = worst_xfsf_to_xmin_ratio(l3_xmin,xfsf,cod_info->block_type,gr,ch); +! print_scalefacs(log_output,scalefac,cod_info->block_type,gr,ch); +! fprintf(log_output,"max_ratio = %6.2f\n",max_ratio ); +! print_ratios(log_output,l3_xmin,xfsf,cod_info->block_type,gr,ch); +! fflush(log_output); + } +! #endif +! return cod_info->part2_3_length; +! } + + + + +- /***************************************************************************/ +- /* inner_loop */ +- /***************************************************************************/ +- /* The code selects the best quantizerStepSize for a particular set +- /* of scalefacs */ +- +- int +- inner_loop( double xr[2][2][576], int l3_enc[2][2][576], int max_bits, +- gr_info *cod_info, int gr, int ch ) +- { +- int bits, c1bits, bvbits; +- double *xrs; /* D576 *xr; */ +- int *ix; /* I576 *ix; */ +- xrs = &xr[gr][ch][0]; +- ix = l3_enc[gr][ch]; + +- assert( max_bits >= 0 ); +- cod_info->quantizerStepSize -= 1.0;; +- do +- { +- do +- { +- cod_info->quantizerStepSize += 1.0; +- quantize( xrs, ix, cod_info ); +- } +- while ( ix_max(ix, 0, 576) > 8191 + 14 ); /* within table range? */ + +- calc_runlen( ix, cod_info ); /*rzero,count1,big_values*/ +- bits = c1bits = count1_bitcount( ix, cod_info ); /*count1_table selection*/ +- subdivide( cod_info ); /* bigvalues sfb division */ +- bigv_tab_select( ix, cod_info ); /* codebook selection*/ +- bits += bvbits = bigv_bitcount( ix, cod_info ); /* bit count */ +- +- #ifdef PERFORM +- if(frameNum == PERFORM) +- fprintf(log_output,"StepSize=%f bits = %d huff_bits = %d\n", +- cod_info->quantizerStepSize,bits,max_bits); +- #endif + +- } +- while ( bits > max_bits ); + +- return bits; + } + + + +- /***************************************************************************/ +- /* calc_scfsi */ +- /***************************************************************************/ +- /* calculation of the scalefactor select information ( scfsi ) */ + +- void calc_scfsi( double xr[576], III_side_info_t *l3_side, +- III_psy_xmin *l3_xmin, int ch, int gr ) +- { +- static int en_tot[2][2]; /* ch,gr */ +- static int en[2][2][21]; +- static int xm[2][2][21]; +- static int xrmax[2][2]; +- +- int en_tot_krit = 10; +- int en_dif_krit = 100; +- int en_scfsi_band_krit = 10; +- int xm_scfsi_band_krit = 10; +- +- int scfsi_band; +- unsigned scfsi_set; +- +- int sfb, start, end, i; +- int condition = 0; +- double temp, log2 = log( 2.0 ); +- gr_info *cod_info = &l3_side->gr[gr].ch[ch].tt; + +! xrmax[gr][ch] = xr_max( xr, 0, 576 ); +! scfsi_set = 0; + +- /* the total energy of the granule */ +- for ( temp = 0.0, i = 0; i < 576; i++ ) +- temp += xr[i] * xr[i]; +- if ( temp == 0.0 ) +- en_tot[gr][ch] = 0.0; +- else +- en_tot[gr][ch] = log( temp ) / log2 ; + +- /* the energy of each scalefactor band, en */ +- /* the allowed distortion of each scalefactor band, xm */ + +- if ( cod_info->window_switching_flag == 0 || +- cod_info->block_type != 2 ) +- for ( sfb = 0; sfb < 21; sfb++ ) +- { +- start = scalefac_band_long[ sfb ]; +- end = scalefac_band_long[ sfb+1 ]; + +- for ( temp = 0.0, i = start; i < end; i++ ) +- temp += xr[i] * xr[i]; +- if ( temp == 0.0 ) +- en[gr][ch][sfb] = 0.0; +- else +- en[gr][ch][sfb] = log( temp )/ log2; + +! if ( l3_xmin->l[gr][ch][sfb] == 0.0 ) +! xm[gr][ch][sfb] = 0.0; +! else +! xm[gr][ch][sfb] = log( l3_xmin->l[gr][ch][sfb] ) / log2; +! } +! if ( gr == 1 ) + { +! int gr2, tp; + +- for ( gr2 = 0; gr2 < 2; gr2++ ) +- { +- /* The spectral values are not all zero */ +- if ( xrmax[ch][gr2] != 0.0 ) +- condition++; +- /* None of the granules contains short blocks */ +- if ( (cod_info->window_switching_flag == 0) || +- (cod_info->block_type != 2) ) +- condition++; +- } +- if ( abs(en_tot[0] - en_tot[1]) < en_tot_krit ) +- condition++; +- for ( tp = 0, sfb = 0; sfb < 21; sfb++ ) +- tp += abs( en[ch][0][sfb] - en[ch][1][sfb] ); +- if ( tp < en_dif_krit ) +- condition++; + +- if ( condition == 6 ) +- { +- for ( scfsi_band = 0; scfsi_band < 4; scfsi_band++ ) +- { +- int sum0 = 0, sum1 = 0; +- l3_side->scfsi[ch][scfsi_band] = 0; +- start = scfsi_band_long[scfsi_band]; +- end = scfsi_band_long[scfsi_band+1]; +- for ( sfb = start; sfb < end; sfb++ ) +- { +- sum0 += abs( en[ch][0][sfb] - en[ch][1][sfb] ); +- sum1 += abs( xm[ch][0][sfb] - xm[ch][1][sfb] ); +- } + +- if ( sum0 < en_scfsi_band_krit && sum1 < xm_scfsi_band_krit ) +- { +- l3_side->scfsi[ch][scfsi_band] = 1; +- scfsi_set |= (1 << scfsi_band); +- } +- else +- l3_side->scfsi[ch][scfsi_band] = 0; +- } /* for scfsi_band */ +- #ifdef DEBUG +- fprintf( stderr, "calc_scfsi: scfsi_set = 0x%02x\n", scfsi_set ); +- #endif +- } /* if condition == 6 */ +- else +- for ( scfsi_band = 0; scfsi_band < 4; scfsi_band++ ) +- l3_side->scfsi[ch][scfsi_band] = 0; +- } /* if gr == 1 */ +- } + + + +--- 170,1028 ---- + + + ++ /* convert from L/R <-> Mid/Side */ ++ void ms_convert(double xr[2][576],double xr_org[2][576]) ++ { ++ int i; ++ for ( i = 0; i < 576; i++ ) { ++ xr[0][i] = (xr_org[0][i]+xr_org[1][i])/SQRT2; ++ xr[1][i] = (xr_org[0][i]-xr_org[1][i])/SQRT2; ++ } ++ } ++ ++ ++ ++ ++ ++ /************************************************************************ ++ * allocate bits among the 4 granules based on PE ++ * mt 6/99 ++ ************************************************************************/ ++ int VBR_on_pe(frame_params *fr_ps, layer *info, III_side_info_t *l3_side, ++ int VBRbits[2][2], ++ double pe[][2], int *mean_bits) ++ ++ { ++ int nshort=0; ++ int index,gr,ch; ++ int bitsPerFrame; ++ int stereo = fr_ps->stereo; ++ int fullframebits; ++ int mode_gr = (info->version == 1) ? 2 : 1; ++ ++ getframebits(info,stereo,&bitsPerFrame,mean_bits); ++ fullframebits= ResvFrameBegin( fr_ps, l3_side, *mean_bits, bitsPerFrame ); ++ ++ ++ for (gr=0;gr<mode_gr;gr++) ++ for (ch=0;ch<2;ch++) { ++ int shortblock = (l3_side->gr[gr].ch[ch].tt.block_type==2); ++ if (shortblock) nshort ++; ++ } ++ index = nshort; ++ if (index>3) index=3; ++ ++ /* impose a minimum bit rate based on number of short blocks */ ++ if (index>0) { ++ /* nshort=1: min 160kbs. (1 shortblock granule) ++ * nshort=2: min 190kbs. (2 shortblock granules) ++ * nshort=3: min 220kbs (3 shortblock granules) ++ * nshort=4: min 256kbs (4 shortblock granules) */ ++ if (info->bitrate_index < 9 + index) info->bitrate_index = 9+index; ++ if (info->bitrate_index > 14) info->bitrate_index=14; ++ ++ ++ getframebits(info,stereo,&bitsPerFrame,mean_bits); ++ fullframebits= ResvFrameBegin( fr_ps, l3_side, *mean_bits, bitsPerFrame ); ++ } ++ ++ /* allocate a minimum 125 bits per channel (approx 32kbs) */ ++ for (gr=0;gr<mode_gr;gr++) { ++ for (ch=0;ch<2;ch++) { ++ VBRbits[gr][ch] = 125; ++ } ++ } ++ fullframebits -= 500; ++ ++ /* divide bits based on PE */ ++ if (fullframebits > 0 ) ++ { ++ double pe2[2][2],pe_tot=.01; ++ for (gr=0;gr<mode_gr;gr++) { ++ for (ch=0;ch<2;ch++) { ++ int shortblock = (l3_side->gr[gr].ch[ch].tt.block_type==2); ++ pe2[gr][ch] = Min(pe[gr][ch],2800.); ++ if (shortblock) pe2[gr][ch] = Max(pe2[gr][ch],1800.0); ++ pe_tot += pe2[gr][ch]; ++ } ++ } ++ for (gr=0;gr<mode_gr;gr++) { ++ for (ch=0;ch<2;ch++) { ++ VBRbits[gr][ch] = (fullframebits*pe2[gr][ch])/pe_tot; ++ } ++ } ++ } ++ /* return value = 1 => do not decrease bit rate further */ ++ if (index) ++ return (info->bitrate_index <= 9 + index); ++ else return 0; ++ } ++ ++ ++ ++ ++ /************************************************************************ ++ * increase or decrease frame bitrate based on max_over = number of scale ++ * factor bands with audible distortion. If bitrate in increased, allocate ++ * the additional bits among the granules based on "over" ++ * mt 6/99 ++ ************************************************************************/ ++ void VBR_on_over(frame_params *fr_ps, layer *info, III_side_info_t *l3_side, ++ int VBRbits[2][2], double pe[2][2], int max_over, ++ int over[2][2], int *mean_bits, int *VBR_no_decrease) ++ ++ { ++ int extrabits=0; ++ int gr,ch; ++ int bitsPerFrame; ++ int stereo = fr_ps->stereo; ++ int fullframebits; ++ int mode_gr = (info->version == 1) ? 2 : 1; ++ ++ if (max_over <= VBR_q) { ++ info->bitrate_index --; ++ if (info->bitrate_index < VBR_min_bitrate) ++ info->bitrate_index=VBR_min_bitrate; ++ *VBR_no_decrease = VBR_on_pe(fr_ps,info,l3_side,VBRbits,pe,mean_bits); ++ return; ++ } ++ ++ getframebits(info,stereo,&bitsPerFrame,mean_bits); ++ fullframebits= ResvFrameBegin( fr_ps, l3_side, *mean_bits, bitsPerFrame ); ++ ++ info->bitrate_index ++; ++ if (info->bitrate_index > VBR_max_bitrate) ++ info->bitrate_index=VBR_max_bitrate; ++ ++ getframebits(info,stereo,&bitsPerFrame,mean_bits); ++ extrabits= ResvFrameBegin( fr_ps, l3_side, *mean_bits, bitsPerFrame ); ++ extrabits -= fullframebits; ++ ++ /* divide extra bits based on OVER */ ++ if (extrabits > 0) { ++ double over2[2][2],over_tot=0; ++ for (gr=0;gr<mode_gr;gr++) { ++ for (ch=0;ch<2;ch++) { ++ over2[gr][ch] = over[gr][ch] + 1; ++ over_tot += over2[gr][ch]; ++ } ++ } ++ for (gr=0;gr<mode_gr;gr++) { ++ for (ch=0;ch<2;ch++) { ++ VBRbits[gr][ch] += (extrabits*over2[gr][ch])/over_tot; ++ } ++ } ++ } ++ for (gr=0;gr<mode_gr;gr++) { ++ for (ch=0;ch<2;ch++) { ++ if (VBRbits[gr][ch]>4095) VBRbits[gr][ch]=4095; ++ } ++ } ++ } ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ + /************************************************************************/ + /* iteration_loop() */ + /************************************************************************/ + void +! iteration_loop( double pe[][2], double ms_ener_ratio[2], +! double xr_org[2][2][576], III_psy_ratio *ratio, + III_side_info_t *l3_side, int l3_enc[2][2][576], +! III_scalefac_t *scalefac, frame_params *fr_ps) +! { +! static int firstcall = 1; +! III_psy_xmin l3_xmin; +! gr_info *cod_info; +! layer *info; +! int VBRbits[2][2]; +! int best_over[2][2]; +! int bitsPerFrame; +! int mean_bits; +! int stereo = fr_ps->stereo; +! int ch, gr, i, mode_gr, bit_rate; +! int VBR_no_decrease; +! int samp_freq; +! double xr[2][2][576]; +! +! if ( firstcall ) { +! l3_side->main_data_begin = 0; +! memset((char *) &l3_xmin, 0, sizeof(l3_xmin)); +! firstcall = 0; +! } +! l3_side->resvDrain = 0; +! info = fr_ps->header; +! mode_gr = (info->version == 1) ? 2 : 1; +! bit_rate = bitrate[info->version][info->lay-1][info->bitrate_index]; +! samp_freq = s_freq[info->version][info->sampling_frequency]*1000 +.5; + +! scalefac_band_long = &sfBandIndex[info->sampling_frequency + (info->version * 3)].l[0]; +! scalefac_band_short = &sfBandIndex[info->sampling_frequency + (info->version * 3)].s[0]; +! + +! convert_mdct=0; +! convert_psy=0; +! reduce_sidechannel=0; +! if (info->mode_ext==2) { +! convert_mdct = 1; +! convert_psy = 1; +! reduce_sidechannel=1; +! } +! if (force_ms) { +! convert_mdct = 0; +! convert_psy = 0; +! reduce_sidechannel=1; +! } +! + +! /* remove coefficients in scalefactor band 21 (12 for short blocks) +! * FhG does this for bps <= 128kbs, so we will too. +! * This amounds to a 16kHz low-pass filter. If that offends you, you +! * probably should not be encoding at 128kbs! +! * There is no ratio[21] or xfsf[21], so when these coefficients are +! * included they are just quantized as is. mt 5/99 +! */ +! if (sfb21 && (samp_freq>33000)) { +! if (bit_rate <= 64*stereo) { +! for (ch =0 ; ch < 2 ; ch++) +! for ( gr = 0; gr < mode_gr; gr++ ) { +! int shortblock = (l3_side->gr[gr].ch[0].tt.block_type==2); +! if (shortblock) { +! int j; +! for (j=0; j<3; j++) { +! int start = scalefac_band_short[ SFB_SMAX-1 ]; +! for ( i = start; i < 192; i++ ) { +! int i0 = 3*i+j; +! xr_org[gr][ch][i0]=0; +! } +! } +! }else{ +! int start = scalefac_band_long[ SFB_LMAX-1 ]; +! for ( i = start; i < 576; i++ ) xr_org[gr][ch][i]=0; +! } +! } +! } +! } +! +! +! +! /* some intializations. */ +! for ( gr = 0; gr < mode_gr; gr++ ){ +! for ( ch = 0; ch < stereo; ch++ ){ +! cod_info = (gr_info *) &(l3_side->gr[gr].ch[ch]); +! gr_deco(cod_info); + } ++ } + + +! /* dont bother with scfsi. */ +! for ( ch = 0; ch < stereo; ch++ ) +! for ( i = 0; i < 4; i++ ) +! l3_side->scfsi[ch][i] = 0; +! +! + ++ VBR_no_decrease=0; ++ if (VBR) { ++ VBR_no_decrease=VBR_on_pe(fr_ps,info,l3_side,VBRbits,pe,&mean_bits); ++ }else{ ++ getframebits(info,stereo,&bitsPerFrame,&mean_bits); + ResvFrameBegin( fr_ps, l3_side, mean_bits, bitsPerFrame ); ++ } ++ + + +! +! /* quantize! */ +! do { +! int max_over; +! +! for ( gr = 0; gr < mode_gr; gr++ ) +! outer_loop( xr, xr_org, mean_bits, VBRbits, bit_rate, best_over[gr], +! &l3_xmin,l3_enc, fr_ps, +! scalefac,gr,stereo, l3_side, ratio, pe, ms_ener_ratio); +! +! if (!VBR) break; +! +! /* see if we should try a higher bitrate quantization */ +! max_over=0; +! for ( gr = 0; gr < mode_gr; gr++ ) +! for (ch=0 ; ch < stereo ; ch ++ ) { +! if (best_over[gr][ch]>max_over) max_over=best_over[gr][ch]; +! } +! +! if ((max_over <= VBR_q) && (info->bitrate_index <= VBR_min_bitrate)) break; +! if ((max_over > VBR_q) && (info->bitrate_index >= VBR_max_bitrate)) break; +! /* do not decrease bit rate if we have ever increased bit rate +! * otherwise infinite loop */ +! if ((max_over <= VBR_q) && (VBR_no_decrease)) break; +! +! /* compute a new bitrate, and allocate bits between granules */ +! if (max_over > VBR_q) VBR_no_decrease=1; +! VBR_on_over(fr_ps,info,l3_side,VBRbits,pe,max_over,best_over,&mean_bits,&VBR_no_decrease); +! } while (1); +! +! if (VBR) { +! /* update reservoir status after FINAL quantization/bitrate */ +! for ( gr = 0; gr < mode_gr; gr++ ) { +! for (ch=0 ; ch < stereo ; ch ++ ) { +! gr_info *cod_info = &l3_side->gr[gr].ch[ch].tt; +! ResvAdjust( fr_ps, cod_info, l3_side, mean_bits ); +! } +! } +! } +! +! /* set the sign of l3_enc */ +! for ( gr = 0; gr < mode_gr; gr++ ) +! for ( ch = 0; ch < stereo; ch++ ) +! { +! int *pi = &l3_enc[gr][ch][0]; +! +! for ( i = 0; i < 576; i++) { +! double pr = xr[gr][ch][i]; +! if ( (pr < 0) && (pi[i] > 0) ) pi[i] *= -1; +! } +! } +! +! +! ResvFrameEnd( fr_ps, l3_side, mean_bits ); + } + + + + /************************************************************************/ +! /* init_outer_loop mt 6/99 */ + /************************************************************************/ +! void init_outer_loop( +! double xr[2][2][576], /* could be L/R OR MID/SIDE */ +! double xr_org[2][2][576], +! III_psy_xmin *l3_xmin, /* the allowed distortion of the scalefactor */ +! III_scalefac_t *scalefac, /* scalefactors */ +! int gr, int stereo, III_side_info_t *l3_side, +! III_psy_ratio *ratio) + { +! int ch,sfb,i; +! gr_info *cod_info[2]; +! cod_info[0] = &l3_side->gr[gr].ch[0].tt; +! cod_info[1] = &l3_side->gr[gr].ch[1].tt; +! +! /* copy data to be quantized into xr */ +! if (convert_mdct) ms_convert(xr[gr],xr_org[gr]); +! else memcpy(xr[gr],xr_org[gr],sizeof(double)*2*576); +! +! +! /* compute max allowed distortion */ +! for ( ch = 0; ch < stereo; ch++ ){ +! calc_xmin( xr_org, ratio, cod_info[ch], l3_xmin, gr, ch ); +! } +! + +! for (ch=0 ; ch < stereo ; ch ++ ){ + +! /* if ( info->version == 1 ) +! calc_scfsi( xr[gr][ch], l3_side, &l3_xmin, ch, gr ); +! */ +! +! +! /* reset of iteration variables */ +! +! for ( sfb = 0; sfb < SFB_LMAX; sfb++ ) +! scalefac->l[gr][ch][sfb] = 0; +! for ( sfb = 0; sfb < SFB_SMAX; sfb++ ) +! for ( i = 0; i < 3; i++ ) +! scalefac->s[gr][ch][sfb][i] = 0; +! +! for ( i = 0; i < 4; i++ ) +! cod_info[ch]->slen[i] = 0; +! cod_info[ch]->sfb_partition_table = &nr_of_sfb_block[0][0][0]; +! +! cod_info[ch]->part2_3_length = 0; +! cod_info[ch]->big_values = 0; +! cod_info[ch]->count1 = 0; +! cod_info[ch]->scalefac_compress = 0; +! cod_info[ch]->table_select[0] = 0; +! cod_info[ch]->table_select[1] = 0; +! cod_info[ch]->table_select[2] = 0; +! cod_info[ch]->subblock_gain[0] = 0; +! cod_info[ch]->subblock_gain[1] = 0; +! cod_info[ch]->subblock_gain[2] = 0; +! cod_info[ch]->region0_count = 0; +! cod_info[ch]->region1_count = 0; +! cod_info[ch]->part2_length = 0; +! cod_info[ch]->preflag = 0; +! cod_info[ch]->scalefac_scale = 0; +! cod_info[ch]->quantizerStepSize = 0.0; +! cod_info[ch]->count1table_select= 0; +! cod_info[ch]->address1 = 0; +! cod_info[ch]->address2 = 0; +! cod_info[ch]->address3 = 0; +! } +! +! +! #define SBGAINXXX +! #ifdef SBGAIN +! /* compute subblock gains */ +! for (ch=0 ; ch < stereo ; ch ++) { +! int j,b; double en[3],mx; +! if ((cod_info[ch]->block_type ==2) && count[ch]) { +! /* estimate energy within each subblock */ +! for (b=0; b<3; b++) en[b]=0; +! for ( i=0,j = 0; j < 192; j++ ) { +! for (b=0; b<3; b++) { +! en[b]+=xr[gr][ch][i]*xr[gr][ch][i]; +! i++; +! } +! } +! mx = 1e-12; +! for (b=0; b<3; b++) mx=Max(mx,en[b]); +! for (b=0; b<3; b++) en[b] = Max(en[b],1e-12)/mx; +! printf("ener = %4.2f %4.2f %4.2f \n",en[0],en[1],en[2]); +! /* pick gain so that 2^(2gain)*en[0] = 1 */ +! /* gain = .5* log( 1/en[0] )/log(2) = -.5*log(en[])/log(2) */ +! for (b=0; b<3; b++) { +! cod_info[ch]->subblock_gain[b]=nint(-.5*log(en[b])/log(2.0)); +! if (cod_info[ch]->subblock_gain[b] > 2) +! cod_info[ch]->subblock_gain[b]=2; +! if (cod_info[ch]->subblock_gain[b] < 0) +! cod_info[ch]->subblock_gain[b]=0; +! } + } +! } + #endif + } + + + + +! + /************************************************************************/ +! /* outer_loop */ + /************************************************************************/ + /* Function: The outer iteration loop controls the masking conditions */ + /* of all scalefactorbands. It computes the best scalefac and */ +! /* global gain. This module calls the inner iteration loop +! * +! * mt 5/99 completely rewritten to allow for bit reservoir control, +! * mid/side channels with L/R or mid/side masking thresholds, +! * and chooses best quantization instead of last quantization when +! * no distortion free quantization can be found. +! * +! * added VBR support mt 5/99 +! ************************************************************************/ +! void outer_loop( +! double xr[2][2][576], /* could be L/R OR MID/SIDE */ +! double xr_org[2][2][576], +! int mean_bits, +! int VBRbits[2][2], +! int bit_rate, +! int best_over[2], + III_psy_xmin *l3_xmin, /* the allowed distortion of the scalefactor */ + int l3_enc[2][2][576], /* vector of quantized values ix(0..575) */ + frame_params *fr_ps, + III_scalefac_t *scalefac, /* scalefactors */ +! int gr, int stereo, III_side_info_t *l3_side, +! III_psy_ratio *ratio, double pe[2][2], double ms_ener_ratio[2]) + { +! int status[2],notdone[2]={0,0},count[2]={0,0},bits_found[2]; +! int targ_bits[2],real_bits[2],tbits,extra_bits; +! int scalesave_l[2][CBLIMIT], scalesave_s[2][CBLIMIT][3]; +! int sfb, bits, huff_bits, save_preflag[2], save_compress[2]; +! double xfsf[2][4][CBLIMIT]; +! double xrpow[2][2][576],temp; +! int distort[2][4][CBLIMIT]; +! int save_l3_enc[2][576]; +! int save_real_bits[2]; +! int i,over[2], iteration, ch, try_scale; +! int better[2]; +! double tot_noise[2]; +! double best_noise[2]; +! gr_info save_cod_info[2]; +! gr_info *cod_info[2]; +! +! cod_info[0] = &l3_side->gr[gr].ch[0].tt; +! cod_info[1] = &l3_side->gr[gr].ch[1].tt; +! for (ch=0 ; ch < stereo ; ch ++) +! best_over[ch] = 100; + ++ init_outer_loop(xr,xr_org,l3_xmin,scalefac,gr,stereo,l3_side,ratio); ++ ++ for (ch=0 ; ch < stereo ; ch ++) { ++ count[ch]=0; ++ for (i=0; i<576; i++) { ++ if ( fabs(xr[gr][ch][i]) > 0 ) count[ch]++; ++ } ++ notdone[ch]=count[ch]; ++ if (count[ch]==0) best_over[ch]=0; ++ } + + +! /****************************************************************** +! * allocate bits for each channel +! ******************************************************************/ +! if (VBR) { +! for (ch=0 ; ch < stereo ; ch ++ ) +! targ_bits[ch]=VBRbits[gr][ch]; +! +! }else { +! int add_bits[2]; +! double bits_needed; + +! /* allocate targ_bits for granule */ +! ResvMaxBits2( mean_bits, &tbits, &extra_bits, gr); + +! for (ch=0 ; ch < stereo ; ch ++ ) +! targ_bits[ch]=tbits/stereo; +! +! // allocate extra bits from reservoir based on PE +! bits=0; +! for (ch=0; ch<stereo; ch++) { +! /* extra bits based on PE > 700 */ +! add_bits[ch]=(pe[gr][ch]-700)/2.0; /* 3.0; */ +! if (convert_psy) bits_needed = (pe[gr][0]+pe[gr][1]-1400)/2.0; +! +! /* short blocks need extra, no matter what the pe */ +! if (cod_info[ch]->block_type==2) +! if (add_bits[ch]<500) add_bits[ch]=500; +! +! if (add_bits[ch] < 0) add_bits[ch]=0; +! bits += add_bits[ch]; +! } +! for (ch=0; ch<stereo; ch++) { +! if (bits > extra_bits) add_bits[ch] = (extra_bits*add_bits[ch])/bits; +! targ_bits[ch] = targ_bits[ch] + add_bits[ch]; +! } +! for (ch=0; ch<stereo; ch++) +! extra_bits -= add_bits[ch]; +! } + + + +! if (reduce_sidechannel) { +! /* ms_ener_ratio = 0: allocate 66/33 mid/side fac=.33 +! * ms_ener_ratio =.5: allocate 50/50 mid/side fac= 0 */ +! /* 75/25 split is fac=.5 */ +! /* float fac = .50*(.5-ms_ener_ratio[gr])/.5;*/ +! float fac = .33*(.5-ms_ener_ratio[gr])/.5; +! if (fac<0) fac=0; +! +! /* dont reduce side channel below 125 bits */ +! if (targ_bits[1]-targ_bits[1]*fac > 125) { +! targ_bits[0] += targ_bits[1]*fac; +! targ_bits[1] -= targ_bits[1]*fac; +! } +! } +! +! /* dont allow to many bits per channel */ +! for (ch=0; ch<stereo; ch++) { +! int max_bits = mean_bits/2 + 1200; +! if (targ_bits[ch] > max_bits) { +! extra_bits += (targ_bits[ch] - max_bits); +! targ_bits[ch] = max_bits; +! } +! } +! +! + +! +! +! /* BEGIN MAIN LOOP */ +! iteration = 0; + +! while ( (notdone[0] || notdone[1]) ) { +! int pre_just_turned_on[2]; +! iteration ++; +! +! if (iteration==1) { +! /* compute initial quantization step */ +! for (ch=0 ; ch < stereo ; ch ++ ) +! if (count[ch]) { +! for(i=0;i<576;i++) { +! temp=fabs(xr[gr][ch][i]); +! xrpow[gr][ch][i]=sqrt(sqrt(temp)*temp); +! } +! bits_found[ch]=bin_search_StepSize2(targ_bits[ch],-211.0,46, +! l3_enc[gr][ch],xr[gr][ch],xrpow[gr][ch],cod_info[ch]); +! } +! } + + +! for (ch=0 ; ch < stereo ; ch ++ ) { +! /* inner_loop starts with the initial quantization step computed above +! * and slowly increases until the bits < huff_bits. +! * Thus is it important not to start with too large of an inital +! * quantization step. Too small is ok, but inner_loop will take longer +! */ +! for (ch=0 ; ch < stereo ; ch ++ ) { +! if (notdone[ch]) { +! cod_info[ch]->part2_length = part2_length( scalefac, fr_ps, gr, ch, l3_side ); +! huff_bits = targ_bits[ch] - cod_info[ch]->part2_length; +! if (huff_bits < 0) { +! if (iteration==1) { +! fprintf(stderr,"ERROR: outer_loop(): huff_bits < 0. \n"); +! exit(-5); +! }else{ +! /* scale factors too large, not enough bits. use previous quantizaton */ +! notdone[ch]=0; +! over[ch]=999; +! } +! }else{ +! /* if this is the first iteration, see if we can reuse the quantization +! * computed in bin_search_StepSize above */ +! if (iteration==1) { +! if(bits_found[ch]>huff_bits) { +! cod_info[ch]->quantizerStepSize+=1.0; +! real_bits[ch] = inner_loop( xr, xrpow, l3_enc, huff_bits, cod_info[ch], gr, ch ); +! } else real_bits[ch]=bits_found[ch]; +! } +! else +! real_bits[ch]=inner_loop( xr, xrpow, l3_enc, huff_bits, cod_info[ch], gr, ch ); +! } +! } +! } +! } + ++ if (fast_mode) { ++ for (ch=0; ch<stereo; ch++) ++ over[ch]=0; ++ }else{ ++ if (convert_psy) { ++ /* mid/side coefficiets, l/r thresholds */ ++ calc_noise2( xr[gr], l3_enc[gr], cod_info, xfsf, ++ distort, l3_xmin,gr,stereo,over,tot_noise); ++ } else { ++ /* coefficients and thresholds both l/r (or both mid/side) */ ++ for (ch=0; ch<stereo; ch++) ++ if (notdone[ch]) ++ over[ch]=calc_noise1( xr[gr][ch], l3_enc[gr][ch], cod_info[ch], ++ xfsf[ch],distort[ch], l3_xmin,gr,ch, tot_noise[ch]); ++ } ++ } + + +! /* check if this quantization is better the our saved quantization */ +! for (ch=0 ; ch < stereo ; ch ++ ) { +! better[ch]=0; +! if (notdone[ch]) { +! if (convert_psy) { +! if (experimentalZ) { +! better[ch] = (over[0]+over[1]) < (best_over[0]+best_over[1]); +! if ((over[0]+over[1])==(best_over[0]+best_over[1])) { +! better[ch] = (tot_noise[0]+tot_noise[1]) < (best_noise[0]+best_noise[1]); +! } +! } else { +! better[ch] = (over[0]+over[1]) <= (best_over[0]+best_over[1]); +! } +! }else{ +! if (experimentalZ) { +! better[ch] = ((over[ch] < best_over[ch]) || +! ((over[ch]==best_over[ch]) && (tot_noise[ch]<=best_noise[ch])) ) ; +! }else{ +! better[ch] = (over[ch] <= best_over[ch]); +! } +! } +! } + } ++ /* save data so we can restore this quantization later */ ++ for (ch=0 ; ch < stereo ; ch ++ ) { ++ if (better[ch]) { ++ best_over[ch]=over[ch]; ++ best_noise[ch]=tot_noise[ch]; ++ if (notdone[ch]) { ++ for ( sfb = 0; sfb < CBLIMIT; sfb++ ) /* save scaling factors */ ++ scalesave_l[ch][sfb] = scalefac->l[gr][ch][sfb]; ++ ++ for ( sfb = 0; sfb < SFB_SMAX; sfb++ ) ++ for ( i = 0; i < 3; i++ ) ++ scalesave_s[ch][sfb][i] = scalefac->s[gr][ch][sfb][i]; ++ ++ save_preflag[ch] = cod_info[ch]->preflag; ++ save_compress[ch] = cod_info[ch]->scalefac_compress; ++ ++ memcpy(save_l3_enc[ch],l3_enc[gr][ch],sizeof(l3_enc[gr][ch])); ++ memcpy(&save_cod_info[ch],cod_info[ch],sizeof(save_cod_info[ch])); ++ save_real_bits[ch]=real_bits[ch]; ++ ++ #ifdef HAVEGTK ++ if (gtkflag) { ++ for ( i = 0; i < 3; i++ ) { ++ for ( sfb = cod_info[ch]->sfb_smax; sfb < 12; sfb++ ) { ++ pinfo->xfsf_s[gr][ch][3*sfb+i] = ++ pinfo->thr_s[gr][ch][3*sfb+i]*xfsf[ch][i+1][sfb]/ ++ (1e-20+l3_xmin->s[gr][ch][sfb][i]); ++ } ++ } ++ for ( sfb = 0; sfb < cod_info[ch]->sfb_lmax; sfb++ ) { ++ pinfo->xfsf[gr][ch][sfb] = ++ pinfo->thr[gr][ch][sfb]*xfsf[ch][0][sfb]/ ++ (1e-20 + l3_xmin->l[gr][ch][sfb]); ++ } ++ pinfo->over[gr][ch]=over[ch]; ++ pinfo->noise[gr][ch]=tot_noise[ch]; ++ } + #endif +! + + } +! } + } + +! /* if no bands with distortion, we are done */ +! for (ch=0 ; ch < stereo ; ch ++ ) +! if (notdone[ch]) { +! if (convert_psy) +! notdone[ch] = (over[0] || over[1]); +! else +! notdone[ch] = over[ch]; +! } +! + + + +! /* see if we should apply preemphasis */ +! for (ch=0 ; ch < stereo ; ch ++ ) { +! pre_just_turned_on[ch]=0; +! if (notdone[ch]) pre_just_turned_on[ch]= +! preemphasis2( xr[gr][ch], xrpow[gr][ch], l3_xmin, +! gr, ch, l3_side, distort[ch]); +! } +! +! +! /* if we didn't just apply pre-emph, let us see if we should +! * amplify some scale factor bands */ +! for (ch=0 ; ch < stereo ; ch ++ ) +! if (notdone[ch] && (!pre_just_turned_on[ch]) ) { +! amp_scalefac_bands2( xr[gr][ch], xrpow[gr][ch], l3_xmin, +! l3_side, scalefac, gr, ch, iteration,distort[ch]); +! } +! + +! /* check to make sure we have not amplified too much */ +! try_scale=0; +! for (ch=0 ; ch < stereo ; ch ++ ) { +! if (notdone[ch]) { +! if ( (status[ch] = loop_break(scalefac, cod_info[ch], gr, ch)) == 0 ) { +! if ( fr_ps->header->version == 1 ) { +! status[ch] = scale_bitcount( scalefac, cod_info[ch], gr, ch ); +! if (status[ch] && (cod_info[ch]->scalefac_scale==0)) try_scale=1; +! }else{ +! status[ch] = scale_bitcount_lsf( scalefac, cod_info[ch], gr, ch ); +! if (status[ch] && (cod_info[ch]->scalefac_scale==0)) try_scale=1; +! } +! } +! notdone[ch] = !status[ch]; +! } + } +! if (try_scale && experimentalY) { +! init_outer_loop(xr,xr_org,l3_xmin,scalefac,gr,stereo,l3_side,ratio); +! for (ch=0 ; ch < stereo ; ch ++ ) { +! iteration=0; +! notdone[ch]=1; +! cod_info[ch]->scalefac_scale=1; +! } +! } +! } /* done with main iteration */ +! + ++ ++ /* restore some data */ ++ for (ch=0 ; ch < stereo ; ch ++ ) { ++ if (count[ch] ) { ++ cod_info[ch]->preflag = save_preflag[ch]; ++ cod_info[ch]->scalefac_compress = save_compress[ch]; ++ ++ for ( sfb = 0; sfb < CBLIMIT; sfb++ ) { ++ scalefac->l[gr][ch][sfb] = scalesave_l[ch][sfb]; ++ } ++ ++ for ( i = 0; i < 3; i++ ) ++ for ( sfb = 0; sfb < SFB_SMAX; sfb++ ) { ++ scalefac->s[gr][ch][sfb][i] = scalesave_s[ch][sfb][i]; ++ } + ++ { ++ real_bits[ch]=save_real_bits[ch]; ++ memcpy(l3_enc[gr][ch],save_l3_enc[ch],sizeof(l3_enc[gr][ch])); ++ memcpy(cod_info[ch],&save_cod_info[ch],sizeof(save_cod_info[ch])); ++ ++ if ( fr_ps->header->version == 1 ) ++ status[ch] = scale_bitcount( scalefac, cod_info[ch], gr, ch ); ++ else ++ status[ch] = scale_bitcount_lsf( scalefac, cod_info[ch], gr, ch ); ++ if (status[ch]) { ++ fprintf(stderr,"Error recomputing scalefac_compress...this should not happen"); ++ exit(-10); ++ } ++ } ++ cod_info[ch]->part2_length = part2_length( scalefac, fr_ps, gr, ch, l3_side ); ++ cod_info[ch]->part2_3_length = cod_info[ch]->part2_length + real_bits[ch]; + ++ #ifdef HAVEGTK ++ if (gtkflag) ++ pinfo->LAMEmainbits[gr][ch]=cod_info[ch]->part2_3_length; ++ #endif ++ } ++ } ++ ++ /* finish up */ ++ for (ch=0 ; ch < stereo ; ch ++ ) { ++ if (!VBR) ResvAdjust( fr_ps, cod_info[ch], l3_side, mean_bits ); ++ cod_info[ch]->global_gain = nint( cod_info[ch]->quantizerStepSize + 210.0 ); ++ assert( cod_info[ch]->global_gain < 256 ); ++ } + + + + + + } + + + + + +! + + + + + +! /*************************************************************************** +! * inner_loop * +! *************************************************************************** +! * The code selects the best quantizerStepSize for a particular set +! * of scalefacs */ +! +! int +! inner_loop( double xr[2][2][576], double xrpow[2][2][576], +! int l3_enc[2][2][576], int max_bits, +! gr_info *cod_info, int gr, int ch ) +! { +! int bits; +! +! assert( max_bits >= 0 ); +! cod_info->quantizerStepSize -= 1.0;; +! do + { +! cod_info->quantizerStepSize += 1.0; +! quantize_xrpow( xrpow[gr][ch], l3_enc[gr][ch], cod_info ); +! bits = count_bits(l3_enc[gr][ch],cod_info); +! } +! while ( bits > max_bits ); +! return bits; +! } + + + + + + +*************** +*** 673,679 **** + int part2_length( III_scalefac_t *scalefac, frame_params *fr_ps, + int gr, int ch, III_side_info_t *si ) + { +! int slen1, slen2, slen3, slen4, bits, sfb, window, partition; + gr_info *gi = &si->gr[gr].ch[ch].tt; + + bits = 0; +--- 1036,1042 ---- + int part2_length( III_scalefac_t *scalefac, frame_params *fr_ps, + int gr, int ch, III_side_info_t *si ) + { +! int slen1, slen2, bits, partition; + gr_info *gi = &si->gr[gr].ch[ch].tt; + + bits = 0; +*************** +*** 687,700 **** + + if ( (gi->window_switching_flag == 1) && (gi->block_type == 2) ) + { +- if ( gi->mixed_block_flag ) +- { +- bits += (8 * slen1) + (9 * slen1) + (18 * slen2); +- } +- else +- { + bits += (18 * slen1) + (18 * slen2); +- } + } + else + { +--- 1050,1056 ---- +*************** +*** 738,749 **** + + static int slen1[16] = { 0, 0, 0, 0, 3, 1, 1, 1, 2, 2, 2, 3, 3, 3, 4, 4 }; + static int slen2[16] = { 0, 1, 2, 3, 0, 1, 2, 3, 1, 2, 3, 1, 2, 3, 2, 3 }; +- static int pow2[5] = { 1, 2, 4, 8, 16 }; + + if ( cod_info->window_switching_flag != 0 && cod_info->block_type == 2 ) + { +- if ( cod_info->mixed_block_flag == 0 ) +- { + /* a = 18; b = 18; */ + for ( i = 0; i < 3; i++ ) + { +--- 1094,1102 ---- +*************** +*** 754,776 **** + if ( scalefac->s[gr][ch][sfb][i] > max_slen2 ) + max_slen2 = scalefac->s[gr][ch][sfb][i]; + } +- } +- else +- {/* mixed_block_flag = 1 */ +- /* a = 17; b = 18; */ +- for ( sfb = 0; sfb < 8; sfb++ ) +- if ( scalefac->l[gr][ch][sfb] > max_slen1 ) +- max_slen1 = scalefac->l[gr][ch][sfb]; +- for ( i = 0; i < 3; i++ ) +- { +- for ( sfb = 3; sfb < 6; sfb++ ) +- if ( scalefac->s[gr][ch][sfb][i] > max_slen1 ) +- max_slen1 = scalefac->s[gr][ch][sfb][i]; +- for ( sfb = 6; sfb < 12; sfb++ ) +- if ( scalefac->s[gr][ch][sfb][i] > max_slen2 ) +- max_slen2 = scalefac->s[gr][ch][sfb][i]; +- } +- } + } + else + { /* block_type == 1,2,or 3 */ +--- 1107,1112 ---- +*************** +*** 785,791 **** + + for ( k = 0; k < 16; k++ ) + { +! if ( (max_slen1 < pow2[slen1[k]]) && (max_slen2 < pow2[slen2[k]]) ) + { + ep = 0; + break; +--- 1121,1127 ---- + + for ( k = 0; k < 16; k++ ) + { +! if ( (max_slen1 < (1<<slen1[k])) && (max_slen2 < (1<<slen2[k])) ) + { + ep = 0; + break; +*************** +*** 794,803 **** + + if ( ep == 0 ) + cod_info->scalefac_compress = k; +- #ifdef DEBUG +- if ( ep != 0 ) +- printf( "---WARNING !! Amplification of some bands over limits\n" ); +- #endif + return ep; + } + +--- 1130,1135 ---- +*************** +*** 818,824 **** + int gr, int ch ) + { + int table_number, row_in_table, partition, nr_sfb, window, over; +! int i, k, sfb, max_sfac[ 4 ]; + unsigned *partition_table; + + /* +--- 1150,1156 ---- + int gr, int ch ) + { + int table_number, row_in_table, partition, nr_sfb, window, over; +! int i, sfb, max_sfac[ 4 ]; + unsigned *partition_table; + + /* +*************** +*** 835,842 **** + + if ( cod_info->window_switching_flag != 0 && cod_info->block_type == 2 ) + { +- if ( cod_info->mixed_block_flag == 0 ) +- { + row_in_table = 1; + partition_table = &nr_of_sfb_block[table_number][row_in_table][0]; + for ( sfb = 0, partition = 0; partition < 4; partition++ ) +--- 1167,1172 ---- +*************** +*** 847,872 **** + if ( scalefac->s[gr][ch][sfb][window] > max_sfac[partition] ) + max_sfac[partition] = scalefac->s[gr][ch][sfb][window]; + } +- } +- else +- {/* mixed_block_flag = 1 */ +- row_in_table = 2; +- partition_table = &nr_of_sfb_block[table_number][row_in_table][0]; +- partition = 0; +- nr_sfb = partition_table[ partition ]; +- for ( i = 0; i < nr_sfb; i++, sfb++ ) +- if ( scalefac->l[gr][ch][sfb] > max_sfac[partition] ) +- max_sfac[partition] = scalefac->l[gr][ch][sfb]; +- +- for ( sfb = 0, partition = 1; partition < 4; partition++ ) +- { +- nr_sfb = partition_table[ partition ] / 3; +- for ( i = 0; i < nr_sfb; i++, sfb++ ) +- for ( window = 0; window < 3; window++ ) +- if ( scalefac->s[gr][ch][sfb][window] > max_sfac[partition] ) +- max_sfac[partition] = scalefac->s[gr][ch][sfb][window]; +- } +- } + } + else + { +--- 1177,1182 ---- +*************** +*** 939,960 **** + } + + + /*************************************************************************/ + /* calc_noise */ + /*************************************************************************/ + +- /* Function: Calculate the distortion introduced by the quantization */ +- /* in each scale factor band. */ +- +- void calc_noise( double xr[576], int ix[576], gr_info *cod_info, +- double xfsf[4][CBLIMIT] ) + { +! int start, end, sfb, l, i; + double sum,step,bw; + + D192_3 *xr_s; + I192_3 *ix_s; + + xr_s = (D192_3 *) xr; + ix_s = (I192_3 *) ix; + +--- 1249,1282 ---- + } + + ++ ++ + /*************************************************************************/ + /* calc_noise */ + /*************************************************************************/ ++ /* mt 5/99: Function: Improved calc_noise for a single channel */ ++ int calc_noise1( double xr[576], int ix[576], gr_info *cod_info, ++ double xfsf[4][CBLIMIT], int distort[4][CBLIMIT], ++ III_psy_xmin *l3_xmin,int gr, int ch, double noise) + + { +! int start, end, sfb, l, i, over=0; + double sum,step,bw; + + D192_3 *xr_s; + I192_3 *ix_s; + ++ #define PRECALC_SIZE 1024 /* WAS 256 !!! */ ++ static double pow43[PRECALC_SIZE]; ++ static int init=0; ++ noise=0; ++ ++ if (init==0) { ++ init++; ++ for(i=0;i<PRECALC_SIZE;i++) ++ pow43[i] = pow((double)i, 4.0/3.0); ++ } ++ + xr_s = (D192_3 *) xr; + ix_s = (I192_3 *) ix; + +*************** +*** 968,982 **** + for ( sum = 0.0, l = start; l < end; l++ ) + { + double temp; +! temp = fabs( xr[l] ) - pow( (double) ix[l], 4.0 / 3.0 ) * step; +! sum += temp * temp; + } + xfsf[0][sfb] = sum / bw; + } + + for ( i = 0; i < 3; i++ ) + { + step = pow( 2.0, (cod_info->quantizerStepSize) * 0.25 ); /* subblock_gain ? */ + for ( sfb = cod_info->sfb_smax; sfb < 12; sfb++ ) + { + start = scalefac_band_short[ sfb ]; +--- 1290,1317 ---- + for ( sum = 0.0, l = start; l < end; l++ ) + { + double temp; +! if (ix[l]<PRECALC_SIZE) +! temp = fabs(xr[l]) - pow43[ix[l]] * step; +! else +! { +! temp = fabs(xr[l]) - pow((double)ix[l],4.0/3.0)*step; +! /* printf("EHHHHHHH !?!?! ---> %d\n",ix[l]); .. table is not big enough */ +! } +! sum += temp * temp; + } + xfsf[0][sfb] = sum / bw; ++ distort[0][sfb] = ( xfsf[0][sfb] > l3_xmin->l[gr][ch][sfb] ); ++ if (distort[0][sfb]) over++; ++ if (distort[0][sfb]) noise += xfsf[0][sfb] - l3_xmin->l[gr][ch][sfb]; + } + ++ + for ( i = 0; i < 3; i++ ) + { + step = pow( 2.0, (cod_info->quantizerStepSize) * 0.25 ); /* subblock_gain ? */ ++ if (cod_info->subblock_gain[i] ) ++ step *= pow(2.0,-2.0*cod_info->subblock_gain[i]); ++ + for ( sfb = cod_info->sfb_smax; sfb < 12; sfb++ ) + { + start = scalefac_band_short[ sfb ]; +*************** +*** 986,1001 **** + for ( sum = 0.0, l = start; l < end; l++ ) + { + double temp; +! temp = fabs( (*xr_s)[l][i] ) - pow( (double) (*ix_s)[l][i], 4.0 / 3.0 ) * step; + sum += temp * temp; + } + xfsf[i+1][sfb] = sum / bw; + } + } + } + + + + + /*************************************************************************/ + /* calc_xmin */ +--- 1321,1469 ---- + for ( sum = 0.0, l = start; l < end; l++ ) + { + double temp; +! if((*ix_s)[l][i]<PRECALC_SIZE) +! temp = fabs((*xr_s)[l][i]) - pow43[(*ix_s)[l][i]] * step; +! else +! { +! temp = fabs((*xr_s)[l][i]) - pow((double)(*ix_s)[l][i],4.0/3.0)*step; +! /* printf("EHHHHHHH !?!?! ---> %d\n",(*ix_s)[l][i]); */ +! } + sum += temp * temp; + } + xfsf[i+1][sfb] = sum / bw; ++ distort[i+1][sfb] = ++ ( xfsf[i+1][sfb] > l3_xmin->s[gr][ch][sfb][i] ); ++ if (distort[i+1][sfb]) over++; ++ if (distort[i+1][sfb]) noise += xfsf[i+1][sfb]-l3_xmin->s[gr][ch][sfb][i]; + } + } ++ return over; + } + + + ++ /*************************************************************************/ ++ /* calc_noise2 */ ++ /*************************************************************************/ ++ /* Improved version of calc_noise for dual channel. This routine is */ ++ /* used when you are quantizaing mid and side channels using masking */ ++ /* thresholds from L and R channels. mt 5/99 */ ++ ++ void calc_noise2( double xr[2][576], int ix[2][576], gr_info *cod_info[2], ++ double xfsf[2][4][CBLIMIT], int distort[2][4][CBLIMIT], ++ III_psy_xmin *l3_xmin,int gr,int stereo, int over[2], ++ double noise[2]) ++ { ++ int start, end, sfb, l, i; ++ double sum[2],step_s[3][2],step[2],bw; ++ ++ D192_3 *xr_s[2]; ++ I192_3 *ix_s[2]; ++ ++ #define PRECALC_SIZE 1024 /* WAS 256 !!! */ ++ static double pow43[PRECALC_SIZE]; ++ int ch; ++ static int init=0; ++ double diff[2], qcoeff; ++ ++ if (init==0) { ++ init++; ++ for(i=0;i<PRECALC_SIZE;i++) ++ pow43[i] = pow((double)i, 4.0/3.0); ++ } ++ ++ ++ ++ /* calc_noise2: we can assume block types of both channels must be the same */ ++ if (cod_info[0]->block_type != 2) { ++ for (ch=0 ; ch < stereo ; ch ++ ) { ++ over[ch]=0; ++ noise[ch]=0; ++ step[ch] = pow( 2.0, (cod_info[ch]->quantizerStepSize) * 0.25 ); ++ } ++ for ( sfb = 0; sfb < SFB_LMAX-1; sfb++ ) { ++ start = scalefac_band_long[ sfb ]; ++ end = scalefac_band_long[ sfb+1 ]; ++ bw = end - start; ++ ++ for (ch=0 ; ch < stereo ; ch ++ ) sum[ch]=0; ++ for ( l = start; l < end; l++ ) { ++ for (ch=0 ; ch < stereo ; ch ++ ) { ++ qcoeff= (ix[ch][l] < PRECALC_SIZE) ? ++ pow43[ix[ch][l]] * step[ch]: ++ pow((double)ix[ch][l],4.0/3.0)*step[ch]; ++ if (xr[ch][l]<0) qcoeff=-qcoeff; ++ diff[ch]=xr[ch][l]-qcoeff; ++ } ++ sum[0] += (diff[0]+diff[1])*(diff[0]+diff[1])/(2.0); ++ sum[1] += (diff[0]-diff[1])*(diff[0]-diff[1])/(2.0); ++ } ++ for (ch=0 ; ch < stereo ; ch ++ ) { ++ xfsf[ch][0][sfb] = sum[ch] / bw; ++ distort[ch][0][sfb] = ( xfsf[ch][0][sfb] > l3_xmin->l[gr][ch][sfb] ); ++ if (distort[ch][0][sfb]) over[ch]++; ++ if (distort[ch][0][sfb]) noise[ch] += xfsf[ch][0][sfb] - l3_xmin->l[gr][ch][sfb]; ++ } ++ ++ /* if there is audible distortion in left or right channel, set flags ++ * to denote distortion in both mid and side channels */ ++ for (ch=0 ; ch < stereo ; ch ++ ) { ++ distort[ch][0][sfb] = (distort[0][0][sfb] || distort[1][0][sfb]); ++ } ++ } ++ } ++ ++ /* calc_noise2: we can assume block types of both channels must be the same */ ++ if (cod_info[0]->block_type == 2) { ++ for (ch=0 ; ch < stereo ; ch ++ ) { ++ ++ for (i=0;i<3;i++){ ++ step_s[i][ch] = pow( 2.0, (cod_info[ch]->quantizerStepSize) * 0.25 ); /* subblock_gain ? */ ++ if (cod_info[ch]->subblock_gain[i] ) ++ step_s[i][ch] *= pow(2.0,-2.0*cod_info[ch]->subblock_gain[i]); ++ } ++ ++ over[ch] = 0; ++ xr_s[ch] = (D192_3 *) xr[ch]; ++ ix_s[ch] = (I192_3 *) ix[ch]; ++ } ++ ++ for ( sfb = 0 ; sfb < SFB_SMAX-1; sfb++ ) { ++ start = scalefac_band_short[ sfb ]; ++ end = scalefac_band_short[ sfb+1 ]; ++ bw = end - start; ++ for ( i = 0; i < 3; i++ ) { ++ for (ch=0 ; ch < stereo ; ch ++ ) sum[ch] = 0.0; ++ for ( l = start; l < end; l++ ) { ++ for (ch=0 ; ch < stereo ; ch ++ ) { ++ qcoeff = ((*ix_s[ch])[l][i]<PRECALC_SIZE) ? ++ pow43[(*ix_s[ch])[l][i]] * step_s[i][ch] : ++ pow((double)(*ix_s[ch])[l][i],4.0/3.0)*step_s[i][ch]; ++ if ((*xr_s[ch])[l][i] < 0) qcoeff=-qcoeff; ++ diff[ch] = (*xr_s[ch])[l][i] - qcoeff; ++ } ++ sum[0] += (diff[0]+diff[1])*(diff[0]+diff[1])/(2.0); ++ sum[1] += (diff[0]-diff[1])*(diff[0]-diff[1])/(2.0); ++ } ++ for (ch=0 ; ch < stereo ; ch ++ ) { ++ xfsf[ch][i+1][sfb] = sum[ch] / bw; ++ distort[ch][i+1][sfb] = ++ ( xfsf[ch][i+1][sfb] > l3_xmin->s[gr][ch][sfb][i] ); ++ if (distort[ch][i+1][sfb]) over[ch]++; ++ if (distort[ch][i+1][sfb]) noise[ch] += xfsf[ch][i+1][sfb]-l3_xmin->s[gr][ch][sfb][i]; ++ } ++ /* if there is audible distortion in left or right channel, set flags ++ * to denote distortion in both mid and side channels */ ++ for (ch=0 ; ch < stereo ; ch ++ ) ++ distort[ch][i+1][sfb] = ++ (distort[0][i+1][sfb] || distort[1][i+1][sfb] ); ++ } ++ } ++ } ++ } ++ ++ ++ + + /*************************************************************************/ + /* calc_xmin */ +*************** +*** 1012,1018 **** + int gr, int ch ) + { + int start, end, sfb, l, b; +! double en, bw; + + D192_3 *xr_s; + +--- 1480,1486 ---- + int gr, int ch ) + { + int start, end, sfb, l, b; +! double en0, bw; + + D192_3 *xr_s; + +*************** +*** 1025,1033 **** + bw = end - start; + for ( b = 0; b < 3; b++ ) + { +! for ( en = 0.0, l = start; l < end; l++ ) +! en += (*xr_s)[l][b] * (*xr_s)[l][b]; +! l3_xmin->s[gr][ch][sfb][b] = ratio->s[gr][ch][sfb][b] * en / bw; + } + } + +--- 1493,1501 ---- + bw = end - start; + for ( b = 0; b < 3; b++ ) + { +! for ( en0 = 0.0, l = start; l < end; l++ ) +! en0 += (*xr_s)[l][b] * (*xr_s)[l][b]; +! l3_xmin->s[gr][ch][sfb][b] = ratio->s[gr][ch][sfb][b] * en0 / bw; + } + } + +*************** +*** 1037,1045 **** + end = scalefac_band_long[ sfb+1 ]; + bw = end - start; + +! for ( en = 0.0, l = start; l < end; l++ ) +! en += xr[gr][ch][l] * xr[gr][ch][l]; +! l3_xmin->l[gr][ch][sfb] = ratio->l[gr][ch][sfb] * en / bw; + } + } + +--- 1505,1513 ---- + end = scalefac_band_long[ sfb+1 ]; + bw = end - start; + +! for ( en0 = 0.0, l = start; l < end; l++ ) +! en0 += xr[gr][ch][l] * xr[gr][ch][l]; +! l3_xmin->l[gr][ch][sfb] = ratio->l[gr][ch][sfb] * en0 / bw; + } + } + +*************** +*** 1056,1080 **** + int loop_break( III_scalefac_t *scalefac, gr_info *cod_info, + int gr, int ch ) + { +! int i, sfb, temp = 1; + + for ( sfb = 0; sfb < cod_info->sfb_lmax; sfb++ ) + if ( scalefac->l[gr][ch][sfb] == 0 ) +! temp = 0; + + for ( sfb = cod_info->sfb_smax; sfb < 12; sfb++ ) + for ( i = 0; i < 3; i++ ) + if ( scalefac->s[gr][ch][sfb][i] == 0 ) +! temp = 0; +! #ifdef DEBUG +! if ( temp != 0 ) +! printf( "---WARNING !! All scalefactor bands amplified\n" ); +! #endif +! return temp; + } + + + + /*************************************************************************/ + /* preemphasis */ + /*************************************************************************/ +--- 1524,1546 ---- + int loop_break( III_scalefac_t *scalefac, gr_info *cod_info, + int gr, int ch ) + { +! int i, sfb; + + for ( sfb = 0; sfb < cod_info->sfb_lmax; sfb++ ) + if ( scalefac->l[gr][ch][sfb] == 0 ) +! return 0; + + for ( sfb = cod_info->sfb_smax; sfb < 12; sfb++ ) + for ( i = 0; i < 3; i++ ) + if ( scalefac->s[gr][ch][sfb][i] == 0 ) +! return 0; +! +! return 1; + } + + + ++ + /*************************************************************************/ + /* preemphasis */ + /*************************************************************************/ +*************** +*** 1082,1091 **** + /* + See ISO 11172-3 section C.1.5.4.3.4 + */ +! +! void preemphasis( double xr[576], double xfsf[4][CBLIMIT], +! III_psy_xmin *l3_xmin, +! int gr, int ch, III_side_info_t *l3_side ) + { + int i, sfb, start, end, scfsi_band, over; + double ifqstep; +--- 1548,1556 ---- + /* + See ISO 11172-3 section C.1.5.4.3.4 + */ +! int preemphasis2( double xr[576], double xrpow[576], +! III_psy_xmin *l3_xmin, +! int gr, int ch, III_side_info_t *l3_side, int distort[4][CBLIMIT] ) + { + int i, sfb, start, end, scfsi_band, over; + double ifqstep; +*************** +*** 1102,1108 **** + if ( l3_side->scfsi[ch][scfsi_band] ) + { + cod_info->preflag = l3_side->gr[0].ch[ch].tt.preflag; +! return; + } + + } +--- 1567,1573 ---- + if ( l3_side->scfsi[ch][scfsi_band] ) + { + cod_info->preflag = l3_side->gr[0].ch[ch].tt.preflag; +! return 0; + } + + } +*************** +*** 1112,1143 **** + bands the actual distortion exceeds the threshold after the + first call of the inner loop + */ + if ( cod_info->block_type != 2 && cod_info->preflag == 0 ) + { +- over = 0; + for ( sfb = 17; sfb < 21; sfb++ ) +! if ( xfsf[0][sfb] > l3_xmin->l[gr][ch][sfb] ) +! over++; + + if (over == 4 ) + { + cod_info->preflag = 1; +! ifqstep = ( cod_info->scalefac_scale == 0 ) ? sqrt(2.) +! : pow( 2.0, (0.5 * (1.0 + (double) cod_info->scalefac_scale)) ); +! +! for ( sfb = 0; sfb < cod_info->sfb_lmax; sfb++ ) + { +! l3_xmin->l[gr][ch][sfb] *= pow( ifqstep, 2.0 * (double) pretab[sfb] ); + start = scalefac_band_long[ sfb ]; + end = scalefac_band_long[ sfb+1 ]; +! for( i = start; i < end; i++ ) +! xr[i] *= pow( ifqstep, (double) pretab[sfb] ); + } + } + } + } + + + /*************************************************************************/ + /* amp_scalefac_bands */ + /*************************************************************************/ +--- 1577,1611 ---- + bands the actual distortion exceeds the threshold after the + first call of the inner loop + */ ++ over = 0; + if ( cod_info->block_type != 2 && cod_info->preflag == 0 ) + { + for ( sfb = 17; sfb < 21; sfb++ ) +! if ( distort[0][sfb] ) over++; + + if (over == 4 ) + { ++ double t,t34; + cod_info->preflag = 1; +! ifqstep = ( cod_info->scalefac_scale == 0 ) ? SQRT2 : 2.0; +! for ( sfb = 11; sfb < cod_info->sfb_lmax; sfb++ ) + { +! t=pow( ifqstep, (double) pretab[sfb] ); +! t34=sqrt(sqrt(t)*t); +! l3_xmin->l[gr][ch][sfb] *= t*t; + start = scalefac_band_long[ sfb ]; + end = scalefac_band_long[ sfb+1 ]; +! for( i = start; i < end; i++ ) xr[i]*=t; +! for( i = start; i < end; i++ ) xrpow[i]*=t34; + } + } + } ++ return (over == 4); + } + + ++ ++ + /*************************************************************************/ + /* amp_scalefac_bands */ + /*************************************************************************/ +*************** +*** 1146,1173 **** + Amplify the scalefactor bands that violate the masking threshold. + See ISO 11172-3 Section C.1.5.4.3.5 + */ +! +! int amp_scalefac_bands( double xr[576], double xfsf[4][CBLIMIT], + III_psy_xmin *l3_xmin, III_side_info_t *l3_side, + III_scalefac_t *scalefac, +! int gr, int ch, int iteration ) + { + int start, end, l, sfb, i, scfsi_band, over = 0; +! double ifqstep, ifqstep2; + D192_3 *xr_s; + gr_info *cod_info, *gr0; + int copySF, preventSF; + cod_info = &l3_side->gr[gr].ch[ch].tt; + gr0 = &l3_side->gr[0].ch[ch].tt; + + xr_s = (D192_3 *) xr; + copySF = 0; + preventSF = 0; + + if ( cod_info->scalefac_scale == 0 ) +! ifqstep = sqrt( 2.0 ); + else +! ifqstep = pow( 2.0, 0.5 * (1.0 + (double) cod_info->scalefac_scale) ); + + if ( gr == 1 ) + { +--- 1614,1643 ---- + Amplify the scalefactor bands that violate the masking threshold. + See ISO 11172-3 Section C.1.5.4.3.5 + */ +! int amp_scalefac_bands2( double xr[576], double xrpow[576], + III_psy_xmin *l3_xmin, III_side_info_t *l3_side, + III_scalefac_t *scalefac, +! int gr, int ch, int iteration, int distort[4][CBLIMIT]) + { + int start, end, l, sfb, i, scfsi_band, over = 0; +! double ifqstep, ifqstep2, ifqstep34; + D192_3 *xr_s; ++ D192_3 *xrpow_s; + gr_info *cod_info, *gr0; + int copySF, preventSF; + cod_info = &l3_side->gr[gr].ch[ch].tt; + gr0 = &l3_side->gr[0].ch[ch].tt; + + xr_s = (D192_3 *) xr; ++ xrpow_s = (D192_3 *) xrpow; + copySF = 0; + preventSF = 0; + ++ + if ( cod_info->scalefac_scale == 0 ) +! ifqstep = SQRT2; + else +! ifqstep = 2.0; + + if ( gr == 1 ) + { +*************** +*** 1183,1191 **** + first granule... + */ + if ( gr0->scalefac_scale == 0 ) +! ifqstep = sqrt( 2.0 ); + else +! ifqstep = pow( 2.0, 0.5 * (1.0 + (double) gr0->scalefac_scale) ); + + if ( iteration == 1 ) + { +--- 1653,1661 ---- + first granule... + */ + if ( gr0->scalefac_scale == 0 ) +! ifqstep = SQRT2; + else +! ifqstep = 2.0; + + if ( iteration == 1 ) + { +*************** +*** 1212,1217 **** +--- 1682,1688 ---- + + ifqstep2 = ifqstep * ifqstep; + scfsi_band = 0; ++ ifqstep34=sqrt(sqrt(ifqstep)*ifqstep); + + for ( sfb = 0; sfb < cod_info->sfb_lmax; sfb++ ) + { +*************** +*** 1226,1240 **** + continue; + } + } +! if ( xfsf[0][sfb] > l3_xmin->l[gr][ch][sfb] ) + { + over++; + l3_xmin->l[gr][ch][sfb] *= ifqstep2; + scalefac->l[gr][ch][sfb]++; + start = scalefac_band_long[sfb]; + end = scalefac_band_long[sfb+1]; +! for ( l = start; l < end; l++ ) +! xr[l] *= ifqstep; + } + } + +--- 1697,1713 ---- + continue; + } + } +! +! +! if ( distort[0][sfb] ) + { + over++; + l3_xmin->l[gr][ch][sfb] *= ifqstep2; + scalefac->l[gr][ch][sfb]++; + start = scalefac_band_long[sfb]; + end = scalefac_band_long[sfb+1]; +! for ( l = start; l < end; l++ ) xr[l] *= ifqstep; +! for ( l = start; l < end; l++ ) xrpow[l] *= ifqstep34; + } + } + +*************** +*** 1243,1250 **** + short blocks + */ + for ( i = 0; i < 3; i++ ) +! for ( sfb = cod_info->sfb_smax; sfb < 12; sfb++ ) +! if ( xfsf[i+1][sfb] > l3_xmin->s[gr][ch][sfb][i] ) + { + over++; + l3_xmin->s[gr][ch][sfb][i] *= ifqstep2; +--- 1716,1723 ---- + short blocks + */ + for ( i = 0; i < 3; i++ ) +! for ( sfb = cod_info->sfb_smax; sfb < 12; sfb++ ) { +! if ( distort[i+1][sfb]) + { + over++; + l3_xmin->s[gr][ch][sfb][i] *= ifqstep2; +*************** +*** 1255,1330 **** + #endif + start = scalefac_band_short[sfb]; + end = scalefac_band_short[sfb+1]; +! for ( l = start; l < end; l++ ) +! (*xr_s)[l][i] *= ifqstep; + } + return over; + } + + + + +! /*************************************************************************/ +! /* quantize */ +! /*************************************************************************/ +! +! /* +! Function: Quantization of the vector xr ( -> ix) +! */ +! +! void quantize( double xr[576], int ix[576], gr_info *cod_info ) +! { +! int i, b, l_end, s_start; +! double step, quantizerStepSize; +! +! D192_3 *xr_s; +! I192_3 *ix_s; +! +! xr_s = (D192_3 *) xr; +! ix_s = (I192_3 *) ix; +! +! quantizerStepSize = (double) cod_info->quantizerStepSize; +! +! for ( i = 0; i < 576; i++ ) +! ix[i] = 0; +! +! if ( cod_info->quantizerStepSize == 0.0 ) +! step = 1.0; +! else +! step = pow ( 2.0, quantizerStepSize * 0.25 ); +! +! if ( cod_info->window_switching_flag != 0 && cod_info->block_type == 2 ) +! if ( cod_info->mixed_block_flag == 0 ) +! { +! l_end = 0; +! s_start = 0; +! } +! else +! { +! l_end = 18 * 2; +! s_start = 6 * 2; +! } +! else + { +! l_end = 576; +! s_start = 192; + } +- +- for ( i = 0; i < l_end; i++ ) +- ix[i] = nint( pow(fabs(xr[i]) / step, 0.75) - 0.0946 ); +- +- if ( s_start < 192 ) +- for ( b = 0; b < 3; b++ ) +- { +- step = pow( 2.0, (quantizerStepSize + 8.0 * (double) cod_info->subblock_gain[b]) * 0.25 ); +- for ( i = s_start; i < 192; i++ ) +- (*ix_s)[i][b] = nint( pow(fabs((*xr_s)[i][b]) / step, 0.75) - 0.0946 ); +- } + } + + + + + /*************************************************************************/ + /* ix_max */ + /*************************************************************************/ +--- 1728,1787 ---- + #endif + start = scalefac_band_short[sfb]; + end = scalefac_band_short[sfb+1]; +! for ( l = start; l < end; l++ ) (*xr_s)[l][i] *= ifqstep; +! for ( l = start; l < end; l++ ) (*xrpow_s)[l][i] *= ifqstep34; + } ++ } + return over; + } + + + + +! +! +! void quantize_xrpow( double xr[576], int ix[576], gr_info *cod_info ) +! { +! /* quantize on xr^(3/4) instead of xr */ +! register int j; +! double quantizerStepSize; +! double istep_l,istep0,istep1,istep2; +! +! quantizerStepSize = cod_info->quantizerStepSize; +! +! istep_l = pow ( 2.0, quantizerStepSize * -0.1875 ); +! +! if ((cod_info->block_type==2)) + { +! istep0 = istep_l * pow(2.0,1.5* (double) cod_info->subblock_gain[0]); +! istep1 = istep_l * pow(2.0,1.5* (double) cod_info->subblock_gain[1]); +! istep2 = istep_l * pow(2.0,1.5* (double) cod_info->subblock_gain[2]); +! } +! else +! { +! istep0 = istep_l; +! istep1 = istep_l; +! istep2 = istep_l; +! } +! +! for (j=192;j>0;j--) +! { +! #if defined(__GNUC__) && defined(__i386__) +! asm ("fistpl %0 ": "=m"(*(ix++)): "t"(istep0*(*(xr++)) - 0.0946): "st"); +! asm ("fistpl %0 ": "=m"(*(ix++)): "t"(istep1*(*(xr++)) - 0.0946): "st"); +! asm ("fistpl %0 ": "=m"(*(ix++)): "t"(istep2*(*(xr++)) - 0.0946): "st"); +! #else +! *(ix++) = (int)( istep0*(*(xr++)) + 0.4054); +! *(ix++) = (int)( istep1*(*(xr++)) + 0.4054); +! *(ix++) = (int)( istep2*(*(xr++)) + 0.4054); +! #endif + } + } + + + + ++ + /*************************************************************************/ + /* ix_max */ + /*************************************************************************/ +*************** +*** 1339,1345 **** + + for ( i = begin; i < end; i++ ) + { +! int x = abs( ix[i] ); + if ( x > max ) + max = x; + } +--- 1796,1802 ---- + + for ( i = begin; i < end; i++ ) + { +! int x = ix[i]; + if ( x > max ) + max = x; + } +*************** +*** 1376,1389 **** + /*************************************************************************/ + + /* +! Function: Calculation of rzero, count1, big_values + (Partitions ix into big values, quadruples and zeros). + */ + + void calc_runlen( int ix[576], gr_info *cod_info ) + { + int i; +- int rzero = 0; + + if ( cod_info->window_switching_flag && (cod_info->block_type == 2) ) + { /* short blocks */ +--- 1833,1845 ---- + /*************************************************************************/ + + /* +! Function: Calculation of count1, big_values + (Partitions ix into big values, quadruples and zeros). + */ + + void calc_runlen( int ix[576], gr_info *cod_info ) + { + int i; + + if ( cod_info->window_switching_flag && (cod_info->block_type == 2) ) + { /* short blocks */ +*************** +*** 1393,1416 **** + else + { + for ( i = 576; i > 1; i -= 2 ) +! if ( ix[i-1] == 0 && ix[i-2] == 0 ) +! rzero++; +! else +! break; +! + cod_info->count1 = 0 ; + for ( ; i > 3; i -= 4 ) +! if ( abs(ix[i-1]) <= 1 +! && abs(ix[i-2]) <= 1 +! && abs(ix[i-3]) <= 1 +! && abs(ix[i-4]) <= 1 ) + cod_info->count1++; + else + break; + + cod_info->big_values = i/2; + } +- assert( (2 * rzero + 4 * cod_info->count1 + 2 * cod_info->big_values) == 576 ); + } + + +--- 1849,1865 ---- + else + { + for ( i = 576; i > 1; i -= 2 ) +! if ( ix[i-1] | ix[i-2] ) break; +! + cod_info->count1 = 0 ; + for ( ; i > 3; i -= 4 ) +! if ( (ix[i-1] | ix[i-2] | ix[i-3] | ix[i-4]) <= 1 ) + cod_info->count1++; + else + break; + + cod_info->big_values = i/2; + } + } + + +*************** +*** 1432,1460 **** + int sum0 = 0, sum1 = 0; + + for ( i = cod_info->big_values * 2, k = 0; k < cod_info->count1; i += 4, k++ ) +! { + v = ix[ i ]; + w = ix[ i + 1 ]; + x = ix[ i + 2 ]; + y = ix[ i + 3 ]; + + abs_and_sign( &v ); + abs_and_sign( &w ); + abs_and_sign( &x ); + abs_and_sign( &y ); +! + p = v + (w << 1) + (x << 2) + (y << 3); + + signbits = 0; + + if ( v != 0 ) +! signbits += 1; + if ( w != 0 ) +! signbits += 1; + if ( x != 0 ) +! signbits += 1; + if ( y != 0 ) +! signbits += 1; + + sum0 += signbits; + sum1 += signbits; +--- 1881,1916 ---- + int sum0 = 0, sum1 = 0; + + for ( i = cod_info->big_values * 2, k = 0; k < cod_info->count1; i += 4, k++ ) +! { +! v = ix[i]; +! w = ix[i+1]; +! x = ix[i+2]; +! y = ix[i+3]; +! +! /* + v = ix[ i ]; + w = ix[ i + 1 ]; + x = ix[ i + 2 ]; + y = ix[ i + 3 ]; + ++ + abs_and_sign( &v ); + abs_and_sign( &w ); + abs_and_sign( &x ); + abs_and_sign( &y ); +! */ + p = v + (w << 1) + (x << 2) + (y << 3); + + signbits = 0; + + if ( v != 0 ) +! signbits ++; + if ( w != 0 ) +! signbits ++; + if ( x != 0 ) +! signbits ++; + if ( y != 0 ) +! signbits ++; + + sum0 += signbits; + sum1 += signbits; +*************** +*** 1570,1576 **** + } + else + { +! if ( (cod_info->block_type == 2) && (cod_info->mixed_block_flag == 0) ) + { + cod_info->region0_count = 8; + cod_info->region1_count = 36; +--- 2026,2032 ---- + } + else + { +! if ( (cod_info->block_type == 2) ) + { + cod_info->region0_count = 8; + cod_info->region1_count = 36; +*************** +*** 1594,1670 **** + + + /*************************************************************************/ +- /* bigv_tab_select */ +- /*************************************************************************/ +- +- /* +- /* Function: Select huffman code tables for bigvalues regions +- */ +- +- void bigv_tab_select( int ix[576], gr_info *cod_info ) +- { +- /* int max; */ +- +- cod_info->table_select[0] = 0; +- cod_info->table_select[1] = 0; +- cod_info->table_select[2] = 0; +- +- if ( cod_info->window_switching_flag && (cod_info->block_type == 2) ) +- { +- /* +- Within each scalefactor band, data is given for successive +- time windows, beginning with window 0 and ending with window 2. +- Within each window, the quantized values are then arranged in +- order of increasing frequency... +- */ +- int sfb, window, line, start, end, max1, max2, x, y; +- int region1Start; +- int *pmax; +- +- region1Start = 12; +- max1 = max2 = 0; +- +- for ( sfb = 0; sfb < 13; sfb++ ) +- { +- start = scalefac_band_short[ sfb ]; +- end = scalefac_band_short[ sfb+1 ]; +- +- if ( start < region1Start ) +- pmax = &max1; +- else +- pmax = &max2; +- +- for ( window = 0; window < 3; window++ ) +- for ( line = start; line < end; line += 2 ) +- { +- assert( line >= 0 ); +- assert( line < 576 ); +- x = abs( ix[ (line * 3) + window ] ); +- y = abs( ix[ ((line + 1) * 3) + window ]); +- *pmax = *pmax > x ? *pmax : x; +- *pmax = *pmax > y ? *pmax : y; +- } +- } +- cod_info->table_select[0] = choose_table( max1 ); +- cod_info->table_select[1] = choose_table( max2 ); +- } +- else +- { +- if ( cod_info->address1 > 0 ) +- cod_info->table_select[0] = new_choose_table( ix, 0, cod_info->address1 ); +- +- if ( cod_info->address2 > cod_info->address1 ) +- cod_info->table_select[1] = new_choose_table( ix, cod_info->address1, cod_info->address2 ); +- +- if ( cod_info->big_values * 2 > cod_info->address2 ) +- cod_info->table_select[2] = new_choose_table( ix, cod_info->address2, cod_info->big_values * 2 ); +- } +- } +- +- +- +- +- /*************************************************************************/ + /* new_choose table */ + /*************************************************************************/ + +--- 2050,2055 ---- +*************** +*** 1677,1683 **** + with any arbitrary tables. + */ + +! int new_choose_table( int ix[576], unsigned int begin, unsigned int end ) + { + int i, max; + int choice[ 2 ]; +--- 2062,2068 ---- + with any arbitrary tables. + */ + +! int new_choose_table( int ix[576], unsigned int begin, unsigned int end, int * s ) + { + int i, max; + int choice[ 2 ]; +*************** +*** 1687,1694 **** + + if ( max == 0 ) + return 0; +- +- max = abs( max ); + + choice[ 0 ] = 0; + choice[ 1 ] = 0; +--- 2072,2077 ---- +*************** +*** 1702,1708 **** + choice[ 0 ] = i; + break; + } +- assert( choice[0] ); + + sum[ 0 ] = count_bit( ix, begin, end, choice[0] ); + +--- 2085,2090 ---- +*************** +*** 1711,1723 **** +--- 2093,2111 ---- + case 2: + sum[ 1 ] = count_bit( ix, begin, end, 3 ); + if ( sum[1] <= sum[0] ) ++ { ++ sum[0]=sum[1]; + choice[ 0 ] = 3; ++ } + break; + + case 5: + sum[ 1 ] = count_bit( ix, begin, end, 6 ); + if ( sum[1] <= sum[0] ) ++ { ++ sum[0]=sum[1]; + choice[ 0 ] = 6; ++ } + break; + + case 7: +*************** +*** 1729,1735 **** +--- 2117,2126 ---- + } + sum[ 1 ] = count_bit( ix, begin, end, 9 ); + if ( sum[1] <= sum[0] ) ++ { ++ sum[0]=sum[1]; + choice[ 0 ] = 9; ++ } + break; + + case 10: +*************** +*** 1741,1753 **** +--- 2132,2150 ---- + } + sum[ 1 ] = count_bit( ix, begin, end, 12 ); + if ( sum[1] <= sum[0] ) ++ { ++ sum[0]=sum[1]; + choice[ 0 ] = 12; ++ } + break; + + case 13: + sum[ 1 ] = count_bit( ix, begin, end, 15 ); + if ( sum[1] <= sum[0] ) ++ { ++ sum[0]=sum[1]; + choice[ 0 ] = 15; ++ } + break; + + default: +*************** +*** 1775,1793 **** + break; + } + } +- assert( choice[0] ); +- assert( choice[1] ); + + sum[ 0 ] = count_bit( ix, begin, end, choice[0] ); + sum[ 1 ] = count_bit( ix, begin, end, choice[1] ); + if ( sum[1] < sum[0] ) + choice[ 0 ] = choice[ 1 ]; + } + return choice[ 0 ]; + } + + +- + /*************************************************************************/ + /* choose table */ + /*************************************************************************/ +--- 2172,2191 ---- + break; + } + } + + sum[ 0 ] = count_bit( ix, begin, end, choice[0] ); + sum[ 1 ] = count_bit( ix, begin, end, choice[1] ); + if ( sum[1] < sum[0] ) ++ { ++ sum[0]=sum[1]; + choice[ 0 ] = choice[ 1 ]; ++ } + } ++ *s=sum[0]; + return choice[ 0 ]; + } + + + /*************************************************************************/ + /* choose table */ + /*************************************************************************/ +*************** +*** 1799,1805 **** + if ( max == 0 ) + return 0; + +- max = abs( max ); + choice = 0; + + if ( max < 15 ) +--- 2197,2202 ---- +*************** +*** 1830,1848 **** + } + + + /*************************************************************************/ + /* bigv_bitcount */ + /*************************************************************************/ + + /* + Function: Count the number of bits necessary to code the bigvalues region. + */ + +! int bigv_bitcount( int ix[576], gr_info *gi ) + { + int bits = 0; + +! if ( gi->window_switching_flag && gi->block_type == 2 ) + { + /* + Within each scalefactor band, data is given for successive +--- 2227,2251 ---- + } + + ++ + /*************************************************************************/ + /* bigv_bitcount */ + /*************************************************************************/ + + /* ++ Function: Select huffman code tables for bigvalues regions + Function: Count the number of bits necessary to code the bigvalues region. + */ + +! int bigv_bitcount( int ix[576], gr_info *cod_info ) + { + int bits = 0; ++ ++ cod_info->table_select[0] = 0; ++ cod_info->table_select[1] = 0; ++ cod_info->table_select[2] = 0; + +! if ( cod_info->window_switching_flag && (cod_info->block_type == 2) ) + { + /* + Within each scalefactor band, data is given for successive +*************** +*** 1850,1867 **** + Within each window, the quantized values are then arranged in + order of increasing frequency... + */ +! int sfb, window, line, start, end; + I192_3 *ix_s; + +- if ( gi->mixed_block_flag ) +- { +- unsigned int table; + +! if ( (table = gi->table_select[0]) != 0 ) +! bits += count_bit( ix, 0, gi->address1, table ); +! sfb = 2; + } +! else + sfb = 0; + + ix_s = (I192_3 *) &ix[0]; +--- 2253,2296 ---- + Within each window, the quantized values are then arranged in + order of increasing frequency... + */ +! int sfb, window, line, start, end, max1, max2, x, y; +! int region1Start; +! int *pmax; + I192_3 *ix_s; + + +! region1Start = 12; +! max1 = max2 = 0; +! +! for ( sfb = 0; sfb < 13; sfb++ ) +! { +! start = scalefac_band_short[ sfb ]; +! end = scalefac_band_short[ sfb+1 ]; +! +! if ( start < region1Start ) +! pmax = &max1; +! else +! pmax = &max2; +! +! for ( window = 0; window < 3; window++ ) +! for ( line = start; line < end; line += 2 ) +! { +! x = ix[ (line * 3) + window ]; +! y = ix[ ((line + 1) * 3) + window ]; +! *pmax = *pmax > x ? *pmax : x; +! *pmax = *pmax > y ? *pmax : y; +! } + } +! cod_info->table_select[0] = choose_table( max1 ); +! cod_info->table_select[1] = choose_table( max2 ); +! +! /* +! Within each scalefactor band, data is given for successive +! time windows, beginning with window 0 and ending with window 2. +! Within each window, the quantized values are then arranged in +! order of increasing frequency... +! */ +! + sfb = 0; + + ix_s = (I192_3 *) &ix[0]; +*************** +*** 1874,1883 **** + end = scalefac_band_short[ sfb+1 ]; + + if ( start < 12 ) +! tableindex = gi->table_select[ 0 ]; + else +! tableindex = gi->table_select[ 1 ]; +! assert( tableindex < 32 ); + + for ( window = 0; window < 3; window++ ) + for ( line = start; line < end; line += 2 ) +--- 2303,2311 ---- + end = scalefac_band_short[ sfb+1 ]; + + if ( start < 12 ) +! tableindex = cod_info->table_select[ 0 ]; + else +! tableindex = cod_info->table_select[ 1 ]; + + for ( window = 0; window < 3; window++ ) + for ( line = start; line < end; line += 2 ) +*************** +*** 1893,1905 **** + else + { + unsigned int table; + +! if( (table=gi->table_select[0]) != 0 ) /* region0 */ +! bits += count_bit(ix, 0, gi->address1, table ); +! if( (table=gi->table_select[1]) != 0 ) /* region1 */ +! bits += count_bit(ix, gi->address1, gi->address2, table ); +! if( (table=gi->table_select[2]) != 0 ) /* region2 */ +! bits += count_bit(ix, gi->address2, gi->address3, table ); + } + return bits; + } +--- 2321,2348 ---- + else + { + unsigned int table; ++ int s; + +! if ( cod_info->address1 > 0 ) +! { +! table = cod_info->table_select[0] = new_choose_table( ix, 0, cod_info->address1, &s ); +! if( table != 0 ) /* region0 */ +! bits += s; +! } +! +! if ( cod_info->address2 > cod_info->address1 ) +! { +! table = cod_info->table_select[1] = new_choose_table( ix, cod_info->address1, cod_info->address2, &s ); +! if( table != 0 ) /* region1 */ +! bits += s; +! } +! +! if ( cod_info->big_values * 2 > cod_info->address2 ) +! { +! table = cod_info->table_select[2] = new_choose_table( ix, cod_info->address2, cod_info->big_values * 2, &s ); +! if( table != 0 ) /* region2 */ +! bits += s; +! } + } + return bits; + } +*************** +*** 1916,1930 **** + + int count_bit( int ix[576], unsigned int start, unsigned int end, unsigned int table ) + { +! int i, sum; + + sum = 0; +! for ( i = start; i < end; i += 2 ) +! { +! unsigned int code, ext; +! int cbits, xbits; +! sum += HuffmanCode( table, ix[i], ix[i+1], &code, &ext, &cbits, &xbits ); + } + return sum; + } + +--- 2359,2414 ---- + + int count_bit( int ix[576], unsigned int start, unsigned int end, unsigned int table ) + { +! unsigned linbits; +! register int i, sum; +! register int x,y; +! struct huffcodetab *h; +! unsigned char *hlen; + ++ if(table==0) return 0; ++ h = &(ht[table]); + sum = 0; +! +! linbits = h->linbits; +! hlen=h->hlen; +! +! if(table>15) +! { /* ESC-table is used */ +! for(i=start;i<end;i+=2) +! { +! x = ix[i]; +! y = ix[i+1]; +! +! if(x>14) +! { +! x = 15; +! sum += linbits; +! } +! if(y>14) +! { +! y = 15; +! sum += linbits; +! } +! +! sum += hlen[(x*16)+y]; +! if(x!=0) sum++; +! if(y!=0) sum++; +! } +! } +! else +! { /* No ESC-words */ +! for(i=start;i<end;i+=2) +! { +! x = ix[i]; +! y = ix[i+1]; +! +! sum += hlen[(x*16)+y]; +! +! if(x!=0) sum++; +! if(y!=0) sum++; +! } + } ++ + return sum; + } + +*************** +*** 1941,1951 **** + + return(temp); + } +- +- double +- aint(double in) { +- return((long) in); +- } + #endif + + +--- 2425,2430 ---- +*************** +*** 1967,1974 **** + to account for wider frequency line.) + */ + +! +! + /*************************************************************************/ + /* gr_deco */ + /*************************************************************************/ +--- 2446,2455 ---- + to account for wider frequency line.) + */ + +! /* mt 4/99: ISO code cannot produces mixed blocks, +! * Fhg Code also never seems to use them, so no need to add them +! * to this code +! */ + /*************************************************************************/ + /* gr_deco */ + /*************************************************************************/ +*************** +*** 1976,1993 **** + void gr_deco( gr_info *cod_info ) + { + if ( cod_info->window_switching_flag != 0 && cod_info->block_type == 2 ) +- if ( cod_info->mixed_block_flag == 0 ) + { + cod_info->sfb_lmax = 0; /* No sb*/ + cod_info->sfb_smax = 0; + } +- else +- { +- cod_info->sfb_lmax = 8; +- cod_info->sfb_smax = 3; +- } + else + { + cod_info->sfb_lmax = SFB_LMAX - 1; + cod_info->sfb_smax = SFB_SMAX - 1; /* No sb */ + } +--- 2457,2469 ---- + void gr_deco( gr_info *cod_info ) + { + if ( cod_info->window_switching_flag != 0 && cod_info->block_type == 2 ) + { + cod_info->sfb_lmax = 0; /* No sb*/ + cod_info->sfb_smax = 0; + } + else + { ++ /* MPEG 1 doesnt use last scalefactor band? */ + cod_info->sfb_lmax = SFB_LMAX - 1; + cod_info->sfb_smax = SFB_SMAX - 1; /* No sb */ + } +*************** +*** 2000,2220 **** + + + +- /* The following optional code written by Seymour Shlien +- will speed up the outer_loop code which is called +- by iteration_loop. When BIN_SEARCH is defined, the +- outer_loop function precedes the call to the function inner_loop +- with a call to bin_search gain defined below, which +- returns a good starting quantizerStepSize. +- */ + +- #if defined(BIN_SEARCH) || defined(PERFORM) + int count_bits(ix,cod_info) + int *ix; /* I576 *ix; */ + gr_info *cod_info; + { +! int bits,max; +! calc_runlen(ix,cod_info); /*rzero,count1,big_values*/ +! max = ix_max( ix, 0,576); +! if(max > 8192) return 100000; /* report unsuitable quantizer */ + bits = count1_bitcount(ix, cod_info); /*count1_table selection*/ + subdivide(cod_info); /* bigvalues sfb division */ +- bigv_tab_select(ix,cod_info); /* codebook selection*/ + bits += bigv_bitcount(ix,cod_info); /* bit count */ + /* printf("\nglobal_gain = %f bits= %d ",cod_info->quantizerStepSize,bits);*/ + return bits; + } +- #endif + + +- #ifdef BIN_SEARCH + +! int bin_search_StepSize(int desired_rate, double start, int *ix, +! double xrs[576], gr_info * cod_info) + { +! double top,bot,next,last; + int bit; + top = start; +- bot = 200; + next = start; + do + { + last = next; +! next = aint((top+bot)/2.0); + cod_info->quantizerStepSize = next; +! quantize(xrs,ix,cod_info); + bit = count_bits(ix,cod_info); + if (bit>desired_rate) top = next; + else bot = next; +! /* printf("\n%f %f %f %d %d",next, top,bot,bit,desired_rate);*/ +! } +! while ((bit != desired_rate) && fabs(last - next) > 1.0); +! return next; +! } +! +! #endif +! +! +! +! +! #ifdef PERFORM +! +! /* The following code is used for exposing some problems with +! the outer_loop code. PERFORM should be defined to the +! frame number you wish to have additional output recorded +! in the file encode.log - Seymour Shlien 14-Jan-97 +! */ +! +! /* +! float worst_xfsf_to_xmin_ratio(l3_xmin,xfsf,block_type,gr,ch) +! double xfsf[4][CBLIMIT]; +! III_psy_xmin *l3_xmin; +! int block_type,gr,ch; +! */ +! float worst_xfsf_to_xmin_ratio(III_psy_xmin *l3_xmin, double xfsf[4][CBLIMIT] +! ,int block_type,int gr,int ch) +! +! { +! float ratio,maxratio; +! int i,j; +! maxratio =-100.0; +! if (block_type != 2) +! for(i=0;i<21;i++) +! { +! ratio = 10.*log10(xfsf[0][i] /l3_xmin->l[gr][ch][i]); +! if (ratio > maxratio) maxratio = ratio; +! } +! else +! { +! for(j=0;j<3;j++) +! /* for(i = cod_info->sfb_smax; i <SFB_SMAX; i++) */ +! for(i = 0; i <11; i++) +! { +! ratio = 10.*log10(xfsf[j+1][i] /l3_xmin->s[gr][ch][i][j]); +! if (ratio > maxratio) maxratio = ratio; +! } +! } +! return maxratio; +! } +! +! +! print_ratios(handle_out,l3_xmin,xfsf,block_type,gr,ch) +! FILE *handle_out; +! double xfsf[4][CBLIMIT]; +! III_psy_xmin *l3_xmin; +! int gr,ch; +! int block_type; +! { +! float ratio; +! int i,j; +! if(block_type !=2) +! for (i=0;i<21;i++) +! { +! ratio = 100.0; /* signals undefined value in output */ +! if(l3_xmin->l[gr][ch][i] >1.0e-20) +! ratio = 10.*log10(xfsf[0][i] /l3_xmin->l[gr][ch][i]); +! fprintf(handle_out,"%6.2f ",ratio); +! if(i%5==4) fprintf(handle_out,"\n"); +! } +! else +! +! for(j=0;j<3;j++) +! { fprintf(handle_out,"\n block %d\n",j); +! for(i = 0; i <11; i++) +! { +! ratio = 10.*log10(xfsf[j+1][i] /l3_xmin->s[gr][ch][i][j]); +! fprintf(handle_out,"%6.2f ",ratio); +! if(i%5==4) fprintf(handle_out,"\n"); +! } +! } +! fprintf(handle_out,"\n"); +! } +! +! +! print_scalefacs(handle_out,scalefac,block_type,gr,ch) +! FILE *handle_out; +! III_scalefac_t *scalefac; +! int gr,ch; +! int block_type; +! { +! int sfb,j; +! +! if(block_type !=2) +! for ( sfb = 0; sfb < 21; sfb++ ) +! { +! fprintf(handle_out,"%6d ", scalefac->l[gr][ch][sfb]); +! if(sfb%5==4) fprintf(handle_out,"\n"); +! } +! else +! for (j=0;j<3;j++) +! { +! fprintf(handle_out,"\n block %d\n",j); +! for (sfb=0;sfb<11;sfb++) +! { +! fprintf(handle_out,"%6d ",scalefac->s[gr][ch][sfb][j]); +! if(sfb%5==4) fprintf(handle_out,"\n"); +! } + } +! fprintf(handle_out,"\n"); +! } + +! +! print_quantized_values(FILE *handle, int ix[576], gr_info *cod_info) +! { +! int sfb,start,end,i,bw; +! for (sfb=0;sfb<cod_info->sfb_lmax;sfb++) +! { +! start = scalefac_band_long[sfb]; +! end = scalefac_band_long[sfb+1]; +! bw = end - start; +! fprintf(handle,"scalefac band %d from %d to %d\n",sfb,start,end); +! for (i=0;i<bw;i++) +! { +! fprintf(handle,"%8d",ix[start+i]); +! if(i%5==4) fprintf(handle,"\n"); +! } +! fprintf(handle,"\n"); +! } + } + + +- test_inner_loop(double xr[2][2][576], int l3_enc[2][2][576], int max_bits, +- gr_info *cod_info, int gr, int ch, double xfsf[4][CBLIMIT], +- III_psy_xmin *l3_xmin) +- { +- int bits, c1bits, bvbits; +- double *xrs; /* D576 *xr; */ +- int *ix; /* I576 *ix; */ +- xrs = &xr[gr][ch][0]; +- ix = l3_enc[gr][ch]; +- +- assert( max_bits >= 0 ); +- cod_info->quantizerStepSize -= 1.0;; +- do +- { +- do +- { +- cod_info->quantizerStepSize += 1.0; +- quantize( xrs, ix, cod_info ); +- } +- while ( ix_max(ix, 0, 576) > 8191 + 14 ); /* within table range? */ +- +- bits = count_bits(ix,cod_info); + +- if(frameNum == PERFORM) +- { +- fprintf(log_output,"StepSize=%f bits = %d huff_bits = %d\n", +- cod_info->quantizerStepSize,bits,max_bits); +- calc_noise( &xr[gr][ch][0], &l3_enc[gr][ch][0], cod_info, xfsf ); +- /* distortion calculation */ +- print_ratios(log_output,l3_xmin,xfsf,cod_info->block_type,gr,ch); +- fprintf(log_output,"\n\n"); +- } +- +- } +- while ( bits > max_bits ); +- +- return bits; +- } +- +- #endif PERFORM +--- 2476,2594 ---- + + + + + int count_bits(ix,cod_info) + int *ix; /* I576 *ix; */ + gr_info *cod_info; + { +! int bits,i; +! for ( i = 0; i < 576; i++ ) +! { +! if ( ix[i] > 8191 + 14) +! return 100000; /* report unsuitable quantizer */ +! } +! calc_runlen(ix,cod_info); /*count1,big_values*/ + bits = count1_bitcount(ix, cod_info); /*count1_table selection*/ + subdivide(cod_info); /* bigvalues sfb division */ + bits += bigv_bitcount(ix,cod_info); /* bit count */ + /* printf("\nglobal_gain = %f bits= %d ",cod_info->quantizerStepSize,bits);*/ + return bits; + } + + + +! static int OldValue = -30; /* guess it or so. */ +! +! typedef enum { +! BINSEARCH_NONE, +! BINSEARCH_UP, +! BINSEARCH_DOWN +! } binsearchDirection_t; +! +! /*-------------------------------------------------------------------------*/ +! int +! bin_search_StepSize2(int desired_rate, +! double start, +! int bot, +! int *ix, +! double xrs[576], +! double xrspow[576], +! gr_info *cod_info) +! /*-------------------------------------------------------------------------*/ +! { +! int flag_GoneOver = 0; +! int CurrentStep = 4; +! int nBits, Counter = 0; +! int StepSize = OldValue; +! binsearchDirection_t Direction = BINSEARCH_NONE; +! do +! { +! cod_info->quantizerStepSize = StepSize; +! quantize_xrpow(xrspow, ix, cod_info); +! nBits = count_bits(ix,cod_info); +! +! if (CurrentStep == 1 || Counter > 50) +! /* the 50 is for ''lock-up'' - havent found any.*/ +! { +! break; /* nothing to adjust anymore */ +! } +! if (flag_GoneOver) +! { +! CurrentStep /= 2; +! } +! if (nBits > desired_rate) /* increase Quantize_StepSize */ +! { +! if (Direction == BINSEARCH_DOWN && !flag_GoneOver) +! { +! flag_GoneOver = 1; +! CurrentStep /= 2; /* late adjust */ +! } +! Direction = BINSEARCH_UP; +! StepSize += CurrentStep; +! } +! else if (nBits < desired_rate) +! { +! if (Direction == BINSEARCH_UP && !flag_GoneOver) +! { +! flag_GoneOver = 1; +! CurrentStep /= 2; /* late adjust */ +! } +! Direction = BINSEARCH_DOWN; +! StepSize -= CurrentStep; +! } +! else break; /* nBits == desired_rate;; most unlikely to happen. +! */ +! } while (1); /* For-ever, break is adjusted. */ +! OldValue = StepSize; +! return nBits; +! } +! +! +! +! +! int bin_search_StepSize(int desired_rate, double start, int bot, int *ix, +! double xrs[576], double xrspow[576], gr_info * cod_info) + { +! int top,next,last; + int bit; + top = start; + next = start; + do + { + last = next; +! next = (top+bot)/2.0; + cod_info->quantizerStepSize = next; +! quantize_xrpow(xrspow,ix,cod_info); + bit = count_bits(ix,cod_info); + if (bit>desired_rate) top = next; + else bot = next; +! // printf("\n%f %f %f %f %d %d",start,next, top,bot,bit,desired_rate); + } +! while ((bit != desired_rate) && abs(last - next) > 1); +! //printf("\n done %f %d %d",next,bit,desired_rate); + +! return bit; + } + + + +diff -r -c -N encoder/loop.h lame3.13/loop.h +*** encoder/loop.h Wed Jan 22 02:43:16 1997 +--- lame3.13/loop.h Sat Jun 5 16:46:15 1999 +*************** +*** 45,55 **** + extern struct scalefac_struct sfBandIndex[]; /* Table B.8 -- in loop.c */ + + +! void iteration_loop( double pe[][2], double xr_org[2][2][576], III_psy_ratio *ratio, +! III_side_info_t *l3_side, int l3_enc[2][2][576], int mean_bits, +! int stereo, double xr_dec[2][2][576], +! III_scalefac_t *scalefac, frame_params *fr_ps, +! int ancillary_pad, int bitsPerFrame ); + + + int nint( double in ); +--- 45,54 ---- + extern struct scalefac_struct sfBandIndex[]; /* Table B.8 -- in loop.c */ + + +! void iteration_loop( double pe[][2], double ms_ratio[2], +! double xr_org[2][2][576], III_psy_ratio *ratio, +! III_side_info_t *l3_side, int l3_enc[2][2][576], +! III_scalefac_t *scalefac, frame_params *fr_ps); + + + int nint( double in ); +diff -r -c -N encoder/makefile.unix lame3.13/makefile.unix +*** encoder/makefile.unix Wed Jan 22 02:43:23 1997 +--- lame3.13/makefile.unix Wed Dec 31 17:00:00 1969 +*************** +*** 1,77 **** +- ############################################################################ +- ## ISO MPEG Audio Subgroup Software Simulation Group (1996) +- ## ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- ## +- ## $Id: makefile.unix,v 1.1 1996/02/14 05:28:25 rowlands Exp $ +- ## +- ## Makefile for encoder for vanilla UNIX platform using standard make. +- ## +- ## $Log: makefile.unix,v $ +- ## Revision 1.1 1996/02/14 05:28:25 rowlands +- ## Initial revision +- ## +- ############################################################################ +- +- # MODE= -O -DUNIX +- MODE= -g -DUNIX +- #MODE= -DMSC60 -AH -Zi -Gt +- OBJ_SUFFIX=.o +- EXE_SUFFIX= +- +- PROGRAM = encode$(EXE_SUFFIX) +- +- all: $(PROGRAM) +- +- $(PROGRAM) : \ +- common.c common.h encode.c encoder.h formatBitstream.c \ +- formatBitstream.h huffman.c huffman.h ieeefloat.c ieeefloat.h \ +- l3bitstream-pvt.h l3bitstream.c l3bitstream.h l3psy.c l3psy.h \ +- l3side.h loop-pvt.h loop.c loop.h mdct.c mdct.h musicin.c \ +- portableio.c portableio.h psy.c reservoir.c reservoir.h subs.c tonal.c +- +- HEDS = \ +- common.h encoder.h formatBitstream.h huffman.h ieeefloat.h \ +- l3bitstream-pvt.h l3bitstream.h l3psy.h l3side.h loop-pvt.h loop.h \ +- mdct.h portableio.h reservoir.h +- +- OBJS = \ +- common$(OBJ_SUFFIX) \ +- encode$(OBJ_SUFFIX) \ +- formatBitstream$(OBJ_SUFFIX) \ +- huffman$(OBJ_SUFFIX) \ +- ieeefloat$(OBJ_SUFFIX) \ +- l3bitstream$(OBJ_SUFFIX) \ +- l3psy$(OBJ_SUFFIX) \ +- loop$(OBJ_SUFFIX) \ +- mdct$(OBJ_SUFFIX) \ +- musicin$(OBJ_SUFFIX) \ +- portableio$(OBJ_SUFFIX) \ +- psy$(OBJ_SUFFIX) \ +- reservoir$(OBJ_SUFFIX) \ +- subs$(OBJ_SUFFIX) \ +- tonal$(OBJ_SUFFIX) +- +- LFLAG = +- +- CC = gcc +- LN = gcc +- LNEXTRA=-lm +- #CC = cl +- #LN = cl +- +- .c$(OBJ_SUFFIX): $(HEDS) +- $(CC) -c $(MODE) $*.c +- +- all: $(PROGRAM) +- +- default: all +- +- +- $(PROGRAM) : $(OBJS) $(HEDS) +- $(LN) -o $(PROGRAM) $(MODE) $(LFLAG) $(OBJS) $(LNEXTRA) +- +- clean: +- rm -f $(OBJS) +- +- veryclean: +- rm -f $(PROGRAM) $(OBJS) +--- 0 ---- +diff -r -c -N encoder/mdct.c lame3.13/mdct.c +*** encoder/mdct.c Wed Jan 22 02:43:17 1997 +--- lame3.13/mdct.c Tue Jun 1 07:38:54 1999 +*************** +*** 54,62 **** + /* + Compensate for inversion in the analysis filter + */ +! for ( band = 0; band < 32; band++ ) +! for ( k = 0; k < 18; k++ ) +! if ( (band & 1) && (k & 1) ) + (*sb_sample)[ch][gr+1][k][band] *= -1.0; + + /* +--- 54,61 ---- + /* + Compensate for inversion in the analysis filter + */ +! for ( band = 1; band < 32; band+=2 ) +! for ( k = 1; k < 18; k+=2 ) + (*sb_sample)[ch][gr+1][k][band] *= -1.0; + + /* +*************** +*** 70,78 **** + mdct_in[k] = (*sb_sample)[ch][ gr ][k][band]; + mdct_in[k+18] = (*sb_sample)[ch][gr+1][k][band]; + } + if ( cod_info->mixed_block_flag && (band < 2) ) + block_type = 0; +! + mdct( mdct_in, &mdct_enc[gr][ch][band][0], block_type ); + } + +--- 69,78 ---- + mdct_in[k] = (*sb_sample)[ch][ gr ][k][band]; + mdct_in[k+18] = (*sb_sample)[ch][gr+1][k][band]; + } ++ #ifdef ALLOW_MIXED + if ( cod_info->mixed_block_flag && (band < 2) ) + block_type = 0; +! #endif + mdct( mdct_in, &mdct_enc[gr][ch][band][0], block_type ); + } + +*************** +*** 123,130 **** + double sum; + static double win[4][36]; + static int init = 0; +! static double cos_s[6][12], cos_l[18][36]; +! + + if ( init == 0 ) + { +--- 123,131 ---- + double sum; + static double win[4][36]; + static int init = 0; +! static double cos_s[6][12], cos_l[18][18]; +! static double fin[18]; +! static double *cos_l0; + + if ( init == 0 ) + { +*************** +*** 149,176 **** + win[3][i] = 1.0; + for ( i = 18; i < 36; i++ ) + win[3][i] = sin( PI/36 * (i + 0.5) ); +- /* type 2*/ +- for ( i = 0; i < 12; i++ ) +- win[2][i] = sin( PI/12 * (i + 0.5) ); +- for ( i = 12; i < 36; i++ ) +- win[2][i] = 0.0; + + N = 12; + for ( m = 0; m < N / 2; m++ ) + for ( k = 0; k < N; k++ ) + cos_s[m][k] = cos( (PI /(2 * N)) * (2 * k + 1 + N / 2) * +! (2 * m + 1) ) / (N / 4); + + N = 36; + for ( m = 0; m < N / 2; m++ ) +! for ( k = 0; k < N; k++ ) + cos_l[m][k] = cos( (PI / (2 * N)) * (2 * k + 1 + N / 2) * + (2 * m + 1) ) / (N / 4); +! + init++; + } + +! if ( block_type == 2 ) + { + N = 12; + for ( l = 0; l < 3; l++ ) +--- 150,177 ---- + win[3][i] = 1.0; + for ( i = 18; i < 36; i++ ) + win[3][i] = sin( PI/36 * (i + 0.5) ); + + N = 12; + for ( m = 0; m < N / 2; m++ ) + for ( k = 0; k < N; k++ ) + cos_s[m][k] = cos( (PI /(2 * N)) * (2 * k + 1 + N / 2) * +! (2 * m + 1) ) / (N / 4) * sin( PI/12 * (k + 0.5) ); + + N = 36; + for ( m = 0; m < N / 2; m++ ) +! { +! for ( k = 0; k < 9; k++ ) +! { + cos_l[m][k] = cos( (PI / (2 * N)) * (2 * k + 1 + N / 2) * + (2 * m + 1) ) / (N / 4); +! cos_l[m][9+k] = cos( (PI / (2 * N)) * (2 * (18 + k) + 1 + N / 2) * +! (2 * m + 1) ) / (N / 4); +! } +! } + init++; + } + +! if ( block_type == 2) + { + N = 12; + for ( l = 0; l < 3; l++ ) +*************** +*** 178,212 **** + for ( m = 0; m < N / 2; m++ ) + { + for ( sum = 0.0, k = 0; k < N; k++ ) +! sum += win[block_type][k] * in[k + 6 * l + 6] * cos_s[m][k]; + out[ 3 * m + l] = sum; + } + } + } +! else + { +! N = 36; +! for ( m = 0; m < N / 2; m++ ) +! { +! for ( sum = 0.0, k = 0; k < N; k++ ) +! sum += win[block_type][k] * in[k] * cos_l[m][k]; +! out[m] = sum; +! } +! } +! } + +! void +! delay( double (*xr)[2][576], int stereo ) +! { +! static double xr_buff[2][576]; +! double xr_buff2[2][576]; +! unsigned int i,j; +! +! for (i=0;i<stereo;i++) +! { +! for (j=0;j<576;j++) xr_buff2[i][j] = xr_buff[i][j]; +! for (j=0;j<576;j++) xr_buff[i][j] = xr[1][i][j]; +! for (j=0;j<576;j++) xr[1][i][j] = xr[0][i][j]; +! for (j=0;j<576;j++) xr[0][i][j] = xr_buff2[i][j]; +! } + } +--- 179,508 ---- + for ( m = 0; m < N / 2; m++ ) + { + for ( sum = 0.0, k = 0; k < N; k++ ) +! sum += in[k + 6 * l + 6] * cos_s[m][k]; + out[ 3 * m + l] = sum; + } + } + } +! else + { +! for (k=0;k<9;k++) +! { +! fin[k] = win[block_type][k] * in[k] - win[block_type][17-k] * in[17-k]; +! fin[9+k] = win[block_type][18+k] * in[18+k] + win[block_type][35-k] * in[35-k]; +! } + +! /* 0 */ +! cos_l0= &cos_l[0][0]; +! sum = ( fin[0] ) * *cos_l0; /* 17 */ +! sum += ( fin[1] ) * *++cos_l0; /* 15 */ +! sum += ( fin[2] ) * *++cos_l0; /* 13 */ +! sum += ( fin[3] ) * *++cos_l0; /* 11 */ +! sum += ( fin[4] ) * *++cos_l0; /* 9 */ +! sum += ( fin[5] ) * *++cos_l0; /* 7 */ +! sum += ( fin[6] ) * *++cos_l0; /* 5 */ +! sum += ( fin[7] ) * *++cos_l0; /* 3 */ +! sum += ( fin[8] ) * *++cos_l0; /* 1 */ +! sum += ( fin[9] ) * *++cos_l0; /* 19*/ +! sum += ( fin[10] ) * *++cos_l0; /* 21 */ +! sum += ( fin[11] ) * *++cos_l0; /* 23 */ +! sum += ( fin[12] ) * *++cos_l0; /*25 */ +! sum += ( fin[13] ) * *++cos_l0; /* 27 */ +! sum += ( fin[14] ) * *++cos_l0; /* 29 */ +! sum += ( fin[15] ) * *++cos_l0; /* 31*/ +! sum += ( fin[16] ) * *++cos_l0; /* 33 */ +! sum += ( fin[17] ) * *++cos_l0; /* 35 */ +! out[0]=sum; +! +! +! /* 1 */ +! sum = ( fin[2]+fin[3]+fin[17] ) * cos_l[1][2]; /* mfc=3 2*/ +! sum += ( fin[1]+fin[4]+fin[16] ) * cos_l[1][1]; /* mfc=9 1*/ +! sum += ( fin[0]+fin[5]+fin[15] ) * *++cos_l0; /* mfc=15 0*/ +! sum += ( fin[6]-fin[9]+fin[14] ) * *(cos_l0+=6); /* mfc=21 6*/ +! sum += ( fin[7]-fin[10]+fin[13] ) * *++cos_l0; /* mfc=27 7*/ +! sum += ( fin[8]-fin[11]+fin[12] ) * *++cos_l0; /* mfc = 28 8*/ +! out[1]=sum; +! +! /* 2 */ +! cos_l0= &cos_l[2][0]; +! sum = ( fin[0] ) * *cos_l0; /* mfc 23 */ +! sum += ( fin[1] ) * *++cos_l0; /* mfc 33 */ +! sum += ( fin[2] ) * *++cos_l0; /* mfc 29 */ +! sum += ( fin[3] ) * *++cos_l0; /* mfc 19 */ +! sum += ( fin[4] ) * *++cos_l0; /* mfc 9 */ +! sum += ( fin[5] ) * *++cos_l0; /* mfc 1 */ +! sum += ( fin[6] ) * *++cos_l0; /* mfc 11 */ +! sum += ( fin[7] ) * *++cos_l0; /* mfc 21 */ +! sum += ( fin[8] ) * *++cos_l0; /* mfc 31 */ +! sum += ( fin[9] ) * *++cos_l0; /* mfc 13 */ +! sum += ( fin[10] ) * *++cos_l0; /* mfc 3 */ +! sum += ( fin[11] ) * *++cos_l0; /* mfc 7 */ +! sum += ( fin[12] ) * *++cos_l0; /* mfc 17 */ +! sum += ( fin[13] ) * *++cos_l0; /* mfc 27 */ +! sum += ( fin[14] ) * *++cos_l0; /* mfc 35 */ +! sum += ( fin[15] ) * *++cos_l0; /* mfc 25 */ +! sum += ( fin[16] ) * *++cos_l0; /* mfc 15 */ +! sum += ( fin[17] ) * *++cos_l0; /* mfc 5 */ +! out[2]=sum; +! +! /* 3 */; +! sum = ( fin[0] ) * *++cos_l0; /* mfc 11 */ +! sum += ( fin[1] ) * *++cos_l0; /* mfc 3 */ +! sum += ( fin[2] ) * *++cos_l0; /* 17 */ +! sum += ( fin[3] ) * *++cos_l0; /* 31 */ +! sum += ( fin[4] ) * *++cos_l0; /* 27 */ +! sum += ( fin[5] ) * *++cos_l0; /* 13 */ +! sum += ( fin[6] ) * *++cos_l0; /* 1 */ +! sum += ( fin[7] ) * *++cos_l0; /* 15 */ +! sum += ( fin[8] ) * *++cos_l0; /* 29 */ +! sum += ( fin[9] ) * *++cos_l0; /* 25 */ +! sum += ( fin[10] ) * *++cos_l0; /* 33 */ +! sum += ( fin[11] ) * *++cos_l0; /* 19 */ +! sum += ( fin[12] ) * *++cos_l0; /* 5 */ +! sum += ( fin[13] ) * *++cos_l0; /* 9 */ +! sum += ( fin[14] ) * *++cos_l0; /* 23 */ +! sum += ( fin[15] ) * *++cos_l0; /* 35 */ +! sum += ( fin[16] ) * *++cos_l0; /* 21 */ +! sum += ( fin[17] ) * *++cos_l0; /* 7 */ +! out[3]=sum; +! +! /* 4 */ +! /* mfc 27 */ +! sum = ( fin[0]-fin[1]-fin[4]+fin[5]+fin[8] \ +! -fin[11]+fin[12]+fin[15]-fin[16] ) * *++cos_l0; +! /* mfc 9 */ +! sum += ( -fin[2]-fin[3]+fin[6]+fin[7]-fin[9]-fin[10] \ +! +fin[13]+fin[14]-fin[17] ) * cos_l[4][6]; +! out[4]=sum; +! +! /* 5 */ +! cos_l0= &cos_l[5][0]; +! sum = ( fin[0] ) * *cos_l0; /* 7 */ +! sum += ( fin[1] ) * *++cos_l0; /* 15 */ +! sum += ( fin[2] ) * *++cos_l0; /* 35 */ +! sum += ( fin[3] ) * *++cos_l0; /* 13 */ +! sum += ( fin[4] ) * *++cos_l0; /* 9 */ +! sum += ( fin[5] ) * *++cos_l0; /* 31 */ +! sum += ( fin[6] ) * *++cos_l0; /* 19 */ +! sum += ( fin[7] ) * *++cos_l0; /* 3 */ +! sum += ( fin[8] ) * *++cos_l0; /* 8 */ +! sum += ( fin[9] ) * *++cos_l0; /* 29 */ +! sum += ( fin[10] ) * *++cos_l0; /* 21 */ +! sum += ( fin[11] ) * *++cos_l0; /* 1 */ +! sum += ( fin[12] ) * *++cos_l0; /* 23 */ +! sum += ( fin[13] ) * *++cos_l0; /* 27 */ +! sum += ( fin[14] ) * *++cos_l0; /* 5 */ +! sum += ( fin[15] ) * *++cos_l0; /* 17 */ +! sum += ( fin[16] ) * *++cos_l0; /* 33 */ +! sum += ( fin[17] ) * *++cos_l0; /* 11 */ +! out[5]=sum; +! +! /* 6 */ +! sum = ( fin[0] ) * *++cos_l0; /* 31 */ +! sum += ( fin[1] ) * *++cos_l0; /* 15 */ +! sum += ( fin[2] ) * *++cos_l0; /* 11 */ +! sum += ( fin[3] ) * *++cos_l0; /* 35 */ +! sum += ( fin[4] ) * *++cos_l0; /* 9 */ +! sum += ( fin[5] ) * *++cos_l0; /* 17 */ +! sum += ( fin[6] ) * *++cos_l0; /* 29 */ +! sum += ( fin[7] ) * *++cos_l0; /* 3 */ +! sum += ( fin[8] ) * *++cos_l0; /* 23 */ +! sum += ( fin[9] ) * *++cos_l0; /* 5 */ +! sum += ( fin[10] ) * *++cos_l0; /* 21 */ +! sum += ( fin[11] ) * *++cos_l0; /* 25 */ +! sum += ( fin[12] ) * *++cos_l0; /* 1 */ +! sum += ( fin[13] ) * *++cos_l0; /* 27 */ +! sum += ( fin[14] ) * *++cos_l0; /* 19 */ +! sum += ( fin[15] ) * *++cos_l0; /* 7 */ +! sum += ( fin[16] ) * *++cos_l0; /* 33 */ +! sum += ( fin[17] ) * *++cos_l0; /* 13 */ +! out[6]=sum; +! +! /* 7 */ +! sum = ( fin[0]+fin[5]+fin[15] ) * *++cos_l0; /* 3 */ +! sum += ( fin[1]+fin[4]+fin[16] ) * *++cos_l0; /* 27*/ +! sum += ( fin[2]+fin[3]+fin[17] ) * *++cos_l0; /* 15 */ +! sum += ( fin[6]-fin[9]+fin[14] ) * *(cos_l0+=4); /* 33 */ +! sum += ( fin[7]-fin[10]+fin[13] ) * *++cos_l0; /* 9 */ +! sum += ( fin[8]-fin[11]+fin[12] ) * *++cos_l0; /* 21 */ +! out[7]=sum; +! +! /* 8 */ +! cos_l0= &cos_l[8][0]; +! sum = ( fin[0] ) * *cos_l0; /* 35 */ +! sum += ( fin[1] ) * *++cos_l0; /* 3 */ +! sum += ( fin[2] ) * *++cos_l0; /* 31 */ +! sum += ( fin[3] ) * *++cos_l0; /* 7 */ +! sum += ( fin[4] ) * *++cos_l0; /* 27*/ +! sum += ( fin[5] ) * *++cos_l0; /* 11 */ +! sum += ( fin[6] ) * *++cos_l0; /* 23 */ +! sum += ( fin[7] ) * *++cos_l0; /* 15*/ +! sum += ( fin[8] ) * *++cos_l0; /* 19 */ +! sum += ( fin[9] ) * *++cos_l0; /* 1 */ +! sum += ( fin[10] ) * *++cos_l0; /* 33 */ +! sum += ( fin[11] ) * *++cos_l0; /* 5 */ +! sum += ( fin[12] ) * *++cos_l0; /* 29*/ +! sum += ( fin[13] ) * *++cos_l0; /* 9 */ +! sum += ( fin[14] ) * *++cos_l0; /* 25 */ +! sum += ( fin[15] ) * *++cos_l0; /* 13*/ +! sum += ( fin[16] ) * *++cos_l0; /* 21 */ +! sum += ( fin[17] ) * *++cos_l0; /* 17 */ +! out[8]=sum; +! +! /* 9 */ +! sum = ( fin[0] ) * *++cos_l0; /* 1 */ +! sum += ( fin[1] ) * *++cos_l0; /* 33 */ +! sum += ( fin[2] ) * *++cos_l0; /* 5 */ +! sum += ( fin[3] ) * *++cos_l0; /* 29*/ +! sum += ( fin[4] ) * *++cos_l0; /* 9 */ +! sum += ( fin[5] ) * *++cos_l0; /* 25 */ +! sum += ( fin[6] ) * *++cos_l0; /* 13 */ +! sum += ( fin[7] ) * *++cos_l0; /* 21*/ +! sum += ( fin[8] ) * *++cos_l0; /* 17 */ +! sum += ( fin[9] ) * *++cos_l0; /* 35*/ +! sum += ( fin[10] ) * *++cos_l0; /* 3 */ +! sum += ( fin[11] ) * *++cos_l0; /* 31 */ +! sum += ( fin[12] ) * *++cos_l0; /* 7 */ +! sum += ( fin[13] ) * *++cos_l0; /* 27 */ +! sum += ( fin[14] ) * *++cos_l0; /* 11 */ +! sum += ( fin[15] ) * *++cos_l0; /* 23*/ +! sum += ( fin[16] ) * *++cos_l0; /* 15 */ +! sum += ( fin[17] ) * *++cos_l0; /* 19 */ +! out[9]=sum; +! +! /* 10 */ +! sum = ( fin[0]+fin[5]+fin[15] ) * *++cos_l0; /* 33 */ +! sum += ( fin[1]+fin[4]+fin[16] ) * *++cos_l0; /* 9 */ +! sum += ( fin[2]+fin[3]+fin[17] ) * *++cos_l0; /* 21 */ +! sum += ( fin[6]-fin[9]+fin[14] ) * *(cos_l0+=4); /* 3 */ +! sum += ( fin[7]-fin[10]+fin[13] ) * *++cos_l0; /* 27 */ +! sum += ( fin[8]-fin[11]+fin[12] ) * *++cos_l0; /* 15 */ +! out[10]=sum; +! +! /* 11 */ +! cos_l0= &cos_l[11][0]; +! sum = ( fin[0] ) * *cos_l0; /* 5 */ +! sum += ( fin[1] ) * *++cos_l0; /* 21 */ +! sum += ( fin[2] ) * *++cos_l0; /* 25 */ +! sum += ( fin[3] ) * *++cos_l0; /* 1 */ +! sum += ( fin[4] ) * *++cos_l0; /*27 */ +! sum += ( fin[5] ) * *++cos_l0; /* 19 */ +! sum += ( fin[6] ) * *++cos_l0; /* 7 */ +! sum += ( fin[7] ) * *++cos_l0; /* 33*/ +! sum += ( fin[8] ) * *++cos_l0; /* 13 */ +! sum += ( fin[9] ) * *++cos_l0; /* 31*/ +! sum += ( fin[10] ) * *++cos_l0; /* 15 */ +! sum += ( fin[11] ) * *++cos_l0; /* 11 */ +! sum += ( fin[12] ) * *++cos_l0; /*35 */ +! sum += ( fin[13] ) * *++cos_l0; /* 9 */ +! sum += ( fin[14] ) * *++cos_l0; /* 17 */ +! sum += ( fin[15] ) * *++cos_l0; /* 29*/ +! sum += ( fin[16] ) * *++cos_l0; /* 3 */ +! sum += ( fin[17] ) * *++cos_l0; /* 23 */ +! out[11]=sum; +! +! /* 12 */ +! sum = ( fin[0] ) * *++cos_l0; /* 29 */ +! sum += ( fin[1] ) * *++cos_l0; /* 21 */ +! sum += ( fin[2] ) * *++cos_l0; /* 1 */ +! sum += ( fin[3] ) * *++cos_l0; /* 23 */ +! sum += ( fin[4] ) * *++cos_l0; /*27 */ +! sum += ( fin[5] ) * *++cos_l0; /* 5 */ +! sum += ( fin[6] ) * *++cos_l0; /* 17 */ +! sum += ( fin[7] ) * *++cos_l0; /* 33*/ +! sum += ( fin[8] ) * *++cos_l0; /* 11 */ +! sum += ( fin[9] ) * *++cos_l0; /* 7 */ +! sum += ( fin[10] ) * *++cos_l0; /* 15 */ +! sum += ( fin[11] ) * *++cos_l0; /* 35 */ +! sum += ( fin[12] ) * *++cos_l0; /*13 */ +! sum += ( fin[13] ) * *++cos_l0; /* 9 */ +! sum += ( fin[14] ) * *++cos_l0; /* 31 */ +! sum += ( fin[15] ) * *++cos_l0; /* 19*/ +! sum += ( fin[16] ) * *++cos_l0; /* 3 */ +! sum += ( fin[17] ) * *++cos_l0; /* 25 */ +! out[12]=sum; +! +! /* 13 */ +! sum = ( -fin[0]+fin[1]+fin[4]-fin[5]-fin[8]+fin[11]-fin[12] \ +! -fin[15]+fin[16] ) * -*++cos_l0; /* 9 */ +! sum += ( -fin[2]-fin[3]+fin[6]+fin[7]-fin[9]-fin[10]+fin[13] \ +! +fin[14]-fin[17] ) * -cos_l[13][2]; /* 27 */ +! out[13]=sum; +! +! /* 14 */ +! cos_l0= &cos_l[14][0]; +! sum = ( fin[0] ) * *cos_l0; /* 25 */ +! sum += ( fin[1] ) * *++cos_l0; /* 33 */ +! sum += ( fin[2] ) * *++cos_l0; /* 19 */ +! sum += ( fin[3] ) * *++cos_l0; /* 5 */ +! sum += ( fin[4] ) * *++cos_l0; /* 9 */ +! sum += ( fin[5] ) * *++cos_l0; /* 23 */ +! sum += ( fin[6] ) * *++cos_l0; /* 35 */ +! sum += ( fin[7] ) * *++cos_l0; /* 21*/ +! sum += ( fin[8] ) * *++cos_l0; /* 7 */ +! sum += ( fin[9] ) * *++cos_l0; /* 11*/ +! sum += ( fin[10] ) * *++cos_l0; /* 3 */ +! sum += ( fin[11] ) * *++cos_l0; /* 17 */ +! sum += ( fin[12] ) * *++cos_l0; /*31 */ +! sum += ( fin[13] ) * *++cos_l0; /* 27 */ +! sum += ( fin[14] ) * *++cos_l0; /* 13 */ +! sum += ( fin[15] ) * *++cos_l0; /* 1 */ +! sum += ( fin[16] ) * *++cos_l0; /* 15 */ +! sum += ( fin[17] ) * *++cos_l0; /* 29 */ +! out[14]=sum; +! +! /* 15 */ +! sum = ( fin[0] ) * *++cos_l0; /* 13 */ +! sum += ( fin[1] ) * *++cos_l0; /* 3 */ +! sum += ( fin[2] ) * *++cos_l0; /* 7 */ +! sum += ( fin[3] ) * *++cos_l0; /* 17 */ +! sum += ( fin[4] ) * *++cos_l0; /* 27*/ +! sum += ( fin[5] ) * *++cos_l0; /* 35 */ +! sum += ( fin[6] ) * *++cos_l0; /* 25 */ +! sum += ( fin[7] ) * *++cos_l0; /* 15*/ +! sum += ( fin[8] ) * *++cos_l0; /* 5 */ +! sum += ( fin[9] ) * *++cos_l0; /* 23*/ +! sum += ( fin[10] ) * *++cos_l0; /* 33 */ +! sum += ( fin[11] ) * *++cos_l0; /* 29 */ +! sum += ( fin[12] ) * *++cos_l0; /*19 */ +! sum += ( fin[13] ) * *++cos_l0; /* 9 */ +! sum += ( fin[14] ) * *++cos_l0; /* 1 */ +! sum += ( fin[15] ) * *++cos_l0; /* 11*/ +! sum += ( fin[16] ) * *++cos_l0; /* 21 */ +! sum += ( fin[17] ) * *++cos_l0; /* 31 */ +! out[15]=sum; +! +! /* 16 */ +! sum = ( fin[0]+fin[5]+fin[15] ) * *++cos_l0; /* 21 */ +! sum += ( fin[1]+fin[4]+fin[16] ) * *++cos_l0; /* 27 */ +! sum += ( fin[2]+fin[3]+fin[17] ) * *++cos_l0; /* 33 */ +! sum += ( fin[6]-fin[9]+fin[14] ) * *(cos_l0+=4); /* 15 */ +! sum += ( fin[7]-fin[10]+fin[13] ) * *++cos_l0; /* 9 */ +! sum += ( fin[8]-fin[11]+fin[12] ) * *++cos_l0; /* 3 */ +! out[16]=sum; +! +! /* 17 */ +! cos_l0= &cos_l[17][0]; +! sum = ( fin[0] ) * *cos_l0; /* 17 */ +! sum += ( fin[1] ) * *++cos_l0; /* 15 */ +! sum += ( fin[2] ) * *++cos_l0; /* 13 */ +! sum += ( fin[3] ) * *++cos_l0; /* 11 */ +! sum += ( fin[4] ) * *++cos_l0; /* 9 */ +! sum += ( fin[5] ) * *++cos_l0; /* 7 */ +! sum += ( fin[6] ) * *++cos_l0; /* 5 */ +! sum += ( fin[7] ) * *++cos_l0; /* 3 */ +! sum += ( fin[8] ) * *++cos_l0; /* 1 */ +! sum += ( fin[9] ) * *++cos_l0; /* 19*/ +! sum += ( fin[10] ) * *++cos_l0; /* 21 */ +! sum += ( fin[11] ) * *++cos_l0; /* 23 */ +! sum += ( fin[12] ) * *++cos_l0; /*25 */ +! sum += ( fin[13] ) * *++cos_l0; /* 27 */ +! sum += ( fin[14] ) * *++cos_l0; /* 29 */ +! sum += ( fin[15] ) * *++cos_l0; /* 31*/ +! sum += ( fin[16] ) * *++cos_l0; /* 33 */ +! sum += ( fin[17] ) * *++cos_l0; /* 35 */ +! out[17]=sum; +! } + } ++ +diff -r -c -N encoder/mlame lame3.13/mlame +*** encoder/mlame Wed Dec 31 17:00:00 1969 +--- lame3.13/mlame Sun Jun 13 12:03:00 1999 +*************** +*** 0 **** +--- 1,81 ---- ++ #!/bin/bash ++ #!/usr/local/bin/bash ++ ############################################################################ ++ # ++ # Run the LAME encoder on multible files, with option to delete .wav files ++ # after encoding. "mlame -h" will give instructions. ++ # ++ # Robert Hegemann <Robert.Hegemann@gmx.de> ++ # ++ ############################################################################ ++ ++ mp3coder="lame" ++ options="-h -m f -b 128" ++ rmsrc=false ++ ++ helptext="\ ++ \nThis script runs the LAME mp3 encoder on multiple files: \n\n\ ++ $0 [options] <file 1> ... <file n>\n\ ++ \n\ ++ options:\n\ ++ -h this help text\n\ ++ -r remove files after encoding\n\ ++ -o \"<lame options>\" overrides script default options \"${options}\"\n\ ++ \n\ ++ example:\n\ ++ $0 -r -o \"-v -V 0 -b 112\" a*.wav z*.aif\n\ ++ \n\ ++ " ++ ++ # process command-line options ++ # this could be extended to fake the ++ # commandline interface of the mp3encoder ++ ++ while getopts ":o:r" optn; do ++ case $optn in ++ o ) options=$OPTARG # replace default options ++ ;; ++ r ) rmsrc=true ++ ;; ++ \? ) printf "$helptext" ++ exit 1 ++ ;; ++ esac ++ done ++ shift $(($OPTIND - 1)) ++ ++ # process input-files ++ ++ for filename in "$@"; do ++ case $filename in ++ *[*?]* ) # means shell couldn�t extend *.wav, etc. ++ echo "warning: no $filename file(s) found" ++ ;; ++ *[.][wW][aA][vV] ) ++ name=${filename%[.][wW][aA][vV]} ++ if $mp3coder $options $filename ${name}.mp3 ++ then ++ if [ $rmsrc = true ]; then ++ rm -f $filename ++ fi ++ fi ++ ;; ++ *[.][aA][iI][fF] ) ++ name=${filename%[.][aA][iI][fF]} ++ if $mp3coder $options $filename ${name}.mp3 ++ then ++ if [ $rmsrc = true ]; then ++ rm -f $filename ++ fi ++ fi ++ ;; ++ * ) ++ if $mp3coder $options $filename ${filename}.mp3 ++ then ++ if [ $rmsrc = true ]; then ++ rm -f $filename ++ fi ++ fi ++ ;; ++ esac ++ done +diff -r -c -N encoder/mpglib/Makefile lame3.13/mpglib/Makefile +*** encoder/mpglib/Makefile Wed Dec 31 17:00:00 1969 +--- lame3.13/mpglib/Makefile Tue Apr 20 22:27:37 1999 +*************** +*** 0 **** +--- 1,22 ---- ++ ++ CC=gcc ++ CFLAGS=-Wall -g ++ ++ all: mpglib ++ ++ ++ *.o: mpg123.h mpglib.h ++ ++ mpglib: common.o dct64_i386.o decode_i386.o layer3.o tabinit.o interface.o main.o ++ $(CC) -o mpglib common.o dct64_i386.o decode_i386.o layer3.o \ ++ tabinit.o interface.o main.o -lm ++ ++ lib: common.o dct64_i386.o decode_i386.o layer3.o tabinit.o interface.o main.o ++ ar -svru libmpg123.a main.o common.o dct64_i386.o decode_i386.o layer3.o \ ++ tabinit.o interface.o ++ ++ ++ clean: ++ rm *.o mpglib ++ ++ +diff -r -c -N encoder/mpglib/README lame3.13/mpglib/README +*** encoder/mpglib/README Wed Dec 31 17:00:00 1969 +--- lame3.13/mpglib/README Tue Apr 20 22:27:37 1999 +*************** +*** 0 **** +--- 1,33 ---- ++ MP3 library ++ ----------- ++ Version 0.2 ++ ++ This decoder is a 'light' version (thrown out all unnecessay parts) ++ from the mpg123 package. I made this for a company. ++ ++ Currently only Layer3 is enabled to save some space. Layer1,2 isn't ++ tested at all. The interface will not change significantly. ++ A backport to the mpg123 package is planed. ++ ++ comiled and tested only on Solaris 2.6 ++ main.c contains a simple demo application for library. ++ ++ COPYING: you may use this source under GPL terms! ++ ++ PLEASE NOTE: This software may contain patented alogrithm (at least ++ patented in some countries). It may be not allowed to sell/use products ++ based on this source code in these countries. Check this out first! ++ ++ COPYRIGHT of MP3 music: ++ Please note, that the duplicating of copyrighted music without explicit ++ permission violates the rights of the owner. ++ ++ SENDING PATCHES: ++ Maybe I change the copyright policy (ie some kind of more free BSD licencse). ++ Please consider this when sending patches/changes. ++ ++ FEEDBACK: ++ I'm interessted to here from you, when you use this package as part ++ of another project. ++ ++ +diff -r -c -N encoder/mpglib/TODO lame3.13/mpglib/TODO +*** encoder/mpglib/TODO Wed Dec 31 17:00:00 1969 +--- lame3.13/mpglib/TODO Tue Apr 20 22:27:37 1999 +*************** +*** 0 **** +--- 1,2 ---- ++ ++ apply 'VBR' bug +diff -r -c -N encoder/mpglib/common.c lame3.13/mpglib/common.c +*** encoder/mpglib/common.c Wed Dec 31 17:00:00 1969 +--- lame3.13/mpglib/common.c Sun Apr 25 22:13:19 1999 +*************** +*** 0 **** +--- 1,263 ---- ++ #ifdef HAVEGTK ++ #include <ctype.h> ++ #include <stdlib.h> ++ #include <signal.h> ++ ++ #include <sys/types.h> ++ #include <sys/stat.h> ++ #include <fcntl.h> ++ ++ #include "mpg123.h" ++ ++ struct parameter param = { 1 , 1 , 0 , 0 }; ++ ++ int tabsel_123[2][3][16] = { ++ { {0,32,64,96,128,160,192,224,256,288,320,352,384,416,448,}, ++ {0,32,48,56, 64, 80, 96,112,128,160,192,224,256,320,384,}, ++ {0,32,40,48, 56, 64, 80, 96,112,128,160,192,224,256,320,} }, ++ ++ { {0,32,48,56,64,80,96,112,128,144,160,176,192,224,256,}, ++ {0,8,16,24,32,40,48,56,64,80,96,112,128,144,160,}, ++ {0,8,16,24,32,40,48,56,64,80,96,112,128,144,160,} } ++ }; ++ ++ long freqs[9] = { 44100, 48000, 32000, ++ 22050, 24000, 16000 , ++ 11025 , 12000 , 8000 }; ++ ++ int bitindex; ++ unsigned char *wordpointer; ++ unsigned char *pcm_sample; ++ int pcm_point = 0; ++ ++ ++ #if 0 ++ static void get_II_stuff(struct frame *fr) ++ { ++ static int translate[3][2][16] = ++ { { { 0,2,2,2,2,2,2,0,0,0,1,1,1,1,1,0 } , ++ { 0,2,2,0,0,0,1,1,1,1,1,1,1,1,1,0 } } , ++ { { 0,2,2,2,2,2,2,0,0,0,0,0,0,0,0,0 } , ++ { 0,2,2,0,0,0,0,0,0,0,0,0,0,0,0,0 } } , ++ { { 0,3,3,3,3,3,3,0,0,0,1,1,1,1,1,0 } , ++ { 0,3,3,0,0,0,1,1,1,1,1,1,1,1,1,0 } } }; ++ ++ int table,sblim; ++ static struct al_table *tables[5] = ++ { alloc_0, alloc_1, alloc_2, alloc_3 , alloc_4 }; ++ static int sblims[5] = { 27 , 30 , 8, 12 , 30 }; ++ ++ if(fr->lsf) ++ table = 4; ++ else ++ table = translate[fr->sampling_frequency][2-fr->stereo][fr->bitrate_index]; ++ sblim = sblims[table]; ++ ++ fr->alloc = tables[table]; ++ fr->II_sblimit = sblim; ++ } ++ #endif ++ ++ #define HDRCMPMASK 0xfffffd00 ++ ++ #if 0 ++ int head_check(unsigned long head) ++ { ++ if( (head & 0xffe00000) != 0xffe00000) ++ return FALSE; ++ if(!((head>>17)&3)) ++ return FALSE; ++ if( ((head>>12)&0xf) == 0xf) ++ return FALSE; ++ if( ((head>>10)&0x3) == 0x3 ) ++ return FALSE; ++ return TRUE; ++ } ++ #endif ++ ++ /* ++ * the code a header and write the information ++ * into the frame structure ++ */ ++ int decode_header(struct frame *fr,unsigned long newhead) ++ { ++ ++ ++ if( newhead & (1<<20) ) { ++ fr->lsf = (newhead & (1<<19)) ? 0x0 : 0x1; ++ fr->mpeg25 = 0; ++ } ++ else { ++ fr->lsf = 1; ++ fr->mpeg25 = 1; ++ } ++ ++ fr->lay = 4-((newhead>>17)&3); ++ if( ((newhead>>10)&0x3) == 0x3) { ++ fprintf(stderr,"Stream error\n"); ++ exit(1); ++ } ++ if(fr->mpeg25) { ++ fr->sampling_frequency = 6 + ((newhead>>10)&0x3); ++ } ++ else ++ fr->sampling_frequency = ((newhead>>10)&0x3) + (fr->lsf*3); ++ fr->error_protection = ((newhead>>16)&0x1)^0x1; ++ ++ if(fr->mpeg25) /* allow Bitrate change for 2.5 ... */ ++ fr->bitrate_index = ((newhead>>12)&0xf); ++ ++ fr->bitrate_index = ((newhead>>12)&0xf); ++ fr->padding = ((newhead>>9)&0x1); ++ fr->extension = ((newhead>>8)&0x1); ++ fr->mode = ((newhead>>6)&0x3); ++ fr->mode_ext = ((newhead>>4)&0x3); ++ fr->copyright = ((newhead>>3)&0x1); ++ fr->original = ((newhead>>2)&0x1); ++ fr->emphasis = newhead & 0x3; ++ ++ fr->stereo = (fr->mode == MPG_MD_MONO) ? 1 : 2; ++ ++ if(!fr->bitrate_index) ++ { ++ fprintf(stderr,"Free format not supported.\n"); ++ return (0); ++ } ++ ++ switch(fr->lay) ++ { ++ case 1: ++ #if 0 ++ fr->do_layer = do_layer1; ++ fr->jsbound = (fr->mode == MPG_MD_JOINT_STEREO) ? ++ (fr->mode_ext<<2)+4 : 32; ++ fr->framesize = (long) tabsel_123[fr->lsf][0][fr->bitrate_index] * 12000; ++ fr->framesize /= freqs[fr->sampling_frequency]; ++ fr->framesize = ((fr->framesize+fr->padding)<<2)-4; ++ #else ++ fprintf(stderr,"Not supported!\n"); ++ #endif ++ break; ++ case 2: ++ #if 0 ++ fr->do_layer = do_layer2; ++ get_II_stuff(fr); ++ fr->jsbound = (fr->mode == MPG_MD_JOINT_STEREO) ? ++ (fr->mode_ext<<2)+4 : fr->II_sblimit; ++ fr->framesize = (long) tabsel_123[fr->lsf][1][fr->bitrate_index] * 144000; ++ fr->framesize /= freqs[fr->sampling_frequency]; ++ fr->framesize += fr->padding - 4; ++ #else ++ fprintf(stderr,"Not supported!\n"); ++ #endif ++ break; ++ case 3: ++ #if 0 ++ fr->do_layer = do_layer3; ++ if(fr->lsf) ++ ssize = (fr->stereo == 1) ? 9 : 17; ++ else ++ ssize = (fr->stereo == 1) ? 17 : 32; ++ #endif ++ ++ #if 0 ++ if(fr->error_protection) ++ ssize += 2; ++ #endif ++ fr->framesize = (long) tabsel_123[fr->lsf][2][fr->bitrate_index] * 144000; ++ fr->framesize /= freqs[fr->sampling_frequency]<<(fr->lsf); ++ fr->framesize = fr->framesize + fr->padding - 4; ++ break; ++ default: ++ fprintf(stderr,"Sorry, unknown layer type.\n"); ++ return (0); ++ } ++ ++ /* print_header(fr); */ ++ ++ return 1; ++ } ++ ++ ++ #if 1 ++ void print_header(struct frame *fr) ++ { ++ static char *modes[4] = { "Stereo", "Joint-Stereo", "Dual-Channel", "Single-Channel" }; ++ static char *layers[4] = { "Unknown" , "I", "II", "III" }; ++ ++ fprintf(stderr,"MPEG %s, Layer: %s, Freq: %ld, mode: %s, modext: %d, BPF : %d\n", ++ fr->mpeg25 ? "2.5" : (fr->lsf ? "2.0" : "1.0"), ++ layers[fr->lay],freqs[fr->sampling_frequency], ++ modes[fr->mode],fr->mode_ext,fr->framesize+4); ++ fprintf(stderr,"Channels: %d, copyright: %s, original: %s, CRC: %s, emphasis: %d.\n", ++ fr->stereo,fr->copyright?"Yes":"No", ++ fr->original?"Yes":"No",fr->error_protection?"Yes":"No", ++ fr->emphasis); ++ fprintf(stderr,"Bitrate: %d Kbits/s, Extension value: %d\n", ++ tabsel_123[fr->lsf][fr->lay-1][fr->bitrate_index],fr->extension); ++ } ++ ++ void print_header_compact(struct frame *fr) ++ { ++ static char *modes[4] = { "stereo", "joint-stereo", "dual-channel", "mono" }; ++ static char *layers[4] = { "Unknown" , "I", "II", "III" }; ++ ++ fprintf(stderr,"MPEG %s layer %s, %d kbit/s, %ld Hz %s\n", ++ fr->mpeg25 ? "2.5" : (fr->lsf ? "2.0" : "1.0"), ++ layers[fr->lay], ++ tabsel_123[fr->lsf][fr->lay-1][fr->bitrate_index], ++ freqs[fr->sampling_frequency], modes[fr->mode]); ++ } ++ ++ #endif ++ ++ unsigned int getbits(int number_of_bits) ++ { ++ unsigned long rval; ++ ++ if(!number_of_bits) ++ return 0; ++ ++ { ++ rval = wordpointer[0]; ++ rval <<= 8; ++ rval |= wordpointer[1]; ++ rval <<= 8; ++ rval |= wordpointer[2]; ++ rval <<= bitindex; ++ rval &= 0xffffff; ++ ++ bitindex += number_of_bits; ++ ++ rval >>= (24-number_of_bits); ++ ++ wordpointer += (bitindex>>3); ++ bitindex &= 7; ++ } ++ return rval; ++ } ++ ++ unsigned int getbits_fast(int number_of_bits) ++ { ++ unsigned long rval; ++ ++ { ++ rval = wordpointer[0]; ++ rval <<= 8; ++ rval |= wordpointer[1]; ++ rval <<= bitindex; ++ rval &= 0xffff; ++ bitindex += number_of_bits; ++ ++ rval >>= (16-number_of_bits); ++ ++ wordpointer += (bitindex>>3); ++ bitindex &= 7; ++ } ++ return rval; ++ } ++ ++ ++ ++ ++ #endif +diff -r -c -N encoder/mpglib/dct64_i386.c lame3.13/mpglib/dct64_i386.c +*** encoder/mpglib/dct64_i386.c Wed Dec 31 17:00:00 1969 +--- lame3.13/mpglib/dct64_i386.c Sun Apr 25 22:13:34 1999 +*************** +*** 0 **** +--- 1,316 ---- ++ #ifdef HAVEGTK ++ /* ++ * Discrete Cosine Tansform (DCT) for subband synthesis ++ * optimized for machines with no auto-increment. ++ * The performance is highly compiler dependend. Maybe ++ * the dct64.c version for 'normal' processor may be faster ++ * even for Intel processors. ++ */ ++ ++ #include "mpg123.h" ++ ++ static void dct64_1(real *out0,real *out1,real *b1,real *b2,real *samples) ++ { ++ ++ { ++ register real *costab = pnts[0]; ++ ++ b1[0x00] = samples[0x00] + samples[0x1F]; ++ b1[0x1F] = (samples[0x00] - samples[0x1F]) * costab[0x0]; ++ ++ b1[0x01] = samples[0x01] + samples[0x1E]; ++ b1[0x1E] = (samples[0x01] - samples[0x1E]) * costab[0x1]; ++ ++ b1[0x02] = samples[0x02] + samples[0x1D]; ++ b1[0x1D] = (samples[0x02] - samples[0x1D]) * costab[0x2]; ++ ++ b1[0x03] = samples[0x03] + samples[0x1C]; ++ b1[0x1C] = (samples[0x03] - samples[0x1C]) * costab[0x3]; ++ ++ b1[0x04] = samples[0x04] + samples[0x1B]; ++ b1[0x1B] = (samples[0x04] - samples[0x1B]) * costab[0x4]; ++ ++ b1[0x05] = samples[0x05] + samples[0x1A]; ++ b1[0x1A] = (samples[0x05] - samples[0x1A]) * costab[0x5]; ++ ++ b1[0x06] = samples[0x06] + samples[0x19]; ++ b1[0x19] = (samples[0x06] - samples[0x19]) * costab[0x6]; ++ ++ b1[0x07] = samples[0x07] + samples[0x18]; ++ b1[0x18] = (samples[0x07] - samples[0x18]) * costab[0x7]; ++ ++ b1[0x08] = samples[0x08] + samples[0x17]; ++ b1[0x17] = (samples[0x08] - samples[0x17]) * costab[0x8]; ++ ++ b1[0x09] = samples[0x09] + samples[0x16]; ++ b1[0x16] = (samples[0x09] - samples[0x16]) * costab[0x9]; ++ ++ b1[0x0A] = samples[0x0A] + samples[0x15]; ++ b1[0x15] = (samples[0x0A] - samples[0x15]) * costab[0xA]; ++ ++ b1[0x0B] = samples[0x0B] + samples[0x14]; ++ b1[0x14] = (samples[0x0B] - samples[0x14]) * costab[0xB]; ++ ++ b1[0x0C] = samples[0x0C] + samples[0x13]; ++ b1[0x13] = (samples[0x0C] - samples[0x13]) * costab[0xC]; ++ ++ b1[0x0D] = samples[0x0D] + samples[0x12]; ++ b1[0x12] = (samples[0x0D] - samples[0x12]) * costab[0xD]; ++ ++ b1[0x0E] = samples[0x0E] + samples[0x11]; ++ b1[0x11] = (samples[0x0E] - samples[0x11]) * costab[0xE]; ++ ++ b1[0x0F] = samples[0x0F] + samples[0x10]; ++ b1[0x10] = (samples[0x0F] - samples[0x10]) * costab[0xF]; ++ } ++ ++ ++ { ++ register real *costab = pnts[1]; ++ ++ b2[0x00] = b1[0x00] + b1[0x0F]; ++ b2[0x0F] = (b1[0x00] - b1[0x0F]) * costab[0]; ++ b2[0x01] = b1[0x01] + b1[0x0E]; ++ b2[0x0E] = (b1[0x01] - b1[0x0E]) * costab[1]; ++ b2[0x02] = b1[0x02] + b1[0x0D]; ++ b2[0x0D] = (b1[0x02] - b1[0x0D]) * costab[2]; ++ b2[0x03] = b1[0x03] + b1[0x0C]; ++ b2[0x0C] = (b1[0x03] - b1[0x0C]) * costab[3]; ++ b2[0x04] = b1[0x04] + b1[0x0B]; ++ b2[0x0B] = (b1[0x04] - b1[0x0B]) * costab[4]; ++ b2[0x05] = b1[0x05] + b1[0x0A]; ++ b2[0x0A] = (b1[0x05] - b1[0x0A]) * costab[5]; ++ b2[0x06] = b1[0x06] + b1[0x09]; ++ b2[0x09] = (b1[0x06] - b1[0x09]) * costab[6]; ++ b2[0x07] = b1[0x07] + b1[0x08]; ++ b2[0x08] = (b1[0x07] - b1[0x08]) * costab[7]; ++ ++ b2[0x10] = b1[0x10] + b1[0x1F]; ++ b2[0x1F] = (b1[0x1F] - b1[0x10]) * costab[0]; ++ b2[0x11] = b1[0x11] + b1[0x1E]; ++ b2[0x1E] = (b1[0x1E] - b1[0x11]) * costab[1]; ++ b2[0x12] = b1[0x12] + b1[0x1D]; ++ b2[0x1D] = (b1[0x1D] - b1[0x12]) * costab[2]; ++ b2[0x13] = b1[0x13] + b1[0x1C]; ++ b2[0x1C] = (b1[0x1C] - b1[0x13]) * costab[3]; ++ b2[0x14] = b1[0x14] + b1[0x1B]; ++ b2[0x1B] = (b1[0x1B] - b1[0x14]) * costab[4]; ++ b2[0x15] = b1[0x15] + b1[0x1A]; ++ b2[0x1A] = (b1[0x1A] - b1[0x15]) * costab[5]; ++ b2[0x16] = b1[0x16] + b1[0x19]; ++ b2[0x19] = (b1[0x19] - b1[0x16]) * costab[6]; ++ b2[0x17] = b1[0x17] + b1[0x18]; ++ b2[0x18] = (b1[0x18] - b1[0x17]) * costab[7]; ++ } ++ ++ { ++ register real *costab = pnts[2]; ++ ++ b1[0x00] = b2[0x00] + b2[0x07]; ++ b1[0x07] = (b2[0x00] - b2[0x07]) * costab[0]; ++ b1[0x01] = b2[0x01] + b2[0x06]; ++ b1[0x06] = (b2[0x01] - b2[0x06]) * costab[1]; ++ b1[0x02] = b2[0x02] + b2[0x05]; ++ b1[0x05] = (b2[0x02] - b2[0x05]) * costab[2]; ++ b1[0x03] = b2[0x03] + b2[0x04]; ++ b1[0x04] = (b2[0x03] - b2[0x04]) * costab[3]; ++ ++ b1[0x08] = b2[0x08] + b2[0x0F]; ++ b1[0x0F] = (b2[0x0F] - b2[0x08]) * costab[0]; ++ b1[0x09] = b2[0x09] + b2[0x0E]; ++ b1[0x0E] = (b2[0x0E] - b2[0x09]) * costab[1]; ++ b1[0x0A] = b2[0x0A] + b2[0x0D]; ++ b1[0x0D] = (b2[0x0D] - b2[0x0A]) * costab[2]; ++ b1[0x0B] = b2[0x0B] + b2[0x0C]; ++ b1[0x0C] = (b2[0x0C] - b2[0x0B]) * costab[3]; ++ ++ b1[0x10] = b2[0x10] + b2[0x17]; ++ b1[0x17] = (b2[0x10] - b2[0x17]) * costab[0]; ++ b1[0x11] = b2[0x11] + b2[0x16]; ++ b1[0x16] = (b2[0x11] - b2[0x16]) * costab[1]; ++ b1[0x12] = b2[0x12] + b2[0x15]; ++ b1[0x15] = (b2[0x12] - b2[0x15]) * costab[2]; ++ b1[0x13] = b2[0x13] + b2[0x14]; ++ b1[0x14] = (b2[0x13] - b2[0x14]) * costab[3]; ++ ++ b1[0x18] = b2[0x18] + b2[0x1F]; ++ b1[0x1F] = (b2[0x1F] - b2[0x18]) * costab[0]; ++ b1[0x19] = b2[0x19] + b2[0x1E]; ++ b1[0x1E] = (b2[0x1E] - b2[0x19]) * costab[1]; ++ b1[0x1A] = b2[0x1A] + b2[0x1D]; ++ b1[0x1D] = (b2[0x1D] - b2[0x1A]) * costab[2]; ++ b1[0x1B] = b2[0x1B] + b2[0x1C]; ++ b1[0x1C] = (b2[0x1C] - b2[0x1B]) * costab[3]; ++ } ++ ++ { ++ register real const cos0 = pnts[3][0]; ++ register real const cos1 = pnts[3][1]; ++ ++ b2[0x00] = b1[0x00] + b1[0x03]; ++ b2[0x03] = (b1[0x00] - b1[0x03]) * cos0; ++ b2[0x01] = b1[0x01] + b1[0x02]; ++ b2[0x02] = (b1[0x01] - b1[0x02]) * cos1; ++ ++ b2[0x04] = b1[0x04] + b1[0x07]; ++ b2[0x07] = (b1[0x07] - b1[0x04]) * cos0; ++ b2[0x05] = b1[0x05] + b1[0x06]; ++ b2[0x06] = (b1[0x06] - b1[0x05]) * cos1; ++ ++ b2[0x08] = b1[0x08] + b1[0x0B]; ++ b2[0x0B] = (b1[0x08] - b1[0x0B]) * cos0; ++ b2[0x09] = b1[0x09] + b1[0x0A]; ++ b2[0x0A] = (b1[0x09] - b1[0x0A]) * cos1; ++ ++ b2[0x0C] = b1[0x0C] + b1[0x0F]; ++ b2[0x0F] = (b1[0x0F] - b1[0x0C]) * cos0; ++ b2[0x0D] = b1[0x0D] + b1[0x0E]; ++ b2[0x0E] = (b1[0x0E] - b1[0x0D]) * cos1; ++ ++ b2[0x10] = b1[0x10] + b1[0x13]; ++ b2[0x13] = (b1[0x10] - b1[0x13]) * cos0; ++ b2[0x11] = b1[0x11] + b1[0x12]; ++ b2[0x12] = (b1[0x11] - b1[0x12]) * cos1; ++ ++ b2[0x14] = b1[0x14] + b1[0x17]; ++ b2[0x17] = (b1[0x17] - b1[0x14]) * cos0; ++ b2[0x15] = b1[0x15] + b1[0x16]; ++ b2[0x16] = (b1[0x16] - b1[0x15]) * cos1; ++ ++ b2[0x18] = b1[0x18] + b1[0x1B]; ++ b2[0x1B] = (b1[0x18] - b1[0x1B]) * cos0; ++ b2[0x19] = b1[0x19] + b1[0x1A]; ++ b2[0x1A] = (b1[0x19] - b1[0x1A]) * cos1; ++ ++ b2[0x1C] = b1[0x1C] + b1[0x1F]; ++ b2[0x1F] = (b1[0x1F] - b1[0x1C]) * cos0; ++ b2[0x1D] = b1[0x1D] + b1[0x1E]; ++ b2[0x1E] = (b1[0x1E] - b1[0x1D]) * cos1; ++ } ++ ++ { ++ register real const cos0 = pnts[4][0]; ++ ++ b1[0x00] = b2[0x00] + b2[0x01]; ++ b1[0x01] = (b2[0x00] - b2[0x01]) * cos0; ++ b1[0x02] = b2[0x02] + b2[0x03]; ++ b1[0x03] = (b2[0x03] - b2[0x02]) * cos0; ++ b1[0x02] += b1[0x03]; ++ ++ b1[0x04] = b2[0x04] + b2[0x05]; ++ b1[0x05] = (b2[0x04] - b2[0x05]) * cos0; ++ b1[0x06] = b2[0x06] + b2[0x07]; ++ b1[0x07] = (b2[0x07] - b2[0x06]) * cos0; ++ b1[0x06] += b1[0x07]; ++ b1[0x04] += b1[0x06]; ++ b1[0x06] += b1[0x05]; ++ b1[0x05] += b1[0x07]; ++ ++ b1[0x08] = b2[0x08] + b2[0x09]; ++ b1[0x09] = (b2[0x08] - b2[0x09]) * cos0; ++ b1[0x0A] = b2[0x0A] + b2[0x0B]; ++ b1[0x0B] = (b2[0x0B] - b2[0x0A]) * cos0; ++ b1[0x0A] += b1[0x0B]; ++ ++ b1[0x0C] = b2[0x0C] + b2[0x0D]; ++ b1[0x0D] = (b2[0x0C] - b2[0x0D]) * cos0; ++ b1[0x0E] = b2[0x0E] + b2[0x0F]; ++ b1[0x0F] = (b2[0x0F] - b2[0x0E]) * cos0; ++ b1[0x0E] += b1[0x0F]; ++ b1[0x0C] += b1[0x0E]; ++ b1[0x0E] += b1[0x0D]; ++ b1[0x0D] += b1[0x0F]; ++ ++ b1[0x10] = b2[0x10] + b2[0x11]; ++ b1[0x11] = (b2[0x10] - b2[0x11]) * cos0; ++ b1[0x12] = b2[0x12] + b2[0x13]; ++ b1[0x13] = (b2[0x13] - b2[0x12]) * cos0; ++ b1[0x12] += b1[0x13]; ++ ++ b1[0x14] = b2[0x14] + b2[0x15]; ++ b1[0x15] = (b2[0x14] - b2[0x15]) * cos0; ++ b1[0x16] = b2[0x16] + b2[0x17]; ++ b1[0x17] = (b2[0x17] - b2[0x16]) * cos0; ++ b1[0x16] += b1[0x17]; ++ b1[0x14] += b1[0x16]; ++ b1[0x16] += b1[0x15]; ++ b1[0x15] += b1[0x17]; ++ ++ b1[0x18] = b2[0x18] + b2[0x19]; ++ b1[0x19] = (b2[0x18] - b2[0x19]) * cos0; ++ b1[0x1A] = b2[0x1A] + b2[0x1B]; ++ b1[0x1B] = (b2[0x1B] - b2[0x1A]) * cos0; ++ b1[0x1A] += b1[0x1B]; ++ ++ b1[0x1C] = b2[0x1C] + b2[0x1D]; ++ b1[0x1D] = (b2[0x1C] - b2[0x1D]) * cos0; ++ b1[0x1E] = b2[0x1E] + b2[0x1F]; ++ b1[0x1F] = (b2[0x1F] - b2[0x1E]) * cos0; ++ b1[0x1E] += b1[0x1F]; ++ b1[0x1C] += b1[0x1E]; ++ b1[0x1E] += b1[0x1D]; ++ b1[0x1D] += b1[0x1F]; ++ } ++ ++ out0[0x10*16] = b1[0x00]; ++ out0[0x10*12] = b1[0x04]; ++ out0[0x10* 8] = b1[0x02]; ++ out0[0x10* 4] = b1[0x06]; ++ out0[0x10* 0] = b1[0x01]; ++ out1[0x10* 0] = b1[0x01]; ++ out1[0x10* 4] = b1[0x05]; ++ out1[0x10* 8] = b1[0x03]; ++ out1[0x10*12] = b1[0x07]; ++ ++ b1[0x08] += b1[0x0C]; ++ out0[0x10*14] = b1[0x08]; ++ b1[0x0C] += b1[0x0a]; ++ out0[0x10*10] = b1[0x0C]; ++ b1[0x0A] += b1[0x0E]; ++ out0[0x10* 6] = b1[0x0A]; ++ b1[0x0E] += b1[0x09]; ++ out0[0x10* 2] = b1[0x0E]; ++ b1[0x09] += b1[0x0D]; ++ out1[0x10* 2] = b1[0x09]; ++ b1[0x0D] += b1[0x0B]; ++ out1[0x10* 6] = b1[0x0D]; ++ b1[0x0B] += b1[0x0F]; ++ out1[0x10*10] = b1[0x0B]; ++ out1[0x10*14] = b1[0x0F]; ++ ++ b1[0x18] += b1[0x1C]; ++ out0[0x10*15] = b1[0x10] + b1[0x18]; ++ out0[0x10*13] = b1[0x18] + b1[0x14]; ++ b1[0x1C] += b1[0x1a]; ++ out0[0x10*11] = b1[0x14] + b1[0x1C]; ++ out0[0x10* 9] = b1[0x1C] + b1[0x12]; ++ b1[0x1A] += b1[0x1E]; ++ out0[0x10* 7] = b1[0x12] + b1[0x1A]; ++ out0[0x10* 5] = b1[0x1A] + b1[0x16]; ++ b1[0x1E] += b1[0x19]; ++ out0[0x10* 3] = b1[0x16] + b1[0x1E]; ++ out0[0x10* 1] = b1[0x1E] + b1[0x11]; ++ b1[0x19] += b1[0x1D]; ++ out1[0x10* 1] = b1[0x11] + b1[0x19]; ++ out1[0x10* 3] = b1[0x19] + b1[0x15]; ++ b1[0x1D] += b1[0x1B]; ++ out1[0x10* 5] = b1[0x15] + b1[0x1D]; ++ out1[0x10* 7] = b1[0x1D] + b1[0x13]; ++ b1[0x1B] += b1[0x1F]; ++ out1[0x10* 9] = b1[0x13] + b1[0x1B]; ++ out1[0x10*11] = b1[0x1B] + b1[0x17]; ++ out1[0x10*13] = b1[0x17] + b1[0x1F]; ++ out1[0x10*15] = b1[0x1F]; ++ } ++ ++ /* ++ * the call via dct64 is a trick to force GCC to use ++ * (new) registers for the b1,b2 pointer to the bufs[xx] field ++ */ ++ void dct64(real *a,real *b,real *c) ++ { ++ real bufs[0x40]; ++ dct64_1(a,b,bufs,bufs+0x20,c); ++ } ++ ++ #endif +diff -r -c -N encoder/mpglib/decode_i386.c lame3.13/mpglib/decode_i386.c +*** encoder/mpglib/decode_i386.c Wed Dec 31 17:00:00 1969 +--- lame3.13/mpglib/decode_i386.c Sun Apr 25 22:13:48 1999 +*************** +*** 0 **** +--- 1,155 ---- ++ #ifdef HAVEGTK ++ /* ++ * Mpeg Layer-1,2,3 audio decoder ++ * ------------------------------ ++ * copyright (c) 1995,1996,1997 by Michael Hipp, All rights reserved. ++ * See also 'README' ++ * ++ * slighlty optimized for machines without autoincrement/decrement. ++ * The performance is highly compiler dependend. Maybe ++ * the decode.c version for 'normal' processor may be faster ++ * even for Intel processors. ++ */ ++ ++ #include <stdlib.h> ++ #include <math.h> ++ #include <string.h> ++ ++ #include "mpg123.h" ++ #include "mpglib.h" ++ ++ extern struct mpstr *gmp; ++ ++ /* old WRITE_SAMPLE */ ++ #define WRITE_SAMPLE(samples,sum,clip) \ ++ if( (sum) > 32767.0) { *(samples) = 0x7fff; (clip)++; } \ ++ else if( (sum) < -32768.0) { *(samples) = -0x8000; (clip)++; } \ ++ else { *(samples) = sum; } ++ ++ int synth_1to1_mono(real *bandPtr,unsigned char *samples,int *pnt) ++ { ++ short samples_tmp[64]; ++ short *tmp1 = samples_tmp; ++ int i,ret; ++ int pnt1 = 0; ++ ++ ret = synth_1to1(bandPtr,0,(unsigned char *) samples_tmp,&pnt1); ++ samples += *pnt; ++ ++ for(i=0;i<32;i++) { ++ *( (short *) samples) = *tmp1; ++ samples += 2; ++ tmp1 += 2; ++ } ++ *pnt += 64; ++ ++ return ret; ++ } ++ ++ ++ int synth_1to1(real *bandPtr,int channel,unsigned char *out,int *pnt) ++ { ++ static const int step = 2; ++ int bo; ++ short *samples = (short *) (out + *pnt); ++ ++ real *b0,(*buf)[0x110]; ++ int clip = 0; ++ int bo1; ++ ++ bo = gmp->synth_bo; ++ ++ if(!channel) { ++ bo--; ++ bo &= 0xf; ++ buf = gmp->synth_buffs[0]; ++ } ++ else { ++ samples++; ++ buf = gmp->synth_buffs[1]; ++ } ++ ++ if(bo & 0x1) { ++ b0 = buf[0]; ++ bo1 = bo; ++ dct64(buf[1]+((bo+1)&0xf),buf[0]+bo,bandPtr); ++ } ++ else { ++ b0 = buf[1]; ++ bo1 = bo+1; ++ dct64(buf[0]+bo,buf[1]+bo+1,bandPtr); ++ } ++ ++ gmp->synth_bo = bo; ++ ++ { ++ register int j; ++ real *window = decwin + 16 - bo1; ++ ++ for (j=16;j;j--,b0+=0x10,window+=0x20,samples+=step) ++ { ++ real sum; ++ sum = window[0x0] * b0[0x0]; ++ sum -= window[0x1] * b0[0x1]; ++ sum += window[0x2] * b0[0x2]; ++ sum -= window[0x3] * b0[0x3]; ++ sum += window[0x4] * b0[0x4]; ++ sum -= window[0x5] * b0[0x5]; ++ sum += window[0x6] * b0[0x6]; ++ sum -= window[0x7] * b0[0x7]; ++ sum += window[0x8] * b0[0x8]; ++ sum -= window[0x9] * b0[0x9]; ++ sum += window[0xA] * b0[0xA]; ++ sum -= window[0xB] * b0[0xB]; ++ sum += window[0xC] * b0[0xC]; ++ sum -= window[0xD] * b0[0xD]; ++ sum += window[0xE] * b0[0xE]; ++ sum -= window[0xF] * b0[0xF]; ++ ++ WRITE_SAMPLE(samples,sum,clip); ++ } ++ ++ { ++ real sum; ++ sum = window[0x0] * b0[0x0]; ++ sum += window[0x2] * b0[0x2]; ++ sum += window[0x4] * b0[0x4]; ++ sum += window[0x6] * b0[0x6]; ++ sum += window[0x8] * b0[0x8]; ++ sum += window[0xA] * b0[0xA]; ++ sum += window[0xC] * b0[0xC]; ++ sum += window[0xE] * b0[0xE]; ++ WRITE_SAMPLE(samples,sum,clip); ++ b0-=0x10,window-=0x20,samples+=step; ++ } ++ window += bo1<<1; ++ ++ for (j=15;j;j--,b0-=0x10,window-=0x20,samples+=step) ++ { ++ real sum; ++ sum = -window[-0x1] * b0[0x0]; ++ sum -= window[-0x2] * b0[0x1]; ++ sum -= window[-0x3] * b0[0x2]; ++ sum -= window[-0x4] * b0[0x3]; ++ sum -= window[-0x5] * b0[0x4]; ++ sum -= window[-0x6] * b0[0x5]; ++ sum -= window[-0x7] * b0[0x6]; ++ sum -= window[-0x8] * b0[0x7]; ++ sum -= window[-0x9] * b0[0x8]; ++ sum -= window[-0xA] * b0[0x9]; ++ sum -= window[-0xB] * b0[0xA]; ++ sum -= window[-0xC] * b0[0xB]; ++ sum -= window[-0xD] * b0[0xC]; ++ sum -= window[-0xE] * b0[0xD]; ++ sum -= window[-0xF] * b0[0xE]; ++ sum -= window[-0x0] * b0[0xF]; ++ ++ WRITE_SAMPLE(samples,sum,clip); ++ } ++ } ++ *pnt += 128; ++ ++ return clip; ++ } ++ ++ #endif +diff -r -c -N encoder/mpglib/huffman.h lame3.13/mpglib/huffman.h +*** encoder/mpglib/huffman.h Wed Dec 31 17:00:00 1969 +--- lame3.13/mpglib/huffman.h Tue Apr 20 22:27:37 1999 +*************** +*** 0 **** +--- 1,332 ---- ++ /* ++ * huffman tables ... recalcualted to work with my optimzed ++ * decoder scheme (MH) ++ * ++ * probably we could save a few bytes of memory, because the ++ * smaller tables are often the part of a bigger table ++ */ ++ ++ struct newhuff ++ { ++ unsigned int linbits; ++ short *table; ++ }; ++ ++ static short tab0[] = ++ { ++ 0 ++ }; ++ ++ static short tab1[] = ++ { ++ -5, -3, -1, 17, 1, 16, 0 ++ }; ++ ++ static short tab2[] = ++ { ++ -15, -11, -9, -5, -3, -1, 34, 2, 18, -1, 33, 32, 17, -1, 1, ++ 16, 0 ++ }; ++ ++ static short tab3[] = ++ { ++ -13, -11, -9, -5, -3, -1, 34, 2, 18, -1, 33, 32, 16, 17, -1, ++ 1, 0 ++ }; ++ ++ static short tab5[] = ++ { ++ -29, -25, -23, -15, -7, -5, -3, -1, 51, 35, 50, 49, -3, -1, 19, ++ 3, -1, 48, 34, -3, -1, 18, 33, -1, 2, 32, 17, -1, 1, 16, ++ 0 ++ }; ++ ++ static short tab6[] = ++ { ++ -25, -19, -13, -9, -5, -3, -1, 51, 3, 35, -1, 50, 48, -1, 19, ++ 49, -3, -1, 34, 2, 18, -3, -1, 33, 32, 1, -1, 17, -1, 16, ++ 0 ++ }; ++ ++ static short tab7[] = ++ { ++ -69, -65, -57, -39, -29, -17, -11, -7, -3, -1, 85, 69, -1, 84, 83, ++ -1, 53, 68, -3, -1, 37, 82, 21, -5, -1, 81, -1, 5, 52, -1, ++ 80, -1, 67, 51, -5, -3, -1, 36, 66, 20, -1, 65, 64, -11, -7, ++ -3, -1, 4, 35, -1, 50, 3, -1, 19, 49, -3, -1, 48, 34, 18, ++ -5, -1, 33, -1, 2, 32, 17, -1, 1, 16, 0 ++ }; ++ ++ static short tab8[] = ++ { ++ -65, -63, -59, -45, -31, -19, -13, -7, -5, -3, -1, 85, 84, 69, 83, ++ -3, -1, 53, 68, 37, -3, -1, 82, 5, 21, -5, -1, 81, -1, 52, ++ 67, -3, -1, 80, 51, 36, -5, -3, -1, 66, 20, 65, -3, -1, 4, ++ 64, -1, 35, 50, -9, -7, -3, -1, 19, 49, -1, 3, 48, 34, -1, ++ 2, 32, -1, 18, 33, 17, -3, -1, 1, 16, 0 ++ }; ++ ++ static short tab9[] = ++ { ++ -63, -53, -41, -29, -19, -11, -5, -3, -1, 85, 69, 53, -1, 83, -1, ++ 84, 5, -3, -1, 68, 37, -1, 82, 21, -3, -1, 81, 52, -1, 67, ++ -1, 80, 4, -7, -3, -1, 36, 66, -1, 51, 64, -1, 20, 65, -5, ++ -3, -1, 35, 50, 19, -1, 49, -1, 3, 48, -5, -3, -1, 34, 2, ++ 18, -1, 33, 32, -3, -1, 17, 1, -1, 16, 0 ++ }; ++ ++ static short tab10[] = ++ { ++ -125,-121,-111, -83, -55, -35, -21, -13, -7, -3, -1, 119, 103, -1, 118, ++ 87, -3, -1, 117, 102, 71, -3, -1, 116, 86, -1, 101, 55, -9, -3, ++ -1, 115, 70, -3, -1, 85, 84, 99, -1, 39, 114, -11, -5, -3, -1, ++ 100, 7, 112, -1, 98, -1, 69, 53, -5, -1, 6, -1, 83, 68, 23, ++ -17, -5, -1, 113, -1, 54, 38, -5, -3, -1, 37, 82, 21, -1, 81, ++ -1, 52, 67, -3, -1, 22, 97, -1, 96, -1, 5, 80, -19, -11, -7, ++ -3, -1, 36, 66, -1, 51, 4, -1, 20, 65, -3, -1, 64, 35, -1, ++ 50, 3, -3, -1, 19, 49, -1, 48, 34, -7, -3, -1, 18, 33, -1, ++ 2, 32, 17, -1, 1, 16, 0 ++ }; ++ ++ static short tab11[] = ++ { ++ -121,-113, -89, -59, -43, -27, -17, -7, -3, -1, 119, 103, -1, 118, 117, ++ -3, -1, 102, 71, -1, 116, -1, 87, 85, -5, -3, -1, 86, 101, 55, ++ -1, 115, 70, -9, -7, -3, -1, 69, 84, -1, 53, 83, 39, -1, 114, ++ -1, 100, 7, -5, -1, 113, -1, 23, 112, -3, -1, 54, 99, -1, 96, ++ -1, 68, 37, -13, -7, -5, -3, -1, 82, 5, 21, 98, -3, -1, 38, ++ 6, 22, -5, -1, 97, -1, 81, 52, -5, -1, 80, -1, 67, 51, -1, ++ 36, 66, -15, -11, -7, -3, -1, 20, 65, -1, 4, 64, -1, 35, 50, ++ -1, 19, 49, -5, -3, -1, 3, 48, 34, 33, -5, -1, 18, -1, 2, ++ 32, 17, -3, -1, 1, 16, 0 ++ }; ++ ++ static short tab12[] = ++ { ++ -115, -99, -73, -45, -27, -17, -9, -5, -3, -1, 119, 103, 118, -1, 87, ++ 117, -3, -1, 102, 71, -1, 116, 101, -3, -1, 86, 55, -3, -1, 115, ++ 85, 39, -7, -3, -1, 114, 70, -1, 100, 23, -5, -1, 113, -1, 7, ++ 112, -1, 54, 99, -13, -9, -3, -1, 69, 84, -1, 68, -1, 6, 5, ++ -1, 38, 98, -5, -1, 97, -1, 22, 96, -3, -1, 53, 83, -1, 37, ++ 82, -17, -7, -3, -1, 21, 81, -1, 52, 67, -5, -3, -1, 80, 4, ++ 36, -1, 66, 20, -3, -1, 51, 65, -1, 35, 50, -11, -7, -5, -3, ++ -1, 64, 3, 48, 19, -1, 49, 34, -1, 18, 33, -7, -5, -3, -1, ++ 2, 32, 0, 17, -1, 1, 16 ++ }; ++ ++ static short tab13[] = ++ { ++ -509,-503,-475,-405,-333,-265,-205,-153,-115, -83, -53, -35, -21, -13, -9, ++ -7, -5, -3, -1, 254, 252, 253, 237, 255, -1, 239, 223, -3, -1, 238, ++ 207, -1, 222, 191, -9, -3, -1, 251, 206, -1, 220, -1, 175, 233, -1, ++ 236, 221, -9, -5, -3, -1, 250, 205, 190, -1, 235, 159, -3, -1, 249, ++ 234, -1, 189, 219, -17, -9, -3, -1, 143, 248, -1, 204, -1, 174, 158, ++ -5, -1, 142, -1, 127, 126, 247, -5, -1, 218, -1, 173, 188, -3, -1, ++ 203, 246, 111, -15, -7, -3, -1, 232, 95, -1, 157, 217, -3, -1, 245, ++ 231, -1, 172, 187, -9, -3, -1, 79, 244, -3, -1, 202, 230, 243, -1, ++ 63, -1, 141, 216, -21, -9, -3, -1, 47, 242, -3, -1, 110, 156, 15, ++ -5, -3, -1, 201, 94, 171, -3, -1, 125, 215, 78, -11, -5, -3, -1, ++ 200, 214, 62, -1, 185, -1, 155, 170, -1, 31, 241, -23, -13, -5, -1, ++ 240, -1, 186, 229, -3, -1, 228, 140, -1, 109, 227, -5, -1, 226, -1, ++ 46, 14, -1, 30, 225, -15, -7, -3, -1, 224, 93, -1, 213, 124, -3, ++ -1, 199, 77, -1, 139, 184, -7, -3, -1, 212, 154, -1, 169, 108, -1, ++ 198, 61, -37, -21, -9, -5, -3, -1, 211, 123, 45, -1, 210, 29, -5, ++ -1, 183, -1, 92, 197, -3, -1, 153, 122, 195, -7, -5, -3, -1, 167, ++ 151, 75, 209, -3, -1, 13, 208, -1, 138, 168, -11, -7, -3, -1, 76, ++ 196, -1, 107, 182, -1, 60, 44, -3, -1, 194, 91, -3, -1, 181, 137, ++ 28, -43, -23, -11, -5, -1, 193, -1, 152, 12, -1, 192, -1, 180, 106, ++ -5, -3, -1, 166, 121, 59, -1, 179, -1, 136, 90, -11, -5, -1, 43, ++ -1, 165, 105, -1, 164, -1, 120, 135, -5, -1, 148, -1, 119, 118, 178, ++ -11, -3, -1, 27, 177, -3, -1, 11, 176, -1, 150, 74, -7, -3, -1, ++ 58, 163, -1, 89, 149, -1, 42, 162, -47, -23, -9, -3, -1, 26, 161, ++ -3, -1, 10, 104, 160, -5, -3, -1, 134, 73, 147, -3, -1, 57, 88, ++ -1, 133, 103, -9, -3, -1, 41, 146, -3, -1, 87, 117, 56, -5, -1, ++ 131, -1, 102, 71, -3, -1, 116, 86, -1, 101, 115, -11, -3, -1, 25, ++ 145, -3, -1, 9, 144, -1, 72, 132, -7, -5, -1, 114, -1, 70, 100, ++ 40, -1, 130, 24, -41, -27, -11, -5, -3, -1, 55, 39, 23, -1, 113, ++ -1, 85, 7, -7, -3, -1, 112, 54, -1, 99, 69, -3, -1, 84, 38, ++ -1, 98, 53, -5, -1, 129, -1, 8, 128, -3, -1, 22, 97, -1, 6, ++ 96, -13, -9, -5, -3, -1, 83, 68, 37, -1, 82, 5, -1, 21, 81, ++ -7, -3, -1, 52, 67, -1, 80, 36, -3, -1, 66, 51, 20, -19, -11, ++ -5, -1, 65, -1, 4, 64, -3, -1, 35, 50, 19, -3, -1, 49, 3, ++ -1, 48, 34, -3, -1, 18, 33, -1, 2, 32, -3, -1, 17, 1, 16, ++ 0 ++ }; ++ ++ static short tab15[] = ++ { ++ -495,-445,-355,-263,-183,-115, -77, -43, -27, -13, -7, -3, -1, 255, 239, ++ -1, 254, 223, -1, 238, -1, 253, 207, -7, -3, -1, 252, 222, -1, 237, ++ 191, -1, 251, -1, 206, 236, -7, -3, -1, 221, 175, -1, 250, 190, -3, ++ -1, 235, 205, -1, 220, 159, -15, -7, -3, -1, 249, 234, -1, 189, 219, ++ -3, -1, 143, 248, -1, 204, 158, -7, -3, -1, 233, 127, -1, 247, 173, ++ -3, -1, 218, 188, -1, 111, -1, 174, 15, -19, -11, -3, -1, 203, 246, ++ -3, -1, 142, 232, -1, 95, 157, -3, -1, 245, 126, -1, 231, 172, -9, ++ -3, -1, 202, 187, -3, -1, 217, 141, 79, -3, -1, 244, 63, -1, 243, ++ 216, -33, -17, -9, -3, -1, 230, 47, -1, 242, -1, 110, 240, -3, -1, ++ 31, 241, -1, 156, 201, -7, -3, -1, 94, 171, -1, 186, 229, -3, -1, ++ 125, 215, -1, 78, 228, -15, -7, -3, -1, 140, 200, -1, 62, 109, -3, ++ -1, 214, 227, -1, 155, 185, -7, -3, -1, 46, 170, -1, 226, 30, -5, ++ -1, 225, -1, 14, 224, -1, 93, 213, -45, -25, -13, -7, -3, -1, 124, ++ 199, -1, 77, 139, -1, 212, -1, 184, 154, -7, -3, -1, 169, 108, -1, ++ 198, 61, -1, 211, 210, -9, -5, -3, -1, 45, 13, 29, -1, 123, 183, ++ -5, -1, 209, -1, 92, 208, -1, 197, 138, -17, -7, -3, -1, 168, 76, ++ -1, 196, 107, -5, -1, 182, -1, 153, 12, -1, 60, 195, -9, -3, -1, ++ 122, 167, -1, 166, -1, 192, 11, -1, 194, -1, 44, 91, -55, -29, -15, ++ -7, -3, -1, 181, 28, -1, 137, 152, -3, -1, 193, 75, -1, 180, 106, ++ -5, -3, -1, 59, 121, 179, -3, -1, 151, 136, -1, 43, 90, -11, -5, ++ -1, 178, -1, 165, 27, -1, 177, -1, 176, 105, -7, -3, -1, 150, 74, ++ -1, 164, 120, -3, -1, 135, 58, 163, -17, -7, -3, -1, 89, 149, -1, ++ 42, 162, -3, -1, 26, 161, -3, -1, 10, 160, 104, -7, -3, -1, 134, ++ 73, -1, 148, 57, -5, -1, 147, -1, 119, 9, -1, 88, 133, -53, -29, ++ -13, -7, -3, -1, 41, 103, -1, 118, 146, -1, 145, -1, 25, 144, -7, ++ -3, -1, 72, 132, -1, 87, 117, -3, -1, 56, 131, -1, 102, 71, -7, ++ -3, -1, 40, 130, -1, 24, 129, -7, -3, -1, 116, 8, -1, 128, 86, ++ -3, -1, 101, 55, -1, 115, 70, -17, -7, -3, -1, 39, 114, -1, 100, ++ 23, -3, -1, 85, 113, -3, -1, 7, 112, 54, -7, -3, -1, 99, 69, ++ -1, 84, 38, -3, -1, 98, 22, -3, -1, 6, 96, 53, -33, -19, -9, ++ -5, -1, 97, -1, 83, 68, -1, 37, 82, -3, -1, 21, 81, -3, -1, ++ 5, 80, 52, -7, -3, -1, 67, 36, -1, 66, 51, -1, 65, -1, 20, ++ 4, -9, -3, -1, 35, 50, -3, -1, 64, 3, 19, -3, -1, 49, 48, ++ 34, -9, -7, -3, -1, 18, 33, -1, 2, 32, 17, -3, -1, 1, 16, ++ 0 ++ }; ++ ++ static short tab16[] = ++ { ++ -509,-503,-461,-323,-103, -37, -27, -15, -7, -3, -1, 239, 254, -1, 223, ++ 253, -3, -1, 207, 252, -1, 191, 251, -5, -1, 175, -1, 250, 159, -3, ++ -1, 249, 248, 143, -7, -3, -1, 127, 247, -1, 111, 246, 255, -9, -5, ++ -3, -1, 95, 245, 79, -1, 244, 243, -53, -1, 240, -1, 63, -29, -19, ++ -13, -7, -5, -1, 206, -1, 236, 221, 222, -1, 233, -1, 234, 217, -1, ++ 238, -1, 237, 235, -3, -1, 190, 205, -3, -1, 220, 219, 174, -11, -5, ++ -1, 204, -1, 173, 218, -3, -1, 126, 172, 202, -5, -3, -1, 201, 125, ++ 94, 189, 242, -93, -5, -3, -1, 47, 15, 31, -1, 241, -49, -25, -13, ++ -5, -1, 158, -1, 188, 203, -3, -1, 142, 232, -1, 157, 231, -7, -3, ++ -1, 187, 141, -1, 216, 110, -1, 230, 156, -13, -7, -3, -1, 171, 186, ++ -1, 229, 215, -1, 78, -1, 228, 140, -3, -1, 200, 62, -1, 109, -1, ++ 214, 155, -19, -11, -5, -3, -1, 185, 170, 225, -1, 212, -1, 184, 169, ++ -5, -1, 123, -1, 183, 208, 227, -7, -3, -1, 14, 224, -1, 93, 213, ++ -3, -1, 124, 199, -1, 77, 139, -75, -45, -27, -13, -7, -3, -1, 154, ++ 108, -1, 198, 61, -3, -1, 92, 197, 13, -7, -3, -1, 138, 168, -1, ++ 153, 76, -3, -1, 182, 122, 60, -11, -5, -3, -1, 91, 137, 28, -1, ++ 192, -1, 152, 121, -1, 226, -1, 46, 30, -15, -7, -3, -1, 211, 45, ++ -1, 210, 209, -5, -1, 59, -1, 151, 136, 29, -7, -3, -1, 196, 107, ++ -1, 195, 167, -1, 44, -1, 194, 181, -23, -13, -7, -3, -1, 193, 12, ++ -1, 75, 180, -3, -1, 106, 166, 179, -5, -3, -1, 90, 165, 43, -1, ++ 178, 27, -13, -5, -1, 177, -1, 11, 176, -3, -1, 105, 150, -1, 74, ++ 164, -5, -3, -1, 120, 135, 163, -3, -1, 58, 89, 42, -97, -57, -33, ++ -19, -11, -5, -3, -1, 149, 104, 161, -3, -1, 134, 119, 148, -5, -3, ++ -1, 73, 87, 103, 162, -5, -1, 26, -1, 10, 160, -3, -1, 57, 147, ++ -1, 88, 133, -9, -3, -1, 41, 146, -3, -1, 118, 9, 25, -5, -1, ++ 145, -1, 144, 72, -3, -1, 132, 117, -1, 56, 131, -21, -11, -5, -3, ++ -1, 102, 40, 130, -3, -1, 71, 116, 24, -3, -1, 129, 128, -3, -1, ++ 8, 86, 55, -9, -5, -1, 115, -1, 101, 70, -1, 39, 114, -5, -3, ++ -1, 100, 85, 7, 23, -23, -13, -5, -1, 113, -1, 112, 54, -3, -1, ++ 99, 69, -1, 84, 38, -3, -1, 98, 22, -1, 97, -1, 6, 96, -9, ++ -5, -1, 83, -1, 53, 68, -1, 37, 82, -1, 81, -1, 21, 5, -33, ++ -23, -13, -7, -3, -1, 52, 67, -1, 80, 36, -3, -1, 66, 51, 20, ++ -5, -1, 65, -1, 4, 64, -1, 35, 50, -3, -1, 19, 49, -3, -1, ++ 3, 48, 34, -3, -1, 18, 33, -1, 2, 32, -3, -1, 17, 1, 16, ++ 0 ++ }; ++ ++ static short tab24[] = ++ { ++ -451,-117, -43, -25, -15, -7, -3, -1, 239, 254, -1, 223, 253, -3, -1, ++ 207, 252, -1, 191, 251, -5, -1, 250, -1, 175, 159, -1, 249, 248, -9, ++ -5, -3, -1, 143, 127, 247, -1, 111, 246, -3, -1, 95, 245, -1, 79, ++ 244, -71, -7, -3, -1, 63, 243, -1, 47, 242, -5, -1, 241, -1, 31, ++ 240, -25, -9, -1, 15, -3, -1, 238, 222, -1, 237, 206, -7, -3, -1, ++ 236, 221, -1, 190, 235, -3, -1, 205, 220, -1, 174, 234, -15, -7, -3, ++ -1, 189, 219, -1, 204, 158, -3, -1, 233, 173, -1, 218, 188, -7, -3, ++ -1, 203, 142, -1, 232, 157, -3, -1, 217, 126, -1, 231, 172, 255,-235, ++ -143, -77, -45, -25, -15, -7, -3, -1, 202, 187, -1, 141, 216, -5, -3, ++ -1, 14, 224, 13, 230, -5, -3, -1, 110, 156, 201, -1, 94, 186, -9, ++ -5, -1, 229, -1, 171, 125, -1, 215, 228, -3, -1, 140, 200, -3, -1, ++ 78, 46, 62, -15, -7, -3, -1, 109, 214, -1, 227, 155, -3, -1, 185, ++ 170, -1, 226, 30, -7, -3, -1, 225, 93, -1, 213, 124, -3, -1, 199, ++ 77, -1, 139, 184, -31, -15, -7, -3, -1, 212, 154, -1, 169, 108, -3, ++ -1, 198, 61, -1, 211, 45, -7, -3, -1, 210, 29, -1, 123, 183, -3, ++ -1, 209, 92, -1, 197, 138, -17, -7, -3, -1, 168, 153, -1, 76, 196, ++ -3, -1, 107, 182, -3, -1, 208, 12, 60, -7, -3, -1, 195, 122, -1, ++ 167, 44, -3, -1, 194, 91, -1, 181, 28, -57, -35, -19, -7, -3, -1, ++ 137, 152, -1, 193, 75, -5, -3, -1, 192, 11, 59, -3, -1, 176, 10, ++ 26, -5, -1, 180, -1, 106, 166, -3, -1, 121, 151, -3, -1, 160, 9, ++ 144, -9, -3, -1, 179, 136, -3, -1, 43, 90, 178, -7, -3, -1, 165, ++ 27, -1, 177, 105, -1, 150, 164, -17, -9, -5, -3, -1, 74, 120, 135, ++ -1, 58, 163, -3, -1, 89, 149, -1, 42, 162, -7, -3, -1, 161, 104, ++ -1, 134, 119, -3, -1, 73, 148, -1, 57, 147, -63, -31, -15, -7, -3, ++ -1, 88, 133, -1, 41, 103, -3, -1, 118, 146, -1, 25, 145, -7, -3, ++ -1, 72, 132, -1, 87, 117, -3, -1, 56, 131, -1, 102, 40, -17, -7, ++ -3, -1, 130, 24, -1, 71, 116, -5, -1, 129, -1, 8, 128, -1, 86, ++ 101, -7, -5, -1, 23, -1, 7, 112, 115, -3, -1, 55, 39, 114, -15, ++ -7, -3, -1, 70, 100, -1, 85, 113, -3, -1, 54, 99, -1, 69, 84, ++ -7, -3, -1, 38, 98, -1, 22, 97, -5, -3, -1, 6, 96, 53, -1, ++ 83, 68, -51, -37, -23, -15, -9, -3, -1, 37, 82, -1, 21, -1, 5, ++ 80, -1, 81, -1, 52, 67, -3, -1, 36, 66, -1, 51, 20, -9, -5, ++ -1, 65, -1, 4, 64, -1, 35, 50, -1, 19, 49, -7, -5, -3, -1, ++ 3, 48, 34, 18, -1, 33, -1, 2, 32, -3, -1, 17, 1, -1, 16, ++ 0 ++ }; ++ ++ static short tab_c0[] = ++ { ++ -29, -21, -13, -7, -3, -1, 11, 15, -1, 13, 14, -3, -1, 7, 5, ++ 9, -3, -1, 6, 3, -1, 10, 12, -3, -1, 2, 1, -1, 4, 8, ++ 0 ++ }; ++ ++ static short tab_c1[] = ++ { ++ -15, -7, -3, -1, 15, 14, -1, 13, 12, -3, -1, 11, 10, -1, 9, ++ 8, -7, -3, -1, 7, 6, -1, 5, 4, -3, -1, 3, 2, -1, 1, ++ 0 ++ }; ++ ++ ++ ++ static struct newhuff ht[] = ++ { ++ { /* 0 */ 0 , tab0 } , ++ { /* 2 */ 0 , tab1 } , ++ { /* 3 */ 0 , tab2 } , ++ { /* 3 */ 0 , tab3 } , ++ { /* 0 */ 0 , tab0 } , ++ { /* 4 */ 0 , tab5 } , ++ { /* 4 */ 0 , tab6 } , ++ { /* 6 */ 0 , tab7 } , ++ { /* 6 */ 0 , tab8 } , ++ { /* 6 */ 0 , tab9 } , ++ { /* 8 */ 0 , tab10 } , ++ { /* 8 */ 0 , tab11 } , ++ { /* 8 */ 0 , tab12 } , ++ { /* 16 */ 0 , tab13 } , ++ { /* 0 */ 0 , tab0 } , ++ { /* 16 */ 0 , tab15 } , ++ ++ { /* 16 */ 1 , tab16 } , ++ { /* 16 */ 2 , tab16 } , ++ { /* 16 */ 3 , tab16 } , ++ { /* 16 */ 4 , tab16 } , ++ { /* 16 */ 6 , tab16 } , ++ { /* 16 */ 8 , tab16 } , ++ { /* 16 */ 10, tab16 } , ++ { /* 16 */ 13, tab16 } , ++ { /* 16 */ 4 , tab24 } , ++ { /* 16 */ 5 , tab24 } , ++ { /* 16 */ 6 , tab24 } , ++ { /* 16 */ 7 , tab24 } , ++ { /* 16 */ 8 , tab24 } , ++ { /* 16 */ 9 , tab24 } , ++ { /* 16 */ 11, tab24 } , ++ { /* 16 */ 13, tab24 } ++ }; ++ ++ static struct newhuff htc[] = ++ { ++ { /* 1 , 1 , */ 0 , tab_c0 } , ++ { /* 1 , 1 , */ 0 , tab_c1 } ++ }; ++ ++ +diff -r -c -N encoder/mpglib/interface.c lame3.13/mpglib/interface.c +*** encoder/mpglib/interface.c Wed Dec 31 17:00:00 1969 +--- lame3.13/mpglib/interface.c Sun Apr 25 22:14:06 1999 +*************** +*** 0 **** +--- 1,219 ---- ++ #ifdef HAVEGTK ++ #include <stdlib.h> ++ #include <stdio.h> ++ ++ #include "mpg123.h" ++ #include "mpglib.h" ++ ++ ++ /* Global mp .. it's a hack */ ++ struct mpstr *gmp; ++ ++ ++ BOOL InitMP3(struct mpstr *mp) ++ { ++ memset(mp,0,sizeof(struct mpstr)); ++ ++ mp->framesize = 0; ++ mp->fsizeold = -1; ++ mp->bsize = 0; ++ mp->head = mp->tail = NULL; ++ mp->fr.single = -1; ++ mp->bsnum = 0; ++ mp->synth_bo = 1; ++ ++ make_decode_tables(32767); ++ init_layer3(SBLIMIT); ++ ++ return !0; ++ } ++ ++ void ExitMP3(struct mpstr *mp) ++ { ++ struct buf *b,*bn; ++ ++ b = mp->tail; ++ while(b) { ++ free(b->pnt); ++ bn = b->next; ++ free(b); ++ b = bn; ++ } ++ } ++ ++ static struct buf *addbuf(struct mpstr *mp,char *buf,int size) ++ { ++ struct buf *nbuf; ++ ++ nbuf = malloc( sizeof(struct buf) ); ++ if(!nbuf) { ++ fprintf(stderr,"Out of memory!\n"); ++ return NULL; ++ } ++ nbuf->pnt = malloc(size); ++ if(!nbuf->pnt) { ++ free(nbuf); ++ return NULL; ++ } ++ nbuf->size = size; ++ memcpy(nbuf->pnt,buf,size); ++ nbuf->next = NULL; ++ nbuf->prev = mp->head; ++ nbuf->pos = 0; ++ ++ if(!mp->tail) { ++ mp->tail = nbuf; ++ } ++ else { ++ mp->head->next = nbuf; ++ } ++ ++ mp->head = nbuf; ++ mp->bsize += size; ++ ++ return nbuf; ++ } ++ ++ static void remove_buf(struct mpstr *mp) ++ { ++ struct buf *buf = mp->tail; ++ ++ mp->tail = buf->next; ++ if(mp->tail) ++ mp->tail->prev = NULL; ++ else { ++ mp->tail = mp->head = NULL; ++ } ++ ++ free(buf->pnt); ++ free(buf); ++ ++ } ++ ++ static int read_buf_byte(struct mpstr *mp) ++ { ++ unsigned int b; ++ ++ int pos; ++ ++ pos = mp->tail->pos; ++ while(pos >= mp->tail->size) { ++ remove_buf(mp); ++ pos = mp->tail->pos; ++ if(!mp->tail) { ++ fprintf(stderr,"Fatal error!\n"); ++ exit(1); ++ } ++ } ++ ++ b = mp->tail->pnt[pos]; ++ mp->bsize--; ++ mp->tail->pos++; ++ ++ ++ return b; ++ } ++ ++ static void read_head(struct mpstr *mp) ++ { ++ unsigned long head; ++ ++ head = read_buf_byte(mp); ++ head <<= 8; ++ head |= read_buf_byte(mp); ++ head <<= 8; ++ head |= read_buf_byte(mp); ++ head <<= 8; ++ head |= read_buf_byte(mp); ++ ++ mp->header = head; ++ } ++ ++ int decodeMP3(struct mpstr *mp,char *in,int isize,char *out, ++ int osize,int *done) ++ { ++ int len; ++ ++ gmp = mp; ++ ++ if(osize < 4608) { ++ fprintf(stderr,"To less out space\n"); ++ return MP3_ERR; ++ } ++ ++ if(in) { ++ if(addbuf(mp,in,isize) == NULL) { ++ return MP3_ERR; ++ } ++ } ++ ++ ++ /* First decode header */ ++ if(mp->framesize == 0) { ++ if(mp->bsize < 4) { ++ return MP3_NEED_MORE; ++ } ++ read_head(mp); ++ decode_header(&mp->fr,mp->header); ++ mp->framesize = mp->fr.framesize; ++ } ++ ++ /* printf(" fr.framesize = %i \n",mp->fr.framesize); ++ printf(" bsize = %i \n",mp->bsize); ++ */ ++ ++ if(mp->fr.framesize > mp->bsize) { ++ return MP3_NEED_MORE; ++ } ++ wordpointer = mp->bsspace[mp->bsnum] + 512; ++ mp->bsnum = (mp->bsnum + 1) & 0x1; ++ bitindex = 0; ++ ++ len = 0; ++ while(len < mp->framesize) { ++ int nlen; ++ int blen = mp->tail->size - mp->tail->pos; ++ if( (mp->framesize - len) <= blen) { ++ nlen = mp->framesize-len; ++ } ++ else { ++ nlen = blen; ++ } ++ memcpy(wordpointer+len,mp->tail->pnt+mp->tail->pos,nlen); ++ len += nlen; ++ mp->tail->pos += nlen; ++ mp->bsize -= nlen; ++ if(mp->tail->pos == mp->tail->size) { ++ remove_buf(mp); ++ } ++ } ++ ++ *done = 0; ++ if(mp->fr.error_protection) ++ getbits(16); ++ do_layer3(&mp->fr,(unsigned char *) out,done); ++ ++ mp->fsizeold = mp->framesize; ++ mp->framesize = 0; ++ return MP3_OK; ++ } ++ ++ int set_pointer(long backstep) ++ { ++ unsigned char *bsbufold; ++ if(gmp->fsizeold < 0 && backstep > 0) { ++ fprintf(stderr,"Can't step back %ld!\n",backstep); ++ return MP3_ERR; ++ } ++ bsbufold = gmp->bsspace[gmp->bsnum] + 512; ++ wordpointer -= backstep; ++ if (backstep) ++ memcpy(wordpointer,bsbufold+gmp->fsizeold-backstep,backstep); ++ bitindex = 0; ++ return MP3_OK; ++ } ++ ++ #endif ++ ++ ++ +diff -r -c -N encoder/mpglib/layer3.c lame3.13/mpglib/layer3.c +*** encoder/mpglib/layer3.c Wed Dec 31 17:00:00 1969 +--- lame3.13/mpglib/layer3.c Tue Jun 22 01:39:00 1999 +*************** +*** 0 **** +--- 1,2088 ---- ++ #ifdef HAVEGTK ++ /* ++ * Mpeg Layer-3 audio decoder ++ * -------------------------- ++ * copyright (c) 1995,1996,1997 by Michael Hipp. ++ * All rights reserved. See also 'README' ++ */ ++ ++ #include <stdlib.h> ++ #include "mpg123.h" ++ #include "mpglib.h" ++ #include "huffman.h" ++ #include "../gtkanal.h" ++ ++ ++ extern struct mpstr *gmp; ++ ++ #define MPEG1 ++ ++ ++ static real ispow[8207]; ++ static real aa_ca[8],aa_cs[8]; ++ static real COS1[12][6]; ++ static real win[4][36]; ++ static real win1[4][36]; ++ static real gainpow2[256+118+4]; ++ static real COS9[9]; ++ static real COS6_1,COS6_2; ++ static real tfcos36[9]; ++ static real tfcos12[3]; ++ ++ struct bandInfoStruct { ++ short longIdx[23]; ++ short longDiff[22]; ++ short shortIdx[14]; ++ short shortDiff[13]; ++ }; ++ ++ int longLimit[9][23]; ++ int shortLimit[9][14]; ++ ++ struct bandInfoStruct bandInfo[9] = { ++ ++ /* MPEG 1.0 */ ++ { {0,4,8,12,16,20,24,30,36,44,52,62,74, 90,110,134,162,196,238,288,342,418,576}, ++ {4,4,4,4,4,4,6,6,8, 8,10,12,16,20,24,28,34,42,50,54, 76,158}, ++ {0,4*3,8*3,12*3,16*3,22*3,30*3,40*3,52*3,66*3, 84*3,106*3,136*3,192*3}, ++ {4,4,4,4,6,8,10,12,14,18,22,30,56} } , ++ ++ { {0,4,8,12,16,20,24,30,36,42,50,60,72, 88,106,128,156,190,230,276,330,384,576}, ++ {4,4,4,4,4,4,6,6,6, 8,10,12,16,18,22,28,34,40,46,54, 54,192}, ++ {0,4*3,8*3,12*3,16*3,22*3,28*3,38*3,50*3,64*3, 80*3,100*3,126*3,192*3}, ++ {4,4,4,4,6,6,10,12,14,16,20,26,66} } , ++ ++ { {0,4,8,12,16,20,24,30,36,44,54,66,82,102,126,156,194,240,296,364,448,550,576} , ++ {4,4,4,4,4,4,6,6,8,10,12,16,20,24,30,38,46,56,68,84,102, 26} , ++ {0,4*3,8*3,12*3,16*3,22*3,30*3,42*3,58*3,78*3,104*3,138*3,180*3,192*3} , ++ {4,4,4,4,6,8,12,16,20,26,34,42,12} } , ++ ++ /* MPEG 2.0 */ ++ { {0,6,12,18,24,30,36,44,54,66,80,96,116,140,168,200,238,284,336,396,464,522,576}, ++ {6,6,6,6,6,6,8,10,12,14,16,20,24,28,32,38,46,52,60,68,58,54 } , ++ {0,4*3,8*3,12*3,18*3,24*3,32*3,42*3,56*3,74*3,100*3,132*3,174*3,192*3} , ++ {4,4,4,6,6,8,10,14,18,26,32,42,18 } } , ++ ++ { {0,6,12,18,24,30,36,44,54,66,80,96,114,136,162,194,232,278,330,394,464,540,576}, ++ {6,6,6,6,6,6,8,10,12,14,16,18,22,26,32,38,46,52,64,70,76,36 } , ++ {0,4*3,8*3,12*3,18*3,26*3,36*3,48*3,62*3,80*3,104*3,136*3,180*3,192*3} , ++ {4,4,4,6,8,10,12,14,18,24,32,44,12 } } , ++ ++ { {0,6,12,18,24,30,36,44,54,66,80,96,116,140,168,200,238,284,336,396,464,522,576}, ++ {6,6,6,6,6,6,8,10,12,14,16,20,24,28,32,38,46,52,60,68,58,54 }, ++ {0,4*3,8*3,12*3,18*3,26*3,36*3,48*3,62*3,80*3,104*3,134*3,174*3,192*3}, ++ {4,4,4,6,8,10,12,14,18,24,30,40,18 } } , ++ /* MPEG 2.5 */ ++ { {0,6,12,18,24,30,36,44,54,66,80,96,116,140,168,200,238,284,336,396,464,522,576} , ++ {6,6,6,6,6,6,8,10,12,14,16,20,24,28,32,38,46,52,60,68,58,54}, ++ {0,12,24,36,54,78,108,144,186,240,312,402,522,576}, ++ {4,4,4,6,8,10,12,14,18,24,30,40,18} }, ++ { {0,6,12,18,24,30,36,44,54,66,80,96,116,140,168,200,238,284,336,396,464,522,576} , ++ {6,6,6,6,6,6,8,10,12,14,16,20,24,28,32,38,46,52,60,68,58,54}, ++ {0,12,24,36,54,78,108,144,186,240,312,402,522,576}, ++ {4,4,4,6,8,10,12,14,18,24,30,40,18} }, ++ { {0,12,24,36,48,60,72,88,108,132,160,192,232,280,336,400,476,566,568,570,572,574,576}, ++ {12,12,12,12,12,12,16,20,24,28,32,40,48,56,64,76,90,2,2,2,2,2}, ++ {0, 24, 48, 72,108,156,216,288,372,480,486,492,498,576}, ++ {8,8,8,12,16,20,24,28,36,2,2,2,26} } , ++ }; ++ ++ static int mapbuf0[9][152]; ++ static int mapbuf1[9][156]; ++ static int mapbuf2[9][44]; ++ static int *map[9][3]; ++ static int *mapend[9][3]; ++ ++ static unsigned int n_slen2[512]; /* MPEG 2.0 slen for 'normal' mode */ ++ static unsigned int i_slen2[256]; /* MPEG 2.0 slen for intensity stereo */ ++ ++ static real tan1_1[16],tan2_1[16],tan1_2[16],tan2_2[16]; ++ static real pow1_1[2][16],pow2_1[2][16],pow1_2[2][16],pow2_2[2][16]; ++ ++ static unsigned int get1bit(void) ++ { ++ unsigned char rval; ++ rval = *wordpointer << bitindex; ++ ++ bitindex++; ++ wordpointer += (bitindex>>3); ++ bitindex &= 7; ++ ++ return rval>>7; ++ } ++ ++ ++ ++ ++ /* ++ * init tables for layer-3 ++ */ ++ void init_layer3(int down_sample_sblimit) ++ { ++ int i,j,k,l; ++ ++ for(i=-256;i<118+4;i++) ++ gainpow2[i+256] = pow((double)2.0,-0.25 * (double) (i+210) ); ++ ++ for(i=0;i<8207;i++) ++ ispow[i] = pow((double)i,(double)4.0/3.0); ++ ++ for (i=0;i<8;i++) ++ { ++ static double Ci[8]={-0.6,-0.535,-0.33,-0.185,-0.095,-0.041,-0.0142,-0.0037}; ++ double sq=sqrt(1.0+Ci[i]*Ci[i]); ++ aa_cs[i] = 1.0/sq; ++ aa_ca[i] = Ci[i]/sq; ++ } ++ ++ for(i=0;i<18;i++) ++ { ++ win[0][i] = win[1][i] = 0.5 * sin( M_PI / 72.0 * (double) (2*(i+0) +1) ) / cos ( M_PI * (double) (2*(i+0) +19) / 72.0 ); ++ win[0][i+18] = win[3][i+18] = 0.5 * sin( M_PI / 72.0 * (double) (2*(i+18)+1) ) / cos ( M_PI * (double) (2*(i+18)+19) / 72.0 ); ++ } ++ for(i=0;i<6;i++) ++ { ++ win[1][i+18] = 0.5 / cos ( M_PI * (double) (2*(i+18)+19) / 72.0 ); ++ win[3][i+12] = 0.5 / cos ( M_PI * (double) (2*(i+12)+19) / 72.0 ); ++ win[1][i+24] = 0.5 * sin( M_PI / 24.0 * (double) (2*i+13) ) / cos ( M_PI * (double) (2*(i+24)+19) / 72.0 ); ++ win[1][i+30] = win[3][i] = 0.0; ++ win[3][i+6 ] = 0.5 * sin( M_PI / 24.0 * (double) (2*i+1) ) / cos ( M_PI * (double) (2*(i+6 )+19) / 72.0 ); ++ } ++ ++ for(i=0;i<9;i++) ++ COS9[i] = cos( M_PI / 18.0 * (double) i); ++ ++ for(i=0;i<9;i++) ++ tfcos36[i] = 0.5 / cos ( M_PI * (double) (i*2+1) / 36.0 ); ++ for(i=0;i<3;i++) ++ tfcos12[i] = 0.5 / cos ( M_PI * (double) (i*2+1) / 12.0 ); ++ ++ COS6_1 = cos( M_PI / 6.0 * (double) 1); ++ COS6_2 = cos( M_PI / 6.0 * (double) 2); ++ ++ for(i=0;i<12;i++) ++ { ++ win[2][i] = 0.5 * sin( M_PI / 24.0 * (double) (2*i+1) ) / cos ( M_PI * (double) (2*i+7) / 24.0 ); ++ for(j=0;j<6;j++) ++ COS1[i][j] = cos( M_PI / 24.0 * (double) ((2*i+7)*(2*j+1)) ); ++ } ++ ++ for(j=0;j<4;j++) { ++ static int len[4] = { 36,36,12,36 }; ++ for(i=0;i<len[j];i+=2) ++ win1[j][i] = + win[j][i]; ++ for(i=1;i<len[j];i+=2) ++ win1[j][i] = - win[j][i]; ++ } ++ ++ for(i=0;i<16;i++) ++ { ++ double t = tan( (double) i * M_PI / 12.0 ); ++ tan1_1[i] = t / (1.0+t); ++ tan2_1[i] = 1.0 / (1.0 + t); ++ tan1_2[i] = M_SQRT2 * t / (1.0+t); ++ tan2_2[i] = M_SQRT2 / (1.0 + t); ++ ++ for(j=0;j<2;j++) { ++ double base = pow(2.0,-0.25*(j+1.0)); ++ double p1=1.0,p2=1.0; ++ if(i > 0) { ++ if( i & 1 ) ++ p1 = pow(base,(i+1.0)*0.5); ++ else ++ p2 = pow(base,i*0.5); ++ } ++ pow1_1[j][i] = p1; ++ pow2_1[j][i] = p2; ++ pow1_2[j][i] = M_SQRT2 * p1; ++ pow2_2[j][i] = M_SQRT2 * p2; ++ } ++ } ++ ++ for(j=0;j<9;j++) ++ { ++ struct bandInfoStruct *bi = &bandInfo[j]; ++ int *mp; ++ int cb,lwin; ++ short *bdf; ++ ++ mp = map[j][0] = mapbuf0[j]; ++ bdf = bi->longDiff; ++ for(i=0,cb = 0; cb < 8 ; cb++,i+=*bdf++) { ++ *mp++ = (*bdf) >> 1; ++ *mp++ = i; ++ *mp++ = 3; ++ *mp++ = cb; ++ } ++ bdf = bi->shortDiff+3; ++ for(cb=3;cb<13;cb++) { ++ int l = (*bdf++) >> 1; ++ for(lwin=0;lwin<3;lwin++) { ++ *mp++ = l; ++ *mp++ = i + lwin; ++ *mp++ = lwin; ++ *mp++ = cb; ++ } ++ i += 6*l; ++ } ++ mapend[j][0] = mp; ++ ++ mp = map[j][1] = mapbuf1[j]; ++ bdf = bi->shortDiff+0; ++ for(i=0,cb=0;cb<13;cb++) { ++ int l = (*bdf++) >> 1; ++ for(lwin=0;lwin<3;lwin++) { ++ *mp++ = l; ++ *mp++ = i + lwin; ++ *mp++ = lwin; ++ *mp++ = cb; ++ } ++ i += 6*l; ++ } ++ mapend[j][1] = mp; ++ ++ mp = map[j][2] = mapbuf2[j]; ++ bdf = bi->longDiff; ++ for(cb = 0; cb < 22 ; cb++) { ++ *mp++ = (*bdf++) >> 1; ++ *mp++ = cb; ++ } ++ mapend[j][2] = mp; ++ ++ } ++ ++ for(j=0;j<9;j++) { ++ for(i=0;i<23;i++) { ++ longLimit[j][i] = (bandInfo[j].longIdx[i] - 1 + 8) / 18 + 1; ++ if(longLimit[j][i] > (down_sample_sblimit) ) ++ longLimit[j][i] = down_sample_sblimit; ++ } ++ for(i=0;i<14;i++) { ++ shortLimit[j][i] = (bandInfo[j].shortIdx[i] - 1) / 18 + 1; ++ if(shortLimit[j][i] > (down_sample_sblimit) ) ++ shortLimit[j][i] = down_sample_sblimit; ++ } ++ } ++ ++ for(i=0;i<5;i++) { ++ for(j=0;j<6;j++) { ++ for(k=0;k<6;k++) { ++ int n = k + j * 6 + i * 36; ++ i_slen2[n] = i|(j<<3)|(k<<6)|(3<<12); ++ } ++ } ++ } ++ for(i=0;i<4;i++) { ++ for(j=0;j<4;j++) { ++ for(k=0;k<4;k++) { ++ int n = k + j * 4 + i * 16; ++ i_slen2[n+180] = i|(j<<3)|(k<<6)|(4<<12); ++ } ++ } ++ } ++ for(i=0;i<4;i++) { ++ for(j=0;j<3;j++) { ++ int n = j + i * 3; ++ i_slen2[n+244] = i|(j<<3) | (5<<12); ++ n_slen2[n+500] = i|(j<<3) | (2<<12) | (1<<15); ++ } ++ } ++ ++ for(i=0;i<5;i++) { ++ for(j=0;j<5;j++) { ++ for(k=0;k<4;k++) { ++ for(l=0;l<4;l++) { ++ int n = l + k * 4 + j * 16 + i * 80; ++ n_slen2[n] = i|(j<<3)|(k<<6)|(l<<9)|(0<<12); ++ } ++ } ++ } ++ } ++ for(i=0;i<5;i++) { ++ for(j=0;j<5;j++) { ++ for(k=0;k<4;k++) { ++ int n = k + j * 4 + i * 20; ++ n_slen2[n+400] = i|(j<<3)|(k<<6)|(1<<12); ++ } ++ } ++ } ++ } ++ ++ /* ++ * read additional side information ++ */ ++ #ifdef MPEG1 ++ static void III_get_side_info_1(struct III_sideinfo *si,int stereo, ++ int ms_stereo,long sfreq,int single) ++ { ++ int ch, gr; ++ int powdiff = (single == 3) ? 4 : 0; ++ ++ si->main_data_begin = getbits(9); ++ if (stereo == 1) ++ si->private_bits = getbits_fast(5); ++ else ++ si->private_bits = getbits_fast(3); ++ ++ for (ch=0; ch<stereo; ch++) { ++ si->ch[ch].gr[0].scfsi = -1; ++ si->ch[ch].gr[1].scfsi = getbits_fast(4); ++ } ++ ++ for (gr=0; gr<2; gr++) ++ { ++ for (ch=0; ch<stereo; ch++) ++ { ++ register struct gr_info_s *gr_info = &(si->ch[ch].gr[gr]); ++ ++ gr_info->part2_3_length = getbits(12); ++ gr_info->big_values = getbits_fast(9); ++ if(gr_info->big_values > 288) { ++ fprintf(stderr,"big_values too large!\n"); ++ gr_info->big_values = 288; ++ } ++ { ++ unsigned int qss = getbits_fast(8); ++ gr_info->pow2gain = gainpow2+256 - qss + powdiff; ++ #ifdef HAVEGTK ++ pinfo->qss[gr][ch]=qss; ++ #endif ++ } ++ if(ms_stereo) ++ gr_info->pow2gain += 2; ++ gr_info->scalefac_compress = getbits_fast(4); ++ /* window-switching flag == 1 for block_Type != 0 .. and block-type == 0 -> win-sw-flag = 0 */ ++ if(get1bit()) ++ { ++ int i; ++ gr_info->block_type = getbits_fast(2); ++ gr_info->mixed_block_flag = get1bit(); ++ gr_info->table_select[0] = getbits_fast(5); ++ gr_info->table_select[1] = getbits_fast(5); ++ ++ ++ /* ++ * table_select[2] not needed, because there is no region2, ++ * but to satisfy some verifications tools we set it either. ++ */ ++ gr_info->table_select[2] = 0; ++ for(i=0;i<3;i++) { ++ unsigned int sbg = (getbits_fast(3)<<3); ++ gr_info->full_gain[i] = gr_info->pow2gain + sbg; ++ #ifdef HAVEGTK ++ pinfo->sub_gain[gr][ch][i]=sbg/8; ++ #endif ++ } ++ ++ ++ if(gr_info->block_type == 0) { ++ fprintf(stderr,"Blocktype == 0 and window-switching == 1 not allowed.\n"); ++ exit(1); ++ } ++ /* region_count/start parameters are implicit in this case. */ ++ gr_info->region1start = 36>>1; ++ gr_info->region2start = 576>>1; ++ } ++ else ++ { ++ int i,r0c,r1c; ++ for (i=0; i<3; i++) ++ gr_info->table_select[i] = getbits_fast(5); ++ r0c = getbits_fast(4); ++ r1c = getbits_fast(3); ++ gr_info->region1start = bandInfo[sfreq].longIdx[r0c+1] >> 1 ; ++ gr_info->region2start = bandInfo[sfreq].longIdx[r0c+1+r1c+1] >> 1; ++ gr_info->block_type = 0; ++ gr_info->mixed_block_flag = 0; ++ } ++ gr_info->preflag = get1bit(); ++ gr_info->scalefac_scale = get1bit(); ++ gr_info->count1table_select = get1bit(); ++ #ifdef HAVEGTK ++ pinfo->scalefac_scale[gr][ch]=gr_info->scalefac_scale; ++ #endif ++ } ++ } ++ } ++ #endif ++ ++ /* ++ * Side Info for MPEG 2.0 / LSF ++ */ ++ static void III_get_side_info_2(struct III_sideinfo *si,int stereo, ++ int ms_stereo,long sfreq,int single) ++ { ++ int ch; ++ int powdiff = (single == 3) ? 4 : 0; ++ ++ si->main_data_begin = getbits(8); ++ if (stereo == 1) ++ si->private_bits = get1bit(); ++ else ++ si->private_bits = getbits_fast(2); ++ ++ for (ch=0; ch<stereo; ch++) ++ { ++ register struct gr_info_s *gr_info = &(si->ch[ch].gr[0]); ++ ++ gr_info->part2_3_length = getbits(12); ++ gr_info->big_values = getbits_fast(9); ++ if(gr_info->big_values > 288) { ++ fprintf(stderr,"big_values too large!\n"); ++ gr_info->big_values = 288; ++ } ++ gr_info->pow2gain = gainpow2+256 - getbits_fast(8) + powdiff; ++ if(ms_stereo) ++ gr_info->pow2gain += 2; ++ gr_info->scalefac_compress = getbits(9); ++ /* window-switching flag == 1 for block_Type != 0 .. and block-type == 0 -> win-sw-flag = 0 */ ++ if(get1bit()) ++ { ++ int i; ++ gr_info->block_type = getbits_fast(2); ++ gr_info->mixed_block_flag = get1bit(); ++ gr_info->table_select[0] = getbits_fast(5); ++ gr_info->table_select[1] = getbits_fast(5); ++ /* ++ * table_select[2] not needed, because there is no region2, ++ * but to satisfy some verifications tools we set it either. ++ */ ++ gr_info->table_select[2] = 0; ++ for(i=0;i<3;i++) { ++ gr_info->full_gain[i] = gr_info->pow2gain + (getbits_fast(3)<<3); ++ } ++ ++ if(gr_info->block_type == 0) { ++ fprintf(stderr,"Blocktype == 0 and window-switching == 1 not allowed.\n"); ++ exit(1); ++ } ++ /* region_count/start parameters are implicit in this case. */ ++ /* check this again! */ ++ if(gr_info->block_type == 2) ++ gr_info->region1start = 36>>1; ++ else if(sfreq == 8) ++ /* check this for 2.5 and sfreq=8 */ ++ gr_info->region1start = 108>>1; ++ else ++ gr_info->region1start = 54>>1; ++ gr_info->region2start = 576>>1; ++ } ++ else ++ { ++ int i,r0c,r1c; ++ for (i=0; i<3; i++) ++ gr_info->table_select[i] = getbits_fast(5); ++ r0c = getbits_fast(4); ++ r1c = getbits_fast(3); ++ gr_info->region1start = bandInfo[sfreq].longIdx[r0c+1] >> 1 ; ++ gr_info->region2start = bandInfo[sfreq].longIdx[r0c+1+r1c+1] >> 1; ++ gr_info->block_type = 0; ++ gr_info->mixed_block_flag = 0; ++ } ++ gr_info->scalefac_scale = get1bit(); ++ gr_info->count1table_select = get1bit(); ++ #ifdef HAVEGTK ++ pinfo->scalefac_scale[0][ch]=gr_info->scalefac_scale; ++ #endif ++ } ++ } ++ ++ /* ++ * read scalefactors ++ */ ++ #ifdef MPEG1 ++ static int III_get_scale_factors_1(int *scf,struct gr_info_s *gr_info) ++ { ++ static unsigned char slen[2][16] = { ++ {0, 0, 0, 0, 3, 1, 1, 1, 2, 2, 2, 3, 3, 3, 4, 4}, ++ {0, 1, 2, 3, 0, 1, 2, 3, 1, 2, 3, 1, 2, 3, 2, 3} ++ }; ++ int numbits; ++ int num0 = slen[0][gr_info->scalefac_compress]; ++ int num1 = slen[1][gr_info->scalefac_compress]; ++ ++ if (gr_info->block_type == 2) ++ { ++ int i=18; ++ numbits = (num0 + num1) * 18; ++ ++ if (gr_info->mixed_block_flag) { ++ for (i=8;i;i--) ++ *scf++ = getbits_fast(num0); ++ i = 9; ++ numbits -= num0; /* num0 * 17 + num1 * 18 */ ++ } ++ ++ for (;i;i--) ++ *scf++ = getbits_fast(num0); ++ for (i = 18; i; i--) ++ *scf++ = getbits_fast(num1); ++ *scf++ = 0; *scf++ = 0; *scf++ = 0; /* short[13][0..2] = 0 */ ++ } ++ else ++ { ++ int i; ++ int scfsi = gr_info->scfsi; ++ ++ if(scfsi < 0) { /* scfsi < 0 => granule == 0 */ ++ for(i=11;i;i--) ++ *scf++ = getbits_fast(num0); ++ for(i=10;i;i--) ++ *scf++ = getbits_fast(num1); ++ numbits = (num0 + num1) * 10 + num0; ++ } ++ else { ++ numbits = 0; ++ if(!(scfsi & 0x8)) { ++ for (i=6;i;i--) ++ *scf++ = getbits_fast(num0); ++ numbits += num0 * 6; ++ } ++ else { ++ *scf++ = 0; *scf++ = 0; *scf++ = 0; /* set to ZERO necessary? */ ++ *scf++ = 0; *scf++ = 0; *scf++ = 0; ++ } ++ ++ if(!(scfsi & 0x4)) { ++ for (i=5;i;i--) ++ *scf++ = getbits_fast(num0); ++ numbits += num0 * 5; ++ } ++ else { ++ *scf++ = 0; *scf++ = 0; *scf++ = 0; /* set to ZERO necessary? */ ++ *scf++ = 0; *scf++ = 0; ++ } ++ ++ if(!(scfsi & 0x2)) { ++ for(i=5;i;i--) ++ *scf++ = getbits_fast(num1); ++ numbits += num1 * 5; ++ } ++ else { ++ *scf++ = 0; *scf++ = 0; *scf++ = 0; /* set to ZERO necessary? */ ++ *scf++ = 0; *scf++ = 0; ++ } ++ ++ if(!(scfsi & 0x1)) { ++ for (i=5;i;i--) ++ *scf++ = getbits_fast(num1); ++ numbits += num1 * 5; ++ } ++ else { ++ *scf++ = 0; *scf++ = 0; *scf++ = 0; /* set to ZERO necessary? */ ++ *scf++ = 0; *scf++ = 0; ++ } ++ } ++ ++ *scf++ = 0; /* no l[21] in original sources */ ++ } ++ return numbits; ++ } ++ #endif ++ ++ static int III_get_scale_factors_2(int *scf,struct gr_info_s *gr_info,int i_stereo) ++ { ++ unsigned char *pnt; ++ int i,j; ++ unsigned int slen; ++ int n = 0; ++ int numbits = 0; ++ ++ static unsigned char stab[3][6][4] = { ++ { { 6, 5, 5,5 } , { 6, 5, 7,3 } , { 11,10,0,0} , ++ { 7, 7, 7,0 } , { 6, 6, 6,3 } , { 8, 8,5,0} } , ++ { { 9, 9, 9,9 } , { 9, 9,12,6 } , { 18,18,0,0} , ++ {12,12,12,0 } , {12, 9, 9,6 } , { 15,12,9,0} } , ++ { { 6, 9, 9,9 } , { 6, 9,12,6 } , { 15,18,0,0} , ++ { 6,15,12,0 } , { 6,12, 9,6 } , { 6,18,9,0} } }; ++ ++ if(i_stereo) /* i_stereo AND second channel -> do_layer3() checks this */ ++ slen = i_slen2[gr_info->scalefac_compress>>1]; ++ else ++ slen = n_slen2[gr_info->scalefac_compress]; ++ ++ gr_info->preflag = (slen>>15) & 0x1; ++ ++ n = 0; ++ if( gr_info->block_type == 2 ) { ++ n++; ++ if(gr_info->mixed_block_flag) ++ n++; ++ } ++ ++ pnt = stab[n][(slen>>12)&0x7]; ++ ++ for(i=0;i<4;i++) { ++ int num = slen & 0x7; ++ slen >>= 3; ++ if(num) { ++ for(j=0;j<(int)(pnt[i]);j++) ++ *scf++ = getbits_fast(num); ++ numbits += pnt[i] * num; ++ } ++ else { ++ for(j=0;j<(int)(pnt[i]);j++) ++ *scf++ = 0; ++ } ++ } ++ ++ n = (n << 1) + 1; ++ for(i=0;i<n;i++) ++ *scf++ = 0; ++ ++ return numbits; ++ } ++ ++ static int pretab1[22] = {0,0,0,0,0,0,0,0,0,0,0,1,1,1,1,2,2,3,3,3,2,0}; ++ static int pretab2[22] = {0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0}; ++ ++ /* ++ * don't forget to apply the same changes to III_dequantize_sample_ms() !!! ++ */ ++ static int III_dequantize_sample(real xr[SBLIMIT][SSLIMIT],int *scf, ++ struct gr_info_s *gr_info,int sfreq,int part2bits) ++ { ++ int shift = 1 + gr_info->scalefac_scale; ++ real *xrpnt = (real *) xr; ++ int l[3],l3; ++ int part2remain = gr_info->part2_3_length - part2bits; ++ int *me; ++ ++ { ++ int bv = gr_info->big_values; ++ int region1 = gr_info->region1start; ++ int region2 = gr_info->region2start; ++ ++ l3 = ((576>>1)-bv)>>1; ++ /* ++ * we may lose the 'odd' bit here !! ++ * check this later again ++ */ ++ if(bv <= region1) { ++ l[0] = bv; l[1] = 0; l[2] = 0; ++ } ++ else { ++ l[0] = region1; ++ if(bv <= region2) { ++ l[1] = bv - l[0]; l[2] = 0; ++ } ++ else { ++ l[1] = region2 - l[0]; l[2] = bv - region2; ++ } ++ } ++ } ++ ++ if(gr_info->block_type == 2) { ++ /* ++ * decoding with short or mixed mode BandIndex table ++ */ ++ int i,max[4]; ++ int step=0,lwin=0,cb=0; ++ register real v = 0.0; ++ register int *m,mc; ++ ++ if(gr_info->mixed_block_flag) { ++ max[3] = -1; ++ max[0] = max[1] = max[2] = 2; ++ m = map[sfreq][0]; ++ me = mapend[sfreq][0]; ++ } ++ else { ++ max[0] = max[1] = max[2] = max[3] = -1; ++ /* max[3] not really needed in this case */ ++ m = map[sfreq][1]; ++ me = mapend[sfreq][1]; ++ } ++ ++ mc = 0; ++ for(i=0;i<2;i++) { ++ int lp = l[i]; ++ struct newhuff *h = ht+gr_info->table_select[i]; ++ for(;lp;lp--,mc--) { ++ register int x,y; ++ if( (!mc) ) { ++ mc = *m++; ++ xrpnt = ((real *) xr) + (*m++); ++ lwin = *m++; ++ cb = *m++; ++ if(lwin == 3) { ++ v = gr_info->pow2gain[(*scf++) << shift]; ++ step = 1; ++ } ++ else { ++ v = gr_info->full_gain[lwin][(*scf++) << shift]; ++ step = 3; ++ } ++ } ++ { ++ register short *val = h->table; ++ while((y=*val++)<0) { ++ if (get1bit()) ++ val -= y; ++ part2remain--; ++ } ++ x = y >> 4; ++ y &= 0xf; ++ } ++ if(x == 15) { ++ max[lwin] = cb; ++ part2remain -= h->linbits+1; ++ x += getbits(h->linbits); ++ if(get1bit()) ++ *xrpnt = -ispow[x] * v; ++ else ++ *xrpnt = ispow[x] * v; ++ } ++ else if(x) { ++ max[lwin] = cb; ++ if(get1bit()) ++ *xrpnt = -ispow[x] * v; ++ else ++ *xrpnt = ispow[x] * v; ++ part2remain--; ++ } ++ else ++ *xrpnt = 0.0; ++ xrpnt += step; ++ if(y == 15) { ++ max[lwin] = cb; ++ part2remain -= h->linbits+1; ++ y += getbits(h->linbits); ++ if(get1bit()) ++ *xrpnt = -ispow[y] * v; ++ else ++ *xrpnt = ispow[y] * v; ++ } ++ else if(y) { ++ max[lwin] = cb; ++ if(get1bit()) ++ *xrpnt = -ispow[y] * v; ++ else ++ *xrpnt = ispow[y] * v; ++ part2remain--; ++ } ++ else ++ *xrpnt = 0.0; ++ xrpnt += step; ++ } ++ } ++ for(;l3 && (part2remain > 0);l3--) { ++ struct newhuff *h = htc+gr_info->count1table_select; ++ register short *val = h->table,a; ++ ++ while((a=*val++)<0) { ++ part2remain--; ++ if(part2remain < 0) { ++ part2remain++; ++ a = 0; ++ break; ++ } ++ if (get1bit()) ++ val -= a; ++ } ++ ++ for(i=0;i<4;i++) { ++ if(!(i & 1)) { ++ if(!mc) { ++ mc = *m++; ++ xrpnt = ((real *) xr) + (*m++); ++ lwin = *m++; ++ cb = *m++; ++ if(lwin == 3) { ++ v = gr_info->pow2gain[(*scf++) << shift]; ++ step = 1; ++ } ++ else { ++ v = gr_info->full_gain[lwin][(*scf++) << shift]; ++ step = 3; ++ } ++ } ++ mc--; ++ } ++ if( (a & (0x8>>i)) ) { ++ max[lwin] = cb; ++ part2remain--; ++ if(part2remain < 0) { ++ part2remain++; ++ break; ++ } ++ if(get1bit()) ++ *xrpnt = -v; ++ else ++ *xrpnt = v; ++ } ++ else ++ *xrpnt = 0.0; ++ xrpnt += step; ++ } ++ } ++ ++ while( m < me ) { ++ if(!mc) { ++ mc = *m++; ++ xrpnt = ((real *) xr) + *m++; ++ if( (*m++) == 3) ++ step = 1; ++ else ++ step = 3; ++ m++; /* cb */ ++ } ++ mc--; ++ *xrpnt = 0.0; ++ xrpnt += step; ++ *xrpnt = 0.0; ++ xrpnt += step; ++ /* we could add a little opt. here: ++ * if we finished a band for window 3 or a long band ++ * further bands could copied in a simple loop without a ++ * special 'map' decoding ++ */ ++ } ++ ++ gr_info->maxband[0] = max[0]+1; ++ gr_info->maxband[1] = max[1]+1; ++ gr_info->maxband[2] = max[2]+1; ++ gr_info->maxbandl = max[3]+1; ++ ++ { ++ int rmax = max[0] > max[1] ? max[0] : max[1]; ++ rmax = (rmax > max[2] ? rmax : max[2]) + 1; ++ gr_info->maxb = rmax ? shortLimit[sfreq][rmax] : longLimit[sfreq][max[3]+1]; ++ } ++ ++ } ++ else { ++ /* ++ * decoding with 'long' BandIndex table (block_type != 2) ++ */ ++ int *pretab = gr_info->preflag ? pretab1 : pretab2; ++ int i,max = -1; ++ int cb = 0; ++ register int *m = map[sfreq][2]; ++ register real v = 0.0; ++ register int mc = 0; ++ #if 0 ++ me = mapend[sfreq][2]; ++ #endif ++ ++ /* ++ * long hash table values ++ */ ++ for(i=0;i<3;i++) { ++ int lp = l[i]; ++ struct newhuff *h = ht+gr_info->table_select[i]; ++ ++ for(;lp;lp--,mc--) { ++ int x,y; ++ ++ if(!mc) { ++ mc = *m++; ++ v = gr_info->pow2gain[((*scf++) + (*pretab++)) << shift]; ++ cb = *m++; ++ } ++ { ++ register short *val = h->table; ++ while((y=*val++)<0) { ++ if (get1bit()) ++ val -= y; ++ part2remain--; ++ } ++ x = y >> 4; ++ y &= 0xf; ++ } ++ if (x == 15) { ++ max = cb; ++ part2remain -= h->linbits+1; ++ x += getbits(h->linbits); ++ if(get1bit()) ++ *xrpnt++ = -ispow[x] * v; ++ else ++ *xrpnt++ = ispow[x] * v; ++ } ++ else if(x) { ++ max = cb; ++ if(get1bit()) ++ *xrpnt++ = -ispow[x] * v; ++ else ++ *xrpnt++ = ispow[x] * v; ++ part2remain--; ++ } ++ else ++ *xrpnt++ = 0.0; ++ ++ if (y == 15) { ++ max = cb; ++ part2remain -= h->linbits+1; ++ y += getbits(h->linbits); ++ if(get1bit()) ++ *xrpnt++ = -ispow[y] * v; ++ else ++ *xrpnt++ = ispow[y] * v; ++ } ++ else if(y) { ++ max = cb; ++ if(get1bit()) ++ *xrpnt++ = -ispow[y] * v; ++ else ++ *xrpnt++ = ispow[y] * v; ++ part2remain--; ++ } ++ else ++ *xrpnt++ = 0.0; ++ } ++ } ++ ++ /* ++ * short (count1table) values ++ */ ++ for(;l3 && (part2remain > 0);l3--) { ++ struct newhuff *h = htc+gr_info->count1table_select; ++ register short *val = h->table,a; ++ ++ while((a=*val++)<0) { ++ part2remain--; ++ if(part2remain < 0) { ++ part2remain++; ++ a = 0; ++ break; ++ } ++ if (get1bit()) ++ val -= a; ++ } ++ ++ for(i=0;i<4;i++) { ++ if(!(i & 1)) { ++ if(!mc) { ++ mc = *m++; ++ cb = *m++; ++ v = gr_info->pow2gain[((*scf++) + (*pretab++)) << shift]; ++ } ++ mc--; ++ } ++ if ( (a & (0x8>>i)) ) { ++ max = cb; ++ part2remain--; ++ if(part2remain < 0) { ++ part2remain++; ++ break; ++ } ++ if(get1bit()) ++ *xrpnt++ = -v; ++ else ++ *xrpnt++ = v; ++ } ++ else ++ *xrpnt++ = 0.0; ++ } ++ } ++ ++ /* ++ * zero part ++ */ ++ for(i=(&xr[SBLIMIT][0]-xrpnt)>>1;i;i--) { ++ *xrpnt++ = 0.0; ++ *xrpnt++ = 0.0; ++ } ++ ++ gr_info->maxbandl = max+1; ++ gr_info->maxb = longLimit[sfreq][gr_info->maxbandl]; ++ } ++ ++ while( part2remain > 16 ) { ++ getbits(16); /* Dismiss stuffing Bits */ ++ part2remain -= 16; ++ } ++ if(part2remain > 0) ++ getbits(part2remain); ++ else if(part2remain < 0) { ++ fprintf(stderr,"mpg123: Can't rewind stream by %d bits!\n",-part2remain); ++ return 1; /* -> error */ ++ } ++ return 0; ++ } ++ ++ #if 0 ++ static int III_dequantize_sample_ms(real xr[2][SBLIMIT][SSLIMIT],int *scf, ++ struct gr_info_s *gr_info,int sfreq,int part2bits) ++ { ++ int shift = 1 + gr_info->scalefac_scale; ++ real *xrpnt = (real *) xr[1]; ++ real *xr0pnt = (real *) xr[0]; ++ int l[3],l3; ++ int part2remain = gr_info->part2_3_length - part2bits; ++ int *me; ++ ++ { ++ int bv = gr_info->big_values; ++ int region1 = gr_info->region1start; ++ int region2 = gr_info->region2start; ++ ++ l3 = ((576>>1)-bv)>>1; ++ /* ++ * we may lose the 'odd' bit here !! ++ * check this later gain ++ */ ++ if(bv <= region1) { ++ l[0] = bv; l[1] = 0; l[2] = 0; ++ } ++ else { ++ l[0] = region1; ++ if(bv <= region2) { ++ l[1] = bv - l[0]; l[2] = 0; ++ } ++ else { ++ l[1] = region2 - l[0]; l[2] = bv - region2; ++ } ++ } ++ } ++ ++ if(gr_info->block_type == 2) { ++ int i,max[4]; ++ int step=0,lwin=0,cb=0; ++ register real v = 0.0; ++ register int *m,mc = 0; ++ ++ if(gr_info->mixed_block_flag) { ++ max[3] = -1; ++ max[0] = max[1] = max[2] = 2; ++ m = map[sfreq][0]; ++ me = mapend[sfreq][0]; ++ } ++ else { ++ max[0] = max[1] = max[2] = max[3] = -1; ++ /* max[3] not really needed in this case */ ++ m = map[sfreq][1]; ++ me = mapend[sfreq][1]; ++ } ++ ++ for(i=0;i<2;i++) { ++ int lp = l[i]; ++ struct newhuff *h = ht+gr_info->table_select[i]; ++ for(;lp;lp--,mc--) { ++ int x,y; ++ ++ if(!mc) { ++ mc = *m++; ++ xrpnt = ((real *) xr[1]) + *m; ++ xr0pnt = ((real *) xr[0]) + *m++; ++ lwin = *m++; ++ cb = *m++; ++ if(lwin == 3) { ++ v = gr_info->pow2gain[(*scf++) << shift]; ++ step = 1; ++ } ++ else { ++ v = gr_info->full_gain[lwin][(*scf++) << shift]; ++ step = 3; ++ } ++ } ++ { ++ register short *val = h->table; ++ while((y=*val++)<0) { ++ if (get1bit()) ++ val -= y; ++ part2remain--; ++ } ++ x = y >> 4; ++ y &= 0xf; ++ } ++ if(x == 15) { ++ max[lwin] = cb; ++ part2remain -= h->linbits+1; ++ x += getbits(h->linbits); ++ if(get1bit()) { ++ real a = ispow[x] * v; ++ *xrpnt = *xr0pnt + a; ++ *xr0pnt -= a; ++ } ++ else { ++ real a = ispow[x] * v; ++ *xrpnt = *xr0pnt - a; ++ *xr0pnt += a; ++ } ++ } ++ else if(x) { ++ max[lwin] = cb; ++ if(get1bit()) { ++ real a = ispow[x] * v; ++ *xrpnt = *xr0pnt + a; ++ *xr0pnt -= a; ++ } ++ else { ++ real a = ispow[x] * v; ++ *xrpnt = *xr0pnt - a; ++ *xr0pnt += a; ++ } ++ part2remain--; ++ } ++ else ++ *xrpnt = *xr0pnt; ++ xrpnt += step; ++ xr0pnt += step; ++ ++ if(y == 15) { ++ max[lwin] = cb; ++ part2remain -= h->linbits+1; ++ y += getbits(h->linbits); ++ if(get1bit()) { ++ real a = ispow[y] * v; ++ *xrpnt = *xr0pnt + a; ++ *xr0pnt -= a; ++ } ++ else { ++ real a = ispow[y] * v; ++ *xrpnt = *xr0pnt - a; ++ *xr0pnt += a; ++ } ++ } ++ else if(y) { ++ max[lwin] = cb; ++ if(get1bit()) { ++ real a = ispow[y] * v; ++ *xrpnt = *xr0pnt + a; ++ *xr0pnt -= a; ++ } ++ else { ++ real a = ispow[y] * v; ++ *xrpnt = *xr0pnt - a; ++ *xr0pnt += a; ++ } ++ part2remain--; ++ } ++ else ++ *xrpnt = *xr0pnt; ++ xrpnt += step; ++ xr0pnt += step; ++ } ++ } ++ ++ for(;l3 && (part2remain > 0);l3--) { ++ struct newhuff *h = htc+gr_info->count1table_select; ++ register short *val = h->table,a; ++ ++ while((a=*val++)<0) { ++ part2remain--; ++ if(part2remain < 0) { ++ part2remain++; ++ a = 0; ++ break; ++ } ++ if (get1bit()) ++ val -= a; ++ } ++ ++ for(i=0;i<4;i++) { ++ if(!(i & 1)) { ++ if(!mc) { ++ mc = *m++; ++ xrpnt = ((real *) xr[1]) + *m; ++ xr0pnt = ((real *) xr[0]) + *m++; ++ lwin = *m++; ++ cb = *m++; ++ if(lwin == 3) { ++ v = gr_info->pow2gain[(*scf++) << shift]; ++ step = 1; ++ } ++ else { ++ v = gr_info->full_gain[lwin][(*scf++) << shift]; ++ step = 3; ++ } ++ } ++ mc--; ++ } ++ if( (a & (0x8>>i)) ) { ++ max[lwin] = cb; ++ part2remain--; ++ if(part2remain < 0) { ++ part2remain++; ++ break; ++ } ++ if(get1bit()) { ++ *xrpnt = *xr0pnt + v; ++ *xr0pnt -= v; ++ } ++ else { ++ *xrpnt = *xr0pnt - v; ++ *xr0pnt += v; ++ } ++ } ++ else ++ *xrpnt = *xr0pnt; ++ xrpnt += step; ++ xr0pnt += step; ++ } ++ } ++ ++ while( m < me ) { ++ if(!mc) { ++ mc = *m++; ++ xrpnt = ((real *) xr[1]) + *m; ++ xr0pnt = ((real *) xr[0]) + *m++; ++ if(*m++ == 3) ++ step = 1; ++ else ++ step = 3; ++ m++; /* cb */ ++ } ++ mc--; ++ *xrpnt = *xr0pnt; ++ xrpnt += step; ++ xr0pnt += step; ++ *xrpnt = *xr0pnt; ++ xrpnt += step; ++ xr0pnt += step; ++ /* we could add a little opt. here: ++ * if we finished a band for window 3 or a long band ++ * further bands could copied in a simple loop without a ++ * special 'map' decoding ++ */ ++ } ++ ++ gr_info->maxband[0] = max[0]+1; ++ gr_info->maxband[1] = max[1]+1; ++ gr_info->maxband[2] = max[2]+1; ++ gr_info->maxbandl = max[3]+1; ++ ++ { ++ int rmax = max[0] > max[1] ? max[0] : max[1]; ++ rmax = (rmax > max[2] ? rmax : max[2]) + 1; ++ gr_info->maxb = rmax ? shortLimit[sfreq][rmax] : longLimit[sfreq][max[3]+1]; ++ } ++ } ++ else { ++ int *pretab = gr_info->preflag ? pretab1 : pretab2; ++ int i,max = -1; ++ int cb = 0; ++ register int mc=0,*m = map[sfreq][2]; ++ register real v = 0.0; ++ #if 0 ++ me = mapend[sfreq][2]; ++ #endif ++ ++ for(i=0;i<3;i++) { ++ int lp = l[i]; ++ struct newhuff *h = ht+gr_info->table_select[i]; ++ ++ for(;lp;lp--,mc--) { ++ int x,y; ++ if(!mc) { ++ mc = *m++; ++ cb = *m++; ++ v = gr_info->pow2gain[((*scf++) + (*pretab++)) << shift]; ++ } ++ { ++ register short *val = h->table; ++ while((y=*val++)<0) { ++ if (get1bit()) ++ val -= y; ++ part2remain--; ++ } ++ x = y >> 4; ++ y &= 0xf; ++ } ++ if (x == 15) { ++ max = cb; ++ part2remain -= h->linbits+1; ++ x += getbits(h->linbits); ++ if(get1bit()) { ++ real a = ispow[x] * v; ++ *xrpnt++ = *xr0pnt + a; ++ *xr0pnt++ -= a; ++ } ++ else { ++ real a = ispow[x] * v; ++ *xrpnt++ = *xr0pnt - a; ++ *xr0pnt++ += a; ++ } ++ } ++ else if(x) { ++ max = cb; ++ if(get1bit()) { ++ real a = ispow[x] * v; ++ *xrpnt++ = *xr0pnt + a; ++ *xr0pnt++ -= a; ++ } ++ else { ++ real a = ispow[x] * v; ++ *xrpnt++ = *xr0pnt - a; ++ *xr0pnt++ += a; ++ } ++ part2remain--; ++ } ++ else ++ *xrpnt++ = *xr0pnt++; ++ ++ if (y == 15) { ++ max = cb; ++ part2remain -= h->linbits+1; ++ y += getbits(h->linbits); ++ if(get1bit()) { ++ real a = ispow[y] * v; ++ *xrpnt++ = *xr0pnt + a; ++ *xr0pnt++ -= a; ++ } ++ else { ++ real a = ispow[y] * v; ++ *xrpnt++ = *xr0pnt - a; ++ *xr0pnt++ += a; ++ } ++ } ++ else if(y) { ++ max = cb; ++ if(get1bit()) { ++ real a = ispow[y] * v; ++ *xrpnt++ = *xr0pnt + a; ++ *xr0pnt++ -= a; ++ } ++ else { ++ real a = ispow[y] * v; ++ *xrpnt++ = *xr0pnt - a; ++ *xr0pnt++ += a; ++ } ++ part2remain--; ++ } ++ else ++ *xrpnt++ = *xr0pnt++; ++ } ++ } ++ ++ for(;l3 && (part2remain > 0);l3--) { ++ struct newhuff *h = htc+gr_info->count1table_select; ++ register short *val = h->table,a; ++ ++ while((a=*val++)<0) { ++ part2remain--; ++ if(part2remain < 0) { ++ part2remain++; ++ a = 0; ++ break; ++ } ++ if (get1bit()) ++ val -= a; ++ } ++ ++ for(i=0;i<4;i++) { ++ if(!(i & 1)) { ++ if(!mc) { ++ mc = *m++; ++ cb = *m++; ++ v = gr_info->pow2gain[((*scf++) + (*pretab++)) << shift]; ++ } ++ mc--; ++ } ++ if ( (a & (0x8>>i)) ) { ++ max = cb; ++ part2remain--; ++ if(part2remain <= 0) { ++ part2remain++; ++ break; ++ } ++ if(get1bit()) { ++ *xrpnt++ = *xr0pnt + v; ++ *xr0pnt++ -= v; ++ } ++ else { ++ *xrpnt++ = *xr0pnt - v; ++ *xr0pnt++ += v; ++ } ++ } ++ else ++ *xrpnt++ = *xr0pnt++; ++ } ++ } ++ for(i=(&xr[1][SBLIMIT][0]-xrpnt)>>1;i;i--) { ++ *xrpnt++ = *xr0pnt++; ++ *xrpnt++ = *xr0pnt++; ++ } ++ ++ gr_info->maxbandl = max+1; ++ gr_info->maxb = longLimit[sfreq][gr_info->maxbandl]; ++ } ++ ++ while ( part2remain > 16 ) { ++ getbits(16); /* Dismiss stuffing Bits */ ++ part2remain -= 16; ++ } ++ if(part2remain > 0 ) ++ getbits(part2remain); ++ else if(part2remain < 0) { ++ fprintf(stderr,"mpg123_ms: Can't rewind stream by %d bits!\n",-part2remain); ++ return 1; /* -> error */ ++ } ++ return 0; ++ } ++ #endif ++ ++ /* ++ * III_stereo: calculate real channel values for Joint-I-Stereo-mode ++ */ ++ static void III_i_stereo(real xr_buf[2][SBLIMIT][SSLIMIT],int *scalefac, ++ struct gr_info_s *gr_info,int sfreq,int ms_stereo,int lsf) ++ { ++ real (*xr)[SBLIMIT*SSLIMIT] = (real (*)[SBLIMIT*SSLIMIT] ) xr_buf; ++ struct bandInfoStruct *bi = &bandInfo[sfreq]; ++ real *tab1,*tab2; ++ ++ if(lsf) { ++ int p = gr_info->scalefac_compress & 0x1; ++ if(ms_stereo) { ++ tab1 = pow1_2[p]; tab2 = pow2_2[p]; ++ } ++ else { ++ tab1 = pow1_1[p]; tab2 = pow2_1[p]; ++ } ++ } ++ else { ++ if(ms_stereo) { ++ tab1 = tan1_2; tab2 = tan2_2; ++ } ++ else { ++ tab1 = tan1_1; tab2 = tan2_1; ++ } ++ } ++ ++ if (gr_info->block_type == 2) ++ { ++ int lwin,do_l = 0; ++ if( gr_info->mixed_block_flag ) ++ do_l = 1; ++ ++ for (lwin=0;lwin<3;lwin++) /* process each window */ ++ { ++ /* get first band with zero values */ ++ int is_p,sb,idx,sfb = gr_info->maxband[lwin]; /* sfb is minimal 3 for mixed mode */ ++ if(sfb > 3) ++ do_l = 0; ++ ++ for(;sfb<12;sfb++) ++ { ++ is_p = scalefac[sfb*3+lwin-gr_info->mixed_block_flag]; /* scale: 0-15 */ ++ if(is_p != 7) { ++ real t1,t2; ++ sb = bi->shortDiff[sfb]; ++ idx = bi->shortIdx[sfb] + lwin; ++ t1 = tab1[is_p]; t2 = tab2[is_p]; ++ for (; sb > 0; sb--,idx+=3) ++ { ++ real v = xr[0][idx]; ++ xr[0][idx] = v * t1; ++ xr[1][idx] = v * t2; ++ } ++ } ++ } ++ ++ #if 1 ++ /* in the original: copy 10 to 11 , here: copy 11 to 12 ++ maybe still wrong??? (copy 12 to 13?) */ ++ is_p = scalefac[11*3+lwin-gr_info->mixed_block_flag]; /* scale: 0-15 */ ++ sb = bi->shortDiff[12]; ++ idx = bi->shortIdx[12] + lwin; ++ #else ++ is_p = scalefac[10*3+lwin-gr_info->mixed_block_flag]; /* scale: 0-15 */ ++ sb = bi->shortDiff[11]; ++ idx = bi->shortIdx[11] + lwin; ++ #endif ++ if(is_p != 7) ++ { ++ real t1,t2; ++ t1 = tab1[is_p]; t2 = tab2[is_p]; ++ for ( ; sb > 0; sb--,idx+=3 ) ++ { ++ real v = xr[0][idx]; ++ xr[0][idx] = v * t1; ++ xr[1][idx] = v * t2; ++ } ++ } ++ } /* end for(lwin; .. ; . ) */ ++ ++ if (do_l) ++ { ++ /* also check l-part, if ALL bands in the three windows are 'empty' ++ * and mode = mixed_mode ++ */ ++ int sfb = gr_info->maxbandl; ++ int idx = bi->longIdx[sfb]; ++ ++ for ( ; sfb<8; sfb++ ) ++ { ++ int sb = bi->longDiff[sfb]; ++ int is_p = scalefac[sfb]; /* scale: 0-15 */ ++ if(is_p != 7) { ++ real t1,t2; ++ t1 = tab1[is_p]; t2 = tab2[is_p]; ++ for ( ; sb > 0; sb--,idx++) ++ { ++ real v = xr[0][idx]; ++ xr[0][idx] = v * t1; ++ xr[1][idx] = v * t2; ++ } ++ } ++ else ++ idx += sb; ++ } ++ } ++ } ++ else /* ((gr_info->block_type != 2)) */ ++ { ++ int sfb = gr_info->maxbandl; ++ int is_p,idx = bi->longIdx[sfb]; ++ for ( ; sfb<21; sfb++) ++ { ++ int sb = bi->longDiff[sfb]; ++ is_p = scalefac[sfb]; /* scale: 0-15 */ ++ if(is_p != 7) { ++ real t1,t2; ++ t1 = tab1[is_p]; t2 = tab2[is_p]; ++ for ( ; sb > 0; sb--,idx++) ++ { ++ real v = xr[0][idx]; ++ xr[0][idx] = v * t1; ++ xr[1][idx] = v * t2; ++ } ++ } ++ else ++ idx += sb; ++ } ++ ++ is_p = scalefac[20]; /* copy l-band 20 to l-band 21 */ ++ if(is_p != 7) ++ { ++ int sb; ++ real t1 = tab1[is_p],t2 = tab2[is_p]; ++ ++ for ( sb = bi->longDiff[21]; sb > 0; sb--,idx++ ) ++ { ++ real v = xr[0][idx]; ++ xr[0][idx] = v * t1; ++ xr[1][idx] = v * t2; ++ } ++ } ++ } /* ... */ ++ } ++ ++ static void III_antialias(real xr[SBLIMIT][SSLIMIT],struct gr_info_s *gr_info) ++ { ++ int sblim; ++ ++ if(gr_info->block_type == 2) ++ { ++ if(!gr_info->mixed_block_flag) ++ return; ++ sblim = 1; ++ } ++ else { ++ sblim = gr_info->maxb-1; ++ } ++ ++ /* 31 alias-reduction operations between each pair of sub-bands */ ++ /* with 8 butterflies between each pair */ ++ ++ { ++ int sb; ++ real *xr1=(real *) xr[1]; ++ ++ for(sb=sblim;sb;sb--,xr1+=10) ++ { ++ int ss; ++ real *cs=aa_cs,*ca=aa_ca; ++ real *xr2 = xr1; ++ ++ for(ss=7;ss>=0;ss--) ++ { /* upper and lower butterfly inputs */ ++ register real bu = *--xr2,bd = *xr1; ++ *xr2 = (bu * (*cs) ) - (bd * (*ca) ); ++ *xr1++ = (bd * (*cs++) ) + (bu * (*ca++) ); ++ } ++ } ++ } ++ } ++ ++ /* ++ DCT insipired by Jeff Tsay's DCT from the maplay package ++ this is an optimized version with manual unroll. ++ ++ References: ++ [1] S. Winograd: "On Computing the Discrete Fourier Transform", ++ Mathematics of Computation, Volume 32, Number 141, January 1978, ++ Pages 175-199 ++ */ ++ ++ static void dct36(real *inbuf,real *o1,real *o2,real *wintab,real *tsbuf) ++ { ++ { ++ register real *in = inbuf; ++ ++ in[17]+=in[16]; in[16]+=in[15]; in[15]+=in[14]; ++ in[14]+=in[13]; in[13]+=in[12]; in[12]+=in[11]; ++ in[11]+=in[10]; in[10]+=in[9]; in[9] +=in[8]; ++ in[8] +=in[7]; in[7] +=in[6]; in[6] +=in[5]; ++ in[5] +=in[4]; in[4] +=in[3]; in[3] +=in[2]; ++ in[2] +=in[1]; in[1] +=in[0]; ++ ++ in[17]+=in[15]; in[15]+=in[13]; in[13]+=in[11]; in[11]+=in[9]; ++ in[9] +=in[7]; in[7] +=in[5]; in[5] +=in[3]; in[3] +=in[1]; ++ ++ ++ { ++ ++ #define MACRO0(v) { \ ++ real tmp; \ ++ out2[9+(v)] = (tmp = sum0 + sum1) * w[27+(v)]; \ ++ out2[8-(v)] = tmp * w[26-(v)]; } \ ++ sum0 -= sum1; \ ++ ts[SBLIMIT*(8-(v))] = out1[8-(v)] + sum0 * w[8-(v)]; \ ++ ts[SBLIMIT*(9+(v))] = out1[9+(v)] + sum0 * w[9+(v)]; ++ #define MACRO1(v) { \ ++ real sum0,sum1; \ ++ sum0 = tmp1a + tmp2a; \ ++ sum1 = (tmp1b + tmp2b) * tfcos36[(v)]; \ ++ MACRO0(v); } ++ #define MACRO2(v) { \ ++ real sum0,sum1; \ ++ sum0 = tmp2a - tmp1a; \ ++ sum1 = (tmp2b - tmp1b) * tfcos36[(v)]; \ ++ MACRO0(v); } ++ ++ register const real *c = COS9; ++ register real *out2 = o2; ++ register real *w = wintab; ++ register real *out1 = o1; ++ register real *ts = tsbuf; ++ ++ real ta33,ta66,tb33,tb66; ++ ++ ta33 = in[2*3+0] * c[3]; ++ ta66 = in[2*6+0] * c[6]; ++ tb33 = in[2*3+1] * c[3]; ++ tb66 = in[2*6+1] * c[6]; ++ ++ { ++ real tmp1a,tmp2a,tmp1b,tmp2b; ++ tmp1a = in[2*1+0] * c[1] + ta33 + in[2*5+0] * c[5] + in[2*7+0] * c[7]; ++ tmp1b = in[2*1+1] * c[1] + tb33 + in[2*5+1] * c[5] + in[2*7+1] * c[7]; ++ tmp2a = in[2*0+0] + in[2*2+0] * c[2] + in[2*4+0] * c[4] + ta66 + in[2*8+0] * c[8]; ++ tmp2b = in[2*0+1] + in[2*2+1] * c[2] + in[2*4+1] * c[4] + tb66 + in[2*8+1] * c[8]; ++ ++ MACRO1(0); ++ MACRO2(8); ++ } ++ ++ { ++ real tmp1a,tmp2a,tmp1b,tmp2b; ++ tmp1a = ( in[2*1+0] - in[2*5+0] - in[2*7+0] ) * c[3]; ++ tmp1b = ( in[2*1+1] - in[2*5+1] - in[2*7+1] ) * c[3]; ++ tmp2a = ( in[2*2+0] - in[2*4+0] - in[2*8+0] ) * c[6] - in[2*6+0] + in[2*0+0]; ++ tmp2b = ( in[2*2+1] - in[2*4+1] - in[2*8+1] ) * c[6] - in[2*6+1] + in[2*0+1]; ++ ++ MACRO1(1); ++ MACRO2(7); ++ } ++ ++ { ++ real tmp1a,tmp2a,tmp1b,tmp2b; ++ tmp1a = in[2*1+0] * c[5] - ta33 - in[2*5+0] * c[7] + in[2*7+0] * c[1]; ++ tmp1b = in[2*1+1] * c[5] - tb33 - in[2*5+1] * c[7] + in[2*7+1] * c[1]; ++ tmp2a = in[2*0+0] - in[2*2+0] * c[8] - in[2*4+0] * c[2] + ta66 + in[2*8+0] * c[4]; ++ tmp2b = in[2*0+1] - in[2*2+1] * c[8] - in[2*4+1] * c[2] + tb66 + in[2*8+1] * c[4]; ++ ++ MACRO1(2); ++ MACRO2(6); ++ } ++ ++ { ++ real tmp1a,tmp2a,tmp1b,tmp2b; ++ tmp1a = in[2*1+0] * c[7] - ta33 + in[2*5+0] * c[1] - in[2*7+0] * c[5]; ++ tmp1b = in[2*1+1] * c[7] - tb33 + in[2*5+1] * c[1] - in[2*7+1] * c[5]; ++ tmp2a = in[2*0+0] - in[2*2+0] * c[4] + in[2*4+0] * c[8] + ta66 - in[2*8+0] * c[2]; ++ tmp2b = in[2*0+1] - in[2*2+1] * c[4] + in[2*4+1] * c[8] + tb66 - in[2*8+1] * c[2]; ++ ++ MACRO1(3); ++ MACRO2(5); ++ } ++ ++ { ++ real sum0,sum1; ++ sum0 = in[2*0+0] - in[2*2+0] + in[2*4+0] - in[2*6+0] + in[2*8+0]; ++ sum1 = (in[2*0+1] - in[2*2+1] + in[2*4+1] - in[2*6+1] + in[2*8+1] ) * tfcos36[4]; ++ MACRO0(4); ++ } ++ } ++ ++ } ++ } ++ ++ /* ++ * new DCT12 ++ */ ++ static void dct12(real *in,real *rawout1,real *rawout2,register real *wi,register real *ts) ++ { ++ #define DCT12_PART1 \ ++ in5 = in[5*3]; \ ++ in5 += (in4 = in[4*3]); \ ++ in4 += (in3 = in[3*3]); \ ++ in3 += (in2 = in[2*3]); \ ++ in2 += (in1 = in[1*3]); \ ++ in1 += (in0 = in[0*3]); \ ++ \ ++ in5 += in3; in3 += in1; \ ++ \ ++ in2 *= COS6_1; \ ++ in3 *= COS6_1; \ ++ ++ #define DCT12_PART2 \ ++ in0 += in4 * COS6_2; \ ++ \ ++ in4 = in0 + in2; \ ++ in0 -= in2; \ ++ \ ++ in1 += in5 * COS6_2; \ ++ \ ++ in5 = (in1 + in3) * tfcos12[0]; \ ++ in1 = (in1 - in3) * tfcos12[2]; \ ++ \ ++ in3 = in4 + in5; \ ++ in4 -= in5; \ ++ \ ++ in2 = in0 + in1; \ ++ in0 -= in1; ++ ++ ++ { ++ real in0,in1,in2,in3,in4,in5; ++ register real *out1 = rawout1; ++ ts[SBLIMIT*0] = out1[0]; ts[SBLIMIT*1] = out1[1]; ts[SBLIMIT*2] = out1[2]; ++ ts[SBLIMIT*3] = out1[3]; ts[SBLIMIT*4] = out1[4]; ts[SBLIMIT*5] = out1[5]; ++ ++ DCT12_PART1 ++ ++ { ++ real tmp0,tmp1 = (in0 - in4); ++ { ++ real tmp2 = (in1 - in5) * tfcos12[1]; ++ tmp0 = tmp1 + tmp2; ++ tmp1 -= tmp2; ++ } ++ ts[(17-1)*SBLIMIT] = out1[17-1] + tmp0 * wi[11-1]; ++ ts[(12+1)*SBLIMIT] = out1[12+1] + tmp0 * wi[6+1]; ++ ts[(6 +1)*SBLIMIT] = out1[6 +1] + tmp1 * wi[1]; ++ ts[(11-1)*SBLIMIT] = out1[11-1] + tmp1 * wi[5-1]; ++ } ++ ++ DCT12_PART2 ++ ++ ts[(17-0)*SBLIMIT] = out1[17-0] + in2 * wi[11-0]; ++ ts[(12+0)*SBLIMIT] = out1[12+0] + in2 * wi[6+0]; ++ ts[(12+2)*SBLIMIT] = out1[12+2] + in3 * wi[6+2]; ++ ts[(17-2)*SBLIMIT] = out1[17-2] + in3 * wi[11-2]; ++ ++ ts[(6+0)*SBLIMIT] = out1[6+0] + in0 * wi[0]; ++ ts[(11-0)*SBLIMIT] = out1[11-0] + in0 * wi[5-0]; ++ ts[(6+2)*SBLIMIT] = out1[6+2] + in4 * wi[2]; ++ ts[(11-2)*SBLIMIT] = out1[11-2] + in4 * wi[5-2]; ++ } ++ ++ in++; ++ ++ { ++ real in0,in1,in2,in3,in4,in5; ++ register real *out2 = rawout2; ++ ++ DCT12_PART1 ++ ++ { ++ real tmp0,tmp1 = (in0 - in4); ++ { ++ real tmp2 = (in1 - in5) * tfcos12[1]; ++ tmp0 = tmp1 + tmp2; ++ tmp1 -= tmp2; ++ } ++ out2[5-1] = tmp0 * wi[11-1]; ++ out2[0+1] = tmp0 * wi[6+1]; ++ ts[(12+1)*SBLIMIT] += tmp1 * wi[1]; ++ ts[(17-1)*SBLIMIT] += tmp1 * wi[5-1]; ++ } ++ ++ DCT12_PART2 ++ ++ out2[5-0] = in2 * wi[11-0]; ++ out2[0+0] = in2 * wi[6+0]; ++ out2[0+2] = in3 * wi[6+2]; ++ out2[5-2] = in3 * wi[11-2]; ++ ++ ts[(12+0)*SBLIMIT] += in0 * wi[0]; ++ ts[(17-0)*SBLIMIT] += in0 * wi[5-0]; ++ ts[(12+2)*SBLIMIT] += in4 * wi[2]; ++ ts[(17-2)*SBLIMIT] += in4 * wi[5-2]; ++ } ++ ++ in++; ++ ++ { ++ real in0,in1,in2,in3,in4,in5; ++ register real *out2 = rawout2; ++ out2[12]=out2[13]=out2[14]=out2[15]=out2[16]=out2[17]=0.0; ++ ++ DCT12_PART1 ++ ++ { ++ real tmp0,tmp1 = (in0 - in4); ++ { ++ real tmp2 = (in1 - in5) * tfcos12[1]; ++ tmp0 = tmp1 + tmp2; ++ tmp1 -= tmp2; ++ } ++ out2[11-1] = tmp0 * wi[11-1]; ++ out2[6 +1] = tmp0 * wi[6+1]; ++ out2[0+1] += tmp1 * wi[1]; ++ out2[5-1] += tmp1 * wi[5-1]; ++ } ++ ++ DCT12_PART2 ++ ++ out2[11-0] = in2 * wi[11-0]; ++ out2[6 +0] = in2 * wi[6+0]; ++ out2[6 +2] = in3 * wi[6+2]; ++ out2[11-2] = in3 * wi[11-2]; ++ ++ out2[0+0] += in0 * wi[0]; ++ out2[5-0] += in0 * wi[5-0]; ++ out2[0+2] += in4 * wi[2]; ++ out2[5-2] += in4 * wi[5-2]; ++ } ++ } ++ ++ /* ++ * III_hybrid ++ */ ++ static void III_hybrid(real fsIn[SBLIMIT][SSLIMIT],real tsOut[SSLIMIT][SBLIMIT], ++ int ch,struct gr_info_s *gr_info) ++ { ++ real *tspnt = (real *) tsOut; ++ real (*block)[2][SBLIMIT*SSLIMIT] = gmp->hybrid_block; ++ int *blc = gmp->hybrid_blc; ++ real *rawout1,*rawout2; ++ int bt; ++ int sb = 0; ++ ++ { ++ int b = blc[ch]; ++ rawout1=block[b][ch]; ++ b=-b+1; ++ rawout2=block[b][ch]; ++ blc[ch] = b; ++ } ++ ++ ++ if(gr_info->mixed_block_flag) { ++ sb = 2; ++ dct36(fsIn[0],rawout1,rawout2,win[0],tspnt); ++ dct36(fsIn[1],rawout1+18,rawout2+18,win1[0],tspnt+1); ++ rawout1 += 36; rawout2 += 36; tspnt += 2; ++ } ++ ++ bt = gr_info->block_type; ++ if(bt == 2) { ++ for (; sb<gr_info->maxb; sb+=2,tspnt+=2,rawout1+=36,rawout2+=36) { ++ dct12(fsIn[sb],rawout1,rawout2,win[2],tspnt); ++ dct12(fsIn[sb+1],rawout1+18,rawout2+18,win1[2],tspnt+1); ++ } ++ } ++ else { ++ for (; sb<gr_info->maxb; sb+=2,tspnt+=2,rawout1+=36,rawout2+=36) { ++ dct36(fsIn[sb],rawout1,rawout2,win[bt],tspnt); ++ dct36(fsIn[sb+1],rawout1+18,rawout2+18,win1[bt],tspnt+1); ++ } ++ } ++ ++ for(;sb<SBLIMIT;sb++,tspnt++) { ++ int i; ++ for(i=0;i<SSLIMIT;i++) { ++ tspnt[i*SBLIMIT] = *rawout1++; ++ *rawout2++ = 0.0; ++ } ++ } ++ } ++ ++ /* ++ * main layer3 handler ++ */ ++ int do_layer3(struct frame *fr,unsigned char *pcm_sample,int *pcm_point) ++ { ++ int gr, ch, ss,clip=0; ++ int scalefacs[39]; /* max 39 for short[13][3] mode, mixed: 38, long: 22 */ ++ struct III_sideinfo sideinfo; ++ int stereo = fr->stereo; ++ int single = fr->single; ++ int ms_stereo,i_stereo; ++ int sfreq = fr->sampling_frequency; ++ int stereo1,granules; ++ ++ ++ ++ if(stereo == 1) { /* stream is mono */ ++ stereo1 = 1; ++ single = 0; ++ } ++ else if(single >= 0) /* stream is stereo, but force to mono */ ++ stereo1 = 1; ++ else ++ stereo1 = 2; ++ ++ if(fr->mode == MPG_MD_JOINT_STEREO) { ++ ms_stereo = fr->mode_ext & 0x2; ++ i_stereo = fr->mode_ext & 0x1; ++ } ++ else ++ ms_stereo = i_stereo = 0; ++ ++ ++ if(fr->lsf) { ++ granules = 1; ++ III_get_side_info_2(&sideinfo,stereo,ms_stereo,sfreq,single); ++ } ++ else { ++ granules = 2; ++ #ifdef MPEG1 ++ III_get_side_info_1(&sideinfo,stereo,ms_stereo,sfreq,single); ++ #else ++ fprintf(stderr,"Not supported\n"); ++ #endif ++ } ++ ++ if(set_pointer(sideinfo.main_data_begin) == MP3_ERR) ++ return 0; ++ ++ ++ for (gr=0;gr<granules;gr++) ++ { ++ static real hybridIn[2][SBLIMIT][SSLIMIT]; ++ static real hybridOut[2][SSLIMIT][SBLIMIT]; ++ ++ { ++ struct gr_info_s *gr_info = &(sideinfo.ch[0].gr[gr]); ++ long part2bits; ++ if(fr->lsf) ++ part2bits = III_get_scale_factors_2(scalefacs,gr_info,0); ++ else { ++ #ifdef MPEG1 ++ part2bits = III_get_scale_factors_1(scalefacs,gr_info); ++ #else ++ fprintf(stderr,"Not supported\n"); ++ #endif ++ } ++ if(III_dequantize_sample(hybridIn[0], scalefacs,gr_info,sfreq,part2bits)) ++ return clip; ++ } ++ if(stereo == 2) { ++ struct gr_info_s *gr_info = &(sideinfo.ch[1].gr[gr]); ++ long part2bits; ++ if(fr->lsf) ++ part2bits = III_get_scale_factors_2(scalefacs,gr_info,i_stereo); ++ else { ++ #ifdef MPEG1 ++ part2bits = III_get_scale_factors_1(scalefacs,gr_info); ++ #else ++ fprintf(stderr,"Not supported\n"); ++ #endif ++ } ++ ++ if(III_dequantize_sample(hybridIn[1],scalefacs,gr_info,sfreq,part2bits)) ++ return clip; ++ ++ if(ms_stereo) { ++ int i; ++ for(i=0;i<SBLIMIT*SSLIMIT;i++) { ++ real tmp0,tmp1; ++ tmp0 = ((real *) hybridIn[0])[i]; ++ tmp1 = ((real *) hybridIn[1])[i]; ++ ((real *) hybridIn[1])[i] = tmp0 - tmp1; ++ ((real *) hybridIn[0])[i] = tmp0 + tmp1; ++ } ++ } ++ ++ if(i_stereo) ++ III_i_stereo(hybridIn,scalefacs,gr_info,sfreq,ms_stereo,fr->lsf); ++ ++ if(ms_stereo || i_stereo || (single == 3) ) { ++ if(gr_info->maxb > sideinfo.ch[0].gr[gr].maxb) ++ sideinfo.ch[0].gr[gr].maxb = gr_info->maxb; ++ else ++ gr_info->maxb = sideinfo.ch[0].gr[gr].maxb; ++ } ++ ++ switch(single) { ++ case 3: ++ { ++ register int i; ++ register real *in0 = (real *) hybridIn[0],*in1 = (real *) hybridIn[1]; ++ for(i=0;i<SSLIMIT*gr_info->maxb;i++,in0++) ++ *in0 = (*in0 + *in1++); /* *0.5 done by pow-scale */ ++ } ++ break; ++ case 1: ++ { ++ register int i; ++ register real *in0 = (real *) hybridIn[0],*in1 = (real *) hybridIn[1]; ++ for(i=0;i<SSLIMIT*gr_info->maxb;i++) ++ *in0++ = *in1++; ++ } ++ break; ++ } ++ } ++ ++ #ifdef HAVEGTK ++ { ++ extern int tabsel_123[2][3][16]; ++ int sb; ++ ++ pinfo->bitrate = ++ tabsel_123[fr->lsf][fr->lay-1][fr->bitrate_index]; ++ pinfo->sampfreq = freqs[sfreq]; ++ pinfo->emph = fr->emphasis; ++ pinfo->stereo = fr->stereo; ++ pinfo->js = (fr->mode == MPG_MD_JOINT_STEREO); ++ pinfo->ms_stereo = ms_stereo; ++ pinfo->i_stereo = i_stereo; ++ pinfo->maindata = sideinfo.main_data_begin; ++ ++ for(ch=0;ch<stereo1;ch++) { ++ struct gr_info_s *gr_info = &(sideinfo.ch[ch].gr[gr]); ++ pinfo->mixed[gr][ch] = gr_info->mixed_block_flag; ++ pinfo->mpg123blocktype[gr][ch]=gr_info->block_type; ++ pinfo->mainbits[gr][ch] = gr_info->part2_3_length; ++ pinfo->scfsi = gr_info->scfsi; ++ } ++ for(ch=0;ch<stereo1;ch++) { ++ int j=0; ++ for (sb=0;sb<SBLIMIT;sb++) ++ for(ss=0;ss<SSLIMIT;ss++,j++) ++ pinfo->mpg123xr[gr][ch][j]=hybridIn[ch][sb][ss]; ++ } ++ } ++ ++ #endif ++ ++ for(ch=0;ch<stereo1;ch++) { ++ struct gr_info_s *gr_info = &(sideinfo.ch[ch].gr[gr]); ++ III_antialias(hybridIn[ch],gr_info); ++ III_hybrid(hybridIn[ch], hybridOut[ch], ch,gr_info); ++ } ++ ++ for(ss=0;ss<SSLIMIT;ss++) { ++ if(single >= 0) { ++ clip += synth_1to1_mono(hybridOut[0][ss],pcm_sample,pcm_point); ++ } ++ else { ++ int p1 = *pcm_point; ++ clip += synth_1to1(hybridOut[0][ss],0,pcm_sample,&p1); ++ clip += synth_1to1(hybridOut[1][ss],1,pcm_sample,pcm_point); ++ } ++ } ++ } ++ ++ return clip; ++ } ++ ++ ++ #endif ++ +diff -r -c -N encoder/mpglib/main.c lame3.13/mpglib/main.c +*** encoder/mpglib/main.c Wed Dec 31 17:00:00 1969 +--- lame3.13/mpglib/main.c Tue Jun 22 00:11:55 1999 +*************** +*** 0 **** +--- 1,142 ---- ++ #ifdef HAVEGTK ++ #include "mpg123.h" ++ #include "mpglib.h" ++ ++ char buf[16384]={16384*(char)0}; ++ struct mpstr mp; ++ ++ #if 0 ++ void main(void) ++ { ++ int size; ++ char out[8192]; ++ int len,ret; ++ ++ ++ InitMP3(&mp); ++ ++ while(1) { ++ len = read(0,buf,16384); ++ if(len <= 0) ++ break; ++ ret = decodeMP3(&mp,buf,len,out,8192,&size); ++ while(ret == MP3_OK) { ++ write(1,out,size); ++ ret = decodeMP3(&mp,NULL,0,out,8192,&size); ++ } ++ } ++ ++ } ++ #endif ++ ++ int is_syncword(char *header) ++ { ++ ++ /* ++ unsigned int s0,s1; ++ s0 = (unsigned char) header[0]; ++ s1 = (unsigned char) header[1] ; ++ printf(" syncword: %2X %2X \n ",s0, s1); ++ */ ++ ++ /* ++ printf(" integer %i \n",(int) ( header[0] == (char) 0xFF)); ++ printf(" integer %i \n",(int) ( (header[1] & (char) 0xF0) == (char) 0xF0)); ++ */ ++ ++ return ++ ((int) ( header[0] == (char) 0xFF)) && ++ ((int) ( (header[1] & (char) 0xF0) == (char) 0xF0)); ++ ++ ++ } ++ ++ ++ ++ int decode1file(int fd,short pcm[][1152],int *stereo) ++ { ++ int size; ++ #define FSIZE 8192 ++ char out[FSIZE]; ++ int outsize=0,j,i,ret,len; ++ static int firstcall=1; ++ ++ size=0; ++ if (firstcall) { ++ InitMP3(&mp); ++ firstcall=0; ++ /* skip RIFF type proprietary headers */ ++ /* look for sync word FFF */ ++ while (!is_syncword(buf)) { ++ buf[0]=buf[1]; ++ if (read(fd,&buf[1],1) <= 0) return 0; ++ } ++ len = 2; ++ } ++ else len = read(fd,buf,64); ++ if (len <=0 ) return 0; ++ ret = decodeMP3(&mp,buf,len,out,FSIZE,&size); ++ ++ ++ /* read more until we get a valid output frame */ ++ while((ret == MP3_NEED_MORE) || !size) { ++ len = read(fd,buf,100); if (len <=0 ) return 0; ++ ret = decodeMP3(&mp,buf,len,out,FSIZE,&size); ++ /* if (ret ==MP3_ERR) return 0; lets ignore errors and keep reading... */ ++ /* ++ printf("ret = %i size= %i %i %i %i \n",ret,size, ++ MP3_NEED_MORE,MP3_ERR,MP3_OK); ++ */ ++ } ++ ++ *stereo=mp.fr.stereo; ++ if (ret == MP3_OK) { ++ /* write(1,out,size); */ ++ outsize = size/(2*(*stereo)); ++ if ((outsize!=576) && (outsize!=1152)) { ++ fprintf(stderr,"Opps: mpg123 returned more than one frame! Cant handle this... \n"); ++ exit(-50); ++ } ++ ++ for (j=0; j<*stereo; j++) ++ for (i=0; i<outsize; i++) ++ pcm[j][i] = ((short *) out)[*stereo*i+j]; ++ } ++ return outsize; ++ } ++ ++ ++ int decode1(char *buf,int len,short pcm[][1152],int stereo) ++ { ++ int size; ++ #define FSIZE 8192 ++ char out[FSIZE]; ++ int outsize=0,j,i,ret; ++ static int firstcall=1; ++ ++ if (firstcall) { ++ InitMP3(&mp); ++ firstcall=0; ++ } ++ ++ ret = decodeMP3(&mp,buf,len,out,FSIZE,&size); ++ if (ret==MP3_OK) { ++ /* printf("mpg123 output one frame out=%i \n",size/4); */ ++ outsize = size/(2*stereo); ++ if (outsize > 1152) { ++ fprintf(stderr,"Opps: mpg123 returned more than one frame! shouldn't happen... \n"); ++ exit(-50); ++ } ++ ++ for (j=0; j<stereo; j++) ++ for (i=0; i<outsize; i++) ++ pcm[j][i] = ((short *) out)[stereo*i+j]; ++ } ++ /* printf("mpg123 returned %i \n",outsize); */ ++ return outsize; ++ } ++ ++ ++ ++ ++ #endif +diff -r -c -N encoder/mpglib/mpg123.h lame3.13/mpglib/mpg123.h +*** encoder/mpglib/mpg123.h Wed Dec 31 17:00:00 1969 +--- lame3.13/mpglib/mpg123.h Tue Apr 20 22:27:37 1999 +*************** +*** 0 **** +--- 1,180 ---- ++ #include <stdio.h> ++ #include <string.h> ++ #include <signal.h> ++ ++ #ifndef WIN32 ++ #include <sys/signal.h> ++ #include <unistd.h> ++ #endif ++ ++ #include <math.h> ++ ++ #ifdef _WIN32 ++ # undef WIN32 ++ # define WIN32 ++ ++ # define M_PI 3.14159265358979323846 ++ # define M_SQRT2 1.41421356237309504880 ++ # define REAL_IS_FLOAT ++ # define NEW_DCT9 ++ ++ # define random rand ++ # define srandom srand ++ ++ #endif ++ ++ #ifdef REAL_IS_FLOAT ++ # define real float ++ #elif defined(REAL_IS_LONG_DOUBLE) ++ # define real long double ++ #else ++ # define real double ++ #endif ++ ++ #ifdef __GNUC__ ++ #define INLINE inline ++ #else ++ #define INLINE ++ #endif ++ ++ /* AUDIOBUFSIZE = n*64 with n=1,2,3 ... */ ++ #define AUDIOBUFSIZE 16384 ++ ++ #define FALSE 0 ++ #define TRUE 1 ++ ++ #define SBLIMIT 32 ++ #define SSLIMIT 18 ++ ++ #define MPG_MD_STEREO 0 ++ #define MPG_MD_JOINT_STEREO 1 ++ #define MPG_MD_DUAL_CHANNEL 2 ++ #define MPG_MD_MONO 3 ++ ++ #define MAXFRAMESIZE 1792 ++ ++ ++ /* Pre Shift fo 16 to 8 bit converter table */ ++ #define AUSHIFT (3) ++ ++ struct frame { ++ int stereo; ++ int jsbound; ++ int single; ++ int lsf; ++ int mpeg25; ++ int header_change; ++ int lay; ++ int error_protection; ++ int bitrate_index; ++ int sampling_frequency; ++ int padding; ++ int extension; ++ int mode; ++ int mode_ext; ++ int copyright; ++ int original; ++ int emphasis; ++ int framesize; /* computed framesize */ ++ }; ++ ++ struct parameter { ++ int quiet; /* shut up! */ ++ int tryresync; /* resync stream after error */ ++ int verbose; /* verbose level */ ++ int checkrange; ++ }; ++ ++ /* extern unsigned int get1bit(void); */ ++ extern unsigned int getbits(int); ++ extern unsigned int getbits_fast(int); ++ extern int set_pointer(long); ++ ++ extern unsigned char *wordpointer; ++ extern int bitindex; ++ ++ extern void make_decode_tables(long scaleval); ++ extern int do_layer3(struct frame *fr,unsigned char *,int *); ++ extern int decode_header(struct frame *fr,unsigned long newhead); ++ ++ ++ ++ struct gr_info_s { ++ int scfsi; ++ unsigned part2_3_length; ++ unsigned big_values; ++ unsigned scalefac_compress; ++ unsigned block_type; ++ unsigned mixed_block_flag; ++ unsigned table_select[3]; ++ unsigned subblock_gain[3]; ++ unsigned maxband[3]; ++ unsigned maxbandl; ++ unsigned maxb; ++ unsigned region1start; ++ unsigned region2start; ++ unsigned preflag; ++ unsigned scalefac_scale; ++ unsigned count1table_select; ++ real *full_gain[3]; ++ real *pow2gain; ++ }; ++ ++ struct III_sideinfo ++ { ++ unsigned main_data_begin; ++ unsigned private_bits; ++ struct { ++ struct gr_info_s gr[2]; ++ } ch[2]; ++ }; ++ ++ extern int synth_1to1 (real *,int,unsigned char *,int *); ++ extern int synth_1to1_8bit (real *,int,unsigned char *,int *); ++ extern int synth_1to1_mono (real *,unsigned char *,int *); ++ extern int synth_1to1_mono2stereo (real *,unsigned char *,int *); ++ extern int synth_1to1_8bit_mono (real *,unsigned char *,int *); ++ extern int synth_1to1_8bit_mono2stereo (real *,unsigned char *,int *); ++ ++ extern int synth_2to1 (real *,int,unsigned char *,int *); ++ extern int synth_2to1_8bit (real *,int,unsigned char *,int *); ++ extern int synth_2to1_mono (real *,unsigned char *,int *); ++ extern int synth_2to1_mono2stereo (real *,unsigned char *,int *); ++ extern int synth_2to1_8bit_mono (real *,unsigned char *,int *); ++ extern int synth_2to1_8bit_mono2stereo (real *,unsigned char *,int *); ++ ++ extern int synth_4to1 (real *,int,unsigned char *,int *); ++ extern int synth_4to1_8bit (real *,int,unsigned char *,int *); ++ extern int synth_4to1_mono (real *,unsigned char *,int *); ++ extern int synth_4to1_mono2stereo (real *,unsigned char *,int *); ++ extern int synth_4to1_8bit_mono (real *,unsigned char *,int *); ++ extern int synth_4to1_8bit_mono2stereo (real *,unsigned char *,int *); ++ ++ extern int synth_ntom (real *,int,unsigned char *,int *); ++ extern int synth_ntom_8bit (real *,int,unsigned char *,int *); ++ extern int synth_ntom_mono (real *,unsigned char *,int *); ++ extern int synth_ntom_mono2stereo (real *,unsigned char *,int *); ++ extern int synth_ntom_8bit_mono (real *,unsigned char *,int *); ++ extern int synth_ntom_8bit_mono2stereo (real *,unsigned char *,int *); ++ ++ extern void rewindNbits(int bits); ++ extern int hsstell(void); ++ extern int get_songlen(struct frame *fr,int no); ++ ++ extern void init_layer3(int); ++ extern void init_layer2(void); ++ extern void make_decode_tables(long scale); ++ extern void make_conv16to8_table(int); ++ extern void dct64(real *,real *,real *); ++ ++ extern void synth_ntom_set_step(long,long); ++ ++ extern unsigned char *conv16to8; ++ extern long freqs[9]; ++ extern real muls[27][64]; ++ extern real decwin[512+32]; ++ extern real *pnts[5]; ++ ++ extern struct parameter param; ++ ++ +diff -r -c -N encoder/mpglib/mpglib.dsp lame3.13/mpglib/mpglib.dsp +*** encoder/mpglib/mpglib.dsp Wed Dec 31 17:00:00 1969 +--- lame3.13/mpglib/mpglib.dsp Mon May 24 15:43:30 1999 +*************** +*** 0 **** +--- 1,118 ---- ++ # Microsoft Developer Studio Project File - Name="mpglib" - Package Owner=<4> ++ # Microsoft Developer Studio Generated Build File, Format Version 5.00 ++ # ** DO NOT EDIT ** ++ ++ # TARGTYPE "Win32 (x86) Static Library" 0x0104 ++ ++ CFG=mpglib - Win32 Debug ++ !MESSAGE This is not a valid makefile. To build this project using NMAKE, ++ !MESSAGE use the Export Makefile command and run ++ !MESSAGE ++ !MESSAGE NMAKE /f "mpglib.mak". ++ !MESSAGE ++ !MESSAGE You can specify a configuration when running NMAKE ++ !MESSAGE by defining the macro CFG on the command line. For example: ++ !MESSAGE ++ !MESSAGE NMAKE /f "mpglib.mak" CFG="mpglib - Win32 Debug" ++ !MESSAGE ++ !MESSAGE Possible choices for configuration are: ++ !MESSAGE ++ !MESSAGE "mpglib - Win32 Release" (based on "Win32 (x86) Static Library") ++ !MESSAGE "mpglib - Win32 Debug" (based on "Win32 (x86) Static Library") ++ !MESSAGE ++ ++ # Begin Project ++ # PROP Scc_ProjName "" ++ # PROP Scc_LocalPath "" ++ CPP=xicl.exe ++ ++ !IF "$(CFG)" == "mpglib - Win32 Release" ++ ++ # PROP BASE Use_MFC 0 ++ # PROP BASE Use_Debug_Libraries 0 ++ # PROP BASE Output_Dir "Release" ++ # PROP BASE Intermediate_Dir "Release" ++ # PROP BASE Target_Dir "" ++ # PROP Use_MFC 0 ++ # PROP Use_Debug_Libraries 0 ++ # PROP Output_Dir "Release" ++ # PROP Intermediate_Dir "Release" ++ # PROP Target_Dir "" ++ # ADD BASE CPP /nologo /W3 /GX /O2 /D "WIN32" /D "NDEBUG" /D "_WINDOWS" /YX /FD /c ++ # ADD CPP /nologo /W3 /GX /O2 /D "WIN32" /D "NDEBUG" /D "_WINDOWS" /D "HAVEGTK" /YX /FD /c ++ BSC32=bscmake.exe ++ # ADD BASE BSC32 /nologo ++ # ADD BSC32 /nologo ++ LIB32=link.exe -lib ++ # ADD BASE LIB32 /nologo ++ # ADD LIB32 /nologo ++ ++ !ELSEIF "$(CFG)" == "mpglib - Win32 Debug" ++ ++ # PROP BASE Use_MFC 0 ++ # PROP BASE Use_Debug_Libraries 1 ++ # PROP BASE Output_Dir "Debug" ++ # PROP BASE Intermediate_Dir "Debug" ++ # PROP BASE Target_Dir "" ++ # PROP Use_MFC 0 ++ # PROP Use_Debug_Libraries 1 ++ # PROP Output_Dir "Debug" ++ # PROP Intermediate_Dir "Debug" ++ # PROP Target_Dir "" ++ # ADD BASE CPP /nologo /W3 /GX /Z7 /Od /D "WIN32" /D "_DEBUG" /D "_WINDOWS" /YX /FD /c ++ # ADD CPP /nologo /W3 /GX /Z7 /Od /D "WIN32" /D "_DEBUG" /D "_WINDOWS" /D "HAVEGTK" /YX /FD /c ++ BSC32=bscmake.exe ++ # ADD BASE BSC32 /nologo ++ # ADD BSC32 /nologo ++ LIB32=link.exe -lib ++ # ADD BASE LIB32 /nologo ++ # ADD LIB32 /nologo ++ ++ !ENDIF ++ ++ # Begin Target ++ ++ # Name "mpglib - Win32 Release" ++ # Name "mpglib - Win32 Debug" ++ # Begin Source File ++ ++ SOURCE=.\common.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\dct64_i386.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\decode_i386.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\huffman.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\interface.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\layer3.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\main.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\mpg123.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\mpglib.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\tabinit.c ++ # End Source File ++ # End Target ++ # End Project +diff -r -c -N encoder/mpglib/mpglib.h lame3.13/mpglib/mpglib.h +*** encoder/mpglib/mpglib.h Wed Dec 31 17:00:00 1969 +--- lame3.13/mpglib/mpglib.h Tue Apr 20 22:27:37 1999 +*************** +*** 0 **** +--- 1,44 ---- ++ ++ struct buf { ++ unsigned char *pnt; ++ long size; ++ long pos; ++ struct buf *next; ++ struct buf *prev; ++ }; ++ ++ struct framebuf { ++ struct buf *buf; ++ long pos; ++ struct frame *next; ++ struct frame *prev; ++ }; ++ ++ struct mpstr { ++ struct buf *head,*tail; ++ int bsize; ++ int framesize; ++ int fsizeold; ++ struct frame fr; ++ unsigned char bsspace[2][MAXFRAMESIZE+512]; /* MAXFRAMESIZE */ ++ real hybrid_block[2][2][SBLIMIT*SSLIMIT]; ++ int hybrid_blc[2]; ++ unsigned long header; ++ int bsnum; ++ real synth_buffs[2][2][0x110]; ++ int synth_bo; ++ ++ }; ++ ++ #define BOOL int ++ ++ #define MP3_ERR -1 ++ #define MP3_OK 0 ++ #define MP3_NEED_MORE 1 ++ ++ ++ BOOL InitMP3(struct mpstr *mp); ++ int decodeMP3(struct mpstr *mp,char *inmemory,int inmemsize, ++ char *outmemory,int outmemsize,int *done); ++ void ExitMP3(struct mpstr *mp); ++ +diff -r -c -N encoder/mpglib/tabinit.c lame3.13/mpglib/tabinit.c +*** encoder/mpglib/tabinit.c Wed Dec 31 17:00:00 1969 +--- lame3.13/mpglib/tabinit.c Sun Apr 25 22:14:54 1999 +*************** +*** 0 **** +--- 1,82 ---- ++ #ifdef HAVEGTK ++ #include <stdlib.h> ++ ++ #include "mpg123.h" ++ ++ real decwin[512+32]; ++ static real cos64[16],cos32[8],cos16[4],cos8[2],cos4[1]; ++ real *pnts[] = { cos64,cos32,cos16,cos8,cos4 }; ++ ++ #if 0 ++ static unsigned char *conv16to8_buf = NULL; ++ unsigned char *conv16to8; ++ #endif ++ ++ static long intwinbase[] = { ++ 0, -1, -1, -1, -1, -1, -1, -2, -2, -2, ++ -2, -3, -3, -4, -4, -5, -5, -6, -7, -7, ++ -8, -9, -10, -11, -13, -14, -16, -17, -19, -21, ++ -24, -26, -29, -31, -35, -38, -41, -45, -49, -53, ++ -58, -63, -68, -73, -79, -85, -91, -97, -104, -111, ++ -117, -125, -132, -139, -147, -154, -161, -169, -176, -183, ++ -190, -196, -202, -208, -213, -218, -222, -225, -227, -228, ++ -228, -227, -224, -221, -215, -208, -200, -189, -177, -163, ++ -146, -127, -106, -83, -57, -29, 2, 36, 72, 111, ++ 153, 197, 244, 294, 347, 401, 459, 519, 581, 645, ++ 711, 779, 848, 919, 991, 1064, 1137, 1210, 1283, 1356, ++ 1428, 1498, 1567, 1634, 1698, 1759, 1817, 1870, 1919, 1962, ++ 2001, 2032, 2057, 2075, 2085, 2087, 2080, 2063, 2037, 2000, ++ 1952, 1893, 1822, 1739, 1644, 1535, 1414, 1280, 1131, 970, ++ 794, 605, 402, 185, -45, -288, -545, -814, -1095, -1388, ++ -1692, -2006, -2330, -2663, -3004, -3351, -3705, -4063, -4425, -4788, ++ -5153, -5517, -5879, -6237, -6589, -6935, -7271, -7597, -7910, -8209, ++ -8491, -8755, -8998, -9219, -9416, -9585, -9727, -9838, -9916, -9959, ++ -9966, -9935, -9863, -9750, -9592, -9389, -9139, -8840, -8492, -8092, ++ -7640, -7134, -6574, -5959, -5288, -4561, -3776, -2935, -2037, -1082, ++ -70, 998, 2122, 3300, 4533, 5818, 7154, 8540, 9975, 11455, ++ 12980, 14548, 16155, 17799, 19478, 21189, 22929, 24694, 26482, 28289, ++ 30112, 31947, 33791, 35640, 37489, 39336, 41176, 43006, 44821, 46617, ++ 48390, 50137, 51853, 53534, 55178, 56778, 58333, 59838, 61289, 62684, ++ 64019, 65290, 66494, 67629, 68692, 69679, 70590, 71420, 72169, 72835, ++ 73415, 73908, 74313, 74630, 74856, 74992, 75038 }; ++ ++ void make_decode_tables(long scaleval) ++ { ++ int i,j,k,kr,divv; ++ real *table,*costab; ++ ++ ++ for(i=0;i<5;i++) ++ { ++ kr=0x10>>i; divv=0x40>>i; ++ costab = pnts[i]; ++ for(k=0;k<kr;k++) ++ costab[k] = 1.0 / (2.0 * cos(M_PI * ((double) k * 2.0 + 1.0) / (double) divv)); ++ } ++ ++ table = decwin; ++ scaleval = -scaleval; ++ for(i=0,j=0;i<256;i++,j++,table+=32) ++ { ++ if(table < decwin+512+16) ++ table[16] = table[0] = (double) intwinbase[j] / 65536.0 * (double) scaleval; ++ if(i % 32 == 31) ++ table -= 1023; ++ if(i % 64 == 63) ++ scaleval = - scaleval; ++ } ++ ++ for( /* i=256 */ ;i<512;i++,j--,table+=32) ++ { ++ if(table < decwin+512+16) ++ table[16] = table[0] = (double) intwinbase[j] / 65536.0 * (double) scaleval; ++ if(i % 32 == 31) ++ table -= 1023; ++ if(i % 64 == 63) ++ scaleval = - scaleval; ++ } ++ } ++ ++ ++ #endif ++ +diff -r -c -N encoder/musicin.c lame3.13/musicin.c +*** encoder/musicin.c Wed Jan 22 02:43:17 1997 +--- lame3.13/musicin.c Thu Jun 24 18:15:58 1999 +*************** +*** 13,401 **** + * + * Received from Mike Coleman + **********************************************************************/ +! /********************************************************************** +! * date programmers comment * +! * 3/01/91 Douglas Wong, start of version 1.1 records * +! * Davis Pan * +! * 3/06/91 Douglas Wong, rename: setup.h to endef.h * +! * removed extraneous variables * +! * 3/21/91 J.Georges Fritsch introduction of the bit-stream * +! * package. This package allows you * +! * to generate the bit-stream in a * +! * binary or ascii format * +! * 3/31/91 Bill Aspromonte replaced the read of the SB matrix * +! * by an "code generated" one * +! * 5/10/91 W. Joseph Carter Ported to Macintosh and Unix. * +! * Incorporated Jean-Georges Fritsch's * +! * "bitstream.c" package. * +! * Modified to strictly adhere to * +! * encoded bitstream specs, including * +! * "Berlin changes". * +! * Modified user interface dialog & code * +! * to accept any input & output * +! * filenames desired. Also added * +! * de-emphasis prompt and final bail-out * +! * opportunity before encoding. * +! * Added AIFF PCM sound file reading * +! * capability. * +! * Modified PCM sound file handling to * +! * process all incoming samples and fill * +! * out last encoded frame with zeros * +! * (silence) if needed. * +! * Located and fixed numerous software * +! * bugs and table data errors. * +! * 27jun91 dpwe (Aware Inc) Used new frame_params struct. * +! * Clear all automatic arrays. * +! * Changed some variable names, * +! * simplified some code. * +! * Track number of bits actually sent. * +! * Fixed padding slot, stereo bitrate * +! * Added joint-stereo : scales L+R. * +! * 6/12/91 Earle Jennings added fix for MS_DOS in obtain_param * +! * 6/13/91 Earle Jennings added stack length adjustment before * +! * main for MS_DOS * +! * 7/10/91 Earle Jennings conversion of all float to FLOAT * +! * port to MsDos from MacIntosh completed* +! * 8/ 8/91 Jens Spille Change for MS-C6.00 * +! * 8/22/91 Jens Spille new obtain_parameters() * +! *10/ 1/91 S.I. Sudharsanan, Ported to IBM AIX platform. * +! * Don H. Lee, * +! * Peter W. Farrett * +! *10/ 3/91 Don H. Lee implemented CRC-16 error protection * +! * newly introduced functions are * +! * I_CRC_calc, II_CRC_calc and encode_CRC* +! * Additions and revisions are marked * +! * with "dhl" for clarity * +! *11/11/91 Katherine Wang Documentation of code. * +! * (variables in documentation are * +! * surround by the # symbol, and an '*'* +! * denotes layer I or II versions) * +! * 2/11/92 W. Joseph Carter Ported new code to Macintosh. Most * +! * important fixes involved changing * +! * 16-bit ints to long or unsigned in * +! * bit alloc routines for quant of 65535 * +! * and passing proper function args. * +! * Removed "Other Joint Stereo" option * +! * and made bitrate be total channel * +! * bitrate, irrespective of the mode. * +! * Fixed many small bugs & reorganized. * +! * 2/25/92 Masahiro Iwadare made code cleaner and more consistent * +! * 8/07/92 Mike Coleman make exit() codes return error status * +! * made slight changes for portability * +! *19 aug 92 Soren H. Nielsen Changed MS-DOS file name extensions. * +! * 8/25/92 Shaun Astarabadi Replaced rint() function with explicit* +! * rounding for portability with MSDOS. * +! * 9/22/92 jddevine@aware.com Fixed _scale_factor_calc() calls. * +! *10/19/92 Masahiro Iwadare added info->mode and info->mode_ext * +! * updates for AIFF format files * +! * 3/10/93 Kevin Peterson In parse_args, only set non default * +! * bit rate if specified in arg list. * +! * Use return value from aiff_read_hdrs * +! * to fseek to start of sound data * +! * 7/26/93 Davis Pan fixed bug in printing info->mode_ext * +! * value for joint stereo condition * +! * 8/27/93 Seymour Shlien, Fixes in Unix and MSDOS ports, * +! * Daniel Lauzon, and * +! * Bill Truerniet * +! * 11/7/95 Soeren H. Nielsen LSF added. Bug fix in MSDOS ext. * +! * 8/02/95 mc@fivebats.com Changed default bitrate selection so * +! * it works with the new LSF stuff * +! *10/01/95 mc@fivebats.com Added layer3 * +! **********************************************************************/ +! +! #ifdef MS_DOS +! #include <dos.h> +! #endif + #include <stdlib.h> + #include "common.h" + #include "encoder.h" + #include "l3psy.h" + #include "mdct.h" + #include "loop.h" + #include "l3bitstream.h" +! #include <assert.h> + +- /* Global variable definitions for "musicin.c" */ + +! FILE *musicin; +! Bit_stream_struc bs; +! char *programName; + +- /* Implementations */ + +! /************************************************************************ +! * +! * obtain_parameters +! * +! * PURPOSE: Prompts for and reads user input for encoding parameters +! * +! * SEMANTICS: The parameters read are: +! * - input and output filenames +! * - sampling frequency (if AIFF file, will read from the AIFF file header) +! * - layer number +! * - mode (stereo, joint stereo, dual channel or mono) +! * - psychoacoustic model (I or II) +! * - total bitrate, irrespective of the mode +! * - de-emphasis, error protection, copyright and original or copy flags +! * +! ************************************************************************/ + +- void +- obtain_parameters(fr_ps,psy,num_samples,original_file_name,encoded_file_name) +- frame_params *fr_ps; +- int *psy; +- unsigned long *num_samples; +- char original_file_name[MAX_NAME_SIZE]; +- char encoded_file_name[MAX_NAME_SIZE]; +- { +- int j; +- long int freq; +- int model, brt; +- char t[50]; +- IFF_AIFF pcm_aiff_data; +- layer *info = fr_ps->header; +- long soundPosition; +- #ifdef MS_DOS +- char temp_str[MAX_NAME_SIZE]; +- #endif +- +- do { +- printf("Enter PCM input file name <required>: "); +- gets(original_file_name); +- if (original_file_name[0] == NULL_CHAR) +- printf("PCM input file name is required.\n"); +- } while (original_file_name[0] == NULL_CHAR); +- printf(">>> PCM input file name is: %s\n", original_file_name); +- +- if ((musicin = fopen(original_file_name, "rb")) == NULL) { +- printf("Could not find \"%s\".\n", original_file_name); +- exit(1); +- } + +! #ifdef MS_DOS +! /* replace old extension with new one, 1992-08-19, 1995-06-12 shn */ +! new_ext(original_file_name, DFLT_EXT, temp_str); +! printf("Enter MPEG encoded output file name <%s>: ", +! temp_str); +! #else +! printf("Enter MPEG encoded output file name <%s%s>: ", +! original_file_name, DFLT_EXT); +! #endif +! +! gets(encoded_file_name); +! +! if (encoded_file_name[0] == NULL_CHAR) { +! #ifdef MS_DOS +! strcpy(encoded_file_name, temp_str); +! #else +! strcat(strcpy(encoded_file_name, original_file_name), DFLT_EXT); + #endif +! } +! + +- printf(">>> MPEG encoded output file name is: %s\n", encoded_file_name); +- +- open_bit_stream_w(&bs, encoded_file_name, BUFFER_SIZE); +- +- if ((soundPosition = aiff_read_headers(musicin, &pcm_aiff_data)) != -1) { + +- printf(">>> Using Audio IFF sound file headers\n"); + +- aiff_check(original_file_name, &pcm_aiff_data, &info->version); + +! if (fseek(musicin, soundPosition, SEEK_SET) != 0) { +! printf("Could not seek to PCM sound data in \"%s\".\n", +! original_file_name); +! exit(1); +! } +! +! info->sampling_frequency = +! SmpFrqIndex((long)pcm_aiff_data.sampleRate, &info->version); +! printf(">>> %f Hz sampling frequency selected\n", +! pcm_aiff_data.sampleRate); +! +! /* Determine number of samples in sound file */ +! #ifndef MS_DOS +! *num_samples = pcm_aiff_data.numChannels * +! pcm_aiff_data.numSampleFrames; + #else +! *num_samples = (long)(pcm_aiff_data.numChannels) * +! (long)(pcm_aiff_data.numSampleFrames); + #endif + +- } +- else { /* Not using Audio IFF sound file headers. */ + +- printf("What is the sampling frequency? <44100>[Hz]: "); +- gets(t); +- freq = atol(t); +- switch (freq) { +- case 48000 : info->sampling_frequency = 1; +- printf(">>> %ld Hz sampling freq selected\n", freq); +- break; +- case 44100 : info->sampling_frequency = 0; +- printf(">>> %ld Hz sampling freq selected\n", freq); +- break; +- case 32000 : info->sampling_frequency = 2; +- printf(">>> %ld Hz sampling freq selected\n", freq); +- break; +- case 24000 : info->sampling_frequency = 1; +- info->version = MPEG_PHASE2_LSF; +- printf(">>> %ld Hz sampling freq selected\n", freq); +- break; +- case 22050 : info->sampling_frequency = 0; +- info->version = MPEG_PHASE2_LSF; +- printf(">>> %ld Hz sampling freq selected\n", freq); +- break; +- case 16000 : info->sampling_frequency = 2; +- info->version = MPEG_PHASE2_LSF; +- printf(">>> %ld Hz sampling freq selected\n", freq); +- break; +- default: info->sampling_frequency = 0; +- printf(">>> Default 44.1 kHz samp freq selected\n"); +- } +- printf(">>> Encoding algorithm is %s\n", version_names[info->version]); +- if (fseek(musicin, 0, SEEK_SET) != 0) { +- printf("Could not seek to PCM sound data in \"%s\".\n", +- original_file_name); +- exit(1); +- } +- +- /* Declare sound file to have "infinite" number of samples. */ +- *num_samples = MAX_U_32_NUM; + +- } + +- printf("Which layer do you want to use?\n"); +- printf("Available: Layer (1), Layer (<2>), Layer (3): "); +- gets(t); +- switch(*t){ +- case '1': info->lay = 1; printf(">>> Using Layer %s\n",t); break; +- case '2': info->lay = 2; printf(">>> Using Layer %s\n",t); break; +- case '3': info->lay = 3; printf(">>> Using Layer %s\n",t); break; +- default: info->lay = 2; printf(">>> Using default Layer 2\n"); break; +- } +- +- printf("Which mode do you want?\n"); +- printf("Available: (<s>)tereo, (j)oint stereo, "); +- printf("(d)ual channel, s(i)ngle Channel: "); +- gets(t); +- switch(*t){ +- case 's': +- case 'S': +- info->mode = MPG_MD_STEREO; info->mode_ext = 0; +- printf(">>> Using mode %s\n",t); +- break; +- case 'j': +- case 'J': +- info->mode = MPG_MD_JOINT_STEREO; +- printf(">>> Using mode %s\n",t); +- break; +- case 'd': +- case 'D': +- info->mode = MPG_MD_DUAL_CHANNEL; info->mode_ext = 0; +- printf(">>> Using mode %s\n",t); +- break; +- case 'i': +- case 'I': +- info->mode = MPG_MD_MONO; info->mode_ext = 0; +- printf(">>> Using mode %s\n",t); +- break; +- default: +- info->mode = MPG_MD_STEREO; info->mode_ext = 0; +- printf(">>> Using default stereo mode\n"); +- break; +- } +- +- printf("Which psychoacoustic model do you want to use? <1>: "); +- gets(t); +- model = atoi(t); +- if (model > 2 || model < 1) { +- printf(">>> Default model 1 selected\n"); +- *psy = 1; +- } +- else { +- *psy = model; +- printf(">>> Using psychoacoustic model %d\n", model); +- } +- +- /* set default bitrate to highest allowed, which is index 14 */ +- brt = bitrate[info->version][info->lay-1][14]; +- printf( "What is the total bitrate? <%u>[kbps]: ", brt ); +- gets( t ); +- brt = atoi( t ); +- if ( brt == 0 ) +- j = 15; +- else +- j = 0; +- while ( j < 15 ) +- { +- if ( bitrate[info->version][info->lay-1][j] == brt ) +- break; +- j++; +- } +- if ( j == 15 ) +- { +- brt = bitrate[info->version][info->lay-1][14]; +- printf( ">>> Using default %u kbps\n", brt ); +- info->bitrate_index = 14; +- } +- else +- { +- info->bitrate_index = j; +- printf( ">>> Bitrate = %d kbps\n", bitrate[info->version][info->lay-1][j] ); +- } +- +- printf("What type of de-emphasis should the decoder use?\n"); +- printf("Available: (<n>)one, (5)0/15 microseconds, (c)citt j.17: "); +- gets(t); +- if (*t != 'n' && *t != '5' && *t != 'c') { +- printf(">>> Using default no de-emphasis\n"); +- info->emphasis = 0; +- } +- else { +- if (*t == 'n') info->emphasis = 0; +- else if (*t == '5') info->emphasis = 1; +- else if (*t == 'c') info->emphasis = 3; +- printf(">>> Using de-emphasis %s\n",t); +- } +- +- /* Start 2. Part changes for CD Ver 3.2; jsp; 22-Aug-1991 */ +- +- printf("Do you want to set the private bit? (y/<n>): "); +- gets(t); +- if (*t == 'y' || *t == 'Y') info->extension = 1; +- else info->extension = 0; +- if(info->extension) printf(">>> Private bit set\n"); +- else printf(">>> Private bit not set\n"); +- +- /* End changes for CD Ver 3.2; jsp; 22-Aug-1991 */ +- +- printf("Do you want error protection? (y/<n>): "); +- gets(t); +- if (*t == 'y' || *t == 'Y') info->error_protection = TRUE; +- else info->error_protection = FALSE; +- if(info->error_protection) printf(">>> Error protection used\n"); +- else printf(">>> Error protection not used\n"); +- +- printf("Is the material copyrighted? (y/<n>): "); +- gets(t); +- if (*t == 'y' || *t == 'Y') info->copyright = 1; +- else info->copyright = 0; +- if(info->copyright) printf(">>> Copyrighted material\n"); +- else printf(">>> Material not copyrighted\n"); +- +- printf("Is this the original? (y/<n>): "); +- gets(t); +- if (*t == 'y' || *t == 'Y') info->original = 1; +- else info->original = 0; +- if(info->original) printf(">>> Original material\n"); +- else printf(">>> Material not original\n"); +- +- printf("Do you wish to exit (last chance before encoding)? (y/<n>): "); +- gets(t); +- if (*t == 'y' || *t == 'Y') exit(0); +- } + + /************************************************************************ + * +--- 13,168 ---- + * + * Received from Mike Coleman + **********************************************************************/ +! #include <sys/types.h> +! #include <sys/stat.h> +! #include <sys/times.h> +! #include <fcntl.h> + #include <stdlib.h> ++ #include <time.h> ++ #include <assert.h> ++ #ifdef _WIN32 ++ #include <io.h> ++ #endif ++ #ifdef __FreeBSD__ ++ #include <floatingpoint.h> ++ #endif ++ #ifdef ABORTFP ++ #include <fpu_control.h> ++ #endif ++ + #include "common.h" ++ #include "globalflags.h" + #include "encoder.h" + #include "l3psy.h" + #include "mdct.h" + #include "loop.h" + #include "l3bitstream.h" +! #include "gtkanal.h" +! #include "version.h" +! #include "VbrTag.h" +! #include "wavheader.h" +! #ifdef LIBSNDFILE +! #include "get_audio.h" +! #endif + + +! /* Global flags. set defaults here, defined extern in common.h */ +! int autoconvert=FALSE; +! int force_ms=FALSE; +! int gpsycho=1; +! int sfb21=1; +! int experimentalX = 0; +! int experimentalY = 0; +! int experimentalZ = 0; +! int VBR=0; +! int VBR_q=4; +! int VBR_min_bitrate=6; /* 80kbs */ +! int VBR_max_bitrate=13; /* 256kbs */ +! #ifdef _BLADEDLL +! int silent=1; +! #else +! int silent=0; +! #endif +! int highq=0; +! int fast_mode=0; +! int allow_diff_short=0; +! #ifdef DISTRIB +! int gtkflag=1; +! #else +! int gtkflag=0; +! #endif +! int g_bWriteVbrTag=1; + + +! void usage(); + + +! +! /* Global variable definitions for "musicin.c" */ +! struct stat sb; +! int firstcall = 1; /* define like this so Dll/BladeMP3EncDll.c can change value */ +! int target_bitrate; +! long totalframes; +! int swapbytes = FALSE; /* force byte swapping */ +! FILE *musicin; +! Bit_stream_struc bs; +! char *programName; +! int iswav=0; +! III_side_info_t l3_side; +! frame_params fr_ps; +! unsigned long num_samples; +! int frameNum=0; +! #ifdef HAVEGTK +! static int MPGLAG=1; + #endif +! int original_file_type; /* 0=WAV, 1=MP3 */ +! int original_file_fd; /* file descriptor */ +! + + + + +! /*********************************************************/ +! /* Timeconvert: convert given time (in secs) to hh:mm:ss */ +! /*********************************************************/ +! void timestatus(int frameNum, int totalframes, int samp_rate) +! { +! clock_t time_so_far; +! clock_t remain, estimated; +! double speedup; +! static int firstcall=1; +! static clock_t last_clock_value=0,reference_clock_value=0; +! clock_t clock_value; +! static int accum_time=0; +! struct tms tms_buf; +! static divider; +! if (firstcall) { +! reference_clock_value = times(&tms_buf); +! #ifdef CLK_TCK +! divider=CLK_TCK; + #else +! divider= HZ; /* for old BSD systems */ + #endif ++ /* divider=CLOCKS_PER_SEC; */ /* used with clock() */ ++ firstcall=0; ++ } ++ ++ ++ /* clock_value = clock();*/ /* cpu time */ ++ clock_value = times(&tms_buf); /* wall clock more usefull than cpu time */ ++ ++ if ((clock_value<0 && last_clock_value>0) || ++ (clock_value>0 && last_clock_value<0)) { ++ accum_time+=(last_clock_value-reference_clock_value)/divider; ++ accum_time+=(clock_value-last_clock_value)/divider; ++ reference_clock_value=clock_value; ++ } ++ last_clock_value = clock_value; ++ time_so_far = (clock_value-reference_clock_value)/divider + accum_time; ++ ++ /* calc estimated total CPU time */ ++ estimated = (int)((float)totalframes/(float)frameNum*(float)time_so_far); ++ remain = estimated - time_so_far; ++ ++ /* estimated encoding time / playing time. (protected from divide by 0) */ ++ if (samp_rate*estimated) { ++ speedup = (totalframes*1.152)/(samp_rate*estimated); ++ } ++ else { ++ speedup = 0.0; ++ } ++ ++ fprintf(stderr, ++ "Frame: [%4d/%4d] Encoding Time %02d:%02d/%02d:%02d (%.2fx) ETA %02d:%02d\r", ++ frameNum, totalframes, (int)((time_so_far/60)),(int)( time_so_far%60), ++ (int)((estimated/60)),(int)( estimated%60), speedup, ++ (int) ( (remain/60)), (int)( remain%60)); ++ fflush(stderr); ++ } + + + + + + /************************************************************************ + * +*************** +*** 405,423 **** + * command line. Default settings are used for parameters + * not specified in the command line. + * +- * SEMANTICS: The command line is parsed according to the following +- * syntax: +- * +- * -l is followed by the layer number +- * -m is followed by the mode +- * -p is followed by the psychoacoustic model number +- * -s is followed by the sampling rate +- * -b is followed by the total bitrate, irrespective of the mode +- * -d is followed by the emphasis flag +- * -c is followed by the copyright/no_copyright flag +- * -o is followed by the original/not_original flag +- * -e is followed by the error_protection on/off flag +- * + * If the input file is in AIFF format, the sampling frequency is read + * from the AIFF header. + * +--- 172,177 ---- +*************** +*** 435,661 **** + char inPath[MAX_NAME_SIZE]; + char outPath[MAX_NAME_SIZE]; + { +! FLOAT srate; +! int brate; +! layer *info = fr_ps->header; +! int err = 0, i = 0; +! IFF_AIFF pcm_aiff_data; +! long samplerate; +! long soundPosition; +! +! /* preset defaults */ +! inPath[0] = '\0'; outPath[0] = '\0'; +! info->lay = DFLT_LAY; +! switch(DFLT_MOD) { +! case 's': info->mode = MPG_MD_STEREO; info->mode_ext = 0; break; +! case 'd': info->mode = MPG_MD_DUAL_CHANNEL; info->mode_ext=0; break; +! case 'j': info->mode = MPG_MD_JOINT_STEREO; break; +! case 'm': info->mode = MPG_MD_MONO; info->mode_ext = 0; break; +! default: +! fprintf(stderr, "%s: Bad mode dflt %c\n", programName, DFLT_MOD); +! abort(); +! } +! *psy = DFLT_PSY; +! if((info->sampling_frequency = SmpFrqIndex((long)(1000*DFLT_SFQ), &info->version)) < 0) { +! fprintf(stderr, "%s: bad sfrq default %.2f\n", programName, DFLT_SFQ); +! abort(); +! } +! info->bitrate_index = 14; + brate = 0; +! switch(DFLT_EMP) { +! case 'n': info->emphasis = 0; break; +! case '5': info->emphasis = 1; break; +! case 'c': info->emphasis = 3; break; +! default: +! fprintf(stderr, "%s: Bad emph dflt %c\n", programName, DFLT_EMP); +! abort(); +! } +! info->copyright = 0; info->original = 0; info->error_protection = FALSE; +! +! /* process args */ +! while(++i<argc && err == 0) { +! char c, *token, *arg, *nextArg; +! int argUsed; +! +! token = argv[i]; +! if(*token++ == '-') { +! if(i+1 < argc) nextArg = argv[i+1]; +! else nextArg = ""; +! argUsed = 0; +! while( (c = *token++) ) { +! if(*token /* NumericQ(token) */) arg = token; +! else arg = nextArg; +! switch(c) { +! case 'l': info->lay = atoi(arg); argUsed = 1; +! if(info->lay<1 || info->lay>3) { +! fprintf(stderr,"%s: -l layer must be 1, 2, or 3, not %s\n", +! programName, arg); +! err = 1; +! } +! break; +! case 'm': argUsed = 1; +! if (*arg == 's') +! { info->mode = MPG_MD_STEREO; info->mode_ext = 0; } +! else if (*arg == 'd') +! { info->mode = MPG_MD_DUAL_CHANNEL; info->mode_ext=0; } +! else if (*arg == 'j') +! { info->mode = MPG_MD_JOINT_STEREO; } +! else if (*arg == 'm') +! { info->mode = MPG_MD_MONO; info->mode_ext = 0; } +! else { +! fprintf(stderr,"%s: -m mode must be s/d/j/m not %s\n", +! programName, arg); +! err = 1; +! } +! break; +! case 'p': *psy = atoi(arg); argUsed = 1; +! if(*psy<1 || *psy>2) { +! fprintf(stderr,"%s: -p model must be 1 or 2, not %s\n", +! programName, arg); +! err = 1; +! } +! break; +! +! case 's': +! argUsed = 1; +! srate = atof( arg ); +! /* samplerate = rint( 1000.0 * srate ); $A */ +! samplerate = (long) (( 1000.0 * srate ) + 0.5); +! if( (info->sampling_frequency = +! SmpFrqIndex((long) samplerate, &info->version)) < 0 ) +! err = 1; +! break; +! +! case 'b': +! argUsed = 1; +! brate = atoi(arg); +! break; +! case 'd': argUsed = 1; +! if (*arg == 'n') info->emphasis = 0; +! else if (*arg == '5') info->emphasis = 1; +! else if (*arg == 'c') info->emphasis = 3; +! else { +! fprintf(stderr,"%s: -d emp must be n/5/c not %s\n", +! programName, arg); +! err = 1; +! } +! break; +! case 'c': info->copyright = 1; break; +! case 'o': info->original = 1; break; +! case 'e': info->error_protection = TRUE; break; +! default: fprintf(stderr,"%s: unrec option %c\n", +! programName, c); +! err = 1; break; +! } +! if(argUsed) { +! if(arg == token) token = ""; /* no more from token */ +! else ++i; /* skip arg we used */ +! arg = ""; argUsed = 0; +! } +! } + } + else { +! if(inPath[0] == '\0') strcpy(inPath, argv[i]); +! else if(outPath[0] == '\0') strcpy(outPath, argv[i]); +! else { +! fprintf(stderr,"%s: excess arg %s\n", programName, argv[i]); +! err = 1; +! } + } +! } + +! if(err || inPath[0] == '\0') usage(); /* never returns */ +! +! if(outPath[0] == '\0') { +! #ifdef MS_DOS +! /* replace old extension with new one, 1992-08-19, 1995-06-12 shn */ +! new_ext(inPath, DFLT_EXT, outPath); +! #else + strcpy(outPath, inPath); + strcat(outPath, DFLT_EXT); +! #endif +! } +! +! if ((musicin = fopen(inPath, "rb")) == NULL) { +! printf("Could not find \"%s\".\n", inPath); + exit(1); +! } +! +! open_bit_stream_w(&bs, outPath, BUFFER_SIZE); + +- if ((soundPosition = aiff_read_headers(musicin, &pcm_aiff_data)) != -1) { + +! printf(">>> Using Audio IFF sound file headers\n"); + + aiff_check(inPath, &pcm_aiff_data, &info->version); +- + if (fseek(musicin, soundPosition, SEEK_SET) != 0) { +! printf("Could not seek to PCM sound data in \"%s\".\n", inPath); +! exit(1); + } +- + info->sampling_frequency = SmpFrqIndex((long)pcm_aiff_data.sampleRate, &info->version); +! printf(">>> %f Hz sampling frequency selected\n", +! pcm_aiff_data.sampleRate); +! + /* Determine number of samples in sound file */ +- #ifndef MS_DOS +- *num_samples = pcm_aiff_data.numChannels * +- pcm_aiff_data.numSampleFrames; +- #else + *num_samples = (long)(pcm_aiff_data.numChannels) * +! (long)(pcm_aiff_data.numSampleFrames); +! #endif + if ( pcm_aiff_data.numChannels == 1 ) { +! info->mode = MPG_MD_MONO; +! info->mode_ext = 0; + } +! } +! else { /* Not using Audio IFF sound file headers. */ + +! if (fseek(musicin, 0, SEEK_SET) != 0) { +! printf("Could not seek to PCM sound data in \"%s\".\n", inPath); +! exit(1); + } +! +! /* Declare sound file to have "infinite" number of samples. */ +! *num_samples = MAX_U_32_NUM; + +! } +! if ( brate == 0 ) +! brate = bitrate[info->version][info->lay-1][14]; +! if( (info->bitrate_index = BitrateIndex(info->lay, brate, info->version)) < 0) err=1; +! if(err || inPath[0] == '\0') usage(); /* never returns */ + + } + + /************************************************************************ +! * +! * print_config +! * +! * PURPOSE: Prints the encoding parameters used +! * +! ************************************************************************/ +! + void print_config( frame_params *fr_ps, int *psy, char *inPath, char *outPath) + { +! layer *info = fr_ps->header; +! +! printf("Encoding configuration:\n"); +! printf("Algorithm=%s\n", version_names[info->version]); +! if(info->mode != MPG_MD_JOINT_STEREO) +! printf("Layer=%s mode=%s extn=%d psy model=%d\n", +! layer_names[info->lay-1], mode_names[info->mode], +! info->mode_ext, *psy); +! else printf("Layer=%s mode=%s extn=data dependant psy model=%d\n", +! layer_names[info->lay-1], mode_names[info->mode], *psy); +! printf("samp frq=%.1f kHz total bitrate=%d kbps\n", +! s_freq[info->version][info->sampling_frequency], +! bitrate[info->version][info->lay-1][info->bitrate_index]); +! printf("de-emph=%d c/right=%d orig=%d errprot=%s\n", +! info->emphasis, info->copyright, info->original, +! ((info->error_protection) ? "on" : "off")); +! printf("input file: '%s' output file: '%s'\n", inPath, outPath); + } + + +--- 189,1020 ---- + char inPath[MAX_NAME_SIZE]; + char outPath[MAX_NAME_SIZE]; + { +! FLOAT srate; +! int brate; +! layer *info = fr_ps->header; +! int err = 0, i = 0; +! int mode_given=0; +! #ifndef LIBSNDFILE +! IFF_AIFF pcm_aiff_data; +! long soundPosition; +! char ckwav[5]={0,0,0,0,0}; +! #endif +! long samplerate; +! WaveHeader wavhead; +! +! /* preset defaults */ +! inPath[0] = '\0'; outPath[0] = '\0'; +! info->lay = DFLT_LAY; +! switch(DFLT_MOD) { +! case 's': info->mode = MPG_MD_STEREO; info->mode_ext = 0; break; +! case 'd': info->mode = MPG_MD_DUAL_CHANNEL; info->mode_ext=0; break; +! case 'j': info->mode = MPG_MD_JOINT_STEREO; info->mode_ext=0; break; +! case 'm': info->mode = MPG_MD_MONO; info->mode_ext = 0; break; +! default: +! fprintf(stderr, "%s: Bad mode dflt %c\n", programName, DFLT_MOD); +! abort(); +! } +! *psy = DFLT_PSY; +! if((info->sampling_frequency = SmpFrqIndex((long)(1000*DFLT_SFQ), &info->version)) < 0) { +! fprintf(stderr, "%s: bad sfrq default %.2f\n", programName, DFLT_SFQ); +! abort(); +! } +! info->bitrate_index = 9; + brate = 0; +! switch(DFLT_EMP) { +! case 'n': info->emphasis = 0; break; +! case '5': info->emphasis = 1; break; +! case 'c': info->emphasis = 3; break; +! default: +! fprintf(stderr, "%s: Bad emph dflt %c\n", programName, DFLT_EMP); +! abort(); +! } +! info->copyright = 0; +! info->original = 1; +! info->error_protection = FALSE; +! +! /* process args */ +! while(++i<argc && err == 0) { +! char c, *token, *arg, *nextArg; +! int argUsed; +! +! token = argv[i]; +! if(*token++ == '-') { +! if(i+1 < argc) nextArg = argv[i+1]; +! else nextArg = ""; +! argUsed = 0; +! if (! *token) { +! /* The user wants to use stdin and/or stdout. */ +! if(inPath[0] == '\0') strcpy(inPath, argv[i]); +! else if(outPath[0] == '\0') strcpy(outPath, argv[i]); +! } +! while( (c = *token++) ) { +! if(*token ) arg = token; +! else arg = nextArg; +! switch(c) { +! case 'm': argUsed = 1; +! mode_given=1; +! if (*arg == 's') +! { info->mode = MPG_MD_STEREO; info->mode_ext = 0; } +! else if (*arg == 'd') +! { info->mode = MPG_MD_DUAL_CHANNEL; info->mode_ext=0; } +! else if (*arg == 'j') +! { info->mode = MPG_MD_JOINT_STEREO; info->mode_ext=0; } +! else if (*arg == 'f') +! { info->mode = MPG_MD_JOINT_STEREO; force_ms=1; info->mode_ext=0; } +! else if (*arg == 'm') +! { info->mode = MPG_MD_MONO; info->mode_ext = 0; } +! else { +! fprintf(stderr,"%s: -m mode must be s/d/j/f/m not %s\n", +! programName, arg); +! err = 1; +! } +! break; +! case 'V': argUsed = 1; VBR = 1; +! if (*arg == '0') +! { VBR_q=0; } +! else if (*arg == '1') +! { VBR_q=1; } +! else if (*arg == '2') +! { VBR_q=2; } +! else if (*arg == '3') +! { VBR_q=3; } +! else if (*arg == '4') +! { VBR_q=4; } +! else if (*arg == '5') +! { VBR_q=5; } +! else if (*arg == '6') +! { VBR_q=6; } +! else if (*arg == '7') +! { VBR_q=7; } +! else if (*arg == '8') +! { VBR_q=8; } +! else if (*arg == '9') +! { VBR_q=9; } +! else { +! fprintf(stderr,"%s: -V n must be 0-9 not %s\n", +! programName, arg); +! err = 1; +! } +! break; +! +! case 's': +! argUsed = 1; +! srate = atof( arg ); +! /* samplerate = rint( 1000.0 * srate ); $A */ +! samplerate = (long) (( 1000.0 * srate ) + 0.5); +! if( (info->sampling_frequency = +! SmpFrqIndex((long) samplerate, &info->version)) < 0 ) +! err = 1; +! break; +! +! case 'b': +! argUsed = 1; +! brate = atoi(arg); +! break; +! +! // AF ADDED +! case 't': /* write VBR tag */ +! g_bWriteVbrTag=0; +! break; +! // AF END +! +! case 'x': /* force byte swapping */ +! swapbytes=TRUE; +! break; +! case 'p': /* (jo) error_protection: add crc16 information to stream */ +! info->error_protection = 1; +! break; +! case 'a': /* autoconvert input file from stereo to mono - for mono mp3 encoding */ +! autoconvert = TRUE; +! break; +! case 'h': +! highq = TRUE; +! break; +! case 'k': +! sfb21 = FALSE; +! break; +! case 'd': +! allow_diff_short = 1; +! break; +! case 'v': +! VBR = 1; +! break; +! case 'S': +! silent = TRUE; +! break; +! case 'X': +! experimentalX = TRUE; +! break; +! case 'Y': +! experimentalY = TRUE; +! break; +! case 'Z': +! experimentalZ = TRUE; +! break; +! case 'f': +! fast_mode = TRUE; +! break; +! #ifdef HAVEGTK +! case 'g': /* turn on gtk analysis */ +! gtkflag = TRUE; +! break; +! #endif +! case 'e': argUsed = 1; +! if (*arg == 'n') info->emphasis = 0; +! else if (*arg == '5') info->emphasis = 1; +! else if (*arg == 'c') info->emphasis = 3; +! else { +! fprintf(stderr,"%s: -e emp must be n/5/c not %s\n", +! programName, arg); +! err = 1; +! } +! break; +! case 'c': info->copyright = 1; break; +! case 'o': info->original = 0; break; +! default: fprintf(stderr,"%s: unrec option %c\n", +! programName, c); +! err = 1; break; +! } +! if(argUsed) { +! if(arg == token) token = ""; /* no more from token */ +! else ++i; /* skip arg we used */ +! arg = ""; argUsed = 0; +! } + } ++ } ++ else { ++ if(inPath[0] == '\0') strcpy(inPath, argv[i]); ++ else if(outPath[0] == '\0') strcpy(outPath, argv[i]); + else { +! fprintf(stderr,"%s: excess arg %s\n", programName, argv[i]); +! err = 1; + } +! } +! } + +! // Do not write VBR tag if VBR flag is not specified +! if (g_bWriteVbrTag==1 && VBR==0) g_bWriteVbrTag=0; +! +! if(err || inPath[0] == '\0') usage(); /* never returns */ +! if (inPath[0]=='-') silent=1; /* turn off status - it's broken for stdin */ +! if(outPath[0] == '\0') { +! if (inPath[0]=='-') { +! /* if input is stdin, default output is stdout */ +! strcpy(outPath,"-"); +! }else { + strcpy(outPath, inPath); + strcat(outPath, DFLT_EXT); +! } +! } +! if (outPath[0]=='-') g_bWriteVbrTag=0; /* turn off VBR tag */ +! +! // SPECIAL FOR BLADE ENCODER +! #ifdef _BLADEDLL +! open_bit_stream_w(&bs, outPath, BUFFER_SIZE); +! +! #else +! // are we encoding to .mp3, or analyzing a .mp3 file: +! original_file_type=!(strcmp((char *) &inPath[strlen(inPath)-4],".mp3")); +! +! if (original_file_type) { +! /* MP3 file. */ +! totalframes = -1; +! original_file_fd = open(inPath,O_RDONLY); +! if (original_file_fd == -1) { +! fprintf(stderr,"Could not find \"%s\".\n", inPath); + exit(1); +! } +! if (!gtkflag) { +! fprintf(stderr,"Use option -g to analyze mp3 files.\n"); +! exit(1); +! } +! +! }else{ +! /* open the output file */ +! #ifdef DISTRIB +! outPath="/dev/null"; +! #endif +! open_bit_stream_w(&bs, outPath, BUFFER_SIZE); + + +! +! /**************************************************************************/ +! /* new code which uses LIBSNDFILE */ +! /**************************************************************************/ +! #ifdef LIBSNDFILE +! if (!strcmp(inPath, "-")) +! { +! /* Read from standard input. */ +! /* so do not open libsnd */ +! } +! else +! { +! // Open snd File +! OpenSndFile(inPath); +! +! // Set sample frequency +! info->sampling_frequency = SmpFrqIndex(GetSndSampleRate(), &info->version); +! +! if (GetSndChannels()==1) +! { +! info->mode = MPG_MD_MONO; +! info->mode_ext = 0; +! } +! +! /* Declare sound file to have "infinite" number of samples. */ +! *num_samples = MAX_U_32_NUM; +! } + ++ #else ++ /***********************************************************************/ ++ /* old code with native AIFF, PCM and WAV support */ ++ /***********************************************************************/ ++ ++ /* Declare sound file to have "infinite" number of samples */ ++ /* as long as we do not know more about the file... */ ++ *num_samples = MAX_U_32_NUM; ++ ++ if (!strcmp(inPath, "-")) { ++ /* Read from standard input. */ ++ musicin = stdin; ++ } else { ++ if ((musicin = fopen(inPath, "rb")) == NULL) { ++ fprintf(stderr, "Could not find \"%s\".\n", inPath); ++ exit(1); ++ } ++ } ++ if ((soundPosition = aiff_read_headers(musicin, &pcm_aiff_data)) != -1) { ++ fprintf(stderr, "Found Audio IFF sound file headers: "); + aiff_check(inPath, &pcm_aiff_data, &info->version); + if (fseek(musicin, soundPosition, SEEK_SET) != 0) { +! fprintf(stderr,"Could not seek to PCM sound data in \"%s\".\n",inPath); +! fprintf(stderr,"Crossing Fingers.....\n"); + } + info->sampling_frequency = SmpFrqIndex((long)pcm_aiff_data.sampleRate, &info->version); +! fprintf(stderr, "%.1f kHz samples\n", +! pcm_aiff_data.sampleRate/1000); + /* Determine number of samples in sound file */ + *num_samples = (long)(pcm_aiff_data.numChannels) * +! (long)(pcm_aiff_data.numSampleFrames); + if ( pcm_aiff_data.numChannels == 1 ) { +! info->mode = MPG_MD_MONO; +! info->mode_ext = 0; +! } +! } else { /* Not using Audio IFF sound file headers. */ +! if (fseek(musicin, 8, SEEK_SET) != 0) { +! fprintf(stderr,"Could not seek to PCM sound data in \"%s\".\n",inPath); +! fprintf(stderr,"Crossing Fingers.....\n"); + } +! fread(ckwav, 4, 1, musicin); + +! if (strncmp(ckwav, "WAVE",4)==0) { +! iswav = 1; +! { +! fseek(musicin, 0, SEEK_SET); +! fread(&wavhead,sizeof(WaveHeader),1,musicin); +! if (DetermineByteOrder()==order_bigEndian) +! endswap_wave_fmt(&wavhead); +! fprintf(stderr,"Found WAV header: %3.1fkHz %s\n", +! wavhead.sample_fq/1000.0, +! (wavhead.modus==WAVE_MONO ? "mono" : "stereo")); +! /* +! fprintf(stderr," Format: %s\n",wavhead.format==PCM_CODE ? "PCM" : "unknown"); +! fprintf(stderr," Mode : %s\n",wavhead.modus==WAVE_STEREO ? "stereo" : "mono"); +! fprintf(stderr," Freq : %d\n",wavhead.sample_fq); +! fprintf(stderr," Bits : %d\n",wavhead.bit_p_spl); +! fprintf(stderr," Len : %d\n",wavhead.data_length); +! fprintf(stderr," FLen : %d\n",wavhead.length); +! */ +! if (wavhead.bit_p_spl != 16) { +! fprintf(stderr," Bits : %d not supported!\n",wavhead.bit_p_spl); +! exit(1); +! } +! +! +! if (wavhead.modus==WAVE_MONO) { +! info->mode = MPG_MD_MONO; +! info->mode_ext = 0; +! } +! info->sampling_frequency = +! SmpFrqIndex((long)wavhead.sample_fq, &info->version); +! *num_samples = (wavhead.length / (wavhead.bit_p_spl / 8)) * (wavhead.modus==WAVE_STEREO ? 2 : 1); +! } + } +! else { +! fseek(musicin, 0, SEEK_SET); +! fprintf(stderr, "Assuming raw pcm input file"); +! if (swapbytes==TRUE) +! fprintf(stderr, " : Forcing byte-swapping\n"); +! else +! fprintf(stderr, "\n"); +! } +! } +! #endif /* NO LIBSNDFILE */ +! } // MP3 or PCM file? +! #endif /* BLADEDLL*/ + +! // AF ADDED +! if (g_bWriteVbrTag) +! { +! // Write initial VBR Header to bitstream +! InitVbrTag(&bs,info->version-1,info->mode,info->sampling_frequency); +! } +! // AF END +! +! +! /* Timing, autoconvert and resampling checks and setup */ +! /* dont do this if analyzing a mp3 file */ +! if (!original_file_type) { +! stat(inPath,&sb); +! totalframes = sb.st_size/4608; +! if ((info->mode == MPG_MD_MONO)&&(autoconvert==FALSE)) +! totalframes *= 2; +! if (autoconvert==TRUE) { +! fprintf(stderr, "Autoconverting from stereo to mono. Setting encoding to mono mode.\n"); +! info->mode = MPG_MD_MONO; +! info->mode_ext = 0; +! } +! } +! +! +! if ( brate == 0 ) { +! brate = bitrate[info->version][info->lay-1][9]; +! info->bitrate_index=9; +! } else { +! if( (info->bitrate_index = BitrateIndex(info->lay, brate, info->version)) < 0) +! err=1; +! /* a bit rate was specified. for VBR, take this to be the minimum */ +! VBR_min_bitrate=info->bitrate_index; +! } +! if ((!mode_given) && (brate >= 192)) { +! /* default is STEREO for higher bitrates, not JSTEREO */ +! if (info->mode != MPG_MD_MONO){ +! info->mode = MPG_MD_STEREO; info->mode_ext = 0; +! } +! } + ++ ++ ++ if(err || inPath[0] == '\0') usage(); /* never returns */ ++ if (info->bitrate_index==14) VBR=0; /* dont bother with VBR at 320kbs */ ++ /* we do not normally allow 320bps frams with VBR, unless: */ ++ if (info->bitrate_index==13) VBR_max_bitrate=14; ++ if (VBR_q == 0) VBR_max_bitrate=14; ++ if (VBR) fast_mode = FALSE; ++ /* MPEG2 specific requirements */ ++ if (info->version==0) { ++ totalframes *= 2; ++ g_bWriteVbrTag=0; /* no MPEG2 Xing VBR tags yet */ ++ } + } + ++ ++ ++ + /************************************************************************ +! * +! * print_config +! * +! * PURPOSE: Prints the encoding parameters used +! * +! ************************************************************************/ + void print_config( frame_params *fr_ps, int *psy, char *inPath, char *outPath) + { +! layer *info = fr_ps->header; +! #if 0 +! fprintf(stderr, "Encoding configuration:\n"); +! fprintf(stderr, "Algorithm=%s\n", version_names[info->version]); +! if(info->mode != MPG_MD_JOINT_STEREO) +! fprintf(stderr, "Layer=III mode=%s extn=%d psy model=%d\n", +! mode_names[info->mode], info->mode_ext, *psy); +! else fprintf(stderr, "Layer=III mode=%s extn=data dependant psy model=%d\n", +! mode_names[info->mode], *psy); +! fprintf(stderr, "samp frq=%.1f kHz total bitrate=%d kbps\n", +! s_freq[info->version][info->sampling_frequency], +! bitrate[info->version][info->lay-1][info->bitrate_index]); +! fprintf(stderr, "de-emph=%d c/right=%d orig=%d\n", +! info->emphasis, info->copyright, info->original); +! fprintf(stderr, "input file: '%s' output file: '%s'\n", inPath, outPath); +! #endif +! if (original_file_type) { +! fprintf(stderr, "Analyzing %s \n",inPath); +! } +! else { +! fprintf(stderr, "Encoding %s to %s\n", +! (strcmp(inPath, "-")? inPath : "stdin"), +! (strcmp(outPath, "-")? outPath : "stdout")); +! if (VBR) +! fprintf(stderr, "Encoding as %.1f kHz VBR(q=%i) %s MPEG%i LayerIII file\n", +! s_freq[info->version][info->sampling_frequency], +! VBR_q,mode_names[info->mode],2-info->version); +! else +! fprintf(stderr, "Encoding as %.1f kHz %d kbps %s MPEG%i LayerIII file\n", +! s_freq[info->version][info->sampling_frequency], +! bitrate[info->version][info->lay-1][info->bitrate_index], +! mode_names[info->mode],2-info->version); +! } +! } +! +! #ifdef HAVEGTK +! int readframe() +! { +! int j,ch,stereo,out=0; +! int adj; +! short mpg123pcm[2][1152]; +! static int frameNum=0; +! +! frameNum++; +! +! /* create a 1 frame lag so MP3 analysis syncs with WAV analysis */ +! if (frameNum == 1 ) out=1152; +! else out=decode1file(original_file_fd,mpg123pcm,&stereo); +! if (!out) return out; +! +! adj=0; +! if (out==576) adj=576; +! +! fr_ps.stereo=stereo; +! /* Compensate for the decoder_delay. add DECDELAY, where +! * decoder_delay + DECDELAY = 1152, so data is delayed +! * by exactly one frame */ +! for ( ch = 0; ch < stereo; ch++ ) { +! for ( j = 0; j < DECDELAY-adj; j++ ) +! pinfo->pcmdata2[ch][j] = pinfo->pcmdata2[ch][j+1152-adj]; +! for ( j = 0; j < 1152-adj; j++ ) +! pinfo->pcmdata2[ch][j+DECDELAY-adj] = mpg123pcm[ch][j]; +! } +! +! MPGLAG = 2; +! pinfo->frameNum = frameNum; +! pinfo->mpglag = MPGLAG; +! return out; +! } +! #endif +! +! +! +! +! +! encodeframe(short int Buffer[2][1152]) +! { +! static unsigned long frameBits; +! static unsigned long bitsPerSlot; +! static double frac_SpF; +! static double slot_lag; +! static int mode_gr; +! static int mpg123count=0; +! static int old_bitrate; +! static unsigned long sentBits = 0; +! static L3SBS FAR l3_sb_sample; +! /* set to 56 to sync with FhG */ +! #define EXTRADELAY 56 +! static int samplesPerFrame; +! static short int mfbuf[2][1152+576+EXTRADELAY]; /* 3 granule internal buffer */ +! double xr[2][2][576]; +! double win_que[2][HAN_SIZE]; +! int l3_enc[2][2][576]; +! +! III_psy_ratio ratio; +! III_scalefac_t scalefac; +! +! int j,ch,gr,mean_bits; +! int bitsPerFrame; +! double pe[2][2]; +! short *win_buf[2]; +! int stereo; +! layer *info; +! int i; +! +! +! #ifdef HAVEGTK +! char mpg123bs[BUFFER_SIZE]; +! short mpg123pcm[2][1152]; +! #endif +! int check_ms_stereo; +! int bit_rate; +! double samp; +! double ms_ener_ratio[2]; +! +! +! stereo = fr_ps.stereo; +! info = fr_ps.header; +! #ifdef HAVEGTK +! if (gtkflag) { +! pinfo->frameNum = frameNum; +! pinfo->sampfreq=s_freq[info->version][info->sampling_frequency]*1000.0; +! } +! #endif +! +! info->mode_ext = 0; +! +! /* use m/s stereo? */ +! check_ms_stereo = ((info->mode == MPG_MD_JOINT_STEREO) && +! (!force_ms) && (info->version == 1) && +! (stereo==2)); +! +! +! if (firstcall) old_bitrate=info->bitrate_index; +! +! /* what bitrate should we try first? */ +! if (VBR){ +! info->bitrate_index = old_bitrate-1; +! if (info->bitrate_index < VBR_min_bitrate) +! info->bitrate_index=VBR_min_bitrate; +! } else { +! info->bitrate_index = old_bitrate; +! } +! +! bit_rate = bitrate[info->version][info->lay-1][info->bitrate_index]; +! samp = s_freq[info->version][info->sampling_frequency]; +! +! if (firstcall) { +! double avg_slots_per_frame; +! int whole_SpF; +! firstcall = 0; +! mpg123count=0; +! sentBits = 0; +! memset((char *) mfbuf, 0, sizeof(mfbuf)); +! +! /* Figure average number of 'slots' per frame. */ +! /* Bitrate means TOTAL for both channels, not per side. */ +! bitsPerSlot = 8; +! samplesPerFrame = info->version == 1 ? 1152 : 576; +! avg_slots_per_frame = ((double)samplesPerFrame /samp) * +! ((double)bit_rate / (double)bitsPerSlot); +! whole_SpF = (int) avg_slots_per_frame; +! frac_SpF = avg_slots_per_frame - (double)whole_SpF; +! slot_lag = -frac_SpF; +! info->padding = 1; +! if (fabs(frac_SpF) < 1e-9) info->padding = 0; +! mode_gr = (info->version == 1) ? 2 : 1; /* mode_gr = 2 */ +! } +! +! +! /* shift in new samples, delayed by 576+EXTRADELAY */ +! for (ch=0; ch<stereo; ch++) +! for (i=0; i<576+EXTRADELAY; i++) +! mfbuf[ch][i]=mfbuf[ch][i+samplesPerFrame]; +! for (ch=0; ch<stereo; ch++) +! for (i=0; i<samplesPerFrame; i++) +! mfbuf[ch][i+576+EXTRADELAY]=Buffer[ch][i]; +! +! +! +! +! /********************** padding *****************************/ +! if (VBR) { +! /* leave info_padding as it was set above */ +! } else { +! if (frac_SpF != 0) { +! if (slot_lag > (frac_SpF-1.0) ) { +! slot_lag -= frac_SpF; +! info->padding = 0; +! } +! else { +! info->padding = 1; +! slot_lag += (1-frac_SpF); +! } +! } +! } +! +! +! +! +! +! /********************** process InputBuffer *****************************/ +! frameNum++; +! +! if (force_ms) +! for (i=0; i< samplesPerFrame; i++) { +! double mid = (mfbuf[0][576+i]+mfbuf[1][576+i])/(2.0); +! double side= (mfbuf[0][576+i]-mfbuf[1][576+i])/(2.0); +! mfbuf[0][576+i]=mid; +! mfbuf[1][576+i]=side; +! } +! +! if (frameNum%10==0 && !gtkflag && !silent) { +! timestatus(frameNum,totalframes,s_freq[info->version][info->sampling_frequency]); +! } +! +! +! /***************************** Layer 3 ********************************** +! * mfbuf contains 3 granules: [0 1 2 ] +! * encoder will encode granules 0 1 +! * psy-model will be fed granules 1 2, and because of its 1 granule delay +! * it will return thresholds for granules 0 1 */ +! +! win_buf[0] = &mfbuf[0][0]; +! win_buf[1] = &mfbuf[1][0]; +! +! +! if (!fast_mode) { +! /* psychoacoustic model +! * psy model adds a 544 delay to sync with the filterbanks +! * in addition to this delay, it also adds a 1 granule (576) delay +! * that we must compensate for (mt 6/99). +! */ +! short int *bufp[2]; /* address of beginning of left & right granule */ +! int blocktype[2]; +! +! for (gr=0; gr < mode_gr ; gr++) { +! for ( ch = 0; ch < stereo; ch++ ) bufp[ch]=&mfbuf[ch][576 + gr*576]; +! L3psycho_anal( bufp, stereo, gr, info, +! s_freq[info->version][info->sampling_frequency] * 1000.0, +! check_ms_stereo,&ms_ener_ratio[gr], +! ratio.l[gr], ratio.s[gr], pe[gr], blocktype); +! for ( ch = 0; ch < stereo; ch++ ) +! l3_side.gr[gr].ch[ch].tt.block_type=blocktype[ch]; +! } +! }else{ +! for (gr=0; gr < mode_gr ; gr++) +! for ( ch = 0; ch < stereo; ch++ ) +! l3_side.gr[gr].ch[ch].tt.block_type=NORM_TYPE; +! } +! +! +! +! +! /* block type flags */ +! for( gr = 0; gr < mode_gr; gr++ ) { +! for ( ch = 0; ch < stereo; ch++ ) { +! gr_info *cod_info = &l3_side.gr[gr].ch[ch].tt; +! cod_info->mixed_block_flag = 0; /* never used by this model */ +! if (cod_info->block_type == NORM_TYPE ) +! cod_info->window_switching_flag = 0; +! else +! cod_info->window_switching_flag = 1; +! } +! } +! +! +! /* polyphase filtering */ +! for( gr = 0; gr < mode_gr; gr++ ) +! for ( ch = 0; ch < stereo; ch++ ) +! for ( j = 0; j < 18; j++ ) { +! window_subband( &win_buf[ch], &win_que[ch][0], ch ); +! filter_subband( &win_que[ch][0], &(l3_sb_sample)[ch][gr+1][j][0] ); } +! +! /* apply mdct to the polyphase outputs */ +! mdct_sub( &l3_sb_sample, xr, stereo, &l3_side, mode_gr ); +! +! +! /* data was scaled by 1/2. fix so effectively it was scaled by 1/sqrt(2) */ +! if (force_ms) { +! for ( gr = 0; gr < mode_gr; gr++ ) +! for ( ch = 0; ch < stereo; ch++ ) +! for (i =0 ; i< 576; i++) +! xr[gr][ch][i] *= sqrt(2.0); +! } +! +! +! if (check_ms_stereo) { +! /* make sure block type is the same in each channel */ +! check_ms_stereo = +! (l3_side.gr[0].ch[0].tt.block_type==l3_side.gr[0].ch[1].tt.block_type) && +! (l3_side.gr[1].ch[0].tt.block_type==l3_side.gr[1].ch[1].tt.block_type); +! } +! if (check_ms_stereo) { +! if ( (ms_ener_ratio[0] + ms_ener_ratio[1])<.70) info->mode_ext = 2; +! } +! +! #ifdef HAVEGTK +! if (gtkflag) { +! for ( gr = 0; gr < mode_gr; gr++ ) +! for ( ch = 0; ch < stereo; ch++ ) { +! pinfo->ms_ratio[gr]=ms_ener_ratio[gr]; +! pinfo->blocktype[gr][ch]= +! l3_side.gr[gr].ch[ch].tt.block_type; +! for ( j = 0; j < 576; j++ ) pinfo->xr[gr][ch][j]=xr[gr][ch][j]; +! } +! } +! #endif +! +! +! /* bit and noise allocation */ +! iteration_loop( pe, ms_ener_ratio, xr, &ratio, &l3_side, l3_enc, +! &scalefac, &fr_ps); +! +! +! /* flag for our ms_stereo with psy-model on mid & side channels */ +! if (force_ms) info->mode_ext = 2; +! +! +! /* write the frame to the bitstream */ +! old_bitrate = info->bitrate_index; +! getframebits(info,stereo,&bitsPerFrame,&mean_bits); +! III_format_bitstream( bitsPerFrame, &fr_ps, l3_enc, &l3_side, +! &scalefac, &bs, xr, NULL, 0 ); +! +! +! frameBits = sstell( &bs ) - sentBits; +! +! +! if ( frameBits % bitsPerSlot ) /* a program failure */ +! fprintf( stderr, "Sent %ld bits = %ld slots plus %ld\n", +! frameBits, frameBits/bitsPerSlot, +! frameBits%bitsPerSlot ); +! sentBits += frameBits; +! +! +! #ifdef HAVEGTK +! /* copy bit buffer into array and send to mpg123 to synthesis the pcm data */ +! if (gtkflag) { +! mpg123count=0; +! for (i=bs.buf_size-1 ; i > bs.buf_byte_idx ; (i-- )) +! mpg123bs[mpg123count++]=bs.buf[i]; +! /* re-synthesize pcm data from mp3 frame */ +! if (!decode1(mpg123bs,mpg123count,&mpg123pcm,stereo)) { +! if (MPGLAG == MAXMPGLAG) +! fprintf(stderr, "MAXMPGLAG set too low. The following frames will be mis-aligned \n"); +! else MPGLAG+=1; +! } +! pinfo->mpglag = MPGLAG; +! } +! +! /* empty data from bit buffer into mp3 file */ +! empty_buffer(&bs, 1+bs.buf_byte_idx); +! bs.buf[bs.buf_byte_idx] = 0; +! +! if (gtkflag) { +! int adj; +! adj=0; +! if (info->version == 0) adj=576; +! for ( ch = 0; ch < stereo; ch++ ) { +! for ( j = 0; j < WINDELAY; j++ ) +! pinfo->pcmdata[ch][j] = pinfo->pcmdata[ch][j+1152-adj]; +! for ( j = 0; j < 1152-adj; j++ ) { +! pinfo->pcmdata[ch][j+WINDELAY] = mfbuf[ch][j]; +! } +! } +! for ( ch = 0; ch < stereo; ch++ ) { +! for ( j = 0; j < DECDELAY-adj; j++ ) +! pinfo->pcmdata2[ch][j] = pinfo->pcmdata2[ch][j+1152-adj]; +! for ( j = 0; j < 1152-adj; j++ ) { +! pinfo->pcmdata2[ch][j+DECDELAY-adj] = mpg123pcm[ch][j]; +! } +! } +! pinfo->frameNum = frameNum; +! } +! #endif +! if (g_bWriteVbrTag) AddVbrFrame(sentBits/8); +! return 1152; +! } +! +! +! /********************************************************************** +! * read one frame and encode it +! **********************************************************************/ +! int makeframe() +! { +! int iread; +! short int Buffer[2][1152]; +! iread = get_audio(musicin,Buffer[0],Buffer[1],num_samples, +! fr_ps.stereo,fr_ps.header); +! /* if iread=0, get_audio hit EOF and returned Buffer=all 0's. call +! * makeframe() one last time to flush all buffers */ +! encodeframe(Buffer); +! return iread; + } + + +*************** +*** 666,673 **** + * + * main + * +! * PURPOSE: MPEG I Encoder supporting layers 1 and 2, and 3, with +! * psychoacoustic models 1 (MUSICAM) and 2 (AT&T) + * + * SEMANTICS: One overlapping frame of audio of up to 2 channels are + * processed at a time in the following order: +--- 1025,1032 ---- + * + * main + * +! * PURPOSE: MPEG-1 Layer III encoder, with +! * psychoacoustic models ISO-2 (AT&T) and GPSYCHO + * + * SEMANTICS: One overlapping frame of audio of up to 2 channels are + * processed at a time in the following order: +*************** +*** 677,1076 **** + * samples per channel. + * (window_subband,filter_subband) + * +! * 2. If joint stereo mode, combine left and right channels +! * for subbands above #jsbound#. +! * (*_combine_LR) +! * +! * 3. Calculate scalefactors for the frame, and if layer 2, + * also calculate scalefactor select information. + * (*_scale_factor_calc) + * +! * 4. Calculate psychoacoustic masking levels using selected + * psychoacoustic model. + * (*_Psycho_One, psycho_anal) + * +! * 5. Perform iterative bit allocation for subbands with low + * mask_to_noise ratios using masking levels from step 4. + * (*_main_bit_allocation) + * +! * 6. If error protection flag is active, add redundancy for +! * error protection. +! * (*_CRC_calc) +! * +! * 7. Pack bit allocation, scalefactors, and scalefactor select +! * information (layer 2) onto bitstream. +! * (*_encode_bit_alloc,*_encode_scale,II_transmission_pattern) + * +! * 8. Quantize subbands and pack them into bitstream + * (*_subband_quantization, *_sample_encoding) + * + ************************************************************************/ + +! int frameNum=0; +! +! void main(argc, argv) + int argc; + char **argv; + { +- typedef double SBS[2][3][SCALE_BLOCK][SBLIMIT]; +- SBS FAR *sb_sample; +- L3SBS FAR *l3_sb_sample; +- typedef double JSBS[3][SCALE_BLOCK][SBLIMIT]; +- JSBS FAR *j_sample; +- typedef double IN[2][HAN_SIZE]; +- IN FAR *win_que; +- typedef unsigned int SUB[2][3][SCALE_BLOCK][SBLIMIT]; +- SUB FAR *subband; + +- frame_params fr_ps; +- layer info; + char original_file_name[MAX_NAME_SIZE]; + char encoded_file_name[MAX_NAME_SIZE]; +! short FAR **win_buf; +! static short FAR buffer[2][1152]; +! static unsigned int bit_alloc[2][SBLIMIT], scfsi[2][SBLIMIT]; +! static unsigned int scalar[2][3][SBLIMIT], j_scale[3][SBLIMIT]; +! static double FAR ltmin[2][SBLIMIT], lgmin[2][SBLIMIT], max_sc[2][SBLIMIT]; +! FLOAT snr32[32]; +! short sam[2][1344]; /* was [1056]; */ +! int whole_SpF, extra_slot = 0; +! double avg_slots_per_frame, frac_SpF, slot_lag; +! int model, stereo, error_protection; +! static unsigned int crc; +! int i, j, k, adb; +! unsigned long bitsPerSlot, samplesPerFrame; +! unsigned long frameBits, sentBits = 0; +! unsigned long num_samples; +! +! #ifdef MACINTOSH +! argc = ccommand( &argv ); + #endif +- + +! /* Most large variables are declared dynamically to ensure +! compatibility with smaller machines */ +! +! sb_sample = (SBS FAR *) mem_alloc(sizeof(SBS), "sb_sample"); +! l3_sb_sample = (L3SBS FAR *) mem_alloc(sizeof(SBS), "l3_sb_sample"); +! j_sample = (JSBS FAR *) mem_alloc(sizeof(JSBS), "j_sample"); +! win_que = (IN FAR *) mem_alloc(sizeof(IN), "Win_que"); +! subband = (SUB FAR *) mem_alloc(sizeof(SUB),"subband"); +! win_buf = (short FAR **) mem_alloc(sizeof(short *)*2, "win_buf"); +! +! /* clear buffers */ +! memset((char *) buffer, 0, sizeof(buffer)); +! memset((char *) bit_alloc, 0, sizeof(bit_alloc)); +! memset((char *) scalar, 0, sizeof(scalar)); +! memset((char *) j_scale, 0, sizeof(j_scale)); +! memset((char *) scfsi, 0, sizeof(scfsi)); +! memset((char *) ltmin, 0, sizeof(ltmin)); +! memset((char *) lgmin, 0, sizeof(lgmin)); +! memset((char *) max_sc, 0, sizeof(max_sc)); +! memset((char *) snr32, 0, sizeof(snr32)); +! memset((char *) sam, 0, sizeof(sam)); +! +! fr_ps.header = &info; +! fr_ps.tab_num = -1; /* no table loaded */ +! fr_ps.alloc = NULL; +! info.version = MPEG_AUDIO_ID; /* Default: MPEG-1 */ +! +! programName = argv[0]; +! if(argc==1) /* no command-line args */ +! obtain_parameters(&fr_ps, &model, &num_samples, +! original_file_name, encoded_file_name); +! else +! parse_args(argc, argv, &fr_ps, &model, &num_samples, +! original_file_name, encoded_file_name); +! print_config(&fr_ps, &model, +! original_file_name, encoded_file_name); +! +! hdr_to_frps(&fr_ps); +! stereo = fr_ps.stereo; +! error_protection = info.error_protection; +! +! if (info.lay == 1) +! { bitsPerSlot = 32; samplesPerFrame = 384; } +! else +! if ( info.lay == 2 ) +! { bitsPerSlot = 8; samplesPerFrame = 1152; } +! else +! { /* layer 3 */ +! bitsPerSlot = 8; +! samplesPerFrame = info.version == 1 ? 1152 : 576; +! +! /* Apologize for missing features */ +! if ( info.mode == MPG_MD_JOINT_STEREO ) +! { +! fprintf( stderr, "Sorry, joint stereo not yet available for layer3\n" ); +! exit( 1 ); +! } +! #if 0 +! if ( info.version != MPEG_AUDIO_ID ) +! { +! fprintf( stderr, "Sorry, MPEG2-LSF not yet available for layer3\n" ); +! exit( 1 ); +! } +! #endif +! if ( model != 2 ) +! { +! fprintf( stderr, "Sorry, psycho model 1 not available for layer3\n" ); +! exit( 1 ); +! } +! } +! /* Figure average number of 'slots' per frame. */ +! /* Bitrate means TOTAL for both channels, not per side. */ +! avg_slots_per_frame = ((double)samplesPerFrame / +! s_freq[info.version][info.sampling_frequency]) * +! ((double)bitrate[info.version][info.lay-1][info.bitrate_index] / +! (double)bitsPerSlot); +! whole_SpF = (int) avg_slots_per_frame; +! printf("slots/frame = %d\n",whole_SpF); +! frac_SpF = avg_slots_per_frame - (double)whole_SpF; +! slot_lag = -frac_SpF; +! printf("frac SpF=%.3f, tot bitrate=%d kbps, s freq=%.1f kHz\n", +! frac_SpF, bitrate[info.version][info.lay-1][info.bitrate_index], +! s_freq[info.version][info.sampling_frequency]); +! +! if (frac_SpF != 0) +! printf("Fractional number of slots, padding required\n"); +! else info.padding = 0; +! +! while ( get_audio(musicin, buffer, num_samples, stereo, &info) > 0 ) + { +! fprintf(stderr, "{%4lu}", frameNum++); fflush(stderr); +! win_buf[0] = &buffer[0][0]; +! win_buf[1] = &buffer[1][0]; +! if (frac_SpF != 0) { +! if (slot_lag > (frac_SpF-1.0) ) { +! slot_lag -= frac_SpF; +! extra_slot = 0; +! info.padding = 0; +! /* printf("No padding for this frame\n"); */ +! } +! else { +! extra_slot = 1; +! info.padding = 1; +! slot_lag += (1-frac_SpF); +! /* printf("Padding for this frame\n"); */ +! } +! } +! adb = (whole_SpF+extra_slot) * bitsPerSlot; +! +! switch (info.lay) +! { +! +! /***************************** Layer I **********************************/ +! +! case 1 : +! for (j=0;j<SCALE_BLOCK;j++) +! for (k=0;k<stereo;k++) { +! window_subband(&win_buf[k], &(*win_que)[k][0], k); +! filter_subband(&(*win_que)[k][0], &(*sb_sample)[k][0][j][0]); +! } +! +! I_scale_factor_calc(*sb_sample, scalar, stereo); +! if(fr_ps.actual_mode == MPG_MD_JOINT_STEREO) { +! I_combine_LR(*sb_sample, *j_sample); +! I_scale_factor_calc(j_sample, &j_scale, 1); +! } +! +! put_scale(scalar, &fr_ps, max_sc); +! +! if (model == 1) I_Psycho_One(buffer, max_sc, ltmin, &fr_ps); +! else { +! for (k=0;k<stereo;k++) { +! psycho_anal(&buffer[k][0],&sam[k][0], k, info.lay, snr32, +! (FLOAT)s_freq[info.version][info.sampling_frequency]*1000); +! for (i=0;i<SBLIMIT;i++) ltmin[k][i] = (double) snr32[i]; +! } +! } +! +! I_main_bit_allocation(ltmin, bit_alloc, &adb, &fr_ps); +! +! if (error_protection) I_CRC_calc(&fr_ps, bit_alloc, &crc); +! +! encode_info(&fr_ps, &bs); +! +! if (error_protection) encode_CRC(crc, &bs); +! +! I_encode_bit_alloc(bit_alloc, &fr_ps, &bs); +! I_encode_scale(scalar, bit_alloc, &fr_ps, &bs); +! I_subband_quantization(scalar, *sb_sample, j_scale, *j_sample, +! bit_alloc, *subband, &fr_ps); +! I_sample_encoding(*subband, bit_alloc, &fr_ps, &bs); +! for (i=0;i<adb;i++) put1bit(&bs, 0); +! break; +! +! /***************************** Layer 2 **********************************/ +! +! case 2 : +! for (i=0;i<3;i++) for (j=0;j<SCALE_BLOCK;j++) +! for (k=0;k<stereo;k++) { +! window_subband(&win_buf[k], &(*win_que)[k][0], k); +! filter_subband(&(*win_que)[k][0], &(*sb_sample)[k][i][j][0]); +! } +! +! II_scale_factor_calc(*sb_sample, scalar, stereo, fr_ps.sblimit); +! pick_scale(scalar, &fr_ps, max_sc); +! if(fr_ps.actual_mode == MPG_MD_JOINT_STEREO) { +! II_combine_LR(*sb_sample, *j_sample, fr_ps.sblimit); +! II_scale_factor_calc(j_sample, &j_scale, 1, fr_ps.sblimit); +! } /* this way we calculate more mono than we need */ +! /* but it is cheap */ +! +! if (model == 1) II_Psycho_One(buffer, max_sc, ltmin, &fr_ps); +! else { +! for (k=0;k<stereo;k++) { +! psycho_anal(&buffer[k][0],&sam[k][0], k, +! info.lay, snr32, +! (FLOAT)s_freq[info.version][info.sampling_frequency]*1000); +! for (i=0;i<SBLIMIT;i++) ltmin[k][i] = (double) snr32[i]; +! } +! } +! +! II_transmission_pattern(scalar, scfsi, &fr_ps); +! II_main_bit_allocation(ltmin, scfsi, bit_alloc, &adb, &fr_ps); +! +! if (error_protection) +! II_CRC_calc(&fr_ps, bit_alloc, scfsi, &crc); +! +! encode_info(&fr_ps, &bs); +! +! if (error_protection) encode_CRC(crc, &bs); +! +! II_encode_bit_alloc(bit_alloc, &fr_ps, &bs); +! II_encode_scale(bit_alloc, scfsi, scalar, &fr_ps, &bs); +! II_subband_quantization(scalar, *sb_sample, j_scale, +! *j_sample, bit_alloc, *subband, &fr_ps); +! II_sample_encoding(*subband, bit_alloc, &fr_ps, &bs); +! for (i=0;i<adb;i++) put1bit(&bs, 0); +! break; +! +! /***************************** Layer 3 **********************************/ +! +! case 3: +! { +! /* +! large "auto" vars are static due to the Macintosh linker +! */ +! static double xr[2][2][576]; +! static double xr_dec[2][2][576]; +! static double pe[2][2]; +! static int l3_enc[2][2][576]; +! static III_psy_ratio ratio; +! static III_side_info_t l3_side; +! static III_scalefac_t scalefac; +! int gr, mode_gr, ch; +! int mean_bits, sideinfo_len; +! +! int bitsPerFrame = 8 * whole_SpF + (info.padding * 8); +! mode_gr = (info.version == 1) ? 2 : 1; +! +! /* +! determine the mean bitrate for main data +! */ +! sideinfo_len = 32; +! if ( info.version == 1 ) +! { /* MPEG 1 */ +! if ( stereo == 1 ) +! sideinfo_len += 136; +! else +! sideinfo_len += 256; +! } +! else +! { /* MPEG 2 */ +! if ( stereo == 1 ) +! sideinfo_len += 72; +! else +! sideinfo_len += 136; +! } +! if ( info.error_protection ) +! sideinfo_len += 16; +! mean_bits = (bitsPerFrame - sideinfo_len) / mode_gr; +! +! /* +! psychoacoustic model +! */ +! for ( gr = 0; gr < mode_gr; gr++ ) +! for ( ch = 0; ch < stereo; ch++ ) +! { +! L3psycho_anal( &buffer[ch][gr*576], &sam[ch][0], ch, info.lay, +! snr32, s_freq[info.version][info.sampling_frequency] * 1000.0, +! &ratio.l[gr][ch][0], &ratio.s[gr][ch][0], +! &pe[gr][ch], &l3_side.gr[gr].ch[ch].tt ); +! } +! +! /* +! polyphase filtering +! */ +! for( gr = 0; gr < mode_gr; gr++ ) +! for ( ch = 0; ch < stereo; ch++ ) +! for ( j = 0; j < 18; j++ ) +! { +! window_subband( &win_buf[ch], &(*win_que)[ch][0], ch ); +! filter_subband( &(*win_que)[ch][0], &(*l3_sb_sample)[ch][gr+1][j][0] ); +! } +! +! /* +! apply mdct to the polyphase outputs +! */ +! mdct_sub( l3_sb_sample, xr, stereo, &l3_side, mode_gr ); +! #if 0 +! delay( xr, stereo ); +! #endif +! /* +! bit and noise allocation +! */ +! iteration_loop( pe, xr, &ratio, &l3_side, l3_enc, mean_bits, +! stereo, xr_dec, &scalefac, &fr_ps, 0, bitsPerFrame ); +! +! /* +! write the frame to the bitstream +! */ +! III_format_bitstream( bitsPerFrame, &fr_ps, l3_enc, &l3_side, &scalefac, &bs, +! xr, NULL, 0 ); +! } +! break; /* end of layer 3 */ +! +! +! } /* end switch */ +! +! frameBits = sstell( &bs ) - sentBits; +! if ( frameBits % bitsPerSlot ) /* a program failure */ +! fprintf( stderr, "Sent %ld bits = %ld slots plus %ld\n", +! frameBits, frameBits/bitsPerSlot, +! frameBits%bitsPerSlot ); +! sentBits += frameBits; +! +! } +! +! if ( info.lay == 3 ) +! III_FlushBitstream(); +! + close_bit_stream_w( &bs ); +! +! printf("Avg slots/frame = %.3f; b/smp = %.2f; br = %.3f kbps\n", +! (FLOAT) sentBits / (frameNum * bitsPerSlot), +! (FLOAT) sentBits / (frameNum * samplesPerFrame), +! (FLOAT) sentBits / (frameNum * samplesPerFrame) * +! s_freq[info.version][info.sampling_frequency]); +! + if (fclose(musicin) != 0){ +! printf("Could not close \"%s\".\n", original_file_name); +! exit(2); + } +- +- #ifdef MACINTOSH +- set_mac_file_attr( encoded_file_name, VOL_REF_NUM, CREATOR_ENCODE, +- FILETYPE_ENCODE ); + #endif +! +! printf("Encoding of \"%s\" with psychoacoustic model %d is finished\n", +! original_file_name, model); +! printf("The MPEG encoded output file name is \"%s\"\n", +! encoded_file_name); +! exit(0); + } + + /************************************************************************ + * +--- 1036,1164 ---- + * samples per channel. + * (window_subband,filter_subband) + * +! * 2. Calculate scalefactors for the frame, and if layer 2, + * also calculate scalefactor select information. + * (*_scale_factor_calc) + * +! * 3. Calculate psychoacoustic masking levels using selected + * psychoacoustic model. + * (*_Psycho_One, psycho_anal) + * +! * 4. Perform iterative bit allocation for subbands with low + * mask_to_noise ratios using masking levels from step 4. + * (*_main_bit_allocation) + * +! * 5. Pack bit allocation and scalefactors onto bitstream. +! * (*_encode_bit_alloc,*_encode_scale) + * +! * 6. Quantize subbands and pack them into bitstream + * (*_subband_quantization, *_sample_encoding) + * + ************************************************************************/ + +! #ifndef _BLADEDLL +! int main(argc, argv) + int argc; + char **argv; + { + + char original_file_name[MAX_NAME_SIZE]; + char encoded_file_name[MAX_NAME_SIZE]; +! int model, stereo; +! layer info; +! +! #ifdef __FreeBSD__ +! { +! fp_except_t mask; +! mask=fpgetmask(); +! /* printf("FreeBSD mask is 0x%x\n",mask); */ +! } +! #endif +! #ifdef ABORTFP +! { +! unsigned int mask; +! _FPU_GETCW(mask); +! /* Set the Linux mask to the FreeBSD default mask */ +! mask &= ~( _FPU_MASK_IM | _FPU_MASK_ZM | _FPU_MASK_OM ); +! _FPU_SETCW(mask); +! } + #endif + +! // Clear info structure +! memset(&info,0,sizeof(info)); +! +! fr_ps.header = &info; +! fr_ps.tab_num = -1; /* no table loaded */ +! fr_ps.alloc = NULL; +! info.version = MPEG_AUDIO_ID; /* =1 Default: MPEG-1 */ +! info.extension = 0; +! +! /* get version string */ +! if (gpsycho && !original_file_type) +! fprintf(stderr,"LAME version %s (www.sulaco.org/mp3) \n",get_lame_version()); +! +! +! programName = argv[0]; +! if(argc==1) /* no command-line args +! obtain_parameters(&fr_ps, &model, &num_samples, +! original_file_name, encoded_file_name);*/ +! usage(); +! else +! parse_args(argc, argv, &fr_ps, &model, &num_samples, +! original_file_name, encoded_file_name); +! print_config(&fr_ps, &model, +! original_file_name, encoded_file_name); +! +! hdr_to_frps(&fr_ps); +! stereo = fr_ps.stereo; +! if (gpsycho && !original_file_type) fprintf(stderr, +! "using GPSYCHO: GPL'd psycho-acoustic model \n"); +! +! #ifdef HAVEGTK +! if (gtkflag) gtk_init (&argc, &argv); +! if (gtkflag) gtkcontrol(totalframes,original_file_type,original_file_name,gpsycho); +! else +! #endif + { +! while (makeframe()); +! } +! +! /* clean up if we were producing an mp3 file */ +! if (!original_file_type) { +! if ( info.lay == 3 ) III_FlushBitstream(); + close_bit_stream_w( &bs ); +! #ifdef LIBSNDFILE +! CloseSndFile(); +! #else + if (fclose(musicin) != 0){ +! fprintf(stderr, "Could not close \"%s\".\n", original_file_name); +! exit(2); + } + #endif +! +! frameNum--; +! timestatus(frameNum,frameNum,s_freq[info.version][info.sampling_frequency]); +! fprintf(stderr,"\n"); +! } +! fflush(stderr); +! +! +! // AF ADDED +! if (g_bWriteVbrTag) +! { +! // Calculate relative quality of VBR stream +! // 0=best, 100=worst +! int nQuality=VBR_q*100/9; +! +! // Write Xing header again +! PutVbrTag(encoded_file_name,nQuality); +! } +! // AF END +! +! return 0; + } ++ #endif // IFNDEF _BLADEDLL ++ + + /************************************************************************ + * +*************** +*** 1082,1111 **** + + void usage() /* print syntax & exit */ + { +! fprintf(stderr, +! "usage: %s queries for all arguments, or\n", +! programName); +! fprintf(stderr, +! " %s [-l lay][-m mode][-p psy][-s sfrq][-b br][-d emp]\n", +! programName); +! fprintf(stderr, +! " [-c][-o][-e] inputPCM [outBS]\n"); +! fprintf(stderr,"where\n"); +! fprintf(stderr," -l lay use layer <lay> coding (dflt %4u)\n",DFLT_LAY); +! fprintf(stderr," -m mode channel mode : s/d/j/m (dflt %4c)\n",DFLT_MOD); +! fprintf(stderr," -p psy psychoacoustic model 1/2 (dflt %4u)\n",DFLT_PSY); +! fprintf(stderr," -s sfrq input smpl rate in kHz (dflt %4.1f)\n",DFLT_SFQ); +! fprintf(stderr," -b br total bitrate in kbps (dflt highest)\n"); +! fprintf(stderr," -d emp de-emphasis n/5/c (dflt %4c)\n",DFLT_EMP); +! fprintf(stderr," -c mark as copyright\n"); +! fprintf(stderr," -o mark as original\n"); +! fprintf(stderr," -e add error protection\n"); +! fprintf(stderr," inputPCM input PCM sound file (standard or AIFF)\n"); +! fprintf(stderr," outBS output bit stream of encoded audio (dflt inName+%s)\n", +! DFLT_EXT); +! exit(1); + } + + /************************************************************************ + * + * aiff_check +--- 1170,1212 ---- + + void usage() /* print syntax & exit */ + { +! fprintf(stderr," LAMER Ain't an Mp3 Encoder (Reconstruction-iso)\n"); +! fprintf(stderr," version %s\n",get_lame_version()); +! fprintf(stderr," with GPSYCHO psycho-acoustic model (version %s). \n\n",get_psy_version()); +! fprintf(stderr,"USAGE : %s [options] <infile> <outfile>\n",programName); +! fprintf(stderr,"OPTIONS :\n"); +! fprintf(stderr," -m mode (s)tereo, (j)oint, (f)orce or (m)ono (default %c)\n",DFLT_MOD); +! fprintf(stderr," force = force ms_stereo on all frames. Faster and\n"); +! fprintf(stderr," uses special Mid & Side masking thresholds\n"); +! fprintf(stderr," -s sample freq frequency of input file (kHz) - default %4.1f\n",DFLT_SFQ); +! fprintf(stderr," -b <bitrate> set the bitrate, default 128kbps\n"); +! fprintf(stderr," (for VBR, this sets the allowed minimum bitrate)\n"); +! fprintf(stderr," -x force byte-swapping of input\n"); +! fprintf(stderr," -a downmix from stereo to mono file for mono encoding\n"); +! fprintf(stderr," -e emp de-emphasis n/5/c (default %4c)\n",DFLT_EMP); +! fprintf(stderr," -p add CRC error protection (now working!)\n"); +! fprintf(stderr," -c mark as copyright\n"); +! fprintf(stderr," -o mark as non-original\n"); +! fprintf(stderr," -f fast mode (low quality)\n"); +! fprintf(stderr," -k disable sfb=21 cutoff\n"); +! fprintf(stderr," -d allow channels to have different blocktypes\n"); +! fprintf(stderr," -v use variable bitrate (VBR)\n"); +! fprintf(stderr," -V n quality setting for VBR. default n=%i\n",VBR_q); +! fprintf(stderr," 0=high quality,bigger files. 9=smaller files\n"); +! fprintf(stderr," -h use (maybe) some experimental quality improvements\n"); +! #ifdef HAVEGTK +! fprintf(stderr," -g run graphical analysis on <infile>\n"); +! #endif +! fprintf(stderr," -S don't print progress report\n"); +! fprintf(stderr," -t Disable Xing VBR Tag (only valid if -v flag is specified\n"); +! fprintf(stderr,"\n<infile> and/or <outfile> can be \"-\", which means stdin/stdout.\n"); +! +! exit(1); + } + ++ ++ ++ #ifndef LIBSNDFILE + /************************************************************************ + * + * aiff_check +*************** +*** 1118,1151 **** + void aiff_check( char *file_name, IFF_AIFF *pcm_aiff_data, int *version) + { + if (pcm_aiff_data->sampleType != IFF_ID_SSND) { +! printf("Sound data is not PCM in \"%s\".\n", file_name); + exit(1); + } + + if(SmpFrqIndex((long)pcm_aiff_data->sampleRate, version) < 0) { +! printf("in \"%s\".\n", file_name); + exit(1); + } + + if (pcm_aiff_data->sampleSize != sizeof(short) * BITS_IN_A_BYTE) { +! printf("Sound data is not %d bits in \"%s\".\n", +! sizeof(short) * BITS_IN_A_BYTE, file_name); + exit(1); + } + + if (pcm_aiff_data->numChannels != MONO && + pcm_aiff_data->numChannels != STEREO) { +! printf("Sound data is not mono or stereo in \"%s\".\n", file_name); + exit(1); + } + + if (pcm_aiff_data->blkAlgn.blockSize != 0) { +! printf("Block size is not %d bytes in \"%s\".\n", 0, file_name); + exit(1); + } + + if (pcm_aiff_data->blkAlgn.offset != 0) { +! printf("Block offset is not %d bytes in \"%s\".\n", 0, file_name); + exit(1); + } + } +--- 1219,1258 ---- + void aiff_check( char *file_name, IFF_AIFF *pcm_aiff_data, int *version) + { + if (pcm_aiff_data->sampleType != IFF_ID_SSND) { +! fprintf(stderr, "Sound data is not PCM in \"%s\".\n", file_name); + exit(1); + } + + if(SmpFrqIndex((long)pcm_aiff_data->sampleRate, version) < 0) { +! fprintf(stderr, "in \"%s\".\n", file_name); + exit(1); + } + + if (pcm_aiff_data->sampleSize != sizeof(short) * BITS_IN_A_BYTE) { +! fprintf(stderr, "Sound data is not %d bits in \"%s\".\n", +! (unsigned int) sizeof(short) * BITS_IN_A_BYTE, file_name); + exit(1); + } + + if (pcm_aiff_data->numChannels != MONO && + pcm_aiff_data->numChannels != STEREO) { +! fprintf(stderr, "Sound data is not mono or stereo in \"%s\".\n", +! file_name); + exit(1); + } + + if (pcm_aiff_data->blkAlgn.blockSize != 0) { +! fprintf(stderr, "Block size is not %d bytes in \"%s\".\n", +! 0, file_name); + exit(1); + } + + if (pcm_aiff_data->blkAlgn.offset != 0) { +! fprintf(stderr, "Block offset is not %d bytes in \"%s\".\n", +! 0, file_name); + exit(1); + } + } ++ ++ #endif ++ +diff -r -c -N encoder/portableio.c lame3.13/portableio.c +*** encoder/portableio.c Wed Jan 22 02:43:12 1997 +--- lame3.13/portableio.c Sat Jun 5 13:45:37 1999 +*************** +*** 1,3 **** +--- 1,4 ---- ++ #ifndef LIBSNDFILE + /* Copyright (C) 1988-1991 Apple Computer, Inc. + * All Rights Reserved. + * +*************** +*** 226,232 **** + char *p; + int n; + { +! while (!feof(fp) & n-- > 0) + *p++ = getc(fp); + } + +--- 227,233 ---- + char *p; + int n; + { +! while (!feof(fp) & (n-- > 0)) + *p++ = getc(fp); + } + +*************** +*** 237,243 **** + { + register char *q = p; + +! while (!feof(fp) & n-- > 0) + *q++ = getc(fp); + + for (q--; p < q; p++, q--){ +--- 238,244 ---- + { + register char *q = p; + +! while (!feof(fp) & (n-- > 0)) + *q++ = getc(fp); + + for (q--; p < q; p++, q--){ +*************** +*** 394,396 **** +--- 395,398 ---- + } + + ++ #endif +diff -r -c -N encoder/psy.c lame3.13/psy.c +*** encoder/psy.c Wed Jan 22 02:43:18 1997 +--- lame3.13/psy.c Wed Dec 31 17:00:00 1969 +*************** +*** 1,454 **** +- /********************************************************************** +- * ISO MPEG Audio Subgroup Software Simulation Group (1996) +- * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- * +- * $Id: psy.c,v 1.1 1996/02/14 04:04:23 rowlands Exp $ +- * +- * $Log: psy.c,v $ +- * Revision 1.1 1996/02/14 04:04:23 rowlands +- * Initial revision +- * +- * Received from Mike Coleman +- **********************************************************************/ +- /********************************************************************** +- * date programmers comment * +- * 2/25/91 Davis Pan start of version 1.0 records * +- * 5/10/91 W. Joseph Carter Ported to Macintosh and Unix. * +- * 7/10/91 Earle Jennings Ported to MsDos. * +- * replace of floats with FLOAT * +- * 2/11/92 W. Joseph Carter Fixed mem_alloc() arg for "absthr". * +- * 7/24/92 M. Iwadare HANN window coefficients modified. * +- * 7/27/92 Masahiro Iwadare Bug fix, FFT modification for Layer 3 * +- * 7/27/92 Masahiro Iwadare Bug fix, "new", "old", and "oldest" * +- * updates * +- * 8/07/92 Mike Coleman Bug fix, read_absthr() * +- * 95/3/21 Jon Rowlands Removed extra debug statements * +- **********************************************************************/ +- +- #include "common.h" +- #include "encoder.h" +- FILE *fpo; /* file pointer */ +- void psycho_anal(buffer,savebuf,chn,lay,snr32,sfreq) +- short int *buffer; +- short int savebuf[1056]; +- int chn, lay; +- FLOAT snr32[32]; +- double sfreq; /* to match prototype : float args are always double */ +- { +- unsigned int i, j, k; +- FLOAT r_prime, phi_prime; +- FLOAT freq_mult, bval_lo, minthres, sum_energy; +- double tb, temp1, temp2, temp3; +- +- /* The static variables "r", "phi_sav", "new", "old" and "oldest" have */ +- /* to be remembered for the unpredictability measure. For "r" and */ +- /* "phi_sav", the first index from the left is the channel select and */ +- /* the second index is the "age" of the data. */ +- +- static int new = 0, old = 1, oldest = 0; +- static int init = 0, flush, sync_flush, syncsize, sfreq_idx; +- +- /* The following static variables are constants. */ +- +- static double nmt = 5.5; +- +- static FLOAT crit_band[27] = {0, 100, 200, 300, 400, 510, 630, 770, +- 920, 1080, 1270,1480,1720,2000,2320, 2700, +- 3150, 3700, 4400,5300,6400,7700,9500,12000, +- 15500,25000,30000}; +- +- static FLOAT bmax[27] = {20.0, 20.0, 20.0, 20.0, 20.0, 17.0, 15.0, +- 10.0, 7.0, 4.4, 4.5, 4.5, 4.5, 4.5, +- 4.5, 4.5, 4.5, 4.5, 4.5, 4.5, 4.5, +- 4.5, 4.5, 4.5, 3.5, 3.5, 3.5}; +- +- /* The following pointer variables point to large areas of memory */ +- /* dynamically allocated by the mem_alloc() function. Dynamic memory */ +- /* allocation is used in order to avoid stack frame or data area */ +- /* overflow errors that otherwise would have occurred at compile time */ +- /* on the Macintosh computer. */ +- +- FLOAT *grouped_c, *grouped_e, *nb, *cb, *ecb, *bc; +- FLOAT *wsamp_r, *wsamp_i, *phi, *energy; +- FLOAT *c, *fthr; +- F32 *snrtmp; +- +- static int *numlines; +- static int *partition; +- static FLOAT *cbval, *rnorm; +- static FLOAT *window; +- static FLOAT *absthr; +- static double *tmn; +- static FCB *s; +- static FHBLK *lthr; +- static F2HBLK *r, *phi_sav; +- +- /* These dynamic memory allocations simulate "automatic" variables */ +- /* placed on the stack. For each mem_alloc() call here, there must be */ +- /* a corresponding mem_free() call at the end of this function. */ +- +- grouped_c = (FLOAT *) mem_alloc(sizeof(FCB), "grouped_c"); +- grouped_e = (FLOAT *) mem_alloc(sizeof(FCB), "grouped_e"); +- nb = (FLOAT *) mem_alloc(sizeof(FCB), "nb"); +- cb = (FLOAT *) mem_alloc(sizeof(FCB), "cb"); +- ecb = (FLOAT *) mem_alloc(sizeof(FCB), "ecb"); +- bc = (FLOAT *) mem_alloc(sizeof(FCB), "bc"); +- wsamp_r = (FLOAT *) mem_alloc(sizeof(FBLK), "wsamp_r"); +- wsamp_i = (FLOAT *) mem_alloc(sizeof(FBLK), "wsamp_i"); +- phi = (FLOAT *) mem_alloc(sizeof(FBLK), "phi"); +- energy = (FLOAT *) mem_alloc(sizeof(FBLK), "energy"); +- c = (FLOAT *) mem_alloc(sizeof(FHBLK), "c"); +- fthr = (FLOAT *) mem_alloc(sizeof(FHBLK), "fthr"); +- snrtmp = (F32 *) mem_alloc(sizeof(F2_32), "snrtmp"); +- +- if(init==0){ +- +- /* These dynamic memory allocations simulate "static" variables placed */ +- /* in the data space. Each mem_alloc() call here occurs only once at */ +- /* initialization time. The mem_free() function must not be called. */ +- +- numlines = (int *) mem_alloc(sizeof(ICB), "numlines"); +- partition = (int *) mem_alloc(sizeof(IHBLK), "partition"); +- fpo = fopen("out.dat", "wb"); +- if(fpo==NULL) { +- puts("\t The attempt to open the output file failed.\n"); +- exit(-1);} +- cbval = (FLOAT *) mem_alloc(sizeof(FCB), "cbval"); +- rnorm = (FLOAT *) mem_alloc(sizeof(FCB), "rnorm"); +- window = (FLOAT *) mem_alloc(sizeof(FBLK), "window"); +- absthr = (FLOAT *) mem_alloc(sizeof(FHBLK), "absthr"); +- tmn = (double *) mem_alloc(sizeof(DCB), "tmn"); +- s = (FCB *) mem_alloc(sizeof(FCBCB), "s"); +- lthr = (FHBLK *) mem_alloc(sizeof(F2HBLK), "lthr"); +- r = (F2HBLK *) mem_alloc(sizeof(F22HBLK), "r"); +- phi_sav = (F2HBLK *) mem_alloc(sizeof(F22HBLK), "phi_sav"); +- +- i = sfreq + 0.5; +- switch(i){ +- case 32000: sfreq_idx = 0; break; +- case 44100: sfreq_idx = 1; break; +- case 48000: sfreq_idx = 2; break; +- default: printf("error, invalid sampling frequency: %d Hz\n",i); +- exit(-1); +- } +- printf("absthr[][] sampling frequency index: %d\n",sfreq_idx); +- read_absthr(absthr, sfreq_idx); +- if(lay==1){ +- flush = 384; +- syncsize = 1024; +- sync_flush = 576; +- } +- else { +- flush = 384*3.0/2.0; +- syncsize = 1056; +- sync_flush = syncsize - flush; +- } +- /* calculate HANN window coefficients */ +- /* for(i=0;i<BLKSIZE;i++)window[i]=0.5*(1-cos(2.0*PI*i/(BLKSIZE-1.0))); */ +- for(i=0;i<BLKSIZE;i++)window[i]=0.5*(1-cos(2.0*PI*(i-0.5)/BLKSIZE)); +- /* reset states used in unpredictability measure */ +- for(i=0;i<HBLKSIZE;i++){ +- r[0][0][i]=r[1][0][i]=r[0][1][i]=r[1][1][i]=0; +- phi_sav[0][0][i]=phi_sav[1][0][i]=0; +- phi_sav[0][1][i]=phi_sav[1][1][i]=0; +- lthr[0][i] = 60802371420160.0; +- lthr[1][i] = 60802371420160.0; +- } +- /***************************************************************************** +- * Initialization: Compute the following constants for use later * +- * partition[HBLKSIZE] = the partition number associated with each * +- * frequency line * +- * cbval[CBANDS] = the center (average) bark value of each * +- * partition * +- * numlines[CBANDS] = the number of frequency lines in each partition * +- * tmn[CBANDS] = tone masking noise * +- *****************************************************************************/ +- /* compute fft frequency multiplicand */ +- freq_mult = sfreq/BLKSIZE; +- +- /* calculate fft frequency, then bval of each line (use fthr[] as tmp storage)*/ +- for(i=0;i<HBLKSIZE;i++){ +- temp1 = i*freq_mult; +- j = 1; +- while(temp1>crit_band[j])j++; +- fthr[i]=j-1+(temp1-crit_band[j-1])/(crit_band[j]-crit_band[j-1]); +- } +- partition[0] = 0; +- /* temp2 is the counter of the number of frequency lines in each partition */ +- temp2 = 1; +- cbval[0]=fthr[0]; +- bval_lo=fthr[0]; +- for(i=1;i<HBLKSIZE;i++){ +- if((fthr[i]-bval_lo)>0.33){ +- partition[i]=partition[i-1]+1; +- cbval[partition[i-1]] = cbval[partition[i-1]]/temp2; +- cbval[partition[i]] = fthr[i]; +- bval_lo = fthr[i]; +- numlines[partition[i-1]] = temp2; +- temp2 = 1; +- } +- else { +- partition[i]=partition[i-1]; +- cbval[partition[i]] += fthr[i]; +- temp2++; +- } +- } +- numlines[partition[i-1]] = temp2; +- cbval[partition[i-1]] = cbval[partition[i-1]]/temp2; +- +- /************************************************************************ +- * Now compute the spreading function, s[j][i], the value of the spread-* +- * ing function, centered at band j, for band i, store for later use * +- ************************************************************************/ +- for(j=0;j<CBANDS;j++){ +- for(i=0;i<CBANDS;i++){ +- temp1 = (cbval[i] - cbval[j])*1.05; +- if(temp1>=0.5 && temp1<=2.5){ +- temp2 = temp1 - 0.5; +- temp2 = 8.0 * (temp2*temp2 - 2.0 * temp2); +- } +- else temp2 = 0; +- temp1 += 0.474; +- temp3 = 15.811389+7.5*temp1-17.5*sqrt((double) (1.0+temp1*temp1)); +- if(temp3 <= -100) s[i][j] = 0; +- else { +- temp3 = (temp2 + temp3)*LN_TO_LOG10; +- s[i][j] = exp(temp3); +- } +- } +- } +- +- /* Calculate Tone Masking Noise values */ +- for(j=0;j<CBANDS;j++){ +- temp1 = 15.5 + cbval[j]; +- tmn[j] = (temp1>24.5) ? temp1 : 24.5; +- /* Calculate normalization factors for the net spreading functions */ +- rnorm[j] = 0; +- for(i=0;i<CBANDS;i++){ +- rnorm[j] += s[j][i]; +- } +- } +- init++; +- } +- +- /************************* End of Initialization *****************************/ +- switch(lay) { +- case 1: +- case 2: +- for(i=0; i<lay; i++){ +- /***************************************************************************** +- * Net offset is 480 samples (1056-576) for layer 2; this is because one must* +- * stagger input data by 256 samples to synchronize psychoacoustic model with* +- * filter bank outputs, then stagger so that center of 1024 FFT window lines * +- * up with center of 576 "new" audio samples. * +- * * +- * For layer 1, the input data still needs to be staggered by 256 samples, * +- * then it must be staggered again so that the 384 "new" samples are centered* +- * in the 1024 FFT window. The net offset is then 576 and you need 448 "new"* +- * samples for each iteration to keep the 384 samples of interest centered * +- *****************************************************************************/ +- for(j=0; j<syncsize; j++){ +- if(j<(sync_flush))savebuf[j] = savebuf[j+flush]; +- else savebuf[j] = *buffer++; +- if(j<BLKSIZE){ +- /**window data with HANN window***********************************************/ +- wsamp_r[j] = window[j]*((FLOAT) savebuf[j]); +- wsamp_i[j] = 0; +- } +- } +- /**Compute FFT****************************************************************/ +- fft(wsamp_r,wsamp_i,energy,phi,1024); +- /***************************************************************************** +- * calculate the unpredictability measure, given energy[f] and phi[f] * +- *****************************************************************************/ +- /*only update data "age" pointers after you are done with both channels */ +- /*for layer 1 computations, for the layer 2 double computations, the pointers*/ +- /*are reset automatically on the second pass */ +- if(lay==2 || (lay==1 && chn==0) ){ +- if(new==0){new = 1; oldest = 1;} +- else {new = 0; oldest = 0;} +- if(old==0)old = 1; else old = 0; +- } +- for(j=0; j<HBLKSIZE; j++){ +- r_prime = 2.0 * r[chn][old][j] - r[chn][oldest][j]; +- phi_prime = 2.0 * phi_sav[chn][old][j] - phi_sav[chn][oldest][j]; +- r[chn][new][j] = sqrt((double) energy[j]); +- phi_sav[chn][new][j] = phi[j]; +- temp1=r[chn][new][j] * cos((double) phi[j]) - r_prime * cos((double) phi_prime); +- temp2=r[chn][new][j] * sin((double) phi[j]) - r_prime * sin((double) phi_prime); +- temp3=r[chn][new][j] + fabs((double)r_prime); +- if(temp3 != 0)c[j]=sqrt(temp1*temp1+temp2*temp2)/temp3; +- else c[j] = 0; +- } +- /***************************************************************************** +- * Calculate the grouped, energy-weighted, unpredictability measure, * +- * grouped_c[], and the grouped energy. grouped_e[] * +- *****************************************************************************/ +- for(j=1;j<CBANDS;j++){ +- grouped_e[j] = 0; +- grouped_c[j] = 0; +- } +- grouped_e[0] = energy[0]; +- grouped_c[0] = energy[0]*c[0]; +- for(j=1;j<HBLKSIZE;j++){ +- grouped_e[partition[j]] += energy[j]; +- grouped_c[partition[j]] += energy[j]*c[j]; +- } +- +- /***************************************************************************** +- * convolve the grouped energy-weighted unpredictability measure * +- * and the grouped energy with the spreading function, s[j][k] * +- *****************************************************************************/ +- for(j=0;j<CBANDS;j++){ +- ecb[j] = 0; +- cb[j] = 0; +- for(k=0;k<CBANDS;k++){ +- if(s[j][k] != 0.0){ +- ecb[j] += s[j][k]*grouped_e[k]; +- cb[j] += s[j][k]*grouped_c[k]; +- } +- } +- if(ecb[j] !=0)cb[j] = cb[j]/ecb[j]; +- else cb[j] = 0; +- } +- +- /***************************************************************************** +- * Calculate the required SNR for each of the frequency partitions * +- * this whole section can be accomplished by a table lookup * +- *****************************************************************************/ +- for(j=0;j<CBANDS;j++){ +- if(cb[j]<.05)cb[j]=0.05; +- else if(cb[j]>.5)cb[j]=0.5; +- tb = -0.434294482*log((double) cb[j])-0.301029996; +- cb[j]=tb; +- bc[j] = tmn[j]*tb + nmt*(1.0-tb); +- k = cbval[j] + 0.5; +- bc[j] = (bc[j] > bmax[k]) ? bc[j] : bmax[k]; +- bc[j] = exp((double) -bc[j]*LN_TO_LOG10); +- } +- +- /***************************************************************************** +- * Calculate the permissible noise energy level in each of the frequency * +- * partitions. Include absolute threshold and pre-echo controls * +- * this whole section can be accomplished by a table lookup * +- *****************************************************************************/ +- for(j=0;j<CBANDS;j++) +- if(rnorm[j] && numlines[j]) +- nb[j] = ecb[j]*bc[j]/(rnorm[j]*numlines[j]); +- else nb[j] = 0; +- for(j=0;j<HBLKSIZE;j++){ +- /*temp1 is the preliminary threshold */ +- temp1=nb[partition[j]]; +- temp1=(temp1>absthr[j])?temp1:absthr[j]; +- /*do not use pre-echo control for layer 2 because it may do bad things to the*/ +- /* MUSICAM bit allocation algorithm */ +- if(lay==1){ +- fthr[j] = (temp1 < lthr[chn][j]) ? temp1 : lthr[chn][j]; +- temp2 = temp1 * 0.00316; +- fthr[j] = (temp2 > fthr[j]) ? temp2 : fthr[j]; +- } +- else fthr[j] = temp1; +- lthr[chn][j] = LXMIN*temp1; +- } +- +- /***************************************************************************** +- * Translate the 512 threshold values to the 32 filter bands of the coder * +- *****************************************************************************/ +- for(j=0;j<193;j += 16){ +- minthres = 60802371420160.0; +- sum_energy = 0.0; +- for(k=0;k<17;k++){ +- if(minthres>fthr[j+k])minthres = fthr[j+k]; +- sum_energy += energy[j+k]; +- } +- snrtmp[i][j/16] = sum_energy/(minthres * 17.0); +- snrtmp[i][j/16] = 4.342944819 * log((double)snrtmp[i][j/16]); +- } +- for(j=208;j<(HBLKSIZE-1);j += 16){ +- minthres = 0.0; +- sum_energy = 0.0; +- for(k=0;k<17;k++){ +- minthres += fthr[j+k]; +- sum_energy += energy[j+k]; +- } +- snrtmp[i][j/16] = sum_energy/minthres; +- snrtmp[i][j/16] = 4.342944819 * log((double)snrtmp[i][j/16]); +- } +- /***************************************************************************** +- * End of Psychoacuostic calculation loop * +- *****************************************************************************/ +- } +- for(i=0; i<32; i++){ +- if(lay==2) +- snr32[i]=(snrtmp[0][i]>snrtmp[1][i])?snrtmp[0][i]:snrtmp[1][i]; +- else snr32[i]=snrtmp[0][i]; +- } +- break; +- case 3: +- printf("layer 3 is not currently supported\n"); +- break; +- default: +- printf("error, invalid MPEG/audio coding layer: %d\n",lay); +- } +- +- /* These mem_free() calls must correspond with the mem_alloc() calls */ +- /* used at the beginning of this function to simulate "automatic" */ +- /* variables placed on the stack. */ +- +- mem_free((void **) &grouped_c); +- mem_free((void **) &grouped_e); +- mem_free((void **) &nb); +- mem_free((void **) &cb); +- mem_free((void **) &ecb); +- mem_free((void **) &bc); +- mem_free((void **) &wsamp_r); +- mem_free((void **) &wsamp_i); +- mem_free((void **) &phi); +- mem_free((void **) &energy); +- mem_free((void **) &c); +- mem_free((void **) &fthr); +- mem_free((void **) &snrtmp); +- } +- +- /****************************************************************************** +- routine to read in absthr table from a file. +- ******************************************************************************/ +- +- void read_absthr(absthr, table) +- FLOAT *absthr; +- int table; +- { +- FILE *fp; +- long j,index; +- float a; +- char t[80]; +- char ta[16]; +- +- strcpy( ta, "absthr_0" ); +- +- switch(table){ +- case 0 : ta[7] = '0'; +- break; +- case 1 : ta[7] = '1'; +- break; +- case 2 : ta[7] = '2'; +- break; +- default : printf("absthr table: Not valid table number\n"); +- } +- if(!(fp = OpenTableFile(ta) ) ){ +- printf("Please check %s table\n", ta); +- exit(1); +- } +- fgets(t, 150, fp); +- sscanf(t, "table %ld", &index); +- if(index != table){ +- printf("error in absthr table %s",ta); +- exit(1); +- } +- for(j=0; j<HBLKSIZE; j++){ +- fgets(t,80,fp); +- sscanf(t,"%f", &a); +- absthr[j] = a; +- } +- fclose(fp); +- } +--- 0 ---- +diff -r -c -N encoder/reservoir.c lame3.13/reservoir.c +*** encoder/reservoir.c Wed Jan 22 02:43:18 1997 +--- lame3.13/reservoir.c Sat Jun 19 17:33:22 1999 +*************** +*** 27,32 **** +--- 27,33 ---- + #include "huffman.h" + #include "l3bitstream.h" + #include "reservoir.h" ++ #include "gtkanal.h" + + /* + Layer3 bit reservoir: +*************** +*** 36,48 **** + static int ResvSize = 0; /* in bits */ + static int ResvMax = 0; /* in bits */ + + /* + ResvFrameBegin: + Called at the beginning of a frame. Updates the maximum + size of the reservoir, and checks to make sure main_data_begin + was set properly by the formatter + */ +! void + ResvFrameBegin( frame_params *fr_ps, III_side_info_t *l3_side, int mean_bits, int frameLength ) + { + layer *info; +--- 37,55 ---- + static int ResvSize = 0; /* in bits */ + static int ResvMax = 0; /* in bits */ + ++ void InitReservoir(void) ++ { ++ ResvSize = 0; ++ ResvMax = 0; ++ } ++ + /* + ResvFrameBegin: + Called at the beginning of a frame. Updates the maximum + size of the reservoir, and checks to make sure main_data_begin + was set properly by the formatter + */ +! int + ResvFrameBegin( frame_params *fr_ps, III_side_info_t *l3_side, int mean_bits, int frameLength ) + { + layer *info; +*************** +*** 71,78 **** + fprintf( stderr, ">>> ResvSize = %d\n", ResvSize ); + #endif + assert( expectedResvSize == ResvSize ); +! +! fullFrameBits = mean_bits * mode_gr; + + /* + determine maximum size of reservoir: +--- 78,84 ---- + fprintf( stderr, ">>> ResvSize = %d\n", ResvSize ); + #endif + assert( expectedResvSize == ResvSize ); +! fullFrameBits = mean_bits * mode_gr + ResvSize; + + /* + determine maximum size of reservoir: +*************** +*** 90,95 **** +--- 96,110 ---- + */ + if ( ResvMax > resvLimit ) + ResvMax = resvLimit; ++ ++ #ifdef HAVEGTK ++ if (gtkflag){ ++ pinfo->mean_bits=mean_bits/2; /* expected bits per channel per granule */ ++ pinfo->resvsize=ResvSize; ++ } ++ #endif ++ ++ return fullFrameBits; + } + + /* +*************** +*** 99,105 **** + and perceptual entropy. + */ + int +! ResvMaxBits( frame_params *fr_ps, III_side_info_t *l3_side, double *pe, int mean_bits ) + { + int more_bits, max_bits, add_bits, over_bits; + +--- 114,120 ---- + and perceptual entropy. + */ + int +! ResvMaxBits( frame_params *fr_ps, III_side_info_t *l3_side, double *pe, int mean_bits,int gr,int ch) + { + int more_bits, max_bits, add_bits, over_bits; + +*************** +*** 113,118 **** +--- 128,134 ---- + return max_bits; + + more_bits = *pe * 3.1 - mean_bits; ++ + add_bits = 0; + if ( more_bits > 100 ) + { +*************** +*** 123,128 **** +--- 139,146 ---- + else + add_bits = more_bits; + } ++ ++ + over_bits = ResvSize - ((ResvMax * 8) / 10) - add_bits; + if ( over_bits > 0 ) + add_bits += over_bits; +*************** +*** 130,139 **** +--- 148,197 ---- + max_bits += add_bits; + if ( max_bits > 4095 ) + max_bits = 4095; ++ + return max_bits; + } + + /* ++ ResvMaxBits2: ++ As above, but now it *really* is bits per granule (both channels). ++ Mark Taylor 4/99 ++ */ ++ void ResvMaxBits2(int mean_bits, int *targ_bits, int *extra_bits, int gr) ++ { ++ int add_bits; ++ ++ ++ *targ_bits = mean_bits ; ++ /* extra bits if the reservoir is almost full */ ++ if (ResvSize > ((ResvMax * 9) / 10)) { ++ add_bits= ResvSize-((ResvMax * 9) / 10); ++ *targ_bits += add_bits; ++ }else { ++ add_bits =0 ; ++ /* build up reservoir. this builds the reservoir a little slower ++ * than FhG. It could simple be mean_bits/15, but this was rigged ++ * to always produce 100 (the old value) at 128kbs */ ++ *targ_bits -= (int) (mean_bits/15.2); ++ } ++ ++ ++ /* amount from the reservoir we are allowed to use */ ++ *extra_bits = ++ (ResvSize < (ResvMax*6)/10 ? ResvSize : (ResvMax*6)/10); ++ *extra_bits -= add_bits; ++ ++ if (*extra_bits < 0) *extra_bits=0; ++ ++ ++ if ( *targ_bits > 4095 ) ++ *targ_bits = 4095; ++ if ( *targ_bits + *extra_bits > 4095 ) ++ *extra_bits = 4095 - *targ_bits; ++ ++ } ++ ++ /* + ResvAdjust: + Called after a granule's bit allocation. Readjusts the size of + the reservoir to reflect the granule's usage. +*************** +*** 143,148 **** +--- 201,207 ---- + { + ResvSize += (mean_bits / fr_ps->stereo) - gi->part2_3_length; + } ++ + + /* + ResvFrameEnd: +diff -r -c -N encoder/reservoir.h lame3.13/reservoir.h +*** encoder/reservoir.h Wed Jan 22 02:43:18 1997 +--- lame3.13/reservoir.h Thu Jun 3 00:09:03 1999 +*************** +*** 22,29 **** + #ifndef RESERVOIR_H + #define RESERVOIR_H + +! void ResvFrameBegin( frame_params *fr_ps, III_side_info_t *l3_side, int mean_bits, int frameLength ); +! int ResvMaxBits( frame_params *fr_ps, III_side_info_t *l3_side, double *pe, int mean_bits ); + void ResvAdjust( frame_params *fr_ps, gr_info *gi, III_side_info_t *l3_side, int mean_bits ); + void ResvFrameEnd( frame_params *fr_ps, III_side_info_t *l3_side, int mean_bits ); + +--- 22,30 ---- + #ifndef RESERVOIR_H + #define RESERVOIR_H + +! int ResvFrameBegin( frame_params *fr_ps, III_side_info_t *l3_side, int mean_bits, int frameLength ); +! int ResvMaxBits( frame_params *fr_ps, III_side_info_t *l3_side, double *pe, int mean_bits, int gr, int ch ); +! void ResvMaxBits2( int mean_bits, int *targ_bits, int *max_bits, int gr); + void ResvAdjust( frame_params *fr_ps, gr_info *gi, III_side_info_t *l3_side, int mean_bits ); + void ResvFrameEnd( frame_params *fr_ps, III_side_info_t *l3_side, int mean_bits ); + +diff -r -c -N encoder/subs.c lame3.13/subs.c +*** encoder/subs.c Wed Jan 22 02:43:18 1997 +--- lame3.13/subs.c Sat Jun 5 16:52:21 1999 +*************** +*** 1,163 **** +! /********************************************************************** +! * ISO MPEG Audio Subgroup Software Simulation Group (1996) +! * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +! * +! * $Id: subs.c,v 1.1 1996/02/14 04:04:23 rowlands Exp $ +! * +! * $Log: subs.c,v $ +! * Revision 1.1 1996/02/14 04:04:23 rowlands +! * Initial revision +! * +! * Received from Mike Coleman +! **********************************************************************/ +! /********************************************************************** +! * date programmers comment * +! * 2/25/91 Davis Pan start of version 1.0 records * +! * 5/10/91 W. Joseph Carter Ported to Macintosh and Unix. * +! * 7/10/91 Earle Jennings Ported to MsDos from Macintosh * +! * Replacement of one float with FLOAT * +! * 2/11/92 W. Joseph Carter Added type casting to memset() args. * +! * 4/27/92 Masahiro Iwadare Added 256 point version for Layer III * +! **********************************************************************/ + + #include "common.h" +- #include "encoder.h" + +! /***************************************************************************** +! ************************** Start of Subroutines ***************************** +! *****************************************************************************/ +! +! /***************************************************************************** +! * FFT computes fast fourier transform of BLKSIZE samples of data * +! * uses decimation-in-frequency algorithm described in "Digital * +! * Signal Processing" by Oppenheim and Schafer, refer to pages 304 * +! * (flow graph) and 330-332 (Fortran program in problem 5) * +! * to get the inverse fft, change line 20 from * +! * w_imag[L] = -sin(PI/le1); * +! * to * +! * w_imag[L] = sin(PI/le1); * +! * * +! * required constants: * +! * #define PI 3.14159265358979 * +! * #define BLKSIZE 1024 * +! * #define LOGBLKSIZE 10 * +! * #define BLKSIZE_S 256 * +! * #define LOGBLKSIZE_S 8 * +! * * +! *****************************************************************************/ +! #define BLKSIZE_S 256 +! #define LOGBLKSIZE_S 8 +! +! void fft(x_real,x_imag, energy, phi, N) +! FLOAT x_real[BLKSIZE], x_imag[BLKSIZE], energy[BLKSIZE], phi[BLKSIZE]; +! int N; + { +! int M,MM1; +! static int init=0; +! int NV2, NM1, MP; +! static double w_real[2][LOGBLKSIZE], w_imag[2][LOGBLKSIZE]; +! int i,j,k,L; +! int ip, le,le1; +! double t_real, t_imag, u_real, u_imag; +! +! if(init==0) { +! memset((char *) w_real, 0, sizeof(w_real)); /* preset statics to 0 */ +! memset((char *) w_imag, 0, sizeof(w_imag)); /* preset statics to 0 */ +! M = LOGBLKSIZE; +! for(L=0; L<M; L++){ +! le = 1 << (M-L); +! le1 = le >> 1; +! w_real[0][L] = cos(PI/le1); +! w_imag[0][L] = -sin(PI/le1); +! } +! M = LOGBLKSIZE_S; +! for(L=0; L<M; L++){ +! le = 1 << (M-L); +! le1 = le >> 1; +! w_real[1][L] = cos(PI/le1); +! w_imag[1][L] = -sin(PI/le1); +! } +! init++; +! } +! switch(N) { +! case BLKSIZE: +! M = LOGBLKSIZE; +! MP = 0; +! break; +! case BLKSIZE_S: +! M = LOGBLKSIZE_S; +! MP = 1; +! break; +! default: printf("Error: Bad FFT Size in subs.c\n"); +! exit(-1); +! } +! MM1 = M-1; +! NV2 = N >> 1; +! NM1 = N - 1; +! for(L=0; L<MM1; L++){ +! le = 1 << (M-L); +! le1 = le >> 1; +! u_real = 1; +! u_imag = 0; +! for(j=0; j<le1; j++){ +! for(i=j; i<N; i+=le){ +! ip = i + le1; +! t_real = x_real[i] + x_real[ip]; +! t_imag = x_imag[i] + x_imag[ip]; +! x_real[ip] = x_real[i] - x_real[ip]; +! x_imag[ip] = x_imag[i] - x_imag[ip]; +! x_real[i] = t_real; +! x_imag[i] = t_imag; +! t_real = x_real[ip]; +! x_real[ip] = x_real[ip]*u_real - x_imag[ip]*u_imag; +! x_imag[ip] = x_imag[ip]*u_real + t_real*u_imag; +! } +! t_real = u_real; +! u_real = u_real*w_real[MP][L] - u_imag*w_imag[MP][L]; +! u_imag = u_imag*w_real[MP][L] + t_real*w_imag[MP][L]; + } + } +! /* special case: L = M-1; all Wn = 1 */ +! for(i=0; i<N; i+=2){ +! ip = i + 1; +! t_real = x_real[i] + x_real[ip]; +! t_imag = x_imag[i] + x_imag[ip]; +! x_real[ip] = x_real[i] - x_real[ip]; +! x_imag[ip] = x_imag[i] - x_imag[ip]; +! x_real[i] = t_real; +! x_imag[i] = t_imag; +! energy[i] = x_real[i]*x_real[i] + x_imag[i]*x_imag[i]; +! if(energy[i] <= 0.0005){phi[i] = 0;energy[i] = 0.0005;} +! else phi[i] = atan2((double) x_imag[i],(double) x_real[i]); +! energy[ip] = x_real[ip]*x_real[ip] + x_imag[ip]*x_imag[ip]; +! if(energy[ip] == 0)phi[ip] = 0; +! else phi[ip] = atan2((double) x_imag[ip],(double) x_real[ip]); + } +! /* this section reorders the data to the correct ordering */ +! j = 0; +! for(i=0; i<NM1; i++){ +! if(i<j){ +! /* use this section only if you need the FFT in complex number form * +! * (and in the correct ordering) */ +! t_real = x_real[j]; +! t_imag = x_imag[j]; +! x_real[j] = x_real[i]; +! x_imag[j] = x_imag[i]; +! x_real[i] = t_real; +! x_imag[i] = t_imag; +! /* reorder the energy and phase, phi */ +! t_real = energy[j]; +! energy[j] = energy[i]; +! energy[i] = t_real; +! t_real = phi[j]; +! phi[j] = phi[i]; +! phi[i] = t_real; +! } +! k=NV2; +! while(k<=j){ +! j = j-k; +! k = k >> 1; +! } +! j = j+k; + } + } +--- 1,246 ---- +! /* +! ** FFT and FHT routines +! ** Copyright 1988, 1993; Ron Mayer +! ** +! ** fht(fz,n); +! ** Does a hartley transform of "n" points in the array "fz". +! ** +! ** NOTE: This routine uses at least 2 patented algorithms, and may be +! ** under the restrictions of a bunch of different organizations. +! ** Although I wrote it completely myself; it is kind of a derivative +! ** of a routine I once authored and released under the GPL, so it +! ** may fall under the free software foundation's restrictions; +! ** it was worked on as a Stanford Univ project, so they claim +! ** some rights to it; it was further optimized at work here, so +! ** I think this company claims parts of it. The patents are +! ** held by R. Bracewell (the FHT algorithm) and O. Buneman (the +! ** trig generator), both at Stanford Univ. +! ** If it were up to me, I'd say go do whatever you want with it; +! ** but it would be polite to give credit to the following people +! ** if you use this anywhere: +! ** Euler - probable inventor of the fourier transform. +! ** Gauss - probable inventor of the FFT. +! ** Hartley - probable inventor of the hartley transform. +! ** Buneman - for a really cool trig generator +! ** Mayer(me) - for authoring this particular version and +! ** including all the optimizations in one package. +! ** Thanks, +! ** Ron Mayer; mayer@acuson.com +! ** +! */ + ++ #include <math.h> + #include "common.h" + +! static FLOAT costab[20]= +! { +! .00000000000000000000000000000000000000000000000000, +! .70710678118654752440084436210484903928483593768847, +! .92387953251128675612818318939678828682241662586364, +! .98078528040323044912618223613423903697393373089333, +! .99518472667219688624483695310947992157547486872985, +! .99879545620517239271477160475910069444320361470461, +! .99969881869620422011576564966617219685006108125772, +! .99992470183914454092164649119638322435060646880221, +! .99998117528260114265699043772856771617391725094433, +! .99999529380957617151158012570011989955298763362218, +! .99999882345170190992902571017152601904826792288976, +! .99999970586288221916022821773876567711626389934930, +! .99999992646571785114473148070738785694820115568892, +! .99999998161642929380834691540290971450507605124278, +! .99999999540410731289097193313960614895889430318945, +! .99999999885102682756267330779455410840053741619428 +! }; +! static FLOAT sintab[20]= +! { +! 1.0000000000000000000000000000000000000000000000000, +! .70710678118654752440084436210484903928483593768846, +! .38268343236508977172845998403039886676134456248561, +! .19509032201612826784828486847702224092769161775195, +! .09801714032956060199419556388864184586113667316749, +! .04906767432741801425495497694268265831474536302574, +! .02454122852291228803173452945928292506546611923944, +! .01227153828571992607940826195100321214037231959176, +! .00613588464915447535964023459037258091705788631738, +! .00306795676296597627014536549091984251894461021344, +! .00153398018628476561230369715026407907995486457522, +! .00076699031874270452693856835794857664314091945205, +! .00038349518757139558907246168118138126339502603495, +! .00019174759731070330743990956198900093346887403385, +! .00009587379909597734587051721097647635118706561284, +! .00004793689960306688454900399049465887274686668768 +! }; +! +! /* This is a simplified version for n an even power of 2 */ +! +! static void fht(FLOAT *fz, int n) + { +! int i,k,k1,k2,k3,k4,kx; +! FLOAT *fi,*fn,*gi; +! FLOAT t_c,t_s; +! +! for (k1=1,k2=0;k1<n;k1++) +! { +! FLOAT a; +! for (k=n>>1; (!((k2^=k)&k)); k>>=1); +! if (k1>k2) +! { +! a=fz[k1];fz[k1]=fz[k2];fz[k2]=a; +! } + } ++ for (fi=fz,fn=fz+n;fi<fn;fi+=4) ++ { ++ FLOAT f0,f1,f2,f3; ++ f1 = fi[0 ]-fi[1 ]; ++ f0 = fi[0 ]+fi[1 ]; ++ f3 = fi[2 ]-fi[3 ]; ++ f2 = fi[2 ]+fi[3 ]; ++ fi[2 ] = (f0-f2); ++ fi[0 ] = (f0+f2); ++ fi[3 ] = (f1-f3); ++ fi[1 ] = (f1+f3); ++ } ++ ++ k=0; ++ do ++ { ++ FLOAT s1,c1; ++ k += 2; ++ k1 = 1 << k; ++ k2 = k1 << 1; ++ k4 = k2 << 1; ++ k3 = k2 + k1; ++ kx = k1 >> 1; ++ fi = fz; ++ gi = fi + kx; ++ fn = fz + n; ++ do ++ { ++ FLOAT g0,f0,f1,g1,f2,g2,f3,g3; ++ f1 = fi[0 ] - fi[k1]; ++ f0 = fi[0 ] + fi[k1]; ++ f3 = fi[k2] - fi[k3]; ++ f2 = fi[k2] + fi[k3]; ++ fi[k2] = f0 - f2; ++ fi[0 ] = f0 + f2; ++ fi[k3] = f1 - f3; ++ fi[k1] = f1 + f3; ++ g1 = gi[0 ] - gi[k1]; ++ g0 = gi[0 ] + gi[k1]; ++ g3 = SQRT2 * gi[k3]; ++ g2 = SQRT2 * gi[k2]; ++ gi[k2] = g0 - g2; ++ gi[0 ] = g0 + g2; ++ gi[k3] = g1 - g3; ++ gi[k1] = g1 + g3; ++ gi += k4; ++ fi += k4; ++ } while (fi<fn); ++ t_c = costab[k]; ++ t_s = sintab[k]; ++ c1 = 1; ++ s1 = 0; ++ for (i=1;i<kx;i++) ++ { ++ FLOAT c2,s2; ++ FLOAT t = c1; ++ c1 = t*t_c - s1*t_s; ++ s1 = t*t_s + s1*t_c; ++ c2 = c1*c1 - s1*s1; ++ s2 = 2*(c1*s1); ++ fn = fz + n; ++ fi = fz +i; ++ gi = fz +k1-i; ++ do ++ { ++ FLOAT a,b,g0,f0,f1,g1,f2,g2,f3,g3; ++ b = s2*fi[k1] - c2*gi[k1]; ++ a = c2*fi[k1] + s2*gi[k1]; ++ f1 = fi[0 ] - a; ++ f0 = fi[0 ] + a; ++ g1 = gi[0 ] - b; ++ g0 = gi[0 ] + b; ++ b = s2*fi[k3] - c2*gi[k3]; ++ a = c2*fi[k3] + s2*gi[k3]; ++ f3 = fi[k2] - a; ++ f2 = fi[k2] + a; ++ g3 = gi[k2] - b; ++ g2 = gi[k2] + b; ++ b = s1*f2 - c1*g3; ++ a = c1*f2 + s1*g3; ++ fi[k2] = f0 - a; ++ fi[0 ] = f0 + a; ++ gi[k3] = g1 - b; ++ gi[k1] = g1 + b; ++ b = c1*g2 - s1*f3; ++ a = s1*g2 + c1*f3; ++ gi[k2] = g0 - a; ++ gi[0 ] = g0 + a; ++ fi[k3] = f1 - b; ++ fi[k1] = f1 + b; ++ gi += k4; ++ fi += k4; ++ } while (fi<fn); ++ } ++ } while (k4<n); ++ } ++ ++ void fft(FLOAT *x_real, FLOAT *energy, FLOAT *ax, FLOAT *bx, int N) ++ { ++ FLOAT a,b; ++ int i,j; ++ ++ fht(x_real,N); ++ ++ ++ energy[0] = x_real[0] * x_real[0]; ++ ax[0] = bx[0] = x_real[0]; ++ ++ for (i=1,j=N-1;i<N/2;i++,j--) { ++ a = ax[i] = x_real[i]; ++ b = bx[i] = x_real[j]; ++ energy[i]=(a*a + b*b)/2; ++ ++ if (energy[i] < 0.0005) { ++ energy[i] = 0.0005; ++ ax[i] = bx[i] = 0.0223606797749978970790696308768019662239; /* was sqrt(0.0005) */ ++ } + } +! energy[N/2] = x_real[N/2] * x_real[N/2]; +! ax[N/2] = bx[N/2] = x_real[N/2]; +! } +! +! +! +! +! double fft_side( FLOAT in[2][1024], int s) +! { +! double energy,t; +! FLOAT a,b; +! int i,j; +! +! if(!s) +! { +! energy = (in[0][0] - in[1][0]) * (in[0][0] - in[1][0])/2; +! s++; + } +! else +! energy=0.0; +! +! for (i=s,j=1024-s;i<512;i++,j--) { +! a = in[0][i] - in[1][i]; +! b = in[0][j] - in[1][j]; +! +! t=(a*a + b*b)/4; +! +! if (t < 0.0005) { +! t = 0.0005; +! } +! energy += t; + } ++ return energy + (in[0][512] - in[1][512]) * (in[0][512] - in[1][512])/2; ++ ++ + } ++ ++ +diff -r -c -N encoder/subs.h lame3.13/subs.h +*** encoder/subs.h Wed Dec 31 17:00:00 1969 +--- lame3.13/subs.h Wed Jun 2 21:37:52 1999 +*************** +*** 0 **** +--- 1,12 ---- ++ #ifndef FFT_H ++ #define FFT_H ++ ++ #include "types.h" ++ ++ void fft(float x_real[BLKSIZE], ++ float energy[BLKSIZE], ++ float ax[BLKSIZE], bx[BLKSIZE], ++ int N); ++ ++ ++ #endif +diff -r -c -N encoder/tables.c lame3.13/tables.c +*** encoder/tables.c Wed Dec 31 17:00:00 1969 +--- lame3.13/tables.c Thu Jun 17 23:30:58 1999 +*************** +*** 0 **** +--- 1,969 ---- ++ double psy_data[] = ++ {48000.0, 61, ++ 0, 1, 24.5, 4.532, 0.970, 0.000, ++ 1, 1, 24.5, 4.532, 0.755, 0.469, ++ 2, 1, 24.5, 4.532, 0.738, 0.938, ++ 3, 1, 24.5, 0.904, 0.730, 1.406, ++ 4, 1, 24.5, 0.904, 0.724, 1.875, ++ 5, 1, 20.0, 0.090, 0.723, 2.344, ++ 6, 1, 20.0, 0.090, 0.723, 2.813, ++ 7, 1, 20.0, 0.029, 0.723, 3.281, ++ 8, 1, 20.0, 0.029, 0.718, 3.750, ++ 9, 1, 20.0, 0.009, 0.690, 4.199, ++ 10, 1, 20.0, 0.009, 0.660, 4.625, ++ 11, 1, 18.0, 0.009, 0.641, 5.047, ++ 12, 1, 18.0, 0.009, 0.600, 5.438, ++ 13, 1, 18.0, 0.009, 0.584, 5.828, ++ 14, 1, 12.0, 0.009, 0.532, 6.188, ++ 15, 1, 12.0, 0.009, 0.537, 6.522, ++ 16, 2, 6.0, 0.018, 0.857, 7.174, ++ 17, 2, 6.0, 0.018, 0.858, 7.801, ++ 18, 2, 3.0, 0.018, 0.853, 8.402, ++ 19, 2, 3.0, 0.018, 0.824, 8.966, ++ 20, 2, 3.0, 0.018, 0.778, 9.484, ++ 21, 2, 3.0, 0.018, 0.740, 9.966, ++ 22, 2, 0.0, 0.018, 0.709, 10.426, ++ 23, 2, 0.0, 0.018, 0.676, 10.866, ++ 24, 2, 0.0, 0.018, 0.632, 11.279, ++ 25, 2, 0.0, 0.018, 0.592, 11.669, ++ 26, 2, 0.0, 0.018, 0.553, 12.042, ++ 27, 2, 0.0, 0.018, 0.510, 12.386, ++ 28, 2, 0.0, 0.018, 0.513, 12.721, ++ 29, 3, 0.0, 0.027, 0.608, 13.115, ++ 30, 3, 0.0, 0.027, 0.673, 13.562, ++ 31, 3, 0.0, 0.027, 0.637, 13.984, ++ 32, 3, 0.0, 0.027, 0.586, 14.371, ++ 33, 3, 0.0, 0.027, 0.571, 14.741, ++ 34, 4, 0.0, 0.036, 0.616, 15.140, ++ 35, 4, 0.0, 0.036, 0.640, 15.563, ++ 36, 4, 0.0, 0.036, 0.598, 15.962, ++ 37, 4, 0.0, 0.036, 0.538, 16.324, ++ 38, 4, 0.0, 0.036, 0.512, 16.665, ++ 39, 5, 0.0, 0.045, 0.528, 17.020, ++ 40, 5, 0.0, 0.045, 0.517, 17.373, ++ 41, 5, 0.0, 0.045, 0.493, 17.708, ++ 42, 6, 0.0, 0.054, 0.499, 18.045, ++ 43, 7, 0.0, 0.063, 0.525, 18.398, ++ 44, 7, 0.0, 0.063, 0.541, 18.762, ++ 45, 8, 0.0, 0.072, 0.528, 19.120, ++ 46, 8, 0.0, 0.072, 0.510, 19.466, ++ 47, 8, 0.0, 0.072, 0.506, 19.807, ++ 48, 10, 0.0, 0.180, 0.525, 20.159, ++ 49, 10, 0.0, 0.180, 0.536, 20.522, ++ 50, 10, 0.0, 0.180, 0.518, 20.874, ++ 51, 13, 0.0, 0.372, 0.501, 21.214, ++ 52, 13, 0.0, 0.372, 0.497, 21.553, ++ 53, 14, 0.0, 0.400, 0.497, 21.892, ++ 54, 18, 0.0, 1.627, 0.495, 22.231, ++ 55, 18, 0.0, 1.627, 0.494, 22.569, ++ 56, 20, 0.0, 1.808, 0.497, 22.909, ++ 57, 25, 0.0, 22.607, 0.494, 23.248, ++ 58, 25, 0.0, 22.607, 0.487, 23.583, ++ 59, 35, 0.0, 31.650, 0.483, 23.915, ++ 60, 67, 0.0, 605.867, 0.482, 24.246, ++ 61, 67, 0.0, 605.867, 0.524, 24.576, ++ 44100.0, 62, ++ 0, 1, 24.5, 4.532, 0.951, 0.000, ++ 1, 1, 24.5, 4.532, 0.700, 0.431, ++ 2, 1, 24.5, 4.532, 0.681, 0.861, ++ 3, 1, 24.5, 0.904, 0.675, 1.292, ++ 4, 1, 24.5, 0.904, 0.667, 1.723, ++ 5, 1, 20.0, 0.090, 0.665, 2.153, ++ 6, 1, 20.0, 0.090, 0.664, 2.584, ++ 7, 1, 20.0, 0.029, 0.664, 3.015, ++ 8, 1, 20.0, 0.029, 0.664, 3.445, ++ 9, 1, 20.0, 0.029, 0.655, 3.876, ++ 10, 1, 20.0, 0.009, 0.616, 4.279, ++ 11, 1, 20.0, 0.009, 0.597, 4.670, ++ 12, 1, 18.0, 0.009, 0.578, 5.057, ++ 13, 1, 18.0, 0.009, 0.541, 5.416, ++ 14, 1, 18.0, 0.009, 0.575, 5.774, ++ 15, 2, 12.0, 0.018, 0.856, 6.422, ++ 16, 2, 6.0, 0.018, 0.846, 7.026, ++ 17, 2, 6.0, 0.018, 0.840, 7.609, ++ 18, 2, 3.0, 0.018, 0.822, 8.168, ++ 19, 2, 3.0, 0.018, 0.800, 8.710, ++ 20, 2, 3.0, 0.018, 0.753, 9.207, ++ 21, 2, 3.0, 0.018, 0.704, 9.662, ++ 22, 2, 0.0, 0.018, 0.674, 10.099, ++ 23, 2, 0.0, 0.018, 0.640, 10.515, ++ 24, 2, 0.0, 0.018, 0.609, 10.917, ++ 25, 2, 0.0, 0.018, 0.566, 11.293, ++ 26, 2, 0.0, 0.018, 0.535, 11.652, ++ 27, 2, 0.0, 0.018, 0.531, 11.997, ++ 28, 3, 0.0, 0.027, 0.615, 12.394, ++ 29, 3, 0.0, 0.027, 0.686, 12.850, ++ 30, 3, 0.0, 0.027, 0.650, 13.277, ++ 31, 3, 0.0, 0.027, 0.612, 13.681, ++ 32, 3, 0.0, 0.027, 0.567, 14.062, ++ 33, 3, 0.0, 0.027, 0.520, 14.411, ++ 34, 3, 0.0, 0.027, 0.513, 14.751, ++ 35, 4, 0.0, 0.036, 0.557, 15.119, ++ 36, 4, 0.0, 0.036, 0.584, 15.508, ++ 37, 4, 0.0, 0.036, 0.570, 15.883, ++ 38, 5, 0.0, 0.045, 0.579, 16.263, ++ 39, 5, 0.0, 0.045, 0.585, 16.654, ++ 40, 5, 0.0, 0.045, 0.548, 17.020, ++ 41, 6, 0.0, 0.054, 0.536, 17.374, ++ 42, 6, 0.0, 0.054, 0.550, 17.744, ++ 43, 7, 0.0, 0.063, 0.532, 18.104, ++ 44, 7, 0.0, 0.063, 0.504, 18.447, ++ 45, 7, 0.0, 0.063, 0.496, 18.782, ++ 46, 9, 0.0, 0.081, 0.517, 19.130, ++ 47, 9, 0.0, 0.081, 0.527, 19.487, ++ 48, 9, 0.0, 0.081, 0.516, 19.838, ++ 49, 10, 0.0, 0.180, 0.497, 20.179, ++ 50, 10, 0.0, 0.180, 0.489, 20.510, ++ 51, 11, 0.0, 0.198, 0.502, 20.852, ++ 52, 14, 0.0, 0.400, 0.501, 21.196, ++ 53, 14, 0.0, 0.400, 0.491, 21.531, ++ 54, 15, 0.0, 0.429, 0.497, 21.870, ++ 55, 20, 0.0, 1.808, 0.504, 22.214, ++ 56, 20, 0.0, 1.808, 0.504, 22.558, ++ 57, 21, 0.0, 1.898, 0.495, 22.898, ++ 58, 27, 0.0, 24.416, 0.486, 23.232, ++ 59, 27, 0.0, 24.416, 0.484, 23.564, ++ 60, 36, 0.0, 32.554, 0.483, 23.897, ++ 61, 73, 0.0, 660.124, 0.475, 24.229, ++ 62, 18, 0.0, 162.770, 0.515, 24.442, ++ 32000.0, 58, ++ 0, 2, 24.5, 4.532, 0.997, 0.313, ++ 1, 2, 24.5, 4.532, 0.893, 0.938, ++ 2, 2, 24.5, 1.809, 0.881, 1.563, ++ 3, 2, 20.0, 0.181, 0.873, 2.188, ++ 4, 2, 20.0, 0.181, 0.872, 2.813, ++ 5, 2, 20.0, 0.057, 0.871, 3.438, ++ 6, 2, 20.0, 0.018, 0.860, 4.045, ++ 7, 2, 20.0, 0.018, 0.839, 4.625, ++ 8, 2, 18.0, 0.018, 0.812, 5.173, ++ 9, 2, 18.0, 0.018, 0.784, 5.698, ++ 10, 2, 12.0, 0.018, 0.741, 6.185, ++ 11, 2, 12.0, 0.018, 0.697, 6.634, ++ 12, 2, 6.0, 0.018, 0.674, 7.070, ++ 13, 2, 6.0, 0.018, 0.651, 7.492, ++ 14, 2, 6.0, 0.018, 0.633, 7.905, ++ 15, 2, 3.0, 0.018, 0.611, 8.305, ++ 16, 2, 3.0, 0.018, 0.589, 8.695, ++ 17, 2, 3.0, 0.018, 0.575, 9.064, ++ 18, 3, 3.0, 0.027, 0.654, 9.484, ++ 19, 3, 3.0, 0.027, 0.724, 9.966, ++ 20, 3, 0.0, 0.027, 0.701, 10.426, ++ 21, 3, 0.0, 0.027, 0.673, 10.866, ++ 22, 3, 0.0, 0.027, 0.631, 11.279, ++ 23, 3, 0.0, 0.027, 0.592, 11.669, ++ 24, 3, 0.0, 0.027, 0.553, 12.042, ++ 25, 3, 0.0, 0.027, 0.510, 12.386, ++ 26, 3, 0.0, 0.027, 0.506, 12.721, ++ 27, 4, 0.0, 0.036, 0.562, 13.091, ++ 28, 4, 0.0, 0.036, 0.598, 13.488, ++ 29, 4, 0.0, 0.036, 0.589, 13.873, ++ 30, 5, 0.0, 0.045, 0.607, 14.268, ++ 31, 5, 0.0, 0.045, 0.620, 14.679, ++ 32, 5, 0.0, 0.045, 0.580, 15.067, ++ 33, 5, 0.0, 0.045, 0.532, 15.424, ++ 34, 5, 0.0, 0.045, 0.517, 15.771, ++ 35, 6, 0.0, 0.054, 0.517, 16.120, ++ 36, 6, 0.0, 0.054, 0.509, 16.466, ++ 37, 6, 0.0, 0.054, 0.506, 16.807, ++ 38, 8, 0.0, 0.072, 0.522, 17.158, ++ 39, 8, 0.0, 0.072, 0.531, 17.518, ++ 40, 8, 0.0, 0.072, 0.519, 17.869, ++ 41, 10, 0.0, 0.090, 0.512, 18.215, ++ 42, 10, 0.0, 0.090, 0.509, 18.563, ++ 43, 10, 0.0, 0.090, 0.498, 18.902, ++ 44, 12, 0.0, 0.109, 0.494, 19.239, ++ 45, 12, 0.0, 0.109, 0.501, 19.580, ++ 46, 13, 0.0, 0.118, 0.508, 19.925, ++ 47, 14, 0.0, 0.252, 0.502, 20.269, ++ 48, 14, 0.0, 0.252, 0.493, 20.606, ++ 49, 16, 0.0, 0.288, 0.497, 20.944, ++ 50, 20, 0.0, 0.572, 0.506, 21.288, ++ 51, 20, 0.0, 0.572, 0.510, 21.635, ++ 52, 23, 0.0, 0.658, 0.504, 21.980, ++ 53, 27, 0.0, 2.441, 0.496, 22.319, ++ 54, 27, 0.0, 2.441, 0.493, 22.656, ++ 55, 32, 0.0, 2.893, 0.490, 22.993, ++ 56, 37, 0.0, 33.458, 0.482, 23.326, ++ 57, 37, 0.0, 33.458, 0.458, 23.656, ++ 58, 12, 0.0, 10.851, 0.500, 23.937, ++ 24000, 59, ++ 0, 2, 15, 17.8250179, 0.697374165, 0.236874461, ++ 1, 2, 15, 17.8250179, 0.455024809, 0.71016103, ++ 2, 2, 15, 1.78250182, 0.431440443, 1.18193281, ++ 3, 2, 15, 1.78250182, 0.42391625, 1.65102732, ++ 4, 2, 13, 0.178250194, 0.418206781, 2.11632562, ++ 5, 2, 13, 0.178250194, 0.41158545, 2.57676744, ++ 6, 2, 13, 0.0563676581, 0.405409157, 3.03136396, ++ 7, 2, 13, 0.0563676581, 0.399695486, 3.47920918, ++ 8, 2, 13, 0.0563676581, 0.393753231, 3.91948748, ++ 9, 2, 12, 0.0178250186, 0.387357473, 4.35147953, ++ 10, 2, 12, 0.0178250186, 0.38045457, 4.77456427, ++ 11, 2, 10, 0.0178250186, 0.373053908, 5.18822002, ++ 12, 2, 10, 0.0178250186, 0.365188122, 5.59202194, ++ 13, 2, 10, 0.0178250186, 0.356897771, 5.98564005, ++ 14, 2, 9, 0.0178250186, 0.348700613, 6.36883163, ++ 15, 2, 9, 0.0178250186, 0.340260029, 6.74143791, ++ 16, 2, 6, 0.0178250186, 0.332341045, 7.10337448, ++ 17, 2, 6, 0.0178250186, 0.330462843, 7.45462418, ++ 18, 2, 6, 0.0178250186, 0.345568359, 7.79523182, ++ 19, 3, 3, 0.0267375279, 0.377859652, 8.20455742, ++ 20, 3, 3, 0.0267375279, 0.396689415, 8.67640114, ++ 21, 3, 3, 0.0267375279, 0.391237885, 9.12561035, ++ 22, 3, 3, 0.0267375279, 0.37761277, 9.55298138, ++ 23, 3, 3, 0.0267375279, 0.362836808, 9.95940971, ++ 24, 3, 0, 0.0267375279, 0.349010617, 10.3458519, ++ 25, 3, 0, 0.0267375279, 0.339673489, 10.7132998, ++ 26, 3, 0, 0.0267375279, 0.343845725, 11.0627575, ++ 27, 4, 0, 0.0356500372, 0.355822682, 11.447506, ++ 28, 4, 0, 0.0356500372, 0.358104348, 11.8627586, ++ 29, 4, 0, 0.0356500372, 0.34745428, 12.2520256, ++ 30, 4, 0, 0.0356500372, 0.334927917, 12.6173973, ++ 31, 4, 0, 0.0356500372, 0.331643254, 12.9608269, ++ 32, 5, 0, 0.0445625484, 0.333368897, 13.3219252, ++ 33, 5, 0, 0.0445625484, 0.332313001, 13.6976833, ++ 34, 5, 0, 0.0445625484, 0.3314417, 14.047802, ++ 35, 6, 0, 0.0534750558, 0.330947191, 14.405302, ++ 36, 6, 0, 0.0534750558, 0.332477689, 14.7684803, ++ 37, 7, 0, 0.062387567, 0.332647532, 15.1315956, ++ 38, 7, 0, 0.062387567, 0.330841452, 15.4940481, ++ 39, 8, 0, 0.0713000745, 0.327769846, 15.8516159, ++ 40, 8, 0, 0.0713000745, 0.324572712, 16.204628, ++ 41, 9, 0, 0.0802125856, 0.323825002, 16.5502281, ++ 42, 10, 0, 0.0891250968, 0.321414798, 16.9067478, ++ 43, 10, 0, 0.0891250968, 0.318189293, 17.2537231, ++ 44, 11, 0, 0.0980376005, 0.315934151, 17.5901108, ++ 45, 12, 0, 0.106950112, 0.315639287, 17.931406, ++ 46, 13, 0, 0.115862623, 0.316569835, 18.2750721, ++ 47, 14, 0, 0.124775134, 0.31656, 18.6191597, ++ 48, 15, 0, 0.133687645, 0.315465957, 18.9621754, ++ 49, 16, 0, 0.142600149, 0.313576341, 19.3029613, ++ 50, 17, 0, 0.151512653, 0.311635971, 19.6405869, ++ 51, 18, 0, 0.160425171, 0.311066717, 19.9742699, ++ 52, 20, 0, 0.355655879, 0.311465651, 20.3115921, ++ 53, 21, 0, 0.373438686, 0.311872005, 20.6507797, ++ 54, 23, 0, 0.409004271, 0.311015964, 20.9890823, ++ 55, 24, 0, 0.676411927, 0.309207708, 21.3251152, ++ 56, 26, 0, 0.732779562, 0.3081128, 21.6565971, ++ 57, 28, 0, 0.789147198, 0.310006589, 21.9881554, ++ 58, 31, 0, 2.76287794, 0.327113092, 22.3222847, ++ 59, 34, 0, 3.03025317, 0.416082352, 22.6605186, ++ 22050, 59, ++ 0, 2, 15, 17.8250179, 0.658683598, 0.217637643, ++ 1, 2, 15, 17.8250179, 0.432554901, 0.652563453, ++ 2, 2, 15, 1.78250182, 0.405113578, 1.08633137, ++ 3, 2, 15, 1.78250182, 0.397231787, 1.51803517, ++ 4, 2, 15, 1.78250182, 0.392088681, 1.94679713, ++ 5, 2, 13, 0.178250194, 0.386788279, 2.37177849, ++ 6, 2, 13, 0.178250194, 0.380574644, 2.79218864, ++ 7, 2, 13, 0.0563676581, 0.375309765, 3.20729256, ++ 8, 2, 13, 0.0563676581, 0.370087624, 3.61641645, ++ 9, 2, 12, 0.0178250186, 0.364568561, 4.01895428, ++ 10, 2, 12, 0.0178250186, 0.358959526, 4.4143672, ++ 11, 2, 12, 0.0178250186, 0.352938265, 4.80218887, ++ 12, 2, 10, 0.0178250186, 0.3465029, 5.18202305, ++ 13, 2, 10, 0.0178250186, 0.33968094, 5.55354261, ++ 14, 2, 10, 0.0178250186, 0.332571507, 5.91648674, ++ 15, 2, 9, 0.0178250186, 0.326015651, 6.27065945, ++ 16, 2, 9, 0.0178250186, 0.325442046, 6.61592293, ++ 17, 2, 9, 0.0178250186, 0.341315031, 6.95219517, ++ 18, 3, 6, 0.0267375279, 0.374984443, 7.3584404, ++ 19, 3, 6, 0.0267375279, 0.396138102, 7.8290925, ++ 20, 3, 3, 0.0267375279, 0.39271906, 8.27975655, ++ 21, 3, 3, 0.0267375279, 0.380755007, 8.71083069, ++ 22, 3, 3, 0.0267375279, 0.367386311, 9.12284088, ++ 23, 3, 3, 0.0267375279, 0.354351997, 9.51640987, ++ 24, 3, 3, 0.0267375279, 0.341508389, 9.89222908, ++ 25, 3, 0, 0.0267375279, 0.333577901, 10.2510386, ++ 26, 3, 0, 0.0267375279, 0.338108748, 10.5936022, ++ 27, 4, 0, 0.0356500372, 0.350744486, 10.9723492, ++ 28, 4, 0, 0.0356500372, 0.354519457, 11.38272, ++ 29, 4, 0, 0.0356500372, 0.345274031, 11.7689981, ++ 30, 4, 0, 0.0356500372, 0.333828837, 12.1329184, ++ 31, 4, 0, 0.0356500372, 0.331436664, 12.4761295, ++ 32, 5, 0, 0.0445625484, 0.334172577, 12.8381901, ++ 33, 5, 0, 0.0445625484, 0.334024847, 13.2160273, ++ 34, 5, 0, 0.0445625484, 0.33392629, 13.5690479, ++ 35, 6, 0, 0.0534750558, 0.334218502, 13.9303951, ++ 36, 6, 0, 0.0534750558, 0.336405039, 14.298193, ++ 37, 7, 0, 0.062387567, 0.337080389, 14.666563, ++ 38, 7, 0, 0.062387567, 0.335603535, 15.0346909, ++ 39, 8, 0, 0.0713000745, 0.332515866, 15.398139, ++ 40, 8, 0, 0.0713000745, 0.327727586, 15.7570457, ++ 41, 9, 0, 0.0802125856, 0.322346836, 16.1083431, ++ 42, 9, 0, 0.0802125856, 0.317575186, 16.4528522, ++ 43, 10, 0, 0.0891250968, 0.31632933, 16.7886105, ++ 44, 11, 0, 0.0980376005, 0.317602783, 17.132, ++ 45, 12, 0, 0.106950112, 0.319945186, 17.4796028, ++ 46, 13, 0, 0.115862623, 0.320881754, 17.8287659, ++ 47, 14, 0, 0.124775134, 0.320346534, 18.1774921, ++ 48, 15, 0, 0.133687645, 0.318628669, 18.5243168, ++ 49, 16, 0, 0.142600149, 0.316125751, 18.8681736, ++ 50, 17, 0, 0.151512653, 0.313746184, 19.2082729, ++ 51, 18, 0, 0.160425171, 0.312971771, 19.5440025, ++ 52, 20, 0, 0.178250194, 0.313278913, 19.8831882, ++ 53, 21, 0, 0.373438686, 0.313735574, 20.224247, ++ 54, 23, 0, 0.409004271, 0.31308493, 20.5646286, ++ 55, 24, 0, 0.426787049, 0.31156227, 20.903141, ++ 56, 26, 0, 0.732779562, 0.310435742, 21.2376747, ++ 57, 28, 0, 0.789147198, 0.31132248, 21.5730591, ++ 58, 30, 0, 0.845514894, 0.32730341, 21.9066811, ++ 59, 33, 0, 2.94112802, 0.414659739, 22.2411156, ++ 16000, 55, ++ 0, 3, 15, 26.7375278, 0.697374165, 0.236874461, ++ 1, 3, 15, 26.7375278, 0.455024809, 0.71016103, ++ 2, 3, 15, 2.67375278, 0.431440443, 1.18193281, ++ 3, 3, 15, 2.67375278, 0.42391625, 1.65102732, ++ 4, 3, 13, 0.26737529, 0.418206781, 2.11632562, ++ 5, 3, 13, 0.26737529, 0.41158545, 2.57676744, ++ 6, 3, 13, 0.0845514908, 0.405409157, 3.03136396, ++ 7, 3, 13, 0.0845514908, 0.399695486, 3.47920918, ++ 8, 3, 13, 0.0845514908, 0.393753231, 3.91948748, ++ 9, 3, 12, 0.0267375279, 0.387357473, 4.35147953, ++ 10, 3, 12, 0.0267375279, 0.38045457, 4.77456427, ++ 11, 3, 10, 0.0267375279, 0.373053908, 5.18822002, ++ 12, 3, 10, 0.0267375279, 0.365188122, 5.59202194, ++ 13, 3, 10, 0.0267375279, 0.356897742, 5.98564005, ++ 14, 3, 9, 0.0267375279, 0.34869957, 6.36883163, ++ 15, 3, 9, 0.0267375279, 0.340241522, 6.74143791, ++ 16, 3, 6, 0.0267375279, 0.332089454, 7.10337448, ++ 17, 3, 6, 0.0267375279, 0.328292668, 7.45462418, ++ 18, 3, 6, 0.0267375279, 0.336574793, 7.79523182, ++ 19, 4, 3, 0.0356500372, 0.354600489, 8.17827797, ++ 20, 4, 3, 0.0356500372, 0.364343345, 8.59994984, ++ 21, 4, 3, 0.0356500372, 0.359369367, 9.00363636, ++ 22, 4, 3, 0.0356500372, 0.347775847, 9.38988018, ++ 23, 4, 3, 0.0356500372, 0.335562587, 9.7592926, ++ 24, 4, 0, 0.0356500372, 0.326988578, 10.1125278, ++ 25, 4, 0, 0.0356500372, 0.327966213, 10.4502735, ++ 26, 5, 0, 0.0445625484, 0.334450752, 10.811614, ++ 27, 5, 0, 0.0445625484, 0.335228145, 11.1935263, ++ 28, 5, 0, 0.0445625484, 0.329595625, 11.5549288, ++ 29, 5, 0, 0.0445625484, 0.326683223, 11.8971443, ++ 30, 6, 0, 0.0534750558, 0.326986551, 12.2520256, ++ 31, 6, 0, 0.0534750558, 0.325072199, 12.6173973, ++ 32, 6, 0, 0.0534750558, 0.323560268, 12.9608269, ++ 33, 7, 0, 0.062387567, 0.322494298, 13.3093863, ++ 34, 7, 0, 0.062387567, 0.323403448, 13.6617231, ++ 35, 8, 0, 0.0713000745, 0.323232353, 14.0134668, ++ 36, 8, 0, 0.0713000745, 0.322662383, 14.3639784, ++ 37, 9, 0, 0.0802125856, 0.324054241, 14.7098465, ++ 38, 10, 0, 0.0891250968, 0.323228806, 15.0686541, ++ 39, 10, 0, 0.0891250968, 0.320751846, 15.4191036, ++ 40, 11, 0, 0.0980376005, 0.318823338, 15.7594051, ++ 41, 12, 0, 0.106950112, 0.318418682, 16.104557, ++ 42, 13, 0, 0.115862623, 0.318762124, 16.451416, ++ 43, 14, 0, 0.124775134, 0.317806393, 16.7975388, ++ 44, 15, 0, 0.133687645, 0.315653771, 17.1411018, ++ 45, 16, 0, 0.142600149, 0.313369036, 17.4808159, ++ 46, 17, 0, 0.151512653, 0.312513858, 17.8158207, ++ 47, 19, 0, 0.169337675, 0.312785119, 18.1543369, ++ 48, 20, 0, 0.178250194, 0.31343773, 18.4948578, ++ 49, 22, 0, 0.196075201, 0.313258767, 18.8350143, ++ 50, 23, 0, 0.20498772, 0.312570423, 19.1740704, ++ 51, 25, 0, 0.222812727, 0.312572777, 19.5104179, ++ 52, 27, 0, 0.240637749, 0.313047856, 19.8497677, ++ 53, 29, 0, 0.515701056, 0.315029174, 20.1900635, ++ 54, 31, 0, 0.551266611, 0.330613613, 20.5294952, ++ 55, 33, 0, 0.586832225, 0.41819948, 20.8664398, ++ 48000.0, 37, ++ 0, 1, 4.532, 1.000, -8.240, 0.000, ++ 1, 1, 0.904, 0.989, -8.240, 1.875, ++ 2, 1, 0.029, 0.989, -8.240, 3.750, ++ 3, 1, 0.009, 0.981, -8.240, 5.438, ++ 4, 1, 0.009, 0.985, -8.240, 6.857, ++ 5, 1, 0.009, 0.984, -8.240, 8.109, ++ 6, 1, 0.009, 0.980, -8.240, 9.237, ++ 7, 1, 0.009, 0.968, -8.240, 10.202, ++ 8, 1, 0.009, 0.954, -8.240, 11.083, ++ 9, 1, 0.009, 0.929, -8.240, 11.865, ++ 10, 1, 0.009, 0.906, -7.447, 12.554, ++ 11, 1, 0.009, 0.883, -7.447, 13.195, ++ 12, 1, 0.009, 0.844, -7.447, 13.781, ++ 13, 1, 0.009, 0.792, -7.447, 14.309, ++ 14, 1, 0.009, 0.747, -7.447, 14.803, ++ 15, 1, 0.009, 0.689, -7.447, 15.250, ++ 16, 1, 0.009, 0.644, -7.447, 15.667, ++ 17, 1, 0.009, 0.592, -7.447, 16.068, ++ 18, 1, 0.009, 0.553, -7.447, 16.409, ++ 19, 2, 0.018, 0.850, -7.447, 17.045, ++ 20, 2, 0.018, 0.811, -6.990, 17.607, ++ 21, 2, 0.018, 0.736, -6.990, 18.097, ++ 22, 2, 0.018, 0.665, -6.990, 18.528, ++ 23, 2, 0.018, 0.610, -6.990, 18.931, ++ 24, 2, 0.018, 0.544, -6.990, 19.295, ++ 25, 2, 0.018, 0.528, -6.990, 19.636, ++ 26, 3, 0.054, 0.621, -6.990, 20.038, ++ 27, 3, 0.054, 0.673, -6.990, 20.486, ++ 28, 3, 0.054, 0.635, -6.990, 20.900, ++ 29, 4, 0.114, 0.626, -6.990, 21.306, ++ 30, 4, 0.114, 0.636, -6.020, 21.722, ++ 31, 5, 0.452, 0.615, -6.020, 22.128, ++ 32, 5, 0.452, 0.579, -6.020, 22.513, ++ 33, 5, 0.452, 0.551, -6.020, 22.877, ++ 34, 7, 6.330, 0.552, -5.229, 23.241, ++ 35, 7, 6.330, 0.559, -5.229, 23.616, ++ 36, 11, 9.947, 0.528, -5.229, 23.974, ++ 37, 17, 153.727, 0.479, -5.229, 24.313, ++ 44100.0, 38, ++ 0, 1, 4.532, 1.000, -8.240, 0.000, ++ 1, 1, 0.904, 0.983, -8.240, 1.723, ++ 2, 1, 0.029, 0.983, -8.240, 3.445, ++ 3, 1, 0.009, 0.982, -8.240, 5.057, ++ 4, 1, 0.009, 0.985, -8.240, 6.422, ++ 5, 1, 0.009, 0.983, -8.240, 7.609, ++ 6, 1, 0.009, 0.978, -8.240, 8.710, ++ 7, 1, 0.009, 0.967, -8.240, 9.662, ++ 8, 1, 0.009, 0.948, -8.240, 10.515, ++ 9, 1, 0.009, 0.930, -8.240, 11.293, ++ 10, 1, 0.009, 0.914, -7.447, 12.009, ++ 11, 1, 0.009, 0.870, -7.447, 12.625, ++ 12, 1, 0.009, 0.845, -7.447, 13.210, ++ 13, 1, 0.009, 0.800, -7.447, 13.748, ++ 14, 1, 0.009, 0.749, -7.447, 14.241, ++ 15, 1, 0.009, 0.701, -7.447, 14.695, ++ 16, 1, 0.009, 0.653, -7.447, 15.125, ++ 17, 1, 0.009, 0.590, -7.447, 15.508, ++ 18, 1, 0.009, 0.616, -7.447, 15.891, ++ 19, 2, 0.018, 0.860, -7.447, 16.537, ++ 20, 2, 0.018, 0.823, -6.990, 17.112, ++ 21, 2, 0.018, 0.762, -6.990, 17.621, ++ 22, 2, 0.018, 0.688, -6.990, 18.073, ++ 23, 2, 0.018, 0.612, -6.990, 18.470, ++ 24, 2, 0.018, 0.594, -6.990, 18.849, ++ 25, 3, 0.027, 0.658, -6.990, 19.271, ++ 26, 3, 0.027, 0.706, -6.990, 19.741, ++ 27, 3, 0.054, 0.660, -6.990, 20.177, ++ 28, 3, 0.054, 0.606, -6.990, 20.576, ++ 29, 3, 0.054, 0.565, -6.990, 20.950, ++ 30, 4, 0.114, 0.560, -6.020, 21.316, ++ 31, 4, 0.114, 0.579, -6.020, 21.699, ++ 32, 5, 0.452, 0.567, -6.020, 22.078, ++ 33, 5, 0.452, 0.534, -6.020, 22.438, ++ 34, 5, 0.452, 0.514, -5.229, 22.782, ++ 35, 7, 6.330, 0.520, -5.229, 23.133, ++ 36, 7, 6.330, 0.518, -5.229, 23.484, ++ 37, 7, 6.330, 0.507, -5.229, 23.828, ++ 38, 19, 171.813, 0.447, -4.559, 24.173, ++ 32000.0, 41, ++ 0, 1, 4.532, 1.000, -8.240, 0.000, ++ 1, 1, 0.904, 0.985, -8.240, 1.250, ++ 2, 1, 0.090, 0.983, -8.240, 2.500, ++ 3, 1, 0.029, 0.983, -8.240, 3.750, ++ 4, 1, 0.009, 0.981, -8.240, 4.909, ++ 5, 1, 0.009, 0.975, -8.240, 5.958, ++ 6, 1, 0.009, 0.959, -8.240, 6.857, ++ 7, 1, 0.009, 0.944, -8.240, 7.700, ++ 8, 1, 0.009, 0.933, -8.240, 8.500, ++ 9, 1, 0.009, 0.920, -8.240, 9.237, ++ 10, 1, 0.009, 0.892, -7.447, 9.895, ++ 11, 1, 0.009, 0.863, -7.447, 10.500, ++ 12, 1, 0.009, 0.839, -7.447, 11.083, ++ 13, 1, 0.009, 0.786, -7.447, 11.604, ++ 14, 1, 0.009, 0.755, -7.447, 12.107, ++ 15, 1, 0.009, 0.698, -7.447, 12.554, ++ 16, 1, 0.009, 0.673, -7.447, 13.000, ++ 17, 1, 0.009, 0.605, -7.447, 13.391, ++ 18, 1, 0.009, 0.629, -7.447, 13.781, ++ 19, 2, 0.018, 0.883, -7.447, 14.474, ++ 20, 2, 0.018, 0.858, -6.990, 15.096, ++ 21, 2, 0.018, 0.829, -6.990, 15.667, ++ 22, 2, 0.018, 0.767, -6.990, 16.177, ++ 23, 2, 0.018, 0.705, -6.990, 16.636, ++ 24, 2, 0.018, 0.637, -6.990, 17.057, ++ 25, 2, 0.018, 0.564, -6.990, 17.429, ++ 26, 2, 0.018, 0.550, -6.990, 17.786, ++ 27, 3, 0.027, 0.603, -6.990, 18.177, ++ 28, 3, 0.027, 0.635, -6.990, 18.597, ++ 29, 3, 0.027, 0.592, -6.990, 18.994, ++ 30, 3, 0.027, 0.533, -6.020, 19.352, ++ 31, 3, 0.027, 0.518, -6.020, 19.693, ++ 32, 4, 0.072, 0.568, -6.020, 20.066, ++ 33, 4, 0.072, 0.594, -6.020, 20.462, ++ 34, 4, 0.072, 0.568, -5.229, 20.841, ++ 35, 5, 0.143, 0.536, -5.229, 21.201, ++ 36, 5, 0.143, 0.522, -5.229, 21.549, ++ 37, 6, 0.172, 0.542, -5.229, 21.911, ++ 38, 7, 0.633, 0.539, -4.559, 22.275, ++ 39, 7, 0.633, 0.519, -4.559, 22.625, ++ 40, 8, 0.723, 0.514, -3.980, 22.971, ++ 41, 10, 9.043, 0.518, -3.980, 23.321, ++ 24000, 44, ++ 0, 1, 8.91250896, 0.971850038, 0.150000006, 0, ++ 1, 1, 8.91250896, 0.874727964, 0.150000006, 0.946573138, ++ 2, 1, 0.891250908, 0.85779953, 0.150000006, 1.88476217, ++ 3, 1, 0.0891250968, 0.839743853, 0.150000006, 2.8056457, ++ 4, 1, 0.028183829, 0.82260257, 0.150000006, 3.70133615, ++ 5, 1, 0.00891250931, 0.80018574, 0.150000006, 4.56532001, ++ 6, 1, 0.00891250931, 0.771475196, 0.150000006, 5.39263105, ++ 7, 1, 0.00891250931, 0.737389982, 0.150000006, 6.17986727, ++ 8, 1, 0.00891250931, 0.701111019, 0.150000006, 6.92507982, ++ 9, 1, 0.00891250931, 0.65977633, 0.150000006, 7.62757969, ++ 10, 1, 0.00891250931, 0.615037441, 0.150000006, 8.28770351, ++ 11, 1, 0.00891250931, 0.568658054, 0.150000006, 8.90657234, ++ 12, 1, 0.00891250931, 0.522260666, 0.180000007, 9.48587132, ++ 13, 1, 0.00891250931, 0.478903115, 0.180000007, 10.0276566, ++ 14, 1, 0.00891250931, 0.43808648, 0.180000007, 10.5341988, ++ 15, 1, 0.00891250931, 0.412505627, 0.180000007, 11.0078659, ++ 16, 1, 0.00891250931, 0.39070797, 0.180000007, 11.4510288, ++ 17, 1, 0.00891250931, 0.371887118, 0.180000007, 11.866004, ++ 18, 1, 0.00891250931, 0.367617637, 0.180000007, 12.2550087, ++ 19, 1, 0.00891250931, 0.422220588, 0.180000007, 12.6201363, ++ 20, 2, 0.0178250186, 0.564990044, 0.180000007, 13.2772083, ++ 21, 2, 0.0178250186, 0.519700944, 0.180000007, 13.871047, ++ 22, 2, 0.0178250186, 0.455360681, 0.200000003, 14.4024391, ++ 23, 2, 0.0178250186, 0.408867925, 0.200000003, 14.8811684, ++ 24, 2, 0.0178250186, 0.381538749, 0.200000003, 15.3153324, ++ 25, 2, 0.0178250186, 0.362357527, 0.200000003, 15.7116165, ++ 26, 2, 0.0178250186, 0.365735918, 0.200000003, 16.0755405, ++ 27, 3, 0.0267375279, 0.38064, 0.200000003, 16.4882088, ++ 28, 3, 0.0267375279, 0.379183382, 0.200000003, 16.9410992, ++ 29, 3, 0.0267375279, 0.360672712, 0.200000003, 17.3513336, ++ 30, 3, 0.0267375279, 0.343065977, 0.200000003, 17.7264423, ++ 31, 3, 0.0267375279, 0.339290261, 0.200000003, 18.0722466, ++ 32, 4, 0.0356500372, 0.342963994, 0.200000003, 18.4426575, ++ 33, 4, 0.0356500372, 0.343128443, 0.200000003, 18.8344078, ++ 34, 4, 0.0356500372, 0.343988508, 0.25, 19.1955795, ++ 35, 5, 0.0445625484, 0.343928397, 0.25, 19.5697021, ++ 36, 5, 0.0445625484, 0.339527696, 0.25, 19.9551182, ++ 37, 5, 0.0889139697, 0.336541563, 0.280000001, 20.3115921, ++ 38, 6, 0.106696762, 0.334955156, 0.280000001, 20.6737747, ++ 39, 6, 0.169102982, 0.335601568, 0.300000012, 21.0404968, ++ 40, 7, 0.1972868, 0.334716886, 0.300000012, 21.4060211, ++ 41, 7, 0.1972868, 0.331676662, 0.300000012, 21.7696877, ++ 42, 8, 0.713000774, 0.328550965, 0.400000006, 22.1267223, ++ 43, 8, 0.713000774, 0.339241952, 0.400000006, 22.4769249, ++ 44, 9, 0.802125871, 0.425207615, 0.400000006, 22.8164864, ++ 22050, 44, ++ 0, 1, 8.91250896, 0.954045713, 0.150000006, 0, ++ 1, 1, 8.91250896, 0.833381653, 0.150000006, 0.869851649, ++ 2, 1, 0.891250908, 0.815945923, 0.150000006, 1.73325908, ++ 3, 1, 0.0891250968, 0.794244766, 0.150000006, 2.58322191, ++ 4, 1, 0.028183829, 0.776486695, 0.150000006, 3.4134295, ++ 5, 1, 0.00891250931, 0.755260408, 0.150000006, 4.21850443, ++ 6, 1, 0.00891250931, 0.731070817, 0.150000006, 4.99414825, ++ 7, 1, 0.00891250931, 0.701775849, 0.150000006, 5.73718691, ++ 8, 1, 0.00891250931, 0.667876124, 0.150000006, 6.44553185, ++ 9, 1, 0.00891250931, 0.630284071, 0.150000006, 7.11807632, ++ 10, 1, 0.00891250931, 0.590170324, 0.150000006, 7.75455618, ++ 11, 1, 0.00891250931, 0.548788548, 0.150000006, 8.3553915, ++ 12, 1, 0.00891250931, 0.507795513, 0.150000006, 8.92152882, ++ 13, 1, 0.00891250931, 0.469515711, 0.180000007, 9.45430183, ++ 14, 1, 0.00891250931, 0.432291716, 0.180000007, 9.95530319, ++ 15, 1, 0.00891250931, 0.411131173, 0.180000007, 10.4262848, ++ 16, 1, 0.00891250931, 0.390771538, 0.180000007, 10.8690758, ++ 17, 1, 0.00891250931, 0.373318017, 0.180000007, 11.2855215, ++ 18, 1, 0.00891250931, 0.36956048, 0.180000007, 11.6774378, ++ 19, 1, 0.00891250931, 0.42595759, 0.180000007, 12.0465794, ++ 20, 2, 0.0178250186, 0.576900065, 0.180000007, 12.7141209, ++ 21, 2, 0.0178250186, 0.533114731, 0.180000007, 13.3197365, ++ 22, 2, 0.0178250186, 0.469967514, 0.180000007, 13.8634901, ++ 23, 2, 0.0178250186, 0.417268544, 0.200000003, 14.3544445, ++ 24, 2, 0.0178250186, 0.389299124, 0.200000003, 14.8002586, ++ 25, 2, 0.0178250186, 0.362824857, 0.200000003, 15.2073727, ++ 26, 2, 0.0178250186, 0.346801281, 0.200000003, 15.5811834, ++ 27, 2, 0.0178250186, 0.349400043, 0.200000003, 15.926218, ++ 28, 3, 0.0267375279, 0.364026934, 0.200000003, 16.3194923, ++ 29, 3, 0.0267375279, 0.36560446, 0.200000003, 16.752903, ++ 30, 3, 0.0267375279, 0.354275256, 0.200000003, 17.1470814, ++ 31, 3, 0.0267375279, 0.351219416, 0.200000003, 17.5086212, ++ 32, 4, 0.0356500372, 0.354364097, 0.200000003, 17.8938141, ++ 33, 4, 0.0356500372, 0.348915905, 0.200000003, 18.2992878, ++ 34, 4, 0.0356500372, 0.337649345, 0.200000003, 18.6713982, ++ 35, 4, 0.0356500372, 0.332076877, 0.25, 19.015646, ++ 36, 5, 0.0445625484, 0.330793113, 0.25, 19.3734016, ++ 37, 5, 0.0445625484, 0.327528268, 0.25, 19.7430382, ++ 38, 5, 0.0889139697, 0.32551071, 0.280000001, 20.0859604, ++ 39, 6, 0.106696762, 0.324436843, 0.280000001, 20.4354992, ++ 40, 6, 0.106696762, 0.325835049, 0.280000001, 20.7905579, ++ 41, 7, 0.1972868, 0.326221824, 0.300000012, 21.1458054, ++ 42, 7, 0.1972868, 0.325960994, 0.300000012, 21.5005951, ++ 43, 8, 0.225470632, 0.339019388, 0.300000012, 21.8504524, ++ 44, 8, 0.713000774, 0.426850349, 0.400000006, 22.1951065, ++ 16000, 45, ++ 0, 1, 8.91250896, 0.834739447, 0.150000006, 0, ++ 1, 1, 8.91250896, 0.623757005, 0.150000006, 0.631518543, ++ 2, 1, 0.891250908, 0.60420388, 0.150000006, 1.2606914, ++ 3, 1, 0.891250908, 0.591974258, 0.150000006, 1.88476217, ++ 4, 1, 0.0891250968, 0.575301588, 0.150000006, 2.50111985, ++ 5, 1, 0.028183829, 0.561547697, 0.150000006, 3.1073606, ++ 6, 1, 0.028183829, 0.546665847, 0.150000006, 3.70133615, ++ 7, 1, 0.00891250931, 0.52986443, 0.150000006, 4.28118753, ++ 8, 1, 0.00891250931, 0.511183441, 0.150000006, 4.84536505, ++ 9, 1, 0.00891250931, 0.490902334, 0.150000006, 5.39263105, ++ 10, 1, 0.00891250931, 0.46938166, 0.150000006, 5.92205667, ++ 11, 1, 0.00891250931, 0.447003782, 0.150000006, 6.43299866, ++ 12, 1, 0.00891250931, 0.428170592, 0.150000006, 6.92507982, ++ 13, 1, 0.00891250931, 0.414536625, 0.150000006, 7.39815664, ++ 14, 1, 0.00891250931, 0.401033074, 0.150000006, 7.85228777, ++ 15, 1, 0.00891250931, 0.38779071, 0.150000006, 8.28770351, ++ 16, 1, 0.00891250931, 0.374230444, 0.150000006, 8.704772, ++ 17, 1, 0.00891250931, 0.360547513, 0.180000007, 9.10397339, ++ 18, 1, 0.00891250931, 0.348256677, 0.180000007, 9.48587132, ++ 19, 1, 0.00891250931, 0.350327015, 0.180000007, 9.85109234, ++ 20, 1, 0.00891250931, 0.406330824, 0.180000007, 10.200304, ++ 21, 2, 0.0178250186, 0.554098248, 0.180000007, 10.846529, ++ 22, 2, 0.0178250186, 0.528312504, 0.180000007, 11.4447651, ++ 23, 2, 0.0178250186, 0.476527005, 0.180000007, 11.9928398, ++ 24, 2, 0.0178250186, 0.428205669, 0.180000007, 12.495945, ++ 25, 2, 0.0178250186, 0.402271926, 0.180000007, 12.9588718, ++ 26, 2, 0.0178250186, 0.378024429, 0.180000007, 13.3859692, ++ 27, 2, 0.0178250186, 0.36254698, 0.180000007, 13.7811394, ++ 28, 2, 0.0178250186, 0.368058592, 0.200000003, 14.1478529, ++ 29, 3, 0.0267375279, 0.385963261, 0.200000003, 14.5674343, ++ 30, 3, 0.0267375279, 0.38640517, 0.200000003, 15.0304852, ++ 31, 3, 0.0267375279, 0.367834061, 0.200000003, 15.4513416, ++ 32, 3, 0.0267375279, 0.349686563, 0.200000003, 15.836277, ++ 33, 3, 0.0267375279, 0.345709383, 0.200000003, 16.1904697, ++ 34, 4, 0.0356500372, 0.34871915, 0.200000003, 16.5683517, ++ 35, 4, 0.0356500372, 0.347054332, 0.200000003, 16.9660263, ++ 36, 4, 0.0356500372, 0.346329987, 0.200000003, 17.3304482, ++ 37, 5, 0.0445625484, 0.344658494, 0.200000003, 17.7055588, ++ 38, 5, 0.0445625484, 0.338779271, 0.200000003, 18.0899811, ++ 39, 5, 0.0445625484, 0.334878683, 0.200000003, 18.4440536, ++ 40, 6, 0.0534750558, 0.332811534, 0.200000003, 18.8030052, ++ 41, 6, 0.0534750558, 0.333717585, 0.25, 19.1665268, ++ 42, 7, 0.062387567, 0.333986402, 0.25, 19.5299358, ++ 43, 7, 0.062387567, 0.334142625, 0.25, 19.8934898, ++ 44, 8, 0.142262354, 0.34677428, 0.280000001, 20.2535706, ++ 45, 8, 0.142262354, 0.436254472, 0.280000001, 20.610569, ++ 48000.0, 20, ++ 0, 3, 0, 4, 1.000, 0.056, ++ 1, 3, 4, 7, 0.944, 0.611, ++ 2, 4, 7, 11, 0.389, 0.167, ++ 3, 3, 11, 14, 0.833, 0.722, ++ 4, 3, 14, 17, 0.278, 0.639, ++ 5, 2, 17, 19, 0.361, 0.417, ++ 6, 3, 19, 22, 0.583, 0.083, ++ 7, 2, 22, 24, 0.917, 0.750, ++ 8, 3, 24, 27, 0.250, 0.417, ++ 9, 3, 27, 30, 0.583, 0.648, ++ 10, 3, 30, 33, 0.352, 0.611, ++ 11, 3, 33, 36, 0.389, 0.625, ++ 12, 4, 36, 40, 0.375, 0.144, ++ 13, 3, 40, 43, 0.856, 0.389, ++ 14, 3, 43, 46, 0.611, 0.160, ++ 15, 3, 46, 49, 0.840, 0.217, ++ 16, 3, 49, 52, 0.783, 0.184, ++ 17, 2, 52, 54, 0.816, 0.886, ++ 18, 3, 54, 57, 0.114, 0.313, ++ 19, 2, 57, 59, 0.687, 0.452, ++ 20, 1, 59, 60, 0.548, 0.908, ++ 44100.0, 20, ++ 0, 3, 0, 4, 1.000, 0.056, ++ 1, 3, 4, 7, 0.944, 0.611, ++ 2, 4, 7, 11, 0.389, 0.167, ++ 3, 3, 11, 14, 0.833, 0.722, ++ 4, 3, 14, 17, 0.278, 0.139, ++ 5, 1, 17, 18, 0.861, 0.917, ++ 6, 3, 18, 21, 0.083, 0.583, ++ 7, 3, 21, 24, 0.417, 0.250, ++ 8, 3, 24, 27, 0.750, 0.805, ++ 9, 3, 27, 30, 0.194, 0.574, ++ 10, 3, 30, 33, 0.426, 0.537, ++ 11, 3, 33, 36, 0.463, 0.819, ++ 12, 4, 36, 40, 0.180, 0.100, ++ 13, 3, 40, 43, 0.900, 0.468, ++ 14, 3, 43, 46, 0.532, 0.623, ++ 15, 3, 46, 49, 0.376, 0.450, ++ 16, 3, 49, 52, 0.550, 0.552, ++ 17, 3, 52, 55, 0.448, 0.403, ++ 18, 2, 55, 57, 0.597, 0.643, ++ 19, 2, 57, 59, 0.357, 0.722, ++ 20, 2, 59, 61, 0.278, 0.960, ++ 32000, 20, ++ 0, 1, 0, 2, 1.000, 0.528, ++ 1, 2, 2, 4, 0.472, 0.305, ++ 2, 2, 4, 6, 0.694, 0.083, ++ 3, 1, 6, 7, 0.917, 0.861, ++ 4, 2, 7, 9, 0.139, 0.639, ++ 5, 2, 9, 11, 0.361, 0.417, ++ 6, 3, 11, 14, 0.583, 0.083, ++ 7, 2, 14, 16, 0.917, 0.750, ++ 8, 3, 16, 19, 0.250, 0.870, ++ 9, 3, 19, 22, 0.130, 0.833, ++ 10, 4, 22, 26, 0.167, 0.389, ++ 11, 4, 26, 30, 0.611, 0.478, ++ 12, 4, 30, 34, 0.522, 0.033, ++ 13, 3, 34, 37, 0.967, 0.917, ++ 14, 4, 37, 41, 0.083, 0.617, ++ 15, 3, 41, 44, 0.383, 0.995, ++ 16, 4, 44, 48, 0.005, 0.274, ++ 17, 3, 48, 51, 0.726, 0.480, ++ 18, 3, 51, 54, 0.519, 0.261, ++ 19, 2, 54, 56, 0.739, 0.884, ++ 20, 2, 56, 58, 0.116, 1.000, ++ 24000, 20, ++ 0, 2, 0, 3, 1, 0.916666746, ++ 1, 3, 3, 6, 0.0833332539, 0.583333492, ++ 2, 3, 6, 9, 0.416666508, 0.25, ++ 3, 2, 9, 11, 0.75, 0.916666985, ++ 4, 3, 11, 14, 0.0833330154, 0.583333969, ++ 5, 3, 14, 17, 0.416666031, 0.25, ++ 6, 3, 17, 20, 0.75, 0.537036896, ++ 7, 3, 20, 23, 0.462963104, 0.5, ++ 8, 4, 23, 27, 0.5, 0.0555559993, ++ 9, 3, 27, 30, 0.944444001, 0.402778625, ++ 10, 3, 30, 33, 0.597221375, 0.766667187, ++ 11, 3, 33, 36, 0.233332828, 0.805555999, ++ 12, 3, 36, 39, 0.194444016, 0.769841909, ++ 13, 3, 39, 42, 0.23015812, 0.611111104, ++ 14, 3, 42, 45, 0.388888896, 0.449494779, ++ 15, 3, 45, 48, 0.550505221, 0.194444954, ++ 16, 2, 48, 50, 0.805555046, 0.913194656, ++ 17, 3, 50, 53, 0.0868053436, 0.580555737, ++ 18, 3, 53, 56, 0.419444263, 0.113426208, ++ 19, 2, 56, 58, 0.886573792, 0.533730626, ++ 20, 2, 58, 60, 0.466269344, 0.691176474, ++ 22050, 20, ++ 0, 2, 0, 3, 1, 0.916666746, ++ 1, 3, 3, 6, 0.0833332539, 0.583333492, ++ 2, 3, 6, 9, 0.416666508, 0.25, ++ 3, 2, 9, 11, 0.75, 0.916666985, ++ 4, 3, 11, 14, 0.0833330154, 0.583333969, ++ 5, 3, 14, 17, 0.416666031, 0.25, ++ 6, 3, 17, 20, 0.75, 0.203703582, ++ 7, 3, 20, 23, 0.796296418, 0.166666687, ++ 8, 3, 23, 26, 0.833333313, 0.722222686, ++ 9, 4, 26, 30, 0.277777344, 0.152778625, ++ 10, 3, 30, 33, 0.847221375, 0.566667199, ++ 11, 3, 33, 36, 0.433332831, 0.93518573, ++ 12, 4, 36, 40, 0.0648142472, 0.118056297, ++ 13, 3, 40, 43, 0.881943703, 0.0925937295, ++ 14, 2, 43, 45, 0.907406271, 0.934344172, ++ 15, 3, 45, 48, 0.0656557977, 0.575398028, ++ 16, 3, 48, 51, 0.424601972, 0.232026935, ++ 17, 2, 51, 53, 0.767973065, 0.758334339, ++ 18, 3, 53, 56, 0.241665646, 0.187501252, ++ 19, 2, 56, 58, 0.812498748, 0.533731699, ++ 20, 2, 58, 60, 0.466268271, 0.257577598, ++ 16000, 20, ++ 0, 1, 0, 2, 1, 0.944444478, ++ 1, 2, 2, 4, 0.0555555038, 0.722222328, ++ 2, 2, 4, 6, 0.277777672, 0.5, ++ 3, 2, 6, 8, 0.5, 0.27777797, ++ 4, 2, 8, 10, 0.72222203, 0.0555559993, ++ 5, 1, 10, 11, 0.944444001, 0.833333313, ++ 6, 3, 11, 14, 0.166666672, 0.203703582, ++ 7, 3, 14, 17, 0.796296418, 0.166666687, ++ 8, 3, 17, 20, 0.833333313, 0.54166698, ++ 9, 3, 20, 23, 0.458333015, 0.652778625, ++ 10, 4, 23, 27, 0.347221375, 0.166667163, ++ 11, 3, 27, 30, 0.833332837, 0.722222924, ++ 12, 4, 30, 34, 0.277777106, 0.277778625, ++ 13, 3, 34, 37, 0.722221375, 0.604167938, ++ 14, 3, 37, 40, 0.395832062, 0.627778649, ++ 15, 3, 40, 43, 0.37222138, 0.542736351, ++ 16, 3, 43, 46, 0.457263649, 0.371528625, ++ 17, 3, 46, 49, 0.628471375, 0.00833433867, ++ 18, 2, 49, 51, 0.991665661, 0.500001311, ++ 19, 2, 51, 53, 0.499998659, 0.886832893, ++ 20, 2, 53, 55, 0.113167092, 0.629034221, ++ 48000.0, 11, ++ 0, 2, 0, 2, 1.000, 0.167, ++ 1, 2, 3, 5, 0.833, 0.833, ++ 2, 3, 5, 8, 0.167, 0.500, ++ 3, 3, 8, 11, 0.500, 0.167, ++ 4, 4, 11, 15, 0.833, 0.167, ++ 5, 4, 15, 19, 0.833, 0.583, ++ 6, 3, 19, 22, 0.417, 0.917, ++ 7, 4, 22, 26, 0.083, 0.944, ++ 8, 4, 26, 30, 0.055, 0.042, ++ 9, 2, 30, 32, 0.958, 0.567, ++ 10, 3, 32, 35, 0.433, 0.167, ++ 11, 2, 35, 37, 0.833, 0.618, ++ 44100.0, 11, ++ 0, 2, 0, 2, 1.000, 0.167, ++ 1, 2, 3, 5, 0.833, 0.833, ++ 2, 3, 5, 8, 0.167, 0.500, ++ 3, 3, 8, 11, 0.500, 0.167, ++ 4, 4, 11, 15, 0.833, 0.167, ++ 5, 5, 15, 20, 0.833, 0.250, ++ 6, 3, 20, 23, 0.750, 0.583, ++ 7, 4, 23, 27, 0.417, 0.055, ++ 8, 3, 27, 30, 0.944, 0.375, ++ 9, 3, 30, 33, 0.625, 0.300, ++ 10, 3, 33, 36, 0.700, 0.167, ++ 11, 2, 36, 38, 0.833, 1.000, ++ 32000, 11, ++ 0, 2, 0, 2, 1.000, 0.167, ++ 1, 2, 3, 5, 0.833, 0.833, ++ 2, 3, 5, 8, 0.167, 0.500, ++ 3, 3, 8, 11, 0.500, 0.167, ++ 4, 4, 11, 15, 0.833, 0.167, ++ 5, 5, 15, 20, 0.833, 0.250, ++ 6, 4, 20, 24, 0.750, 0.250, ++ 7, 5, 24, 29, 0.750, 0.055, ++ 8, 4, 29, 33, 0.944, 0.375, ++ 9, 4, 33, 37, 0.625, 0.472, ++ 10, 3, 37, 40, 0.528, 0.937, ++ 11, 1, 40, 41, 0.062, 1.000, ++ 24000, 11, ++ 0, 3, 0, 4, 1, 0.166666746, ++ 1, 2, 4, 6, 0.833333254, 0.833333492, ++ 2, 3, 6, 9, 0.166666508, 0.5, ++ 3, 4, 9, 13, 0.5, 0.5, ++ 4, 5, 13, 18, 0.5, 0.833333969, ++ 5, 5, 18, 23, 0.166666031, 0.25, ++ 6, 4, 23, 27, 0.75, 0.25, ++ 7, 3, 27, 30, 0.75, 0.611111999, ++ 8, 4, 30, 34, 0.388888031, 0.208333969, ++ 9, 3, 34, 37, 0.791666031, 0.766667187, ++ 10, 4, 37, 41, 0.233332828, 0.45238167, ++ 11, 4, 41, 45, 0.54761833, 0.277778625, ++ 22050, 11, ++ 0, 3, 0, 4, 1, 0.166666746, ++ 1, 2, 4, 6, 0.833333254, 0.833333492, ++ 2, 3, 6, 9, 0.166666508, 0.5, ++ 3, 4, 9, 13, 0.5, 0.5, ++ 4, 4, 13, 17, 0.5, 0.5, ++ 5, 4, 17, 21, 0.5, 0.916666985, ++ 6, 4, 21, 25, 0.0833330154, 0.25, ++ 7, 4, 25, 29, 0.75, 0.611111999, ++ 8, 4, 29, 33, 0.388888031, 0.458333969, ++ 9, 4, 33, 37, 0.541666031, 0.633334339, ++ 10, 4, 37, 41, 0.366665661, 0.583334565, ++ 11, 4, 41, 45, 0.416665405, 0.437500954, ++ 16000, 11, ++ 0, 3, 0, 4, 1, 0.166666746, ++ 1, 2, 4, 6, 0.833333254, 0.833333492, ++ 2, 3, 6, 9, 0.166666508, 0.5, ++ 3, 4, 9, 13, 0.5, 0.5, ++ 4, 5, 13, 18, 0.5, 0.833333969, ++ 5, 5, 18, 23, 0.166666031, 0.75, ++ 6, 4, 23, 27, 0.25, 0.75, ++ 7, 4, 27, 31, 0.25, 0.611111999, ++ 8, 4, 31, 35, 0.388888031, 0.458333969, ++ 9, 4, 35, 39, 0.541666031, 0.166667163, ++ 10, 3, 39, 42, 0.833332837, 0.805555999, ++ 11, 4, 42, 46, 0.194444016, 0.4375}; ++ ++ double enwindow[] = ++ {0.000000000, -0.000000477, -0.000000477, -0.000000477, ++ -0.000000477, -0.000000477, -0.000000477, -0.000000954, ++ -0.000000954, -0.000000954, -0.000000954, -0.000001431, ++ -0.000001431, -0.000001907, -0.000001907, -0.000002384, ++ -0.000002384, -0.000002861, -0.000003338, -0.000003338, ++ -0.000003815, -0.000004292, -0.000004768, -0.000005245, ++ -0.000006199, -0.000006676, -0.000007629, -0.000008106, ++ -0.000009060, -0.000010014, -0.000011444, -0.000012398, ++ -0.000013828, -0.000014782, -0.000016689, -0.000018120, ++ -0.000019550, -0.000021458, -0.000023365, -0.000025272, ++ -0.000027657, -0.000030041, -0.000032425, -0.000034809, ++ -0.000037670, -0.000040531, -0.000043392, -0.000046253, ++ -0.000049591, -0.000052929, -0.000055790, -0.000059605, ++ -0.000062943, -0.000066280, -0.000070095, -0.000073433, ++ -0.000076771, -0.000080585, -0.000083923, -0.000087261, ++ -0.000090599, -0.000093460, -0.000096321, -0.000099182, ++ 0.000101566, 0.000103951, 0.000105858, 0.000107288, ++ 0.000108242, 0.000108719, 0.000108719, 0.000108242, ++ 0.000106812, 0.000105381, 0.000102520, 0.000099182, ++ 0.000095367, 0.000090122, 0.000084400, 0.000077724, ++ 0.000069618, 0.000060558, 0.000050545, 0.000039577, ++ 0.000027180, 0.000013828, -0.000000954, -0.000017166, ++ -0.000034332, -0.000052929, -0.000072956, -0.000093937, ++ -0.000116348, -0.000140190, -0.000165462, -0.000191212, ++ -0.000218868, -0.000247478, -0.000277042, -0.000307560, ++ -0.000339031, -0.000371456, -0.000404358, -0.000438213, ++ -0.000472546, -0.000507355, -0.000542164, -0.000576973, ++ -0.000611782, -0.000646591, -0.000680923, -0.000714302, ++ -0.000747204, -0.000779152, -0.000809669, -0.000838757, ++ -0.000866413, -0.000891685, -0.000915051, -0.000935555, ++ -0.000954151, -0.000968933, -0.000980854, -0.000989437, ++ -0.000994205, -0.000995159, -0.000991821, -0.000983715, ++ 0.000971317, 0.000953674, 0.000930786, 0.000902653, ++ 0.000868797, 0.000829220, 0.000783920, 0.000731945, ++ 0.000674248, 0.000610352, 0.000539303, 0.000462532, ++ 0.000378609, 0.000288486, 0.000191689, 0.000088215, ++ -0.000021458, -0.000137329, -0.000259876, -0.000388145, ++ -0.000522137, -0.000661850, -0.000806808, -0.000956535, ++ -0.001111031, -0.001269817, -0.001432419, -0.001597881, ++ -0.001766682, -0.001937389, -0.002110004, -0.002283096, ++ -0.002457142, -0.002630711, -0.002803326, -0.002974033, ++ -0.003141880, -0.003306866, -0.003467083, -0.003622532, ++ -0.003771782, -0.003914356, -0.004048824, -0.004174709, ++ -0.004290581, -0.004395962, -0.004489899, -0.004570484, ++ -0.004638195, -0.004691124, -0.004728317, -0.004748821, ++ -0.004752159, -0.004737377, -0.004703045, -0.004649162, ++ -0.004573822, -0.004477024, -0.004357815, -0.004215240, ++ -0.004049301, -0.003858566, -0.003643036, -0.003401756, ++ 0.003134727, 0.002841473, 0.002521515, 0.002174854, ++ 0.001800537, 0.001399517, 0.000971317, 0.000515938, ++ 0.000033379, -0.000475883, -0.001011848, -0.001573563, ++ -0.002161503, -0.002774239, -0.003411293, -0.004072189, ++ -0.004756451, -0.005462170, -0.006189346, -0.006937027, ++ -0.007703304, -0.008487225, -0.009287834, -0.010103703, ++ -0.010933399, -0.011775017, -0.012627602, -0.013489246, ++ -0.014358521, -0.015233517, -0.016112804, -0.016994476, ++ -0.017876148, -0.018756866, -0.019634247, -0.020506859, ++ -0.021372318, -0.022228718, -0.023074150, -0.023907185, ++ -0.024725437, -0.025527000, -0.026310921, -0.027073860, ++ -0.027815342, -0.028532982, -0.029224873, -0.029890060, ++ -0.030526638, -0.031132698, -0.031706810, -0.032248020, ++ -0.032754898, -0.033225536, -0.033659935, -0.034055710, ++ -0.034412861, -0.034730434, -0.035007000, -0.035242081, ++ -0.035435200, -0.035586357, -0.035694122, -0.035758972, ++ 0.035780907, 0.035758972, 0.035694122, 0.035586357, ++ 0.035435200, 0.035242081, 0.035007000, 0.034730434, ++ 0.034412861, 0.034055710, 0.033659935, 0.033225536, ++ 0.032754898, 0.032248020, 0.031706810, 0.031132698, ++ 0.030526638, 0.029890060, 0.029224873, 0.028532982, ++ 0.027815342, 0.027073860, 0.026310921, 0.025527000, ++ 0.024725437, 0.023907185, 0.023074150, 0.022228718, ++ 0.021372318, 0.020506859, 0.019634247, 0.018756866, ++ 0.017876148, 0.016994476, 0.016112804, 0.015233517, ++ 0.014358521, 0.013489246, 0.012627602, 0.011775017, ++ 0.010933399, 0.010103703, 0.009287834, 0.008487225, ++ 0.007703304, 0.006937027, 0.006189346, 0.005462170, ++ 0.004756451, 0.004072189, 0.003411293, 0.002774239, ++ 0.002161503, 0.001573563, 0.001011848, 0.000475883, ++ -0.000033379, -0.000515938, -0.000971317, -0.001399517, ++ -0.001800537, -0.002174854, -0.002521515, -0.002841473, ++ 0.003134727, 0.003401756, 0.003643036, 0.003858566, ++ 0.004049301, 0.004215240, 0.004357815, 0.004477024, ++ 0.004573822, 0.004649162, 0.004703045, 0.004737377, ++ 0.004752159, 0.004748821, 0.004728317, 0.004691124, ++ 0.004638195, 0.004570484, 0.004489899, 0.004395962, ++ 0.004290581, 0.004174709, 0.004048824, 0.003914356, ++ 0.003771782, 0.003622532, 0.003467083, 0.003306866, ++ 0.003141880, 0.002974033, 0.002803326, 0.002630711, ++ 0.002457142, 0.002283096, 0.002110004, 0.001937389, ++ 0.001766682, 0.001597881, 0.001432419, 0.001269817, ++ 0.001111031, 0.000956535, 0.000806808, 0.000661850, ++ 0.000522137, 0.000388145, 0.000259876, 0.000137329, ++ 0.000021458, -0.000088215, -0.000191689, -0.000288486, ++ -0.000378609, -0.000462532, -0.000539303, -0.000610352, ++ -0.000674248, -0.000731945, -0.000783920, -0.000829220, ++ -0.000868797, -0.000902653, -0.000930786, -0.000953674, ++ 0.000971317, 0.000983715, 0.000991821, 0.000995159, ++ 0.000994205, 0.000989437, 0.000980854, 0.000968933, ++ 0.000954151, 0.000935555, 0.000915051, 0.000891685, ++ 0.000866413, 0.000838757, 0.000809669, 0.000779152, ++ 0.000747204, 0.000714302, 0.000680923, 0.000646591, ++ 0.000611782, 0.000576973, 0.000542164, 0.000507355, ++ 0.000472546, 0.000438213, 0.000404358, 0.000371456, ++ 0.000339031, 0.000307560, 0.000277042, 0.000247478, ++ 0.000218868, 0.000191212, 0.000165462, 0.000140190, ++ 0.000116348, 0.000093937, 0.000072956, 0.000052929, ++ 0.000034332, 0.000017166, 0.000000954, -0.000013828, ++ -0.000027180, -0.000039577, -0.000050545, -0.000060558, ++ -0.000069618, -0.000077724, -0.000084400, -0.000090122, ++ -0.000095367, -0.000099182, -0.000102520, -0.000105381, ++ -0.000106812, -0.000108242, -0.000108719, -0.000108719, ++ -0.000108242, -0.000107288, -0.000105858, -0.000103951, ++ 0.000101566, 0.000099182, 0.000096321, 0.000093460, ++ 0.000090599, 0.000087261, 0.000083923, 0.000080585, ++ 0.000076771, 0.000073433, 0.000070095, 0.000066280, ++ 0.000062943, 0.000059605, 0.000055790, 0.000052929, ++ 0.000049591, 0.000046253, 0.000043392, 0.000040531, ++ 0.000037670, 0.000034809, 0.000032425, 0.000030041, ++ 0.000027657, 0.000025272, 0.000023365, 0.000021458, ++ 0.000019550, 0.000018120, 0.000016689, 0.000014782, ++ 0.000013828, 0.000012398, 0.000011444, 0.000010014, ++ 0.000009060, 0.000008106, 0.000007629, 0.000006676, ++ 0.000006199, 0.000005245, 0.000004768, 0.000004292, ++ 0.000003815, 0.000003338, 0.000003338, 0.000002861, ++ 0.000002384, 0.000002384, 0.000001907, 0.000001907, ++ 0.000001431, 0.000001431, 0.000000954, 0.000000954, ++ 0.000000954, 0.000000954, 0.000000477, 0.000000477, ++ 0.000000477, 0.000000477, 0.000000477, 0.000000477}; ++ +diff -r -c -N encoder/tonal.c lame3.13/tonal.c +*** encoder/tonal.c Wed Jan 22 02:43:23 1997 +--- lame3.13/tonal.c Wed Dec 31 17:00:00 1969 +*************** +*** 1,1051 **** +- /********************************************************************** +- * ISO MPEG Audio Subgroup Software Simulation Group (1996) +- * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- * +- * $Id: tonal.c,v 1.1 1996/02/14 04:04:23 rowlands Exp $ +- * +- * $Log: tonal.c,v $ +- * Revision 1.1 1996/02/14 04:04:23 rowlands +- * Initial revision +- * +- * Received from Mike Coleman +- **********************************************************************/ +- /********************************************************************** +- * date programmers comment * +- * 2/25/91 Douglas Wong start of version 1.1 records * +- * 3/06/91 Douglas Wong rename: setup.h to endef.h * +- * updated I_psycho_one and II_psycho_one* +- * 3/11/91 W. J. Carter Added Douglas Wong's updates dated * +- * 3/9/91 for I_Psycho_One() and for * +- * II_Psycho_One(). * +- * 5/10/91 W. Joseph Carter Ported to Macintosh and Unix. * +- * Located and fixed numerous software * +- * bugs and table data errors. * +- * 6/11/91 Davis Pan corrected several bugs * +- * based on comments from H. Fuchs * +- * 01jul91 dpwe (Aware Inc.) Made pow() args float * +- * Removed logical bug in I_tonal_label: * +- * Sometimes *tone returned == STOP * +- * 7/10/91 Earle Jennings no change necessary in port to MsDos * +- * 11sep91 dpwe@aware.com Subtracted 90.3dB from II_f_f_t peaks * +- * 10/1/91 Peter W. Farrett Updated II_Psycho_One(),I_Psycho_One()* +- * to include comments. * +- *11/29/91 Masahiro Iwadare Bug fix regarding POWERNORM * +- * fixed several other miscellaneous bugs* +- * 2/11/92 W. Joseph Carter Ported new code to Macintosh. Most * +- * important fixes involved changing * +- * 16-bit ints to long or unsigned in * +- * bit alloc routines for quant of 65535 * +- * and passing proper function args. * +- * Removed "Other Joint Stereo" option * +- * and made bitrate be total channel * +- * bitrate, irrespective of the mode. * +- * Fixed many small bugs & reorganized. * +- * 2/12/92 Masahiro Iwadare Fixed some potential bugs in * +- * Davis Pan subsampling() * +- * 2/25/92 Masahiro Iwadare Fixed some more potential bugs * +- * 6/24/92 Tan Ah Peng Modified window for FFT * +- * (denominator N-1 to N) * +- * Updated all critical band rate & * +- * absolute threshold tables and critical* +- * boundaries for use with Layer I & II * +- * Corrected boundary limits for tonal * +- * component computation * +- * Placement of non-tonal component at * +- * geometric mean of critical band * +- * (previous placement method commented * +- * out - can be used if desired) * +- * 3/01/93 Mike Li Infinite looping fix in noise_label() * +- * 3/19/93 Jens Spille fixed integer overflow problem in * +- * psychoacoutic model 1 * +- * 3/19/93 Giorgio Dimino modifications to better account for * +- * tonal and non-tonal components * +- * 5/28/93 Sriram Jayasimha "London" mod. to psychoacoustic model1* +- * 8/05/93 Masahiro Iwadare noise_label modification "option" * +- * 1/21/94 Seymore Shlien fixed another infinite looping problem* +- * 7/12/95 Soeren H. Nielsen Changes for LSF, new tables * +- **********************************************************************/ +- +- #include "common.h" +- #include "encoder.h" +- #define LONDON /* enable "LONDON" modification */ +- #define MAKE_SENSE /* enable "MAKE_SENSE" modification */ +- #define MI_OPTION /* enable "MI_OPTION" modification */ +- /********************************************************************** +- * +- * This module implements the psychoacoustic model I for the +- * MPEG encoder layer II. It uses simplified tonal and noise masking +- * threshold analysis to generate SMR for the encoder bit allocation +- * routine. +- * +- **********************************************************************/ +- +- int crit_band; +- int FAR *cbound; +- int sub_size; +- +- void read_cbound(lay,freq) /* this function reads in critical */ +- int lay, freq; /* band boundaries */ +- { +- int i,j,k; +- FILE *fp; +- char r[16], t[80]; +- +- strcpy(r, "2cb1"); +- r[0] = (char) lay + '0'; +- r[3] = (char) freq + '0'; +- if( !(fp = OpenTableFile(r)) ){ /* check boundary values */ +- printf("Please check %s boundary table\n",r); +- exit(1); +- } +- fgets(t,80,fp); /* read input for critical bands */ +- sscanf(t,"%d\n",&crit_band); +- cbound = (int FAR *) mem_alloc(sizeof(int) * crit_band, "cbound"); +- for(i=0;i<crit_band;i++){ /* continue to read input for */ +- fgets(t,80,fp); /* critical band boundaries */ +- sscanf(t,"%d %d\n",&j, &k); +- if(i==j) cbound[j] = k; +- else { /* error */ +- printf("Please check index %d in cbound table %s\n",i,r); +- exit(1); +- } +- } +- fclose(fp); +- } +- +- void read_freq_band(ltg,lay,freq) /* this function reads in */ +- int lay, freq; /* frequency bands and bark */ +- g_ptr FAR *ltg; /* values */ +- { +- int i,j, k; +- double b,c; +- FILE *fp; +- char r[16], t[80]; +- +- strcpy(r, "2th1"); +- r[0] = (char) lay + '0'; +- r[3] = (char) freq + '0'; +- if( !(fp = OpenTableFile(r)) ){ /* check freq. values */ +- printf("Please check frequency and cband table %s\n",r); +- exit(1); +- } +- fgets(t,80,fp); /* read input for freq. subbands */ +- sscanf(t,"%d\n",&sub_size); +- *ltg = (g_ptr FAR ) mem_alloc(sizeof(g_thres) * sub_size, "ltg"); +- (*ltg)[0].line = 0; /* initialize global masking threshold */ +- (*ltg)[0].bark = 0; +- (*ltg)[0].hear = 0; +- for(i=1;i<sub_size;i++){ /* continue to read freq. subband */ +- fgets(t,80,fp); /* and assign */ +- sscanf(t,"%d %d %lf %lf\n",&j, &k, &b, &c); +- if(i == j){ +- (*ltg)[j].line = k; +- (*ltg)[j].bark = b; +- (*ltg)[j].hear = c; +- } +- else { /* error */ +- printf("Please check index %d in freq-cb table %s\n",i,r); +- exit(1); +- } +- } +- fclose(fp); +- } +- +- void make_map(power, ltg) /* this function calculates the */ +- mask FAR power[HAN_SIZE]; /* global masking threshold */ +- g_thres FAR *ltg; +- { +- int i,j; +- +- for(i=1;i<sub_size;i++) for(j=ltg[i-1].line;j<=ltg[i].line;j++) +- power[j].map = i; +- } +- +- double add_db(a,b) +- double a,b; +- { +- a = pow(10.0,a/10.0); +- b = pow(10.0,b/10.0); +- return 10 * log10(a+b); +- } +- +- /**************************************************************** +- * +- * Fast Fourier transform of the input samples. +- * +- ****************************************************************/ +- +- void II_f_f_t(sample, power) /* this function calculates an */ +- double FAR sample[FFT_SIZE]; /* FFT analysis for the freq. */ +- mask FAR power[HAN_SIZE]; /* domain */ +- { +- int i,j,k,L,l=0; +- int ip, le, le1; +- double t_r, t_i, u_r, u_i; +- static int M, MM1, init = 0, N; +- double *x_r, *x_i, *energy; +- static int *rev; +- static double *w_r, *w_i; +- +- x_r = (double *) mem_alloc(sizeof(DFFT), "x_r"); +- x_i = (double *) mem_alloc(sizeof(DFFT), "x_i"); +- energy = (double *) mem_alloc(sizeof(DFFT), "energy"); +- for(i=0;i<FFT_SIZE;i++) x_r[i] = x_i[i] = energy[i] = 0; +- if(!init){ +- rev = (int *) mem_alloc(sizeof(IFFT), "rev"); +- w_r = (double *) mem_alloc(sizeof(D10), "w_r"); +- w_i = (double *) mem_alloc(sizeof(D10), "w_i"); +- M = 10; +- MM1 = 9; +- N = FFT_SIZE; +- for(L=0;L<M;L++){ +- le = 1 << (M-L); +- le1 = le >> 1; +- w_r[L] = cos(PI/le1); +- w_i[L] = -sin(PI/le1); +- } +- for(i=0;i<FFT_SIZE;rev[i] = l,i++) for(j=0,l=0;j<10;j++){ +- k=(i>>j) & 1; +- l |= (k<<(9-j)); +- } +- init = 1; +- } +- memcpy( (char *) x_r, (char *) sample, sizeof(double) * FFT_SIZE); +- for(L=0;L<MM1;L++){ +- le = 1 << (M-L); +- le1 = le >> 1; +- u_r = 1; +- u_i = 0; +- for(j=0;j<le1;j++){ +- for(i=j;i<N;i+=le){ +- ip = i + le1; +- t_r = x_r[i] + x_r[ip]; +- t_i = x_i[i] + x_i[ip]; +- x_r[ip] = x_r[i] - x_r[ip]; +- x_i[ip] = x_i[i] - x_i[ip]; +- x_r[i] = t_r; +- x_i[i] = t_i; +- t_r = x_r[ip]; +- x_r[ip] = x_r[ip] * u_r - x_i[ip] * u_i; +- x_i[ip] = x_i[ip] * u_r + t_r * u_i; +- } +- t_r = u_r; +- u_r = u_r * w_r[L] - u_i * w_i[L]; +- u_i = u_i * w_r[L] + t_r * w_i[L]; +- } +- } +- for(i=0;i<N;i+=2){ +- ip = i + 1; +- t_r = x_r[i] + x_r[ip]; +- t_i = x_i[i] + x_i[ip]; +- x_r[ip] = x_r[i] - x_r[ip]; +- x_i[ip] = x_i[i] - x_i[ip]; +- x_r[i] = t_r; +- x_i[i] = t_i; +- energy[i] = x_r[i] * x_r[i] + x_i[i] * x_i[i]; +- } +- for(i=0;i<FFT_SIZE;i++) if(i<rev[i]){ +- t_r = energy[i]; +- energy[i] = energy[rev[i]]; +- energy[rev[i]] = t_r; +- } +- for(i=0;i<HAN_SIZE;i++){ /* calculate power density spectrum */ +- if (energy[i] < 1E-20) energy[i] = 1E-20; +- power[i].x = 10 * log10(energy[i]) + POWERNORM; +- power[i].next = STOP; +- power[i].type = FALSE; +- } +- mem_free((void **) &x_r); +- mem_free((void **) &x_i); +- mem_free((void **) &energy); +- } +- +- /**************************************************************** +- * +- * Window the incoming audio signal. +- * +- ****************************************************************/ +- +- void II_hann_win(sample) /* this function calculates a */ +- double FAR sample[FFT_SIZE]; /* Hann window for PCM (input) */ +- { /* samples for a 1024-pt. FFT */ +- register int i; +- register double sqrt_8_over_3; +- static int init = 0; +- static double FAR *window; +- +- if(!init){ /* calculate window function for the Fourier transform */ +- window = (double FAR *) mem_alloc(sizeof(DFFT), "window"); +- sqrt_8_over_3 = pow(8.0/3.0, 0.5); +- for(i=0;i<FFT_SIZE;i++){ +- /* Hann window formula */ +- window[i]=sqrt_8_over_3*0.5*(1-cos(2.0*PI*i/(FFT_SIZE)))/FFT_SIZE; +- } +- init = 1; +- } +- for(i=0;i<FFT_SIZE;i++) sample[i] *= window[i]; +- } +- +- /******************************************************************* +- * +- * This function finds the maximum spectral component in each +- * subband and return them to the encoder for time-domain threshold +- * determination. +- * +- *******************************************************************/ +- #ifndef LONDON +- void II_pick_max(power, spike) +- double FAR spike[SBLIMIT]; +- mask FAR power[HAN_SIZE]; +- { +- double max; +- int i,j; +- +- for(i=0;i<HAN_SIZE;spike[i>>4] = max, i+=16) /* calculate the */ +- for(j=0, max = DBMIN;j<16;j++) /* maximum spectral */ +- max = (max>power[i+j].x) ? max : power[i+j].x; /* component in each */ +- } /* subband from bound */ +- /* 4-16 */ +- #else +- void II_pick_max(power, spike) +- double FAR spike[SBLIMIT]; +- mask FAR power[HAN_SIZE]; +- { +- double sum; +- int i,j; +- +- for(i=0;i<HAN_SIZE;spike[i>>4] = 10.0*log10(sum), i+=16) +- /* calculate the */ +- for(j=0, sum = pow(10.0,0.1*DBMIN);j<16;j++) /* sum of spectral */ +- sum += pow(10.0,0.1*power[i+j].x); /* component in each */ +- } /* subband from bound */ +- /* 4-16 */ +- #endif +- +- /**************************************************************** +- * +- * This function labels the tonal component in the power +- * spectrum. +- * +- ****************************************************************/ +- +- void II_tonal_label(power, tone) /* this function extracts (tonal) */ +- mask FAR power[HAN_SIZE]; /* sinusoidals from the spectrum */ +- int *tone; +- { +- int i,j, last = LAST, first, run, last_but_one = LAST; /* dpwe */ +- double max; +- +- *tone = LAST; +- for(i=2;i<HAN_SIZE-12;i++){ +- if(power[i].x>power[i-1].x && power[i].x>=power[i+1].x){ +- power[i].type = TONE; +- power[i].next = LAST; +- if(last != LAST) power[last].next = i; +- else first = *tone = i; +- last = i; +- } +- } +- last = LAST; +- first = *tone; +- *tone = LAST; +- while(first != LAST){ /* the conditions for the tonal */ +- if(first<3 || first>500) run = 0;/* otherwise k+/-j will be out of bounds */ +- else if(first<63) run = 2; /* components in layer II, which */ +- else if(first<127) run = 3; /* are the boundaries for calc. */ +- else if(first<255) run = 6; /* the tonal components */ +- else run = 12; +- max = power[first].x - 7; /* after calculation of tonal */ +- for(j=2;j<=run;j++) /* components, set to local max */ +- if(max < power[first-j].x || max < power[first+j].x){ +- power[first].type = FALSE; +- break; +- } +- if(power[first].type == TONE){ /* extract tonal components */ +- int help=first; +- if(*tone==LAST) *tone = first; +- while((power[help].next!=LAST)&&(power[help].next-first)<=run) +- help=power[help].next; +- help=power[help].next; +- power[first].next=help; +- if((first-last)<=run){ +- if(last_but_one != LAST) power[last_but_one].next=first; +- } +- if(first>1 && first<500){ /* calculate the sum of the */ +- double tmp; /* powers of the components */ +- tmp = add_db(power[first-1].x, power[first+1].x); +- power[first].x = add_db(power[first].x, tmp); +- } +- for(j=1;j<=run;j++){ +- power[first-j].x = power[first+j].x = DBMIN; +- power[first-j].next = power[first+j].next = STOP; +- power[first-j].type = power[first+j].type = FALSE; +- } +- last_but_one=last; +- last = first; +- first = power[first].next; +- } +- else { +- int ll; +- if(last == LAST); /* *tone = power[first].next; dpwe */ +- else power[last].next = power[first].next; +- ll = first; +- first = power[first].next; +- power[ll].next = STOP; +- } +- } +- } +- +- /**************************************************************** +- * +- * This function groups all the remaining non-tonal +- * spectral lines into critical band where they are replaced by +- * one single line. +- * +- ****************************************************************/ +- +- void noise_label(power, noise, ltg) +- g_thres FAR *ltg; +- mask FAR *power; +- int *noise; +- { +- int i,j, centre, last = LAST; +- double index, weight, sum; +- /* calculate the remaining spectral */ +- for(i=0;i<crit_band-1;i++){ /* lines for non-tonal components */ +- for(j=cbound[i],weight = 0.0,sum = DBMIN;j<cbound[i+1];j++){ +- if(power[j].type != TONE){ +- if(power[j].x != DBMIN){ +- sum = add_db(power[j].x,sum); +- /* the line below and others under the "MAKE_SENSE" condition are an alternate +- interpretation of "geometric mean". This approach may make more sense but +- it has not been tested with hardware. */ +- #ifdef MAKE_SENSE +- /* weight += pow(10.0, power[j].x/10.0) * (ltg[power[j].map].bark-i); +- bad code [SS] 21-1-93 +- */ +- weight += pow(10.0,power[j].x/10.0) * (double) (j-cbound[i]) / +- (double) (cbound[i+1]-cbound[i]); /* correction */ +- #endif +- power[j].x = DBMIN; +- } +- } /* check to see if the spectral line is low dB, and if */ +- } /* so replace the center of the critical band, which is */ +- /* the center freq. of the noise component */ +- +- #ifdef MAKE_SENSE +- if(sum <= DBMIN) centre = (cbound[i+1]+cbound[i]) /2; +- else { +- index = weight/pow(10.0,sum/10.0); +- centre = cbound[i] + (int) (index * (double) (cbound[i+1]-cbound[i]) ); +- } +- #else +- index = (double)( ((double)cbound[i]) * ((double)(cbound[i+1]-1)) ); +- centre = (int)(pow(index,0.5)+0.5); +- #endif +- +- /* locate next non-tonal component until finished; */ +- /* add to list of non-tonal components */ +- #ifdef MI_OPTION +- /* Masahiro Iwadare's fix for infinite looping problem? */ +- if(power[centre].type == TONE) +- if (power[centre+1].type == TONE) centre++; else centre--; +- #else +- /* Mike Li's fix for infinite looping problem */ +- if(power[centre].type == FALSE) centre++; +- +- if(power[centre].type == NOISE){ +- if(power[centre].x >= ltg[power[i].map].hear){ +- if(sum >= ltg[power[i].map].hear) sum = add_db(power[j].x,sum); +- else +- sum = power[centre].x; +- } +- } +- #endif +- if(last == LAST) *noise = centre; +- else { +- power[centre].next = LAST; +- power[last].next = centre; +- } +- power[centre].x = sum; +- power[centre].type = NOISE; +- last = centre; +- } +- } +- +- /**************************************************************** +- * +- * This function reduces the number of noise and tonal +- * component for further threshold analysis. +- * +- ****************************************************************/ +- +- void subsampling(power, ltg, tone, noise) +- mask FAR power[HAN_SIZE]; +- g_thres FAR *ltg; +- int *tone, *noise; +- { +- int i, old; +- +- i = *tone; old = STOP; /* calculate tonal components for */ +- while(i!=LAST){ /* reduction of spectral lines */ +- if(power[i].x < ltg[power[i].map].hear){ +- power[i].type = FALSE; +- power[i].x = DBMIN; +- if(old == STOP) *tone = power[i].next; +- else power[old].next = power[i].next; +- } +- else old = i; +- i = power[i].next; +- } +- i = *noise; old = STOP; /* calculate non-tonal components for */ +- while(i!=LAST){ /* reduction of spectral lines */ +- if(power[i].x < ltg[power[i].map].hear){ +- power[i].type = FALSE; +- power[i].x = DBMIN; +- if(old == STOP) *noise = power[i].next; +- else power[old].next = power[i].next; +- } +- else old = i; +- i = power[i].next; +- } +- i = *tone; old = STOP; +- while(i != LAST){ /* if more than one */ +- if(power[i].next == LAST)break; /* tonal component */ +- if(ltg[power[power[i].next].map].bark - /* is less than .5 */ +- ltg[power[i].map].bark < 0.5) { /* bark, take the */ +- if(power[power[i].next].x > power[i].x ){/* maximum */ +- if(old == STOP) *tone = power[i].next; +- else power[old].next = power[i].next; +- power[i].type = FALSE; +- power[i].x = DBMIN; +- i = power[i].next; +- } +- else { +- power[power[i].next].type = FALSE; +- power[power[i].next].x = DBMIN; +- power[i].next = power[power[i].next].next; +- old = i; +- } +- } +- else { +- old = i; +- i = power[i].next; +- } +- } +- } +- +- /**************************************************************** +- * +- * This function calculates the individual threshold and +- * sum with the quiet threshold to find the global threshold. +- * +- ****************************************************************/ +- +- void threshold(power, ltg, tone, noise, bit_rate) +- mask FAR power[HAN_SIZE]; +- g_thres FAR *ltg; +- int *tone, *noise, bit_rate; +- { +- int k, t; +- double dz, tmps, vf; +- +- for(k=1;k<sub_size;k++){ +- ltg[k].x = DBMIN; +- t = *tone; /* calculate individual masking threshold for */ +- while(t != LAST){ /* components in order to find the global */ +- if(ltg[k].bark-ltg[power[t].map].bark >= -3.0 && /*threshold (LTG)*/ +- ltg[k].bark-ltg[power[t].map].bark <8.0){ +- dz = ltg[k].bark-ltg[power[t].map].bark; /* distance of bark value*/ +- tmps = -1.525-0.275*ltg[power[t].map].bark - 4.5 + power[t].x; +- /* masking function for lower & upper slopes */ +- if(-3<=dz && dz<-1) vf = 17*(dz+1)-(0.4*power[t].x +6); +- else if(-1<=dz && dz<0) vf = (0.4 *power[t].x + 6) * dz; +- else if(0<=dz && dz<1) vf = (-17*dz); +- else if(1<=dz && dz<8) vf = -(dz-1) * (17-0.15 *power[t].x) - 17; +- tmps += vf; +- ltg[k].x = add_db(ltg[k].x, tmps); +- } +- t = power[t].next; +- } +- +- t = *noise; /* calculate individual masking threshold */ +- while(t != LAST){ /* for non-tonal components to find LTG */ +- if(ltg[k].bark-ltg[power[t].map].bark >= -3.0 && +- ltg[k].bark-ltg[power[t].map].bark <8.0){ +- dz = ltg[k].bark-ltg[power[t].map].bark; /* distance of bark value */ +- tmps = -1.525-0.175*ltg[power[t].map].bark -0.5 + power[t].x; +- /* masking function for lower & upper slopes */ +- if(-3<=dz && dz<-1) vf = 17*(dz+1)-(0.4*power[t].x +6); +- else if(-1<=dz && dz<0) vf = (0.4 *power[t].x + 6) * dz; +- else if(0<=dz && dz<1) vf = (-17*dz); +- else if(1<=dz && dz<8) vf = -(dz-1) * (17-0.15 *power[t].x) - 17; +- tmps += vf; +- ltg[k].x = add_db(ltg[k].x, tmps); +- } +- t = power[t].next; +- } +- if(bit_rate<96)ltg[k].x = add_db(ltg[k].hear, ltg[k].x); +- else ltg[k].x = add_db(ltg[k].hear-12.0, ltg[k].x); +- } +- } +- +- /**************************************************************** +- * +- * This function finds the minimum masking threshold and +- * return the value to the encoder. +- * +- ****************************************************************/ +- +- void II_minimum_mask(ltg,ltmin,sblimit) +- g_thres FAR *ltg; +- double FAR ltmin[SBLIMIT]; +- int sblimit; +- { +- double min; +- int i,j; +- +- j=1; +- for(i=0;i<sblimit;i++) +- if(j>=sub_size-1) /* check subband limit, and */ +- ltmin[i] = ltg[sub_size-1].hear; /* calculate the minimum masking */ +- else { /* level of LTMIN for each subband*/ +- min = ltg[j].x; +- while(ltg[j].line>>4 == i && j < sub_size){ +- if(min>ltg[j].x) min = ltg[j].x; +- j++; +- } +- ltmin[i] = min; +- } +- } +- +- /***************************************************************** +- * +- * This procedure is called in musicin to pick out the +- * smaller of the scalefactor or threshold. +- * +- *****************************************************************/ +- +- void II_smr(ltmin, spike, scale, sblimit) +- double FAR spike[SBLIMIT], scale[SBLIMIT], ltmin[SBLIMIT]; +- int sblimit; +- { +- int i; +- double max; +- +- for(i=0;i<sblimit;i++){ /* determine the signal */ +- max = 20 * log10(scale[i] * 32768) - 10; /* level for each subband */ +- if(spike[i]>max) max = spike[i]; /* for the maximum scale */ +- max -= ltmin[i]; /* factors */ +- ltmin[i] = max; +- } +- } +- +- /**************************************************************** +- * +- * This procedure calls all the necessary functions to +- * complete the psychoacoustic analysis. +- * +- ****************************************************************/ +- +- void II_Psycho_One(buffer, scale, ltmin, fr_ps) +- short FAR buffer[2][1152]; +- double FAR scale[2][SBLIMIT], ltmin[2][SBLIMIT]; +- frame_params *fr_ps; +- { +- layer *info = fr_ps->header; +- int stereo = fr_ps->stereo; +- int sblimit = fr_ps->sblimit; +- int k,i, tone=0, noise=0; +- static char init = 0; +- static int off[2] = {256,256}; +- double *sample; +- DSBL *spike; +- static D1408 *fft_buf; +- static mask_ptr FAR power; +- static g_ptr FAR ltg; +- +- sample = (double *) mem_alloc(sizeof(DFFT), "sample"); +- spike = (DSBL *) mem_alloc(sizeof(D2SBL), "spike"); +- /* call functions for critical boundaries, freq. */ +- if(!init){ /* bands, bark values, and mapping */ +- fft_buf = (D1408 *) mem_alloc((long) sizeof(D1408) * 2, "fft_buf"); +- power = (mask_ptr FAR ) mem_alloc(sizeof(mask) * HAN_SIZE, "power"); +- if (info->version == MPEG_AUDIO_ID) { +- read_cbound(info->lay, info->sampling_frequency); +- read_freq_band(<g, info->lay, info->sampling_frequency); +- } else { +- read_cbound(info->lay, info->sampling_frequency + 4); +- read_freq_band(<g, info->lay, info->sampling_frequency + 4); +- } +- make_map(power,ltg); +- for (i=0;i<1408;i++) fft_buf[0][i] = fft_buf[1][i] = 0; +- init = 1; +- } +- for(k=0;k<stereo;k++){ /* check pcm input for 3 blocks of 384 samples */ +- for(i=0;i<1152;i++) fft_buf[k][(i+off[k])%1408]= (double)buffer[k][i]/SCALE; +- for(i=0;i<FFT_SIZE;i++) sample[i] = fft_buf[k][(i+1216+off[k])%1408]; +- off[k] += 1152; +- off[k] %= 1408; +- /* call functions for windowing PCM samples,*/ +- II_hann_win(sample); /* location of spectral components in each */ +- for(i=0;i<HAN_SIZE;i++) power[i].x = DBMIN; /*subband with labeling*/ +- II_f_f_t(sample, power); /*locate remaining non-*/ +- II_pick_max(power, &spike[k][0]); /*tonal sinusoidals, */ +- II_tonal_label(power, &tone); /*reduce noise & tonal */ +- noise_label(power, &noise, ltg); /*components, find */ +- subsampling(power, ltg, &tone, &noise); /*global & minimal */ +- threshold(power, ltg, &tone, &noise, /*threshold, and sgnl- */ +- bitrate[info->version][info->lay-1][info->bitrate_index]/stereo); /*to-mask ratio*/ +- II_minimum_mask(ltg, <min[k][0], sblimit); +- II_smr(<min[k][0], &spike[k][0], &scale[k][0], sblimit); +- } +- mem_free((void **) &sample); +- mem_free((void **) &spike); +- } +- +- /********************************************************************** +- * +- * This module implements the psychoacoustic model I for the +- * MPEG encoder layer I. It uses simplified tonal and noise masking +- * threshold analysis to generate SMR for the encoder bit allocation +- * routine. +- * +- **********************************************************************/ +- +- /**************************************************************** +- * +- * Fast Fourier transform of the input samples. +- * +- ****************************************************************/ +- +- void I_f_f_t(sample, power) /* this function calculates */ +- double FAR sample[FFT_SIZE/2]; /* an FFT analysis for the */ +- mask FAR power[HAN_SIZE/2]; /* freq. domain */ +- { +- int i,j,k,L,l=0; +- int ip, le, le1; +- double t_r, t_i, u_r, u_i; +- static int M, MM1, init = 0, N; +- double *x_r, *x_i, *energy; +- static int *rev; +- static double *w_r, *w_i; +- +- x_r = (double *) mem_alloc(sizeof(DFFT2), "x_r"); +- x_i = (double *) mem_alloc(sizeof(DFFT2), "x_i"); +- energy = (double *) mem_alloc(sizeof(DFFT2), "energy"); +- for(i=0;i<FFT_SIZE/2;i++) x_r[i] = x_i[i] = energy[i] = 0; +- if(!init){ +- rev = (int *) mem_alloc(sizeof(IFFT2), "rev"); +- w_r = (double *) mem_alloc(sizeof(D9), "w_r"); +- w_i = (double *) mem_alloc(sizeof(D9), "w_i"); +- M = 9; +- MM1 = 8; +- N = FFT_SIZE/2; +- for(L=0;L<M;L++){ +- le = 1 << (M-L); +- le1 = le >> 1; +- w_r[L] = cos(PI/le1); +- w_i[L] = -sin(PI/le1); +- } +- for(i=0;i<FFT_SIZE/2;rev[i] = l,i++) for(j=0,l=0;j<9;j++){ +- k=(i>>j) & 1; +- l |= (k<<(8-j)); +- } +- init = 1; +- } +- memcpy( (char *) x_r, (char *) sample, sizeof(double) * FFT_SIZE/2); +- for(L=0;L<MM1;L++){ +- le = 1 << (M-L); +- le1 = le >> 1; +- u_r = 1; +- u_i = 0; +- for(j=0;j<le1;j++){ +- for(i=j;i<N;i+=le){ +- ip = i + le1; +- t_r = x_r[i] + x_r[ip]; +- t_i = x_i[i] + x_i[ip]; +- x_r[ip] = x_r[i] - x_r[ip]; +- x_i[ip] = x_i[i] - x_i[ip]; +- x_r[i] = t_r; +- x_i[i] = t_i; +- t_r = x_r[ip]; +- x_r[ip] = x_r[ip] * u_r - x_i[ip] * u_i; +- x_i[ip] = x_i[ip] * u_r + t_r * u_i; +- } +- t_r = u_r; +- u_r = u_r * w_r[L] - u_i * w_i[L]; +- u_i = u_i * w_r[L] + t_r * w_i[L]; +- } +- } +- for(i=0;i<N;i+=2){ +- ip = i + 1; +- t_r = x_r[i] + x_r[ip]; +- t_i = x_i[i] + x_i[ip]; +- x_r[ip] = x_r[i] - x_r[ip]; +- x_i[ip] = x_i[i] - x_i[ip]; +- x_r[i] = t_r; +- x_i[i] = t_i; +- energy[i] = x_r[i] * x_r[i] + x_i[i] * x_i[i]; +- } +- for(i=0;i<FFT_SIZE/2;i++) if(i<rev[i]){ +- t_r = energy[i]; +- energy[i] = energy[rev[i]]; +- energy[rev[i]] = t_r; +- } +- for(i=0;i<HAN_SIZE/2;i++){ /* calculate power */ +- if(energy[i] < 1E-20) energy[i] = 1E-20; /* density spectrum */ +- power[i].x = 10 * log10(energy[i]) + POWERNORM; +- power[i].next = STOP; +- power[i].type = FALSE; +- } +- mem_free((void **) &x_r); +- mem_free((void **) &x_i); +- mem_free((void **) &energy); +- } +- +- /**************************************************************** +- * +- * Window the incoming audio signal. +- * +- ****************************************************************/ +- +- void I_hann_win(sample) /* this function calculates a */ +- double FAR sample[FFT_SIZE/2]; /* Hann window for PCM (input) */ +- { /* samples for a 512-pt. FFT */ +- register int i; +- register double sqrt_8_over_3; +- static int init = 0; +- static double FAR *window; +- +- if(!init){ /* calculate window function for the Fourier transform */ +- window = (double FAR *) mem_alloc(sizeof(DFFT2), "window"); +- sqrt_8_over_3 = pow(8.0/3.0, 0.5); +- for(i=0;i<FFT_SIZE/2;i++){ +- /* Hann window formula */ +- window[i]=sqrt_8_over_3*0.5*(1-cos(2.0*PI*i/(FFT_SIZE/2)))/(FFT_SIZE/2); +- } +- init = 1; +- } +- for(i=0;i<FFT_SIZE/2;i++) sample[i] *= window[i]; +- } +- +- /******************************************************************* +- * +- * This function finds the maximum spectral component in each +- * subband and return them to the encoder for time-domain threshold +- * determination. +- * +- *******************************************************************/ +- #ifndef LONDON +- void I_pick_max(power, spike) +- double FAR spike[SBLIMIT]; +- mask FAR power[HAN_SIZE/2]; +- { +- double max; +- int i,j; +- +- /* calculate the spectral component in each subband */ +- for(i=0;i<HAN_SIZE/2;spike[i>>3] = max, i+=8) +- for(j=0, max = DBMIN;j<8;j++) max = (max>power[i+j].x) ? max : power[i+j].x; +- } +- #else +- void I_pick_max(power, spike) +- double FAR spike[SBLIMIT]; +- mask FAR power[HAN_SIZE]; +- { +- double sum; +- int i,j; +- +- for(i=0;i<HAN_SIZE/2;spike[i>>3] = 10.0*log10(sum), i+=8) +- /* calculate the */ +- for(j=0, sum = pow(10.0,0.1*DBMIN);j<8;j++) /* sum of spectral */ +- sum += pow(10.0,0.1*power[i+j].x); /* component in each */ +- } /* subband from bound */ +- #endif +- /**************************************************************** +- * +- * This function labels the tonal component in the power +- * spectrum. +- * +- ****************************************************************/ +- +- void I_tonal_label(power, tone) /* this function extracts */ +- mask FAR power[HAN_SIZE/2]; /* (tonal) sinusoidals from */ +- int *tone; /* the spectrum */ +- { +- int i,j, last = LAST, first, run; +- double max; +- int last_but_one= LAST; +- +- *tone = LAST; +- for(i=2;i<HAN_SIZE/2-6;i++){ +- if(power[i].x>power[i-1].x && power[i].x>=power[i+1].x){ +- power[i].type = TONE; +- power[i].next = LAST; +- if(last != LAST) power[last].next = i; +- else first = *tone = i; +- last = i; +- } +- } +- last = LAST; +- first = *tone; +- *tone = LAST; +- while(first != LAST){ /* conditions for the tonal */ +- if(first<3 || first>250) run = 0; /* otherwise k+/-j will be out of bounds*/ +- else if(first<63) run = 2; /* components in layer I, which */ +- else if(first<127) run = 3; /* are the boundaries for calc. */ +- else run = 6; /* the tonal components */ +- max = power[first].x - 7; +- for(j=2;j<=run;j++) /* after calc. of tonal components, set to loc.*/ +- if(max < power[first-j].x || max < power[first+j].x){ /* max */ +- power[first].type = FALSE; +- break; +- } +- if(power[first].type == TONE){ /* extract tonal components */ +- int help=first; +- if(*tone == LAST) *tone = first; +- while((power[help].next!=LAST)&&(power[help].next-first)<=run) +- help=power[help].next; +- help=power[help].next; +- power[first].next=help; +- if((first-last)<=run){ +- if(last_but_one != LAST) power[last_but_one].next=first; +- } +- if(first>1 && first<255){ /* calculate the sum of the */ +- double tmp; /* powers of the components */ +- tmp = add_db(power[first-1].x, power[first+1].x); +- power[first].x = add_db(power[first].x, tmp); +- } +- for(j=1;j<=run;j++){ +- power[first-j].x = power[first+j].x = DBMIN; +- power[first-j].next = power[first+j].next = STOP; /*dpwe: 2nd was .x*/ +- power[first-j].type = power[first+j].type = FALSE; +- } +- last_but_one=last; +- last = first; +- first = power[first].next; +- } +- else { +- int ll; +- if(last == LAST) ; /* *tone = power[first].next; dpwe */ +- else power[last].next = power[first].next; +- ll = first; +- first = power[first].next; +- power[ll].next = STOP; +- } +- } +- } +- +- /**************************************************************** +- * +- * This function finds the minimum masking threshold and +- * return the value to the encoder. +- * +- ****************************************************************/ +- +- void I_minimum_mask(ltg,ltmin) +- g_thres FAR *ltg; +- double FAR ltmin[SBLIMIT]; +- { +- double min; +- int i,j; +- +- j=1; +- for(i=0;i<SBLIMIT;i++) +- if(j>=sub_size-1) /* check subband limit, and */ +- ltmin[i] = ltg[sub_size-1].hear; /* calculate the minimum masking */ +- else { /* level of LTMIN for each subband*/ +- min = ltg[j].x; +- while(ltg[j].line>>3 == i && j < sub_size){ +- if (min>ltg[j].x) min = ltg[j].x; +- j++; +- } +- ltmin[i] = min; +- } +- } +- +- /***************************************************************** +- * +- * This procedure is called in musicin to pick out the +- * smaller of the scalefactor or threshold. +- * +- *****************************************************************/ +- +- void I_smr(ltmin, spike, scale) +- double FAR spike[SBLIMIT], scale[SBLIMIT], ltmin[SBLIMIT]; +- { +- int i; +- double max; +- +- for(i=0;i<SBLIMIT;i++){ /* determine the signal */ +- max = 20 * log10(scale[i] * 32768) - 10; /* level for each subband */ +- if(spike[i]>max) max = spike[i]; /* for the scalefactor */ +- max -= ltmin[i]; +- ltmin[i] = max; +- } +- } +- +- /**************************************************************** +- * +- * This procedure calls all the necessary functions to +- * complete the psychoacoustic analysis. +- * +- ****************************************************************/ +- +- void I_Psycho_One(buffer, scale, ltmin, fr_ps) +- short FAR buffer[2][1152]; +- double FAR scale[2][SBLIMIT], ltmin[2][SBLIMIT]; +- frame_params *fr_ps; +- { +- int stereo = fr_ps->stereo; +- the_layer info = fr_ps->header; +- int k,i, tone=0, noise=0; +- static char init = 0; +- static int off[2] = {256,256}; +- double *sample; +- DSBL *spike; +- static D640 *fft_buf; +- static mask_ptr FAR power; +- static g_ptr FAR ltg; +- +- sample = (double *) mem_alloc(sizeof(DFFT2), "sample"); +- spike = (DSBL *) mem_alloc(sizeof(D2SBL), "spike"); +- /* call functions for critical boundaries, freq. */ +- if(!init){ /* bands, bark values, and mapping */ +- fft_buf = (D640 *) mem_alloc(sizeof(D640) * 2, "fft_buf"); +- power = (mask_ptr FAR ) mem_alloc(sizeof(mask) * HAN_SIZE/2, "power"); +- if (info->version == MPEG_AUDIO_ID) { +- read_cbound(info->lay, info->sampling_frequency); +- read_freq_band(<g, info->lay, info->sampling_frequency); +- } else { +- read_cbound(info->lay, info->sampling_frequency + 4); +- read_freq_band(<g, info->lay, info->sampling_frequency + 4); +- } +- make_map(power,ltg); +- for(i=0;i<640;i++) fft_buf[0][i] = fft_buf[1][i] = 0; +- init = 1; +- } +- for(k=0;k<stereo;k++){ /* check PCM input for a block of */ +- for(i=0;i<384;i++) /* 384 samples for a 512-pt. FFT */ +- fft_buf[k][(i+off[k])%640]= (double) buffer[k][i]/SCALE; +- for(i=0;i<FFT_SIZE/2;i++) +- sample[i] = fft_buf[k][(i+448+off[k])%640]; +- off[k] += 384; +- off[k] %= 640; +- /* call functions for windowing PCM samples, */ +- I_hann_win(sample); /* location of spectral components in each */ +- for(i=0;i<HAN_SIZE/2;i++) power[i].x = DBMIN; /* subband with */ +- I_f_f_t(sample, power); /* labeling, locate remaining */ +- I_pick_max(power, &spike[k][0]); /* non-tonal sinusoidals, */ +- I_tonal_label(power, &tone); /* reduce noise & tonal com., */ +- noise_label(power, &noise, ltg); /* find global & minimal */ +- subsampling(power, ltg, &tone, &noise); /* threshold, and sgnl- */ +- threshold(power, ltg, &tone, &noise, /* to-mask ratio */ +- bitrate[info->version][info->lay-1][info->bitrate_index]/stereo); +- I_minimum_mask(ltg, <min[k][0]); +- I_smr(<min[k][0], &spike[k][0], &scale[k][0]); +- } +- mem_free((void **) &sample); +- mem_free((void **) &spike); +- } +--- 0 ---- +diff -r -c -N encoder/version.c lame3.13/version.c +*** encoder/version.c Wed Dec 31 17:00:00 1969 +--- lame3.13/version.c Mon May 24 15:24:32 1999 +*************** +*** 0 **** +--- 1,57 ---- ++ /* ++ * Version numbering for LAME. ++ * ++ * Copyright (c) 1999 A.L. Faber ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License as published by ++ * the Free Software Foundation; either version 2, or (at your option) ++ * any later version. ++ * ++ * This program is distributed in the hope that it will be useful, ++ * but WITHOUT ANY WARRANTY; without even the implied warranty of ++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the ++ * GNU General Public License for more details. ++ * ++ * You should have received a copy of the GNU General Public License ++ * along with this program; see the file COPYING. If not, write to ++ * the Free Software Foundation, 675 Mass Ave, Cambridge, MA 02139, USA. ++ */ ++ ++ #include "version.h" ++ #include <stdio.h> ++ ++ static char lpszVersion[80]; ++ ++ char* get_lame_version() ++ { ++ if (LAME_ALPHAVERSION>0) ++ sprintf(lpszVersion,"%d.%02d (alpha %d)",LAME_MAJOR_VERSION,LAME_MINOR_VERSION,LAME_ALPHAVERSION); ++ else if (LAME_BETAVERSION>0) ++ sprintf(lpszVersion,"%d.%02d (beta %d)",LAME_MAJOR_VERSION,LAME_MINOR_VERSION,LAME_BETAVERSION); ++ else ++ sprintf(lpszVersion,"%d.%02d",LAME_MAJOR_VERSION,LAME_MINOR_VERSION); ++ return lpszVersion; ++ } ++ ++ char* get_psy_version() ++ { ++ if (PSY_ALPHAVERSION>0) ++ sprintf(lpszVersion,"%d.%02d (alpha %d)",PSY_MAJOR_VERSION,PSY_MINOR_VERSION,PSY_ALPHAVERSION); ++ else if (PSY_BETAVERSION>0) ++ sprintf(lpszVersion,"%d.%02d (beta %d)",PSY_MAJOR_VERSION,PSY_MINOR_VERSION,PSY_BETAVERSION); ++ else ++ sprintf(lpszVersion,"%d.%02d",PSY_MAJOR_VERSION,PSY_MINOR_VERSION); ++ return lpszVersion; ++ } ++ ++ char* get_mp3x_version() ++ { ++ if (MP3X_ALPHAVERSION>0) ++ sprintf(lpszVersion,"%d:%02d (alpha %d)",MP3X_MAJOR_VERSION,MP3X_MINOR_VERSION,MP3X_ALPHAVERSION); ++ else if (MP3X_BETAVERSION>0) ++ sprintf(lpszVersion,"%d:%02d (beta %d)",MP3X_MAJOR_VERSION,MP3X_MINOR_VERSION,MP3X_BETAVERSION); ++ else ++ sprintf(lpszVersion,"%d:%02d",MP3X_MAJOR_VERSION,MP3X_MINOR_VERSION); ++ return lpszVersion; ++ } +diff -r -c -N encoder/version.h lame3.13/version.h +*** encoder/version.h Wed Dec 31 17:00:00 1969 +--- lame3.13/version.h Thu Jun 24 18:30:20 1999 +*************** +*** 0 **** +--- 1,44 ---- ++ /* ++ * Version numbering for LAME. ++ * ++ * Copyright (c) 1999 A.L. Faber ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License as published by ++ * the Free Software Foundation; either version 2, or (at your option) ++ * any later version. ++ * ++ * This program is distributed in the hope that it will be useful, ++ * but WITHOUT ANY WARRANTY; without even the implied warranty of ++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the ++ * GNU General Public License for more details. ++ * ++ * You should have received a copy of the GNU General Public License ++ * along with this program; see the file COPYING. If not, write to ++ * the Free Software Foundation, 675 Mass Ave, Cambridge, MA 02139, USA. ++ */ ++ ++ #ifndef LAME_H_INCLUDED ++ #define LAME_H_INCLUDED ++ ++ #define LAME_MAJOR_VERSION 3 /* Major version number */ ++ #define LAME_MINOR_VERSION 13 /* Minor version number */ ++ #define LAME_ALPHAVERSION 0 /* Set number if this is an alpha version, otherwise zero */ ++ #define LAME_BETAVERSION 0 /* Set number if this is a beta version, otherwise zero */ ++ ++ #define PSY_MAJOR_VERSION 0 /* Major version number */ ++ #define PSY_MINOR_VERSION 50 /* Minor version number */ ++ #define PSY_ALPHAVERSION 0 /* Set number if this is an alpha version, otherwise zero */ ++ #define PSY_BETAVERSION 0 /* Set number if this is a beta version, otherwise zero */ ++ ++ #define MP3X_MAJOR_VERSION 0 /* Major version number */ ++ #define MP3X_MINOR_VERSION 50 /* Minor version number */ ++ #define MP3X_ALPHAVERSION 0 /* Set number if this is an alpha version, otherwise zero */ ++ #define MP3X_BETAVERSION 0 /* Set number if this is a beta version, otherwise zero */ ++ ++ ++ char* get_lame_version(); /* returns lame version number string */ ++ char* get_psy_version(); /* returns psy model version number string */ ++ char* get_mp3x_version(); /* returns mp3x version number string */ ++ ++ #endif +diff -r -c -N encoder/wavheader.h lame3.13/wavheader.h +*** encoder/wavheader.h Wed Dec 31 17:00:00 1969 +--- lame3.13/wavheader.h Mon Jun 21 12:21:44 1999 +*************** +*** 0 **** +--- 1,70 ---- ++ /* ++ * WavHeader for LAME. ++ * ++ * Ripped for lame from Michael Beck's vplay sources (found in sndkit) ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License as published by ++ * the Free Software Foundation; either version 2, or (at your option) ++ * any later version. ++ * ++ * This program is distributed in the hope that it will be useful, ++ * but WITHOUT ANY WARRANTY; without even the implied warranty of ++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the ++ * GNU General Public License for more details. ++ * ++ * You should have received a copy of the GNU General Public License ++ * along with this program; see the file COPYING. If not, write to ++ * the Free Software Foundation, 675 Mass Ave, Cambridge, MA 02139, USA. ++ */ ++ ++ #ifndef WAVHEADER_H_INCLUDED ++ #define WAVHEADER_H_INCLUDED ++ ++ /* Definitions for Microsoft WAVE format */ ++ ++ #define RIFF 0x46464952 ++ #define WAVE 0x45564157 ++ #define FMT 0x20746D66 ++ #define DATA 0x61746164 ++ #define PCM_CODE 1 ++ #define WAVE_MONO 1 ++ #define WAVE_STEREO 2 ++ ++ #define u_int unsigned int ++ #define u_short unsigned short ++ #ifdef _WIN32 ++ #pragma pack(push) ++ #pragma pack(1) ++ #endif ++ ++ /* it's in chunks like .voc and AMIGA iff, but my source say there ++ are in only in this combination, so I combined them in one header; ++ it works on all WAVE-file I have ++ */ ++ typedef struct _waveheader { ++ u_int main_chunk; /* 'RIFF' */ ++ u_int length; /* filelen */ ++ u_int chunk_type; /* 'WAVE' */ ++ ++ u_int sub_chunk; /* 'fmt ' */ ++ u_int sc_len; /* length of sub_chunk, =16 */ ++ u_short format; /* should be 1 for PCM-code */ ++ u_short modus; /* 1 Mono, 2 Stereo */ ++ u_int sample_fq; /* frequence of sample */ ++ u_int byte_p_sec; ++ u_short byte_p_spl; /* samplesize; 1 or 2 bytes */ ++ u_short bit_p_spl; /* 8, 12 or 16 bit */ ++ ++ u_int data_chunk; /* 'data' */ ++ u_int data_length; /* samplecount */ ++ } WaveHeader; ++ ++ ++ void endswap_wave_fmt (WaveHeader *wavfmt); ++ ++ #ifdef _WIN32 ++ #pragma pack(pop) ++ #endif ++ ++ #endif diff --git a/lame3.50.patch b/lame3.50.patch new file mode 100644 index 0000000..fb4a351 --- /dev/null +++ b/lame3.50.patch @@ -0,0 +1,35315 @@ +diff -r -c -N encoder/COPYING lame3.50/COPYING +*** encoder/COPYING Wed Dec 31 17:00:00 1969 +--- lame3.50/COPYING Mon Nov 1 18:17:59 1999 +*************** +*** 0 **** +--- 1,340 ---- ++ GNU GENERAL PUBLIC LICENSE ++ Version 2, June 1991 ++ ++ Copyright (C) 1989, 1991 Free Software Foundation, Inc. ++ 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA ++ Everyone is permitted to copy and distribute verbatim copies ++ of this license document, but changing it is not allowed. ++ ++ Preamble ++ ++ The licenses for most software are designed to take away your ++ freedom to share and change it. By contrast, the GNU General Public ++ License is intended to guarantee your freedom to share and change free ++ software--to make sure the software is free for all its users. This ++ General Public License applies to most of the Free Software ++ Foundation's software and to any other program whose authors commit to ++ using it. (Some other Free Software Foundation software is covered by ++ the GNU Library General Public License instead.) You can apply it to ++ your programs, too. ++ ++ When we speak of free software, we are referring to freedom, not ++ price. Our General Public Licenses are designed to make sure that you ++ have the freedom to distribute copies of free software (and charge for ++ this service if you wish), that you receive source code or can get it ++ if you want it, that you can change the software or use pieces of it ++ in new free programs; and that you know you can do these things. ++ ++ To protect your rights, we need to make restrictions that forbid ++ anyone to deny you these rights or to ask you to surrender the rights. ++ These restrictions translate to certain responsibilities for you if you ++ distribute copies of the software, or if you modify it. ++ ++ For example, if you distribute copies of such a program, whether ++ gratis or for a fee, you must give the recipients all the rights that ++ you have. You must make sure that they, too, receive or can get the ++ source code. 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To prevent this, we have made it clear that any ++ patent must be licensed for everyone's free use or not licensed at all. ++ ++ The precise terms and conditions for copying, distribution and ++ modification follow. ++ ++ GNU GENERAL PUBLIC LICENSE ++ TERMS AND CONDITIONS FOR COPYING, DISTRIBUTION AND MODIFICATION ++ ++ 0. This License applies to any program or other work which contains ++ a notice placed by the copyright holder saying it may be distributed ++ under the terms of this General Public License. The "Program", below, ++ refers to any such program or work, and a "work based on the Program" ++ means either the Program or any derivative work under copyright law: ++ that is to say, a work containing the Program or a portion of it, ++ either verbatim or with modifications and/or translated into another ++ language. (Hereinafter, translation is included without limitation in ++ the term "modification".) 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See the ++ GNU General Public License for more details. ++ ++ You should have received a copy of the GNU General Public License ++ along with this program; if not, write to the Free Software ++ Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA ++ ++ ++ Also add information on how to contact you by electronic and paper mail. ++ ++ If the program is interactive, make it output a short notice like this ++ when it starts in an interactive mode: ++ ++ Gnomovision version 69, Copyright (C) 19yy name of author ++ Gnomovision comes with ABSOLUTELY NO WARRANTY; for details type `show w'. ++ This is free software, and you are welcome to redistribute it ++ under certain conditions; type `show c' for details. ++ ++ The hypothetical commands `show w' and `show c' should show the appropriate ++ parts of the General Public License. Of course, the commands you use may ++ be called something other than `show w' and `show c'; they could even be ++ mouse-clicks or menu items--whatever suits your program. ++ ++ You should also get your employer (if you work as a programmer) or your ++ school, if any, to sign a "copyright disclaimer" for the program, if ++ necessary. Here is a sample; alter the names: ++ ++ Yoyodyne, Inc., hereby disclaims all copyright interest in the program ++ `Gnomovision' (which makes passes at compilers) written by James Hacker. ++ ++ <signature of Ty Coon>, 1 April 1989 ++ Ty Coon, President of Vice ++ ++ This General Public License does not permit incorporating your program into ++ proprietary programs. If your program is a subroutine library, you may ++ consider it more useful to permit linking proprietary applications with the ++ library. If this is what you want to do, use the GNU Library General ++ Public License instead of this License. +diff -r -c -N encoder/Dll/BladeMP3EncDLL.c lame3.50/Dll/BladeMP3EncDLL.c +*** encoder/Dll/BladeMP3EncDLL.c Wed Dec 31 17:00:00 1969 +--- lame3.50/Dll/BladeMP3EncDLL.c Mon Oct 18 12:25:28 1999 +*************** +*** 0 **** +--- 1,474 ---- ++ /* ++ * Blade DLL Interface for LAME. ++ * ++ * Copyright (c) 1999 A.L. Faber ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License as published by ++ * the Free Software Foundation; either version 2, or (at your option) ++ * any later version. ++ * ++ * This program is distributed in the hope that it will be useful, ++ * but WITHOUT ANY WARRANTY; without even the implied warranty of ++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the ++ * GNU General Public License for more details. ++ * ++ * You should have received a copy of the GNU General Public License ++ * along with this program; see the file COPYING. If not, write to ++ * the Free Software Foundation, 675 Mass Ave, Cambridge, MA 02139, USA. ++ */ ++ ++ #include "machine.h" ++ #include "BladeMP3EncDLL.h" ++ #include <assert.h> ++ #include "util.h" ++ #include "version.h" ++ #include "VbrTag.h" ++ #include "lame.h" ++ #include "get_audio.h" ++ #include "globalflags.h" ++ ++ ++ ++ const int MAJORVERSION=1; ++ const int MINORVERSION=02; ++ ++ static short int InputBuffer[2][1152]; ++ ++ int nBladeBufferSize; ++ ++ // Local variables ++ static int nPsychoModel=2; ++ static BOOL bFirstFrame=TRUE; ++ static DWORD dwSampleBufferSize=0; ++ ++ ++ #ifdef _DEBUG ++ void dump_config( frame_params *fr_ps, int *psy, char *inPath, char *outPath); ++ #endif ++ ++ // Taken from main.c ++ static char original_file_name[MAX_NAME_SIZE]; ++ static char encoded_file_name[MAX_NAME_SIZE]; ++ static int stereo, error_protection; ++ static layer info; ++ ++ ++ extern Bit_stream_struc bs; ++ extern III_side_info_t l3_side; ++ extern frame_params fr_ps; ++ ++ static void InitParams() ++ { ++ // clear buffers ++ memset(InputBuffer, 0,sizeof(InputBuffer)); ++ bFirstFrame=TRUE; ++ // Initialize output buffer ++ bs.pbtOutBuf=NULL; ++ bs.nOutBufPos=0; ++ lame_init(1); /* 1 means LAME will do no output file I/O */ ++ // InitSndFile(); /* only needed because we will not be calling OpenSndFile */ ++ ++ } ++ ++ #define NORMAL_QUALITY 0 ++ #define LOW_QUALITY 1 ++ #define HIGH_QUALITY 2 ++ ++ #define GPL_PSYCHOMODEL 0 ++ #define ISO_PSYCHOMODEL 1 ++ ++ __declspec(dllexport) BE_ERR beInitStream(PBE_CONFIG pbeConfig, PDWORD dwSamples, PDWORD dwBufferSize, PHBE_STREAM phbeStream) ++ { ++ #define MAX_ARGV 20 ++ ++ char strTmp[255]; ++ int nDllArgC=0; ++ char DllArgV[20][80]; ++ char* argv[MAX_ARGV]; ++ int i; ++ int nCRC=pbeConfig->format.mp3.bCRC; ++ int nVBR; ++ int nQuality; ++ ++ // Get VBR setting from fourth nibble ++ nVBR=(nCRC>>12)&0x0F; ++ // Get Quality from third nibble ++ nQuality=(nCRC>>8)&0x0F; ++ // Get model from lower nibble (to be compatible with standard definition) ++ nCRC=(nCRC&0x01); ++ ++ for (i=0;i<MAX_ARGV;i++) ++ argv[i]=DllArgV[i]; ++ ++ // Clear the external and local paramters ++ InitParams(); ++ ++ // Clear argument array ++ memset(&DllArgV[0][0],0x00,sizeof(DllArgV)); ++ ++ // Not used, always assign stream 1 ++ *phbeStream=1; ++ ++ // Set MP3 buffer size ++ *dwBufferSize=BUFFER_SIZE*2; ++ ++ ++ // Set number of input samples depending on the number of samples ++ if ((pbeConfig->format.mp3.byMode&0x0F)== BE_MP3_MODE_MONO) ++ { ++ // Mono channel, thus MPEGFRAMSIZE samples needed ++ *dwSamples=1152; ++ } ++ else ++ { ++ // For Stereo.Joint stereo and dual channel, need 2*MPEGFRAMESIZE samples ++ *dwSamples=1152*2; ++ } ++ ++ // Set the input sample buffer size, so we know what we can expect ++ dwSampleBufferSize=*dwSamples; ++ ++ // --------------- Set arguments for ParseArg function ------------------------- ++ ++ // Set zero argument, the filename ++ strcpy(DllArgV[nDllArgC++],"LameDLLEncoder"); ++ ++ switch (pbeConfig->format.mp3.byMode) ++ { ++ case BE_MP3_MODE_STEREO: ++ strcpy(DllArgV[nDllArgC++],"-ms"); ++ break; ++ case BE_MP3_MODE_JSTEREO: ++ strcpy(DllArgV[nDllArgC++],"-mj"); ++ break; ++ case BE_MP3_MODE_MONO: ++ strcpy(DllArgV[nDllArgC++],"-mm"); ++ break; ++ case BE_MP3_MODE_DUALCHANNEL: ++ strcpy(DllArgV[nDllArgC++],"-mf"); ++ break; ++ default: ++ return BE_ERR_INVALID_FORMAT_PARAMETERS; ++ } ++ ++ switch (nQuality) ++ { ++ case NORMAL_QUALITY: // Nothing special ++ break; ++ case LOW_QUALITY: // -f flag ++ strcpy(DllArgV[nDllArgC++],"-f"); ++ break; ++ case HIGH_QUALITY: // -k flag for high qualtiy ++ strcpy(DllArgV[nDllArgC++],"-k"); ++ break; ++ } ++ ++ if (nVBR) ++ { ++ // 0=no vbr 1..10 is VBR quality setting -1 ++ sprintf(DllArgV[nDllArgC++],"-v"); ++ sprintf(DllArgV[nDllArgC++],"-V%d",nVBR-1); ++ } ++ ++ // Set frequency ++ sprintf(strTmp,"-s %f",pbeConfig->format.mp3.dwSampleRate/1000.0); ++ strcpy(DllArgV[nDllArgC++],strTmp); ++ ++ // Set bitrate ++ sprintf(strTmp,"-b %d",pbeConfig->format.mp3.wBitrate); ++ strcpy(DllArgV[nDllArgC++],strTmp); ++ ++ // Set copyright flag? ++ if (pbeConfig->format.mp3.bCopyright) ++ strcpy(DllArgV[nDllArgC++],"-c"); ++ ++ // Do we have to tag it as non original ++ if (!pbeConfig->format.mp3.bOriginal) ++ strcpy(DllArgV[nDllArgC++],"-o"); ++ ++ // Add CRC? ++ if (nCRC) ++ strcpy(DllArgV[nDllArgC++],"-p"); ++ ++ // Add input filename ++ strcpy(DllArgV[nDllArgC++],"NULL.wav"); ++ ++ // Add output filename ++ strcpy(DllArgV[nDllArgC++],"NULL.mp3"); ++ ++ ++ // Set the encoder variables ++ lame_parse_args(nDllArgC,argv); ++ ++ // Set private bit? ++ if (pbeConfig->format.mp3.bPrivate) ++ { ++ info.extension = 0; ++ } ++ else ++ { ++ info.extension = 1; ++ } ++ ++ ++ //hdr_to_frps(&fr_ps); /* now called in parse_args */ ++ ++ stereo = fr_ps.stereo; ++ ++ error_protection = info.error_protection; ++ ++ if (info.lay != 3) ++ { ++ printf("Only Layer 3 supported\n"); ++ return BE_ERR_INVALID_FORMAT_PARAMETERS; ++ } ++ ++ #ifdef _DEBUG ++ dump_config(&fr_ps,&nPsychoModel,original_file_name,encoded_file_name); ++ #endif ++ ++ // Everything went OK, thus return SUCCESSFUL ++ return BE_ERR_SUCCESSFUL; ++ } ++ ++ ++ ++ __declspec(dllexport) BE_ERR beDeinitStream(HBE_STREAM hbeStream, PBYTE pOutput, PDWORD pdwOutput) ++ { ++ // Set output buffer ++ bs.pbtOutBuf=pOutput; ++ bs.nOutBufPos=0; ++ ++ // Crank out the last part of the buffer ++ if (fr_ps.header->lay == 3 ) ++ III_FlushBitstream(); ++ ++ // close the stream ++ //close_bit_stream_w( &bs ); /* routine removed */ ++ ++ // Flush last bytes ++ //write_buffer(&bs, 1+bs.buf_byte_idx); /* ouput to mp3 file */ ++ *pdwOutput = copy_buffer((char*)&bs.pbtOutBuf[bs.nOutBufPos],&bs); ++ empty_buffer(&bs); /* empty buffer */ ++ ++ // Number of bytes in output buffer ++ bs.nOutBufPos= *pdwOutput; ++ ++ // Deallocate all buffers ++ desalloc_buffer(&bs); ++ ++ // Everything went OK, thus return SUCCESSFUL ++ return BE_ERR_SUCCESSFUL; ++ } ++ ++ ++ __declspec(dllexport) BE_ERR beCloseStream(HBE_STREAM hbeStream) ++ { ++ // DeInit encoder ++ // return DeInitEncoder(); ++ return BE_ERR_SUCCESSFUL; ++ } ++ ++ ++ ++ __declspec(dllexport) VOID beVersion(PBE_VERSION pbeVersion) ++ { ++ // DLL Release date ++ char lpszDate[20]; ++ char lpszTemp[5]; ++ ++ ++ // Set DLL interface version ++ pbeVersion->byDLLMajorVersion=MAJORVERSION; ++ pbeVersion->byDLLMinorVersion=MINORVERSION; ++ ++ // Set Engine version number (Same as Lame version) ++ pbeVersion->byMajorVersion=LAME_MAJOR_VERSION; ++ pbeVersion->byMinorVersion=LAME_MINOR_VERSION; ++ ++ // Get compilation date ++ strcpy(lpszDate,__DATE__); ++ ++ // Get the first three character, which is the month ++ strncpy(lpszTemp,lpszDate,3); ++ ++ // Set month ++ if (strcmp(lpszTemp,"Jan")==0) pbeVersion->byMonth=1; ++ if (strcmp(lpszTemp,"Feb")==0) pbeVersion->byMonth=2; ++ if (strcmp(lpszTemp,"Mar")==0) pbeVersion->byMonth=3; ++ if (strcmp(lpszTemp,"Apr")==0) pbeVersion->byMonth=4; ++ if (strcmp(lpszTemp,"May")==0) pbeVersion->byMonth=5; ++ if (strcmp(lpszTemp,"Jun")==0) pbeVersion->byMonth=6; ++ if (strcmp(lpszTemp,"Jul")==0) pbeVersion->byMonth=7; ++ if (strcmp(lpszTemp,"Aug")==0) pbeVersion->byMonth=8; ++ if (strcmp(lpszTemp,"Sep")==0) pbeVersion->byMonth=9; ++ if (strcmp(lpszTemp,"Oct")==0) pbeVersion->byMonth=10; ++ if (strcmp(lpszTemp,"Nov")==0) pbeVersion->byMonth=11; ++ if (strcmp(lpszTemp,"Dec")==0) pbeVersion->byMonth=12; ++ ++ // Get day of month string (char [4..5]) ++ pbeVersion->byDay=atoi(lpszDate+4); ++ ++ // Get year of compilation date (char [7..10]) ++ pbeVersion->wYear=atoi(lpszDate+7); ++ ++ memset(pbeVersion->zHomepage,0x00,BE_MAX_HOMEPAGE); ++ ++ strcpy(pbeVersion->zHomepage,"http://www.surf.to/cdex"); ++ } ++ ++ ++ __declspec(dllexport) BE_ERR beEncodeChunk(HBE_STREAM hbeStream, DWORD nSamples, PSHORT pSamples, PBYTE pOutput, PDWORD pdwOutput) ++ { ++ int iSampleIndex; ++ ++ // Set output buffer ++ bs.pbtOutBuf=pOutput; ++ bs.nOutBufPos=0; ++ /* ++ if (g_bFirstFrame) ++ { ++ // Write initial VBR Header to bitstream ++ InitVbrTag(&bs,info->version-1,info->mode,info->sampling_frequency); ++ if (!lame_nowrite) ++ write_buffer(&bs); ++ empty_buffer(&bs); ++ } ++ */ ++ // Is this the last (incomplete) frame ++ if (nSamples<dwSampleBufferSize) ++ { ++ // Padd with zeros ++ memset(pSamples+nSamples,0x00,(dwSampleBufferSize-nSamples)*sizeof(SHORT)); ++ } ++ ++ // Set buffer size, in number of bytes ++ nBladeBufferSize=nSamples*sizeof(SHORT); ++ ++ if (stereo==2) ++ { ++ for (iSampleIndex=0;iSampleIndex<1152;iSampleIndex++) ++ { ++ // Copy new sample data into InputBuffer ++ InputBuffer[0][iSampleIndex]=*pSamples++; ++ InputBuffer[1][iSampleIndex]=*pSamples++; ++ } ++ } ++ else ++ { ++ // Mono, only put it data into buffer[0] (=left channel) ++ for (iSampleIndex=0;iSampleIndex<1152;iSampleIndex++) ++ { ++ // Copy new sample data into InputBuffer ++ InputBuffer[0][iSampleIndex]=*pSamples++; ++ } ++ } ++ ++ ++ // Encode it ++ *pdwOutput=lame_encode(InputBuffer,(char*)&bs.pbtOutBuf[bs.nOutBufPos]); ++ ++ // Set number of output bytes ++ bs.nOutBufPos=*pdwOutput; ++ ++ return BE_ERR_SUCCESSFUL; ++ } ++ ++ ++ __declspec(dllexport) BE_ERR beWriteVBRHeader(LPCSTR lpszFileName) ++ { ++ if (g_bWriteVbrTag) ++ { ++ // Calculate relative quality of VBR stream ++ // 0=best, 100=worst ++ int nQuality=VBR_q*100/9; ++ ++ // Write Xing header again ++ return PutVbrTag((LPSTR)lpszFileName,nQuality); ++ } ++ return BE_ERR_INVALID_FORMAT_PARAMETERS; ++ } ++ ++ ++ BOOL APIENTRY DllMain(HANDLE hModule, ++ DWORD ul_reason_for_call, ++ LPVOID lpReserved) ++ { ++ switch( ul_reason_for_call ) ++ { ++ case DLL_PROCESS_ATTACH: ++ OutputDebugString("Attach Process \n"); ++ break; ++ case DLL_THREAD_ATTACH: ++ OutputDebugString("Attach Thread \n"); ++ break; ++ case DLL_THREAD_DETACH: ++ OutputDebugString("Detach Thread \n"); ++ break; ++ case DLL_PROCESS_DETACH: ++ OutputDebugString("Detach Process \n"); ++ break; ++ } ++ return TRUE; ++ } ++ ++ ++ #ifdef _DEBUG ++ void dump_config( frame_params *fr_ps, int *psy, char *inPath, char *outPath) ++ { ++ layer *info = fr_ps->header; ++ char strTmp[255]; ++ ++ OutputDebugString("Encoding configuration:\n"); ++ ++ sprintf(strTmp,"Algorithm=%s\n", version_names[info->version]); ++ OutputDebugString(strTmp); ++ ++ if(info->mode != MPG_MD_JOINT_STEREO) ++ { ++ sprintf(strTmp,"Layer=%s mode=%s extn=%d psy model=%d\n",layer_names[info->lay-1], mode_names[info->mode],info->mode_ext, *psy); ++ OutputDebugString(strTmp); ++ } ++ else ++ { ++ sprintf(strTmp,"Layer=%s mode=%s extn=data dependant psy model=%d\n",layer_names[info->lay-1], mode_names[info->mode], *psy); ++ OutputDebugString(strTmp); ++ } ++ ++ sprintf(strTmp,"samp frq=%.1f kHz total bitrate=%d kbps\n",s_freq[info->version][info->sampling_frequency],bitrate[info->version][info->lay-1][info->bitrate_index]); ++ OutputDebugString(strTmp); ++ ++ sprintf(strTmp,"de-emph=%d c/right=%d orig=%d errprot=%s\n",info->emphasis, info->copyright, info->original,((info->error_protection) ? "on" : "off")); ++ OutputDebugString(strTmp); ++ ++ sprintf(strTmp,"16 Khz cut off is %s\n",(sfb21)?"enabled":"disabled"); ++ OutputDebugString(strTmp); ++ ++ sprintf(strTmp,"Fast mode is %s\n",(fast_mode)?"enabled":"disabled"); ++ OutputDebugString(strTmp); ++ ++ sprintf(strTmp,"Force ms %s\n",(force_ms)?"enabled":"disabled"); ++ OutputDebugString(strTmp); ++ ++ sprintf(strTmp,"GPsycho acoustic model is %s\n",(gpsycho)?"enabled":"disabled"); ++ OutputDebugString(strTmp); ++ ++ ++ sprintf(strTmp,"input file: '%s' output file: '%s'\n", inPath, outPath); ++ OutputDebugString(strTmp); ++ ++ sprintf(strTmp,"Encoding %s to %s\n",inPath,outPath); ++ OutputDebugString(strTmp); ++ ++ sprintf(strTmp,"Encoding as %.1f kHz %d kbps %s MPEG-1 LayerIII file\n",s_freq[info->version][info->sampling_frequency],bitrate[info->version][info->lay-1][info->bitrate_index],mode_names[info->mode]); ++ OutputDebugString(strTmp); ++ } ++ ++ ++ void DispErr(LPSTR strErr) ++ { ++ MessageBox(NULL,strErr,"",MB_OK); ++ } ++ ++ #endif +diff -r -c -N encoder/Dll/BladeMP3EncDLL.def lame3.50/Dll/BladeMP3EncDLL.def +*** encoder/Dll/BladeMP3EncDLL.def Wed Dec 31 17:00:00 1969 +--- lame3.50/Dll/BladeMP3EncDLL.def Mon Oct 18 12:27:22 1999 +*************** +*** 0 **** +--- 1,9 ---- ++ LIBRARY lame_enc.DLL ++ EXPORTS ++ ++ beInitStream @1 ++ beEncodeChunk @2 ++ beDeinitStream @3 ++ beCloseStream @4 ++ beVersion @5 ++ beWriteVBRHeader @6 +\ No newline at end of file +diff -r -c -N encoder/Dll/BladeMP3EncDLL.h lame3.50/Dll/BladeMP3EncDLL.h +*** encoder/Dll/BladeMP3EncDLL.h Wed Dec 31 17:00:00 1969 +--- lame3.50/Dll/BladeMP3EncDLL.h Fri Sep 10 09:52:00 1999 +*************** +*** 0 **** +--- 1,150 ---- ++ /* ++ ++ bladedll.h ++ ++ +++++++++++++++++++++++++++ ++ + Blade's Encoder DLL + ++ +++++++++++++++++++++++++++ ++ ++ ------------------------------------------------------ ++ - Version 1.00 (7 November 1998) - Jukka Poikolainen - ++ ------------------------------------------------------ ++ ++ Initial version ++ ++ ------------------------------------------------------ ++ - Version x.xx (x xxxxxxxx xxxx) - xxxxx xxxxxxxxxxx - ++ ------------------------------------------------------ ++ ++ */ ++ ++ #ifndef ___BLADEDLL_H_INCLUDED___ ++ #define ___BLADEDLL_H_INCLUDED___ ++ ++ #pragma pack(push) ++ #pragma pack(1) ++ ++ /* encoding formats */ ++ ++ #define BE_CONFIG_MP3 0 ++ ++ /* type definitions */ ++ ++ typedef unsigned long HBE_STREAM; ++ typedef HBE_STREAM *PHBE_STREAM; ++ typedef unsigned long BE_ERR; ++ ++ /* error codes */ ++ ++ #define BE_ERR_SUCCESSFUL 0x00000000 ++ #define BE_ERR_INVALID_FORMAT 0x00000001 ++ #define BE_ERR_INVALID_FORMAT_PARAMETERS 0x00000002 ++ #define BE_ERR_NO_MORE_HANDLES 0x00000003 ++ #define BE_ERR_INVALID_HANDLE 0x00000004 ++ ++ /* other constants */ ++ ++ #define BE_MAX_HOMEPAGE 256 ++ ++ /* format specific variables */ ++ ++ #define BE_MP3_MODE_STEREO 0 ++ #define BE_MP3_MODE_JSTEREO 1 ++ #define BE_MP3_MODE_DUALCHANNEL 2 ++ #define BE_MP3_MODE_MONO 3 ++ ++ #ifdef _BLADEDLL ++ #undef FLOAT ++ #include <Windows.h> ++ #endif ++ ++ typedef struct { ++ ++ DWORD dwConfig; // BE_CONFIG_XXXXX ++ // Currently only BE_CONFIG_MP3 is supported ++ union { ++ ++ struct { ++ ++ DWORD dwSampleRate; // 48000, 44100 and 32000 allowed ++ BYTE byMode; // BE_MP3_MODE_STEREO, BE_MP3_MODE_DUALCHANNEL, BE_MP3_MODE_MONO ++ WORD wBitrate; // 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256 and 320 allowed ++ BOOL bPrivate; ++ BOOL bCRC; ++ BOOL bCopyright; ++ BOOL bOriginal; ++ ++ } mp3; // BE_CONFIG_MP3 ++ ++ struct { ++ ++ DWORD dwSampleRate; ++ BYTE byMode; ++ WORD wBitrate; ++ BYTE byEncodingMethod; ++ ++ } aac; ++ ++ } format; ++ ++ } BE_CONFIG, *PBE_CONFIG; ++ ++ typedef struct { ++ ++ // BladeEnc DLL Version number ++ ++ BYTE byDLLMajorVersion; ++ BYTE byDLLMinorVersion; ++ ++ // BladeEnc Engine Version Number ++ ++ BYTE byMajorVersion; ++ BYTE byMinorVersion; ++ ++ // DLL Release date ++ ++ BYTE byDay; ++ BYTE byMonth; ++ WORD wYear; ++ ++ // BladeEnc Homepage URL ++ ++ CHAR zHomepage[BE_MAX_HOMEPAGE + 1]; ++ ++ } BE_VERSION, *PBE_VERSION; ++ ++ #ifndef _BLADEDLL ++ ++ typedef BE_ERR (*BEINITSTREAM) (PBE_CONFIG, PDWORD, PDWORD, PHBE_STREAM); ++ typedef BE_ERR (*BEENCODECHUNK) (HBE_STREAM, DWORD, PSHORT, PBYTE, PDWORD); ++ typedef BE_ERR (*BEDEINITSTREAM) (HBE_STREAM, PBYTE, PDWORD); ++ typedef BE_ERR (*BECLOSESTREAM) (HBE_STREAM); ++ typedef VOID (*BEVERSION) (PBE_VERSION); ++ ++ #define TEXT_BEINITSTREAM "beInitStream" ++ #define TEXT_BEENCODECHUNK "beEncodeChunk" ++ #define TEXT_BEDEINITSTREAM "beDeinitStream" ++ #define TEXT_BECLOSESTREAM "beCloseStream" ++ #define TEXT_BEVERSION "beVersion" ++ ++ /* ++ BE_ERR beInitStream(PBE_CONFIG pbeConfig, PDWORD dwSamples, PDWORD dwBufferSize, PHBE_STREAM phbeStream); ++ BE_ERR beEncodeChunk(HBE_STREAM hbeStream, DWORD nSamples, PSHORT pSamples, PBYTE pOutput, PDWORD pdwOutput); ++ BE_ERR beDeinitStream(HBE_STREAM hbeStream, PBYTE pOutput, PDWORD pdwOutput); ++ BE_ERR beCloseStream(HBE_STREAM hbeStream); ++ VOID beVersion(PBE_VERSION pbeVersion); ++ */ ++ ++ #else ++ ++ __declspec(dllexport) BE_ERR beInitStream(PBE_CONFIG pbeConfig, PDWORD dwSamples, PDWORD dwBufferSize, PHBE_STREAM phbeStream); ++ __declspec(dllexport) BE_ERR beEncodeChunk(HBE_STREAM hbeStream, DWORD nSamples, PSHORT pSamples, PBYTE pOutput, PDWORD pdwOutput); ++ __declspec(dllexport) BE_ERR beDeinitStream(HBE_STREAM hbeStream, PBYTE pOutput, PDWORD pdwOutput); ++ __declspec(dllexport) BE_ERR beCloseStream(HBE_STREAM hbeStream); ++ __declspec(dllexport) VOID beVersion(PBE_VERSION pbeVersion); ++ ++ #endif ++ ++ #pragma pack(pop) ++ ++ #endif +diff -r -c -N encoder/Dll/MP3EncDll.dsp lame3.50/Dll/MP3EncDll.dsp +*** encoder/Dll/MP3EncDll.dsp Wed Dec 31 17:00:00 1969 +--- lame3.50/Dll/MP3EncDll.dsp Mon Oct 18 12:36:24 1999 +*************** +*** 0 **** +--- 1,273 ---- ++ # Microsoft Developer Studio Project File - Name="MP3EncDll" - Package Owner=<4> ++ # Microsoft Developer Studio Generated Build File, Format Version 5.00 ++ # ** DO NOT EDIT ** ++ ++ # TARGTYPE "Win32 (x86) Dynamic-Link Library" 0x0102 ++ ++ CFG=MP3EncDll - Win32 Debug ++ !MESSAGE This is not a valid makefile. To build this project using NMAKE, ++ !MESSAGE use the Export Makefile command and run ++ !MESSAGE ++ !MESSAGE NMAKE /f "MP3EncDll.mak". ++ !MESSAGE ++ !MESSAGE You can specify a configuration when running NMAKE ++ !MESSAGE by defining the macro CFG on the command line. For example: ++ !MESSAGE ++ !MESSAGE NMAKE /f "MP3EncDll.mak" CFG="MP3EncDll - Win32 Debug" ++ !MESSAGE ++ !MESSAGE Possible choices for configuration are: ++ !MESSAGE ++ !MESSAGE "MP3EncDll - Win32 Release" (based on\ ++ "Win32 (x86) Dynamic-Link Library") ++ !MESSAGE "MP3EncDll - Win32 Debug" (based on\ ++ "Win32 (x86) Dynamic-Link Library") ++ !MESSAGE ++ ++ # Begin Project ++ # PROP Scc_ProjName "" ++ # PROP Scc_LocalPath "" ++ CPP=xicl.exe ++ MTL=midl.exe ++ RSC=rc.exe ++ ++ !IF "$(CFG)" == "MP3EncDll - Win32 Release" ++ ++ # PROP BASE Use_MFC 0 ++ # PROP BASE Use_Debug_Libraries 0 ++ # PROP BASE Output_Dir "Release" ++ # PROP BASE Intermediate_Dir "Release" ++ # PROP BASE Target_Dir "" ++ # PROP Use_MFC 0 ++ # PROP Use_Debug_Libraries 0 ++ # PROP Output_Dir "Release" ++ # PROP Intermediate_Dir "Release" ++ # PROP Ignore_Export_Lib 0 ++ # PROP Target_Dir "" ++ # ADD BASE CPP /nologo /MT /W3 /GX /O2 /D "WIN32" /D "NDEBUG" /D "_WINDOWS" /YX /FD /c ++ # ADD CPP /nologo /Zp2 /MT /W3 /GX /O2 /Ob2 /I "..\\" /D "NDEBUG" /D "WIN32" /D "_WINDOWS" /D "_BLADEDLL" /YX /FD -Qmem -Qip /c ++ # ADD BASE MTL /nologo /D "NDEBUG" /mktyplib203 /o NUL /win32 ++ # ADD MTL /nologo /D "NDEBUG" /mktyplib203 /o NUL /win32 ++ # ADD BASE RSC /l 0x409 /d "NDEBUG" ++ # ADD RSC /l 0x409 /d "NDEBUG" ++ BSC32=bscmake.exe ++ # ADD BASE BSC32 /nologo ++ # ADD BSC32 /nologo ++ LINK32=xilink.exe ++ # ADD BASE LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:windows /dll /machine:I386 ++ # ADD LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:windows /dll /map /machine:I386 /out:"..\..\lame_enc.dll" ++ ++ !ELSEIF "$(CFG)" == "MP3EncDll - Win32 Debug" ++ ++ # PROP BASE Use_MFC 0 ++ # PROP BASE Use_Debug_Libraries 1 ++ # PROP BASE Output_Dir "Debug" ++ # PROP BASE Intermediate_Dir "Debug" ++ # PROP BASE Target_Dir "" ++ # PROP Use_MFC 0 ++ # PROP Use_Debug_Libraries 1 ++ # PROP Output_Dir "Debug" ++ # PROP Intermediate_Dir "Debug" ++ # PROP Ignore_Export_Lib 0 ++ # PROP Target_Dir "" ++ # ADD BASE CPP /nologo /MTd /W3 /Gm /GX /Zi /Od /D "WIN32" /D "_DEBUG" /D "_WINDOWS" /YX /FD /c ++ # ADD CPP /nologo /Zp2 /MTd /W3 /Gm /GX /Zi /Od /I "..\\" /D "_DEBUG" /D "WIN32" /D "_WINDOWS" /D "_BLADEDLL" /YX /FD /c ++ # ADD BASE MTL /nologo /D "_DEBUG" /mktyplib203 /o NUL /win32 ++ # ADD MTL /nologo /D "_DEBUG" /mktyplib203 /o NUL /win32 ++ # ADD BASE RSC /l 0x409 /d "_DEBUG" ++ # ADD RSC /l 0x409 /d "_DEBUG" ++ BSC32=bscmake.exe ++ # ADD BASE BSC32 /nologo ++ # ADD BSC32 /nologo ++ LINK32=xilink.exe ++ # ADD BASE LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:windows /dll /debug /machine:I386 /pdbtype:sept ++ # ADD LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:windows /dll /debug /machine:I386 /out:"..\..\lame_enc.dll" /pdbtype:sept ++ # SUBTRACT LINK32 /map ++ ++ !ENDIF ++ ++ # Begin Target ++ ++ # Name "MP3EncDll - Win32 Release" ++ # Name "MP3EncDll - Win32 Debug" ++ # Begin Source File ++ ++ SOURCE=.\BladeMP3EncDLL.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\BladeMP3EncDLL.def ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\BladeMP3EncDLL.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\encoder.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\fastmdct.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\fastmdct.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\fft.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\fft.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\filterbank.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\formatBitstream.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\formatBitstream.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\globalflags.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\gpkplotting.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\gpkplotting.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\gtkanal.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\gtkanal.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\huffman.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\ieeefloat.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\ieeefloat.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE="..\l3bitstream-pvt.h" ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\l3bitstream.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\l3bitstream.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\l3side.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\lame.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\lame.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\portableio.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\portableio.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\psymodel.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\psymodel.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE="..\quantize-pvt.h" ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\quantize.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\quantize.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\reservoir.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\reservoir.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\sqrttab.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\tables.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\tables.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\takehiro.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\util.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\util.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\VbrTag.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\VbrTag.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\version.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\version.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\wavheader.h ++ # End Source File ++ # End Target ++ # End Project +diff -r -c -N encoder/Dll/MP3EncDll.dsw lame3.50/Dll/MP3EncDll.dsw +*** encoder/Dll/MP3EncDll.dsw Wed Dec 31 17:00:00 1969 +--- lame3.50/Dll/MP3EncDll.dsw Fri Sep 10 09:52:00 1999 +*************** +*** 0 **** +--- 1,29 ---- ++ Microsoft Developer Studio Workspace File, Format Version 5.00 ++ # WARNING: DO NOT EDIT OR DELETE THIS WORKSPACE FILE! ++ ++ ############################################################################### ++ ++ Project: "MP3EncDll"=.\MP3EncDll.dsp - Package Owner=<4> ++ ++ Package=<5> ++ {{{ ++ }}} ++ ++ Package=<4> ++ {{{ ++ }}} ++ ++ ############################################################################### ++ ++ Global: ++ ++ Package=<5> ++ {{{ ++ }}} ++ ++ Package=<3> ++ {{{ ++ }}} ++ ++ ############################################################################### ++ +diff -r -c -N encoder/INSTALL lame3.50/INSTALL +*** encoder/INSTALL Wed Jan 22 02:43:12 1997 +--- lame3.50/INSTALL Mon Sep 27 15:10:17 1999 +*************** +*** 1,36 **** + Building the MPEG audio encoding software: + +- If you have gnu make and are on a Unix-type system, you should only have to +- type: + +- ./configure +- make + +! This will create a Makefile customized to your environment, overwriting the +! previous Makefile, using Makefile.in as a template. + +- For other systems you may have to modify the Makefile. Some flags of +- interest are: + +- BS_FORMAT +- Set to BINARY to work with most MPEG audio decoders (and save space). You +- can set it to ASCII, but your bitstream will only work with musicout(). + +! HAVE_NINT +! Define this if your math library already has nint(). + +- NDEBUG +- Define this to turn off all the assertions. Don't define this if you +- are developing code. + +- DEBUG +- Define this to get debugging messages printed to stdout. + +- For non-Unix systems you may have to define other things. Be sure to look +- at common.h. If you build this on a Macintosh, be sure to use a fairly +- recent set of Universal Headers. I have built this code on SunOS 4.1 and +- Linux 1.3 (using gcc 2.6.3), and on the Power Macintosh (using Metrowerks +- CodeWarrior 6). + + +- <mc@fivebats.com> 1995/10/16 +--- 1,53 ---- ++ LAME 3.xx 9/99 Mark Taylor (http://www.sulaco.org/mp3) ++ ++ ======================================================================= + Building the MPEG audio encoding software: ++ ======================================================================= ++ If you are reading this, you have already applied the patch to the ++ dist10 ISO demonstration source (or downloaded a full source ++ distrubtion). If you do not have GTK 1.1.16 or higher, or do not want ++ the graphical frame analyzer (it uses a lot of memory), edit the ++ Makefile and dont define GTK and GTKLIBS. If you have trouble with ++ libncurses/libtermcap, you can also disable that in the Makefile. ++ ++ Now just type: ++ ++ % make ++ ++ If it doesn't work, figure out why and send me the fix! ++ If you are using a newer version of EGCS, check the Makefile. ++ There are some flags which may improve performance. ++ ++ LAME has built in support for 16bit raw pcm, aiff and simple wav ++ formats. Some complex .wav headers will confuse LAME If you have ++ trouble with this, or want to encode from other sound formats, you can ++ now compile LAME to use Erik de Castro Lopo's libsndfile. To use ++ libsndfile in LAME: ++ ++ 1. install libsndfile (see http://www.zip.com.au/~erikd/libsndfile/). ++ 2. set SNDLIB and LIBSNDFILE as shown in the LAME Makefile. ++ 3. make clean ; make ++ ++ Thanks to Albert Faber for adding the libsndfile support! ++ ++ LAME can also be compiled into a BladeEnc compatiable MS Windows .dll ++ file. See the Dll/ directory for details. + + + +! ======================================================================= +! To encode: +! ======================================================================= +! see the file "USAGE" for some usage guidlines. + + + +! ======================================================================= +! Portability +! ======================================================================= +! LAME 3.x has been run on every modern OS. see the Makefile. + + + + + +diff -r -c -N encoder/Makefile lame3.50/Makefile +*** encoder/Makefile Wed Jan 22 02:43:23 1997 +--- lame3.50/Makefile Mon Nov 1 18:23:02 1999 +*************** +*** 1,68 **** +! # Generated automatically from Makefile.in by configure. +! ############################################################################ +! ## ISO MPEG Audio Subgroup Software Simulation Group (1996) +! ## ISO 13818-3 MPEG-2 Audio Decoder - Lower Sampling Frequency Extension +! ## +! ## $Id: Makefile,v 1.2 1996/08/14 23:18:34 rowlands Exp $ +! ## +! ## Makefile for encoder. Requies GNU make. +! ## +! ## $Log: Makefile,v $ +! ## Revision 1.2 1996/08/14 23:18:34 rowlands +! ## Cleanups. +! ## +! ## Revision 1.2 1996/02/14 05:18:05 rowlands +! ## Cleanups. +! ## +! ## Revision 1.1 1996/02/14 04:04:23 rowlands +! ## Initial revision +! ## +! ## Received from Mike Coleman +! ############################################################################ + + CC = gcc + + c_sources = \ +- common.c \ +- encode.c \ + formatBitstream.c \ +! huffman.c \ +! ieeefloat.c \ + l3bitstream.c \ +! l3psy.c \ +! loop.c \ +! mdct.c \ +! musicin.c \ + portableio.c \ +! psy.c \ + reservoir.c \ +! subs.c \ +! tonal.c + + OBJ = $(c_sources:.c=.o) + DEP = $(c_sources:.c=.d) + +- NINT_SWITCH = +- +- CC_SWITCHES = -g -O -DUNIX -DBS_FORMAT=BINARY $(NINT_SWITCH) -DNDEBUG +- +- PGM = encode +- +- LIBS = -lm + + + %.o: %.c + $(CC) $(CC_SWITCHES) -c $< -o $@ + + %.d: %.c +! $(SHELL) -ec '$(CC) -M $(CC_SWITCHES) $< | sed '\''s/$*.o/& $@/g'\'' > $@' + + +! $(PGM): $(OBJ) Makefile +! $(CC) -o $(PGM) $(OBJ) $(LIBS) + + clean: +! -rm $(OBJ) $(DEP) + + tags: TAGS + +--- 1,254 ---- +! # Makefile for LAME 3.xx +! # +! # LAME is reported to work under: +! # Linux (i86), NetBSD 1.3.2 (StrongARM), FreeBSD (i86) +! # Compaq Alpha(OSF, Linux, Tru64 Unix), Sun Solaris, SGI IRIX, +! # OS2 Warp, Macintosh PPC, BeOS, Amiga and even VC++ +! # +! UNAME = $(shell uname) +! ARCH = $(shell uname -m) + ++ ++ # defaults: ++ PGM = lame + CC = gcc ++ CC_OPTS = -O ++ GTK = ++ GTKLIBS = ++ SNDLIB = ++ LIBSNDFILE = ++ LIBS = -lm ++ MAKEDEP = -M ++ BRHIST_SWITCH = ++ LIBTERMCAP = ++ ++ ++ ########################################################################## ++ # Define these to produce a VBR bitrate histogram. Requires libtermcap. ++ # You can also use the more modern ncurses but it doesn't work as well. ++ # If you have any trouble, just dont use termcap/ncurses. ++ ########################################################################## ++ #BRHIST_SWITCH = -DBRHIST ++ #LIBTERMCAP = -ltermcap ++ ++ ++ ########################################################################## ++ # define these to use Erik de Castro Lopo's libsndfile ++ # http://www.zip.com.au/~erikd/libsndfile/ ++ # otherwise LAME can only read 16bit wav, aiff and pcm inputfiles. ++ # Note: at present, libsndfile does not support input from stdin. ++ ########################################################################## ++ #SNDLIB = -DLIBSNDFILE ++ #LIBSNDFILE=-lsndfile ++ # if libsndfile is in a custom location, try: ++ #LIBSNDFILE=-L $(LIBSNDHOME) -lsndfile -I $(LIBSNDHOME) ++ ++ ++ ########################################################################## ++ # define these to use compile in support for the GTK mp3 frame analyzer ++ ########################################################################## ++ #GTK = -DHAVEGTK `gtk-config --cflags` ++ #GTKLIBS = `gtk-config --libs` ++ ++ ++ ########################################################################## ++ # LINUX ++ ########################################################################## ++ ifeq ($(UNAME),Linux) ++ # remove if you do not have GTK or do not want the GTK frame analyzer ++ GTK = -DHAVEGTK `gtk-config --cflags` ++ GTKLIBS = `gtk-config --libs` ++ # Comment out next 2 lines if you want to remove VBR histogram capability ++ BRHIST_SWITCH = -DBRHIST ++ LIBTERMCAP = -ltermcap ++ ++ # for debugging: ++ # CC_OPTS = -UNDEBUG -O -Wall -g -DABORTFP ++ ++ # for lots of debugging: ++ # CC_OPTS = -DDEBUG -UNDEBUG -O -Wall -g -DABORTFP -DNO_LEONID ++ ++ ++ # suggested for gcc-2.7.x ++ # CC_OPTS = -O3 -fomit-frame-pointer -funroll-loops -ffast-math \ ++ # -finline-functions ++ ++ # these options were suggested with egcs-990524 ++ # CC = egcs ++ # CC_OPTS = -O9 -fomit-frame-pointer -march=pentiumpro \ ++ # -ffast-math -funroll-loops \ ++ # -fprofile-arcs -fbranch-probabilities ++ ++ ++ ++ ########################################################################## ++ # LINUX on Digital/Compaq Alpha CPUs ++ ########################################################################## ++ ifeq ($(ARCH),alpha) ++ # double is faster than float on Alpha ++ CC_OPTS = -O4 -Wall -fomit-frame-pointer -ffast-math -funroll-loops \ ++ -mfp-regs -fschedule-insns -fschedule-insns2 \ ++ -finline-functions \ ++ -DFLOAT=double ++ # standard Linux libm ++ LIBS = -lm ++ # optimized libffm (free fast math library) ++ #LIBS = -lffm ++ # Compaq's fast math library ++ #LIBS = -lcpml ++ endif ++ endif ++ ++ ++ ++ ########################################################################## ++ # FreeBSD ++ ########################################################################## ++ ifeq ($(UNAME),FreeBSD) ++ # remove if you do not have GTK or do not want the GTK frame analyzer ++ GTK = -DHAVEGTK `gtk12-config --cflags` ++ GTKLIBS = `gtk12-config --libs` ++ # Comment out next 2 lines if you want to remove VBR histogram capability ++ BRHIST_SWITCH = -DBRHIST ++ LIBTERMCAP = -lncurses ++ ++ endif ++ ++ ++ ########################################################################## ++ # SunOS ++ ########################################################################## ++ ifeq ($(UNAME),SunOS) ++ CC = cc ++ CC_OPTS = -O -xCC ++ MAKEDEP = -xM ++ endif ++ ++ ++ ########################################################################## ++ # SGI ++ ########################################################################## ++ ifeq ($(UNAME),IRIX64) ++ CC = cc ++ endif ++ ++ ++ ########################################################################## ++ # Compaq Alpha running Dec Unix (OSF) ++ ########################################################################## ++ ifeq ($(UNAME),OSF1) ++ CC = cc ++ CC_OPTS = -fast -O3 -std -g3 -non_shared ++ endif ++ ++ ########################################################################## ++ # BeOS ++ ########################################################################## ++ ifeq ($(UNAME),BeOS) ++ CC = $(BE_C_COMPILER) ++ LIBS = ++ ifeq ($(ARCH),BePC) ++ CC_OPTS = -O9 -fomit-frame-pointer -march=pentium \ ++ -mcpu=pentium -ffast-math -funroll-loops \ ++ -fprofile-arcs -fbranch-probabilities ++ else ++ CC_OPTS = -opt all ++ MAKEDEP = -make ++ endif ++ endif ++ ++ ########################################################################## ++ # OS/2 ++ ########################################################################## ++ # Properly installed EMX runtime & development package is a prerequisite. ++ # tools I used: make 3.76.1, uname 1.12, sed 2.05, PD-ksh 5.2.13 ++ # ++ ########################################################################## ++ ifeq ($(UNAME),OS/2) ++ SHELL=sh ++ CC = gcc ++ CC_OPTS = -O3 ++ PGM = lame.exe ++ LIBS = ++ ++ # I use the following for slightly better performance on my Pentium-II ++ # using pgcc-2.91.66: ++ # CC_OPTS = -O6 -ffast-math -funroll-loops -mpentiumpro -march=pentiumpro ++ ++ # Comment out next 2 lines if you want to remove VBR histogram capability ++ BRHIST_SWITCH = -DBRHIST ++ LIBTERMCAP = -ltermcap ++ ++ # Uncomment & inspect the 2 GTK lines to use MP3x GTK frame analyzer. ++ # Properly installed XFree86/devlibs & GTK+ is a prerequisite. ++ # The following works for me using Xfree86/OS2 3.3.5 and GTK+ 1.2.3: ++ # GTK = -DHAVEGTK -IC:/XFree86/include/gtk12 -Zmt -D__ST_MT_ERRNO__ -IC:/XFree86/include/glib12 -IC:/XFree86/include ++ # GTKLIBS = -LC:/XFree86/lib -Zmtd -Zsysv-signals -Zbin-files -lgtk12 -lgdk12 -lgmodule -lglib12 -lXext -lX11 -lshm -lbsd -lsocket -lm ++ endif ++ ++ ++ + ++ # 10/99 added -D__NO_MATH_INLINES to fix a bug in *all* versions of ++ # gcc 2.8+ as of 10/99. ++ ++ CC_SWITCHES = -DNDEBUG -D__NO_MATH_INLINES $(CC_OPTS) $(SNDLIB) $(GTK) $(BRHIST_SWITCH) + c_sources = \ + formatBitstream.c \ +! fastmdct.c \ +! filterbank.c \ +! fft.c \ +! get_audio.c \ + l3bitstream.c \ +! id3tag.c \ +! ieeefloat.c \ +! lame.c \ +! psymodel.c \ + portableio.c \ +! quantize.c \ + reservoir.c \ +! tables.c \ +! takehiro.c \ +! timestatus.c \ +! util.c \ +! VbrTag.c \ +! version.c \ +! gtkanal.c \ +! gpkplotting.c \ +! mpglib/common.c \ +! mpglib/dct64_i386.c \ +! mpglib/decode_i386.c \ +! mpglib/layer3.c \ +! mpglib/tabinit.c \ +! mpglib/interface.c \ +! mpglib/main.c + + OBJ = $(c_sources:.c=.o) + DEP = $(c_sources:.c=.d) + + + + %.o: %.c + $(CC) $(CC_SWITCHES) -c $< -o $@ + + %.d: %.c +! $(SHELL) -ec '$(CC) $(MAKEDEP) $(CC_SWITCHES) $< | sed '\''s;$*.o;& $@;g'\'' > $@' +! +! $(PGM): main.o $(OBJ) Makefile +! $(CC) -o $(PGM) main.o $(OBJ) $(LIBS) $(LIBSNDFILE) $(GTKLIBS) $(LIBTERMCAP) + ++ #$(PGM): main.o libmp3lame.a Makefile ++ # $(CC) -o $(PGM) main.o -L. -lmp3lame $(LIBS) $(LIBSNDFILE) $(GTKLIBS) $(LIBTERMCAP) + +! mp3x: mp3x.o $(OBJ) Makefile +! $(CC) -o mp3x mp3x.o $(OBJ) $(LIBS) $(LIBSNDFILE) $(GTKLIBS) $(LIBTERMCAP) +! +! libmp3lame.a: $(OBJ) Makefile +! ar cr libmp3lame.a $(OBJ) + + clean: +! -rm $(OBJ) $(DEP) $(PGM) main.o mp3x.o mp3x libmp3lame.a \ +! mp3resample.o mp3resample + + tags: TAGS + +*************** +*** 70,72 **** +--- 256,274 ---- + etags -T ${c_sources} + + -include $(DEP) ++ ++ test15: $(PGM) ++ ./lame ../test/castanets.wav ++ cmp -l ../test/castanets.wav.mp3 ../test/castanets.ref15.mp3 | head ++ test15h: $(PGM) ++ ./lame -h ../test/castanets.wav ++ cmp -l ../test/castanets.wav.mp3 ../test/castanets.ref15h.mp3 | head ++ test14: $(PGM) ++ ./lame ../test/castanets.wav ++ cmp -l ../test/castanets.wav.mp3 ../test/castanets.ref14.mp3 | head ++ test14h: $(PGM) ++ ./lame -h ../test/castanets.wav ++ cmp -l ../test/castanets.wav.mp3 ../test/castanets.ref14h.mp3 | head ++ testg: $(PGM) ++ ./lame -h -g ../test/castanets.wav ++ +diff -r -c -N encoder/Makefile.MSVC lame3.50/Makefile.MSVC +*** encoder/Makefile.MSVC Wed Dec 31 17:00:00 1969 +--- lame3.50/Makefile.MSVC Tue Oct 12 21:19:42 1999 +*************** +*** 0 **** +--- 1,79 ---- ++ # Makefile.MSVC: MSVC Makefile for LAME 3.34 ++ ++ PGM = lame ++ CC = cl ++ LN = link ++ ++ # debugging options ++ # CC_OPTS = /nologo /Zi ++ # LN_OPTS = /nologo /debug:full /debugtype:cv /fixed:no ++ ++ # profiling options ++ # CC_OPTS = /nologo /Zi /O2ab2gitwy /G6AFs /DNDEBUG ++ # LN_OPTS = /nologo /debug:full /debugtype:cv /fixed:no /profile ++ ++ # release options ++ CC_OPTS = /nologo /O2ab2gitwy /G6AFs /DNDEBUG ++ LN_OPTS = /nologo ++ ++ GTK = ++ GTKLIBS = ++ SNDLIB = ++ LIBSNDFILE = ++ LIBS = ++ MAKEDEP = ++ TIMER_SWITCH = ++ BRHIST_SWITCH = ++ LIBTERMCAP = ++ ++ CC_SWITCHES = $(CC_OPTS) $(DISTRIB) $(SNDLIB) $(GTK) \ ++ /DBS_FORMAT=BINARY $(TIMER_SWITCH) $(BRHIST_SWITCH) ++ LN_SWITCHES = $(LN_OPTS) ++ ++ c_sources = \ ++ main.c \ ++ formatBitstream.c \ ++ fastmdct.c \ ++ filterbank.c \ ++ fft.c \ ++ get_audio.c \ ++ l3bitstream.c \ ++ id3tag.c \ ++ ieeefloat.c \ ++ lame.c \ ++ psymodel.c \ ++ portableio.c \ ++ quantize.c \ ++ reservoir.c \ ++ tables.c \ ++ takehiro.c \ ++ timestatus.c \ ++ util.c \ ++ VbrTag.c \ ++ version.c \ ++ gtkanal.c \ ++ gpkplotting.c \ ++ mpglib/common.c \ ++ mpglib/dct64_i386.c \ ++ mpglib/decode_i386.c \ ++ mpglib/layer3.c \ ++ mpglib/tabinit.c \ ++ mpglib/interface.c \ ++ mpglib/main.c ++ ++ OBJ = $(c_sources:.c=.obj) ++ ++ .c.obj: ++ @$(CC) $(CC_SWITCHES) /c $< /Fo$@ ++ ++ $(PGM).exe: $(OBJ) Makefile Makefile.MSVC ++ @echo $(PGM).exe ++ @$(LN) $(LN_SWITCHES) $(OBJ) $(LIBS) $(LIBSNDFILE) $(GTKLIBS) \ ++ $(LIBTERMCAP) /out:$(PGM).exe /map:$(PGM).map ++ ++ clean: ++ @-del *.obj ++ @-del dll\*.obj ++ @-del mpglib\*.obj ++ ++ rebuild: clean $(PGM).exe +diff -r -c -N encoder/Makefile.in lame3.50/Makefile.in +*** encoder/Makefile.in Wed Jan 22 02:43:12 1997 +--- lame3.50/Makefile.in Wed Dec 31 17:00:00 1969 +*************** +*** 1,68 **** +- ############################################################################ +- ## ISO MPEG Audio Subgroup Software Simulation Group (1996) +- ## ISO 13818-3 MPEG-2 Audio Decoder - Lower Sampling Frequency Extension +- ## +- ## $Id: Makefile.in,v 1.2 1996/02/14 05:18:05 rowlands Exp $ +- ## +- ## Makefile for encoder. Requies GNU make. +- ## +- ## $Log: Makefile.in,v $ +- ## Revision 1.2 1996/02/14 05:18:05 rowlands +- ## Cleanups. +- ## +- ## Revision 1.1 1996/02/14 04:04:23 rowlands +- ## Initial revision +- ## +- ## Received from Mike Coleman +- ############################################################################ +- +- CC = gcc +- +- c_sources = \ +- common.c \ +- encode.c \ +- formatBitstream.c \ +- huffman.c \ +- ieeefloat.c \ +- l3bitstream.c \ +- l3psy.c \ +- loop.c \ +- mdct.c \ +- musicin.c \ +- portableio.c \ +- psy.c \ +- reservoir.c \ +- subs.c \ +- tonal.c +- +- OBJ = $(c_sources:.c=.o) +- DEP = $(c_sources:.c=.d) +- +- NINT_SWITCH = @NINTSW@ +- +- CC_SWITCHES = -g -O -DUNIX -DBS_FORMAT=BINARY $(NINT_SWITCH) -DNDEBUG +- +- PGM = encode +- +- LIBS = @LIBS@ @MATHLIBS@ +- +- +- %.o: %.c +- $(CC) $(CC_SWITCHES) -c $< -o $@ +- +- %.d: %.c +- $(SHELL) -ec '$(CC) -M $(CC_SWITCHES) $< | sed '\''s/$*.o/& $@/g'\'' > $@' +- +- +- $(PGM): $(OBJ) Makefile +- $(CC) -o $(PGM) $(OBJ) $(LIBS) +- +- clean: +- -rm $(OBJ) $(DEP) +- +- tags: TAGS +- +- TAGS: ${c_sources} +- etags -T ${c_sources} +- +- -include $(DEP) +--- 0 ---- +diff -r -c -N encoder/README lame3.50/README +*** encoder/README Wed Dec 31 17:00:00 1969 +--- lame3.50/README Mon Nov 1 21:04:53 1999 +*************** +*** 0 **** +--- 1,44 ---- ++ LAME 3.xx ++ LAME Ain't an MP3 Encoder ++ http://www.sulaco.org/mp3 ++ September 1999 ++ ++ Written by Mike Cheng (www.uq.net.au/~zzmcheng). Modified ++ and maintained by Mark Taylor (www.sulaco.org/mp3). ++ ++ This is a patch agains the ISO MPEG1 demonstration source. The ++ modifications are distributed under the GNU GENERAL PUBLIC LICENSE ++ (see the file COPYING for details). ++ ++ The graphical frame analyzer uses the MPGLIB decoding engine, from the ++ MPG123 package, written by: Michael Hipp (www.mpg123.de) MPGLIB is ++ released under the GPL, while MPG123 is released under a more ++ restrictive agreement. ++ ++ Various components of the patch are: ++ ++ Copyrights (c) 1999 by Mark Taylor: new psy-model, quantization and bit allocation ++ Copyrights (c) 1998 by Michael Cheng: new transforms and many other improvements ++ Copyrights (c) 1995,1996,1997 by Michael Hipp: mpglib ++ ++ ++ The ISO demonstration source (distributed separately) has its own ++ patent, copyright and distribution issues. The code is freely ++ available, but any commercial use (including distributing free ++ encoders) may require a patent license from Thomson Multimedia. ++ (see http://www.iis.fhg.de/amm/legal/) ++ ++ ============================================================================ ++ ++ see the file "INSTALL" for installation instructions. ++ see the file "USAGE" for the most up-to-date guide to the command ++ line options. ++ ++ There is HTML documentation and a man page in the doc directory. ++ The HTML docs were contributed by Gabriel Bouvigne (www.mp3tech.org). ++ See www.mp3tech.org for more audio compression information and links. ++ ++ ++ ++ ++ +diff -r -c -N encoder/README.WINGTK lame3.50/README.WINGTK +*** encoder/README.WINGTK Wed Dec 31 17:00:00 1969 +--- lame3.50/README.WINGTK Sun Oct 24 21:31:56 1999 +*************** +*** 0 **** +--- 1,56 ---- ++ ++ If you got the LAME source from http://www.sulaco.org/mp3, the ++ libraries and binaries mentioned below are not included. You will ++ need to get the GLIB and GTK developer libraries, available from: ++ http://user.sgic.fi/~tml/gimp/win32/ ++ (or see http://www.gtk.org, and click on the GTK under win32 link). ++ ++ -Mark Taylor ++ May 24,1999 ++ ++ ++ *************************************************************************** ++ ++ ++ ++ May 24,1999 ++ ++ Release notes MP3X for WIN2 ++ ++ Installation instructions: ++ ++ Copy the DLL files: (in GtkWin32Dll.zip) ++ ++ gdk-1.3.dll ++ glib-1.2.dll ++ gtk-1.3.dll ++ gmodule-1.2.dll ++ ++ ++ either into the current working direcory you are running LAME, or ++ copy them in the windows system directory. ++ ++ ++ This is basically it, you can run LAME with the Graphic Psyco Analysis as follows: ++ ++ Lame -g my.wav ++ ++ (just enter Lame at the dos prompt to get all the options) Of course you can ommit the ++ -g optiont to use lame without GPA. ++ ++ ++ ++ How to compile Lame with GPA for Windows 32 platforms ++ ++ In order to compile Lame with the Graphic Psyco Analysis (GPA) with MSVC, you have to have the ++ proper GTK and GLIB include files and lib files. I've put these files together in a ++ zip file (named GtkWin32Lib.zip). Unzip these files into the Lame source directory, make ++ sure you unpack it with the directory options. The files are created in the WinGTK subdirecory ++ Furthermore, you should have at least Lame version 3.05 beta 2. ++ Open the project files in MSVC 5, and just hit the build button. There will be a few warnings, ++ but you can ignore them. ++ ++ ++ ++ Albert L. Faber ++ +diff -r -c -N encoder/TODO lame3.50/TODO +*** encoder/TODO Wed Dec 31 17:00:00 1969 +--- lame3.50/TODO Sat Oct 9 15:15:43 1999 +*************** +*** 0 **** +--- 1,37 ---- ++ When all sfb's have energy below ATH, (analog silence) ++ use a 32kbs frame even if min VBR bandwidth is larger. ++ ++ User specified min & max bandwidth (scalefactor bands) ++ ++ ++ ++ ++ Does stdin work when LAME is compiled to use libsndfile? ++ (new version of libsndfile supports stdin - try this out) ++ ++ automatic downsampling of input file. For example, a 22.05kHz input ++ file encoded at 24kbs will sound better if it is first downsampled ++ to 16kHz. This can be done with sox, but would be nice to have ++ built into LAME. Use linear resample code from sox 12.16 or higher. ++ ++ ++ make use of subblock_gain (crude algorithm enabled with -Z) ++ One method: loop over all possible subblock_gains and ++ see which combination gives best distortion? But this puts a lot ++ of faith in the short block psycho acoustics ++ ++ make use of scalefactor_select (crude algorithm enabled with -Y) ++ ++ 10Hz high-pass filter. Probably wont help much. ++ ++ ++ Modularization: ++ ++ lame_parse_args(): main.c should parse the argument list, and then ++ call lame_setflags() or some other libmp3lame.a routine to ++ set all the various flags. ++ ++ ++ ++ ++ +diff -r -c -N encoder/USAGE lame3.50/USAGE +*** encoder/USAGE Wed Dec 31 17:00:00 1969 +--- lame3.50/USAGE Tue Oct 12 15:30:46 1999 +*************** +*** 0 **** +--- 1,525 ---- ++ ++ % lame [options] inputfile [outputfile] ++ ++ ======================================================================= ++ Examples: ++ ======================================================================= ++ fixed bit rate jstereo 128kbs encoding: ++ % lame sample.wav sample.mp3 ++ ++ fixed bit rate jstereo 128kbs encoding, highest quality: (recommended) ++ % lame -h sample.wav sample.mp3 ++ ++ To disable joint stereo encoding (slightly faster, but less quality at bitrates<=128kbs) ++ % lame -m s sample.wav sample.mp3 ++ ++ Fast encode, low quality (no psycho-acoustics) ++ % lame -f sample.wav sample.mp3 ++ ++ Variable Bitrate (VBR): (use -V n to adjust quality/filesize) ++ % lame -h -v sample.wav sample.mp3 ++ ++ Note: VBR is currently under heavy development. Right now it can ++ often result in too much compression. I would recommend using VBR ++ with a minimum bitrate of 112kbs. This will let LAME increase ++ the bitrate for difficult-to-encode frames, but prevent LAME from ++ being too aggressive for simple frames: ++ ++ % lame -h -v -b 112 sample.wav sample.mp3 ++ ++ ++ ======================================================================= ++ LOW BITRATES ++ ======================================================================= ++ At lower bitrates, (like 24kbs per channel), it is recommended that ++ you use a 16kHz sampling rate. Some commercial encoders (FhG, Xing) ++ will do this automatically. With LAME, you can use Sox, but you have ++ to be careful to get good results. ++ ++ The following information is from John Hayward-Warburton ++ (john@billabong-media.com, http://billabong-media.com) ++ ++ There seem to be two very important points: ++ ++ 1) make sure you have the very latest sox: the `resample' routine ++ was broken in all previous versions before 12.16, and was really ++ quite awful. (latest version: http://home.sprynet.com/~cbagwell/sox.html) ++ ++ 2) use the resample options. The defaults are not good for encoding ++ music or speech. ++ ++ Here are some options which work well, but more experimentation is ++ needed. When available, I'll update this with recommended settings ++ for speech and music. ++ ++ ++ % sox input.wav -c 1 -r 16000 output.wav resample 0.66 ++ ++ % sox INFILE.wav -c 1 -r 22050 OUTFILE.wav polyphase -cutoff 0.45 ++ ++ ++ ++ ++ ++ ======================================================================= ++ STREAMING EXAMPLES ++ ======================================================================= ++ Streaming mono 22.05kHz raw pcm, 24kbs output: ++ % cat inputfile | lame -r -m m -b 24 -s 22.05 - - > output ++ ++ Streaming mono 22.05kHz raw pcm, with downsampling to 16kHz: ++ % cat inputfile | sox -t raw -x -w -s -c 1 -r 22050 - -t raw -x -w -s -c 1 -r 16000 - resample 0.66 | lame -r -m m -b 24 -s 16 - - > output ++ ++ You may or may not need one or both of the "-x" (swap bytes) option in Sox. ++ ++ ++ ++ ++ ======================================================================= ++ ++ For more options, just type: ++ % lame ++ ++ Scripts are included to run lame on multiple files: ++ ++ bach script: mlame Run "mlame -h" for instructions. ++ sh script: auenc Run auenc for instructions ++ ++ ++ ++ ======================================================================= ++ options guide: ++ ======================================================================= ++ These options are explained in detail below. ++ ++ ++ Quality related: ++ ++ -m m/s/j/f mode selection ++ -f disable psycho-acoustics. Encoding much faster but lower quality ++ -k disable scalefactor band 21 cutoff ++ -d allow block types to differ between channels ++ -h higher quality but slower ++ -b n set bitrate (8,16,24,...,320) ++ --athonly ignore psy-model output, only use masking from the ATH ++ --voice experimental voice encoding mode ++ ++ Variable Bit Rate (VBR) ++ -v VBR ++ -V n VBR quality setting (0=highest quality, 9=lowest) ++ -b n specify a minimu allowed bitrate (8,16,24,...,320) ++ -t disable Xing VBR informational tag ++ --nohist disable display of VBR bitrate histogram ++ ++ ++ Experimental (undocumented): may work better or worse: ++ ++ -X n try different quality measures (when comparing quantizations) ++ -Y try to use scalefac_select ++ -Z try to use subblock_gain ++ ++ ++ Operational: ++ ++ -r assume input file is raw PCM ++ -s n input sampling frequency in kHz (for raw PCM input files) ++ --resample n output sampling frequency ++ --mp3input input file is an MP3 file. decode using mpglib/mpg123 ++ -x swap bytes of input file ++ -a downmix stereo input file to mono .mp3 ++ -e n/5/c de-emphasis ++ -p add CRC error protection ++ -c mark the encoded file as copyrighted ++ -o mark the encoded file as a copy ++ -S don't print progress report, VBR histogram ++ -g run MP3x, the graphical frame analyzer ++ ++ ++ id3 tagging: ++ ++ --tt <title> title of song (max 30 chars) ++ --ta <artist> artist who did the song (max 30 chars) ++ --tl <album> album where it came from (max 30 chars) ++ --ty <year> year in which the song/album was made (max 4 chars) ++ --tc <comment> additional info (max 30 chars) ++ ++ ++ ++ ++ ++ ++ ======================================================================= ++ Detailed description of all options in alphabetical order ++ ======================================================================= ++ ++ ++ ======================================================================= ++ downmix ++ ======================================================================= ++ -a ++ ++ mix the stereo input file to mono and encode as mono. ++ ++ This option is only needed in the case of raw PCM stereo input ++ (because LAME cannot determine the number of channels in the input file). ++ To encode a stereo PCM input file as mono, use "lame -m s -a" ++ ++ For WAV and AIFF input files, using "-m m" will always produce a ++ mono .mp3 file from both mono and stereo input. ++ ++ ======================================================================= ++ ATH only ++ ======================================================================= ++ --athonly ++ ++ This option causes LAME to ignore the output of the psy-model and ++ only use masking from the ATH. Might be useful at very high bitrates ++ or for testing the ATH. ++ ++ ++ ++ ++ ======================================================================= ++ bitrate ++ ======================================================================= ++ -b n ++ ++ For MPEG1 (sampling frequencies of 32, 44.1 and 48kHz) ++ n = 32,40,48,56,64,80,96,112,128,160,192,224,256,320 ++ ++ For MPEG2 (sampling frequencies of 16, 22.05 and 24kHz) ++ n = 8,16,24,32,40,48,56,64,80,96,112,128,144,160 ++ ++ ++ The bitrate to be used. Default is 128kbs MPEG1, 80kbs MPEG2. ++ ++ When used with variable bitrate encodings (VBR), -b specifies the ++ minimum bitrate to use. This is useful to prevent LAME VBR from ++ using some very aggressive compression which can cause some distortion ++ due to small flaws in the psycho-acoustic model. ++ ++ ++ ++ ======================================================================= ++ copyright ++ ======================================================================= ++ -c ++ ++ mark the encoded file as copyrighted ++ ++ ++ ++ ======================================================================= ++ block type control ++ ======================================================================= ++ -d ++ ++ Allows the left and right channels to use different block types. ++ Normally this is not allowed, only because the FhG encoder does ++ not seem to allow it either. If anyone finds a sample where -d ++ produces better results, let me know. (mt@sulaco.org) ++ ++ ++ ++ ======================================================================= ++ de-emphasis ++ ======================================================================= ++ -e n/5/c ++ ++ n = (none, default) ++ 5 = 0/15 microseconds ++ c = citt j.17 ++ ++ All this does is set a flag in the bitstream. If you have a PCM ++ input file where one of the above types of (obsolete) emphasis has ++ been applied, you can set this flag in LAME. Then the mp3 decoder ++ should de-emphasize the output during playback, although most ++ decoders ignore this flag. ++ ++ A better solution would be to apply the de-emphasis with a standalone ++ utility before encoding. ++ ++ ++ ++ ======================================================================= ++ fast mode ++ ======================================================================= ++ -f ++ ++ disable psycho-acoustics. Encoding much faster but lower quality ++ ++ ++ ++ ======================================================================= ++ graphical frame analyzer ++ ======================================================================= ++ -g ++ ++ run MP3x, the graphical frame analyzer analysis on the inputfile. The ++ inputfile can be either an .mp3 file or uncompressed audio file. MP3x ++ support must be compiled into LAME, and requires GTK 1.2. ++ Documentation is under the About pull down menu. ++ ++ ++ ++ ======================================================================= ++ high quality ++ ======================================================================= ++ -h ++ ++ use (maybe) some quality improvements ++ ++ LAME 3.21 and up: -h enables specialized mid/side masking thresholds to ++ be used in jstereo mode. Will sound better in jstereo mode ++ but is 20% slower. No effect for mono files ++ ++ ++ ++ ++ ======================================================================= ++ sfb=21 cutoff ++ ======================================================================= ++ -k ++ ++ At 128kbs and less, LAME will remove data in scalefactor band 21. ++ This amounts to a low pass 16kHz filter. -k will disable this ++ feature. However, these coefficients can take up a lot of bits that ++ would be better used for lower, more important frequencies. ++ Furthermore, the MPEG standard does not allow scalefactors or ++ use psycho-acoustic information for this band. ++ ++ ++ ++ ======================================================================= ++ Modes: ++ ======================================================================= ++ ++ -m m mono. ++ -m s stereo ++ -m j jstereo ++ -m f forced mid/side stereo ++ ++ mono is the default mode for mono input files. If "-m m" is specified ++ for a stereo input file, the two channels will be averaged into a mono ++ signal. ++ ++ jstereo is the default mode for stereo files with VBR or fixed bitrates ++ of 160kbs or less. At higher fixed bitrates, the default is stereo. ++ ++ jstereo means the encoder can use (on a frame by frame bases) either ++ regular stereo (just encode left and right channels independently) ++ or mid/side stereo. In mid/side stereo, the mid (L+R) and side (L-R) ++ channels are encoded, and more bits are allocated to the mid channel ++ than the side channel. This will effectively increase the bandwidth ++ if the signal does not have too much stereo separation. ++ ++ Mid/side stereo is basically a trick to increase bandwidth. At 128kbs, ++ it is clearly worth while. At higher bitrates it is less usefull. ++ ++ Using mid/side stereo inappropriately can result in audible ++ compression artifacts. To much switching between mid/side and regular ++ stereo can also sound bad. To determine when to switch to mid/side ++ stereo, LAME uses a much more sophisticated algorithm than that ++ described in the ISO documentation. ++ ++ -m f (forced mid/side stereo) is slightly faster then jstereo. But ++ it should only be used if you are sure every frame of the input file ++ has very little stereo seperation. ++ ++ ++ ++ ======================================================================= ++ MP3 input file ++ ======================================================================= ++ --mp3input ++ ++ Assume the input file is a MP3 file. Usefull for downsampling ++ from one mp3 to another. If the filename ends in ".mp3" LAME will ++ assume it is an MP3. For stdin or MP3 files which dont end in .mp3 ++ you need to use this switch. ++ ++ ++ ======================================================================= ++ disable historgram display ++ ======================================================================= ++ --nohist ++ ++ By default, LAME will display a bitrate histogram while producing ++ VBR mp3 files. This will disable that feature. ++ ++ ++ ======================================================================= ++ non-original ++ ======================================================================= ++ -o ++ ++ mark the encoded file as a copy ++ ++ ++ ======================================================================= ++ CRC error protection ++ ======================================================================= ++ -p ++ ++ turn on CRC error protection. ++ Yes this really does work correctly in LAME. However, it takes ++ 16 bits per frame that would otherwise be used for encoding. ++ ++ ++ ++ ======================================================================= ++ input file is raw pcm ++ ======================================================================= ++ -r ++ ++ Assume the input file is raw pcm. Sampling rate and mono/stereo/jstereo ++ must be specified on the command line. Without -r, LAME will perform ++ several fseek()'s on the input file looking for WAV and AIFF headers. ++ ++ Not supported if LAME is compiled to use LIBSNDFILE. ++ ++ ++ ++ ======================================================================= ++ output sampling frequency in kHZ ++ ======================================================================= ++ --resample n ++ ++ where n = 16, 22.05, 24, 32, 44.1, 48 ++ ++ Output sampling frequency. Resample the input if necessary. ++ If not specified, defaults to the input sampling frequency. ++ (resampling code not yet implemented) ++ ++ ======================================================================= ++ sampling frequency in kHZ ++ ======================================================================= ++ -s n ++ ++ where n = 16, 22.05, 24, 32, 44.1, 48 ++ ++ Required for raw PCM input files. Otherwise it will be determined ++ from the header information in the input file. ++ ++ ++ ++ ======================================================================= ++ silent operation ++ ======================================================================= ++ -S ++ ++ don't print progress report ++ ++ ++ ++ ======================================================================= ++ disable Xing VBR tag ++ ======================================================================= ++ -t ++ ++ Disable writing of the Xing VBR Tag (only valid if -v flag is ++ specified) This tag in embedded in frame 0 of the MP3 file. It lets ++ VBR aware players correctly seek and compute playing times of VBR ++ files. ++ ++ ++ ++ ======================================================================= ++ variable bit rate (VBR) ++ ======================================================================= ++ -v ++ ++ Turn on VBR. There are several ways you can use VBR. I personally ++ like using VBR to get files slightly bigger than 128kbs files, where ++ the extra bits are used for the occasional difficult-to-encode frame. ++ For this, try specifying a minimum bitrate to use with VBR: ++ ++ lame -v -b 112 input.wav output.mp3 ++ ++ If the file is too big, use -V n, where n=0..9 ++ ++ lame -v -V n -b 112 input.wav output.mp3 ++ ++ ++ If you wan to use VBR to get the maximum compression possible, ++ and for this, you can try: ++ ++ lame -v input.wav output.mp3 ++ lame -v -V n input.wav output.mp3 (to very quality/filesize) ++ ++ ++ ++ ++ ======================================================================= ++ VBR quality setting ++ ======================================================================= ++ -V n ++ ++ n=0..9. Specifies the value of VBR_q. default=4. 0=highest quality. ++ ++ ++ ++ How is VBR_q used? ++ ++ OVER = number of scalefactor bands with distortion that exceeds the ++ allowed distortion given by the masking thresholds. OVER is computed ++ by outer_loop, and the masking thresholds are computed by the ++ psycho-acoustic model. ++ ++ VBR_q = the minimum value of OVER which is to be allowed. ++ LAME will choose the smallest bitrate for which OVER <= VBR_q. ++ (a minimum allowed bitrate can be set with -b. default=64kbs) ++ ++ If the frame contains short blocks, then the minimum bitrate is made ++ much larger since the OVER does not adequately measure distortion ++ caused by pre-echo. LAME uses bitrates of at least 160kbs for short ++ blocks to make sure they sound good. ++ ++ *NOTE* No psy-model is perfect, so there can often be distortion which ++ is audible even though the psy-model claims it is not! Thus using a ++ small minimum bitrate can result in some aggressive compression and ++ audible distortion even with -V 0. Thus using -V 0 does not sound ++ better than a fixed 256kbs encoding. For example: suppose in the 1kHz ++ frequency band the psy-model claims 20db of distortion will not be ++ detectable by the human ear, so LAME VBR-0 will compress that ++ frequency band as much as possible and introduce at most 20db of ++ distortion. Using a fixed 256kbit framesize, LAME could end up ++ introducing only 2db of distortion. If the psy-model was correct, ++ they will both sound the same. If the psy-model was wrong, the VBR-0 ++ result can sound worse. ++ ++ ++ ======================================================================= ++ voice encoding mode ++ ======================================================================= ++ --voice ++ ++ An experimental voice encoding mode. Tuned for 44.1kHz input files. ++ ++ ++ ======================================================================= ++ swapbytes ++ ======================================================================= ++ -x ++ ++ swap bytes in the input file. for sorting out little endian/big endian ++ type problems. If your encodings sound like static, try this first. ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ +diff -r -c -N encoder/VbrTag.c lame3.50/VbrTag.c +*** encoder/VbrTag.c Wed Dec 31 17:00:00 1969 +--- lame3.50/VbrTag.c Wed Oct 27 01:15:22 1999 +*************** +*** 0 **** +--- 1,489 ---- ++ /* ++ * Version numbering for LAME. ++ * ++ * Copyright (c) 1999 A.L. Faber ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License as published by ++ * the Free Software Foundation; either version 2, or (at your option) ++ * any later version. ++ * ++ * This program is distributed in the hope that it will be useful, ++ * but WITHOUT ANY WARRANTY; without even the implied warranty of ++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the ++ * GNU General Public License for more details. ++ * ++ * You should have received a copy of the GNU General Public License ++ * along with this program; see the file COPYING. If not, write to ++ * the Free Software Foundation, 675 Mass Ave, Cambridge, MA 02139, USA. ++ */ ++ ++ #include "machine.h" ++ #include <math.h> ++ #include "VbrTag.h" ++ #include "version.h" ++ ++ #ifdef _DEBUG ++ // #define DEBUG_VBRTAG ++ #endif ++ ++ ++ int SizeOfEmptyFrame[2][2]= ++ { ++ {32,17}, ++ {17,9}, ++ }; ++ ++ static u_char pbtStreamBuffer[216]; ++ static long g_Position[NUMTOCENTRIES]; ++ static int nZeroStreamSize=0; ++ static int TotalFrameSize=0; ++ static char VBRTag[]={"Xing"}; ++ //static char VBRTag[]={"Lame"}; ++ ++ ++ int* pVbrFrames=NULL; ++ int nVbrNumFrames=0; ++ int nVbrFrameBufferSize=0; ++ ++ /**************************************************************************** ++ * AddVbrFrame: Add VBR entry, used to fill the VBR the TOC entries ++ * Paramters: ++ * nStreamPos: how many bytes did we write to the bitstream so far ++ * (in Bytes NOT Bits) ++ **************************************************************************** ++ */ ++ void AddVbrFrame(int nStreamPos) ++ { ++ // Simple exponential growing buffer ++ if (pVbrFrames==NULL || nVbrFrameBufferSize==0) ++ { ++ // Start with 100 frames ++ nVbrFrameBufferSize=100; ++ ++ // Allocate them ++ pVbrFrames=(int*)malloc((size_t)(nVbrFrameBufferSize*sizeof(int))); ++ } ++ ++ // Is buffer big enough to store this new frame ++ if (nVbrNumFrames==nVbrFrameBufferSize) ++ { ++ // Guess not, double th e buffer size ++ nVbrFrameBufferSize*=2; ++ ++ // Allocate new buffer ++ pVbrFrames=(int*)realloc(pVbrFrames,(size_t)(nVbrFrameBufferSize*sizeof(int))); ++ } ++ ++ // Store values ++ pVbrFrames[nVbrNumFrames++]=nStreamPos; ++ } ++ ++ ++ /*-------------------------------------------------------------*/ ++ static int ExtractI4(unsigned char *buf) ++ { ++ int x; ++ // big endian extract ++ x = buf[0]; ++ x <<= 8; ++ x |= buf[1]; ++ x <<= 8; ++ x |= buf[2]; ++ x <<= 8; ++ x |= buf[3]; ++ return x; ++ } ++ ++ void CreateI4(unsigned char *buf, int nValue) ++ { ++ // big endian create ++ buf[0]=(nValue>>24)&0xff; ++ buf[1]=(nValue>>16)&0xff; ++ buf[2]=(nValue>> 8)&0xff; ++ buf[3]=(nValue )&0xff; ++ } ++ ++ ++ /*-------------------------------------------------------------*/ ++ int GetVbrTag(VBRTAGDATA *pTagData, unsigned char *buf) ++ { ++ int i, head_flags; ++ int h_id, h_mode, h_sr_index; ++ static int sr_table[4] = { 44100, 48000, 32000, 99999 }; ++ ++ // get Vbr header data ++ pTagData->flags = 0; // clear to null incase fail ++ ++ // get selected MPEG header data ++ h_id = (buf[1] >> 3) & 1; ++ h_sr_index = (buf[2] >> 2) & 3; ++ h_mode = (buf[3] >> 6) & 3; ++ ++ // determine offset of header ++ if( h_id ) ++ { ++ // mpeg1 ++ if( h_mode != 3 ) buf+=(32+4); ++ else buf+=(17+4); ++ } ++ else ++ { ++ // mpeg2 ++ if( h_mode != 3 ) buf+=(17+4); ++ else buf+=(9+4); ++ } ++ ++ if( buf[0] != VBRTag[0] ) return 0; // fail ++ if( buf[1] != VBRTag[1] ) return 0; // header not found ++ if( buf[2] != VBRTag[2] ) return 0; ++ if( buf[3] != VBRTag[3] ) return 0; ++ ++ buf+=4; ++ ++ pTagData->h_id = h_id; ++ ++ pTagData->samprate = sr_table[h_sr_index]; ++ ++ if( h_id == 0 ) ++ pTagData->samprate >>= 1; ++ ++ head_flags = pTagData->flags = ExtractI4(buf); buf+=4; // get flags ++ ++ if( head_flags & FRAMES_FLAG ) ++ { ++ pTagData->frames = ExtractI4(buf); buf+=4; ++ } ++ ++ if( head_flags & BYTES_FLAG ) ++ { ++ pTagData->bytes = ExtractI4(buf); buf+=4; ++ } ++ ++ if( head_flags & TOC_FLAG ) ++ { ++ if( pTagData->toc != NULL ) ++ { ++ for(i=0;i<NUMTOCENTRIES;i++) ++ pTagData->toc[i] = buf[i]; ++ } ++ buf+=NUMTOCENTRIES; ++ } ++ ++ pTagData->vbr_scale = -1; ++ ++ if( head_flags & VBR_SCALE_FLAG ) ++ { ++ pTagData->vbr_scale = ExtractI4(buf); buf+=4; ++ } ++ ++ #ifdef DEBUG_VBRTAG ++ printf("\n\n********************* VBR TAG INFO *****************\n"); ++ printf("tag :%s\n",VBRTag); ++ printf("head_flags :%d\n",head_flags); ++ printf("bytes :%d\n",pTagData->bytes); ++ printf("frames :%d\n",pTagData->frames); ++ printf("VBR Scale :%d\n",pTagData->vbr_scale); ++ printf("toc:\n"); ++ if( pTagData->toc != NULL ) ++ { ++ for(i=0;i<NUMTOCENTRIES;i++) ++ { ++ if( (i%10) == 0 ) printf("\n"); ++ printf(" %3d", (int)(pTagData->toc[i])); ++ } ++ } ++ printf("\n***************** END OF VBR TAG INFO ***************\n"); ++ #endif ++ return 1; // success ++ } ++ ++ ++ /**************************************************************************** ++ * InitVbrTag: Initializes the header, and write empty frame to stream ++ * Paramters: ++ * fpStream: pointer to output file stream ++ * nVersion: 0= MPEG1 1=MPEG2 ++ * nMode : Channel Mode: 0=STEREO 1=JS 2=DS 3=MONO ++ **************************************************************************** ++ */ ++ int InitVbrTag(Bit_stream_struc* pBs,int nVersion, int nMode, int SampIndex) ++ { ++ int i; ++ ++ // Clear Frame position array variables ++ pVbrFrames=NULL; ++ nVbrNumFrames=0; ++ nVbrFrameBufferSize=0; ++ ++ // Clear struct ++ memset(g_Position,0x00,sizeof(g_Position)); ++ ++ // Clear stream buffer ++ memset(pbtStreamBuffer,0x00,sizeof(pbtStreamBuffer)); ++ ++ // Set TOC values to 255 ++ for (i=0;i<NUMTOCENTRIES;i++) ++ { ++ g_Position[i]=-1; ++ } ++ ++ ++ ++ // Reserve the proper amount of bytes ++ if (nMode==3) ++ { ++ nZeroStreamSize=SizeOfEmptyFrame[nVersion][1]+4; ++ } ++ else ++ { ++ nZeroStreamSize=SizeOfEmptyFrame[nVersion][0]+4; ++ } ++ ++ ++ // Xing VBR pretends to be a 48kbs layer III frame. (at 44.1kHz). ++ // what do they do at 48kHz? frame not big enough for Xing header. ++ // let's always embed Xing header inside a 64kbs layer III frame. ++ // this gives us enough room for a LAME version string too. ++ // size determined by sampling frequency (MPEG1) ++ // 32kHz: 216 bytes@48kbs 288bytes@ 64kbs ++ // 44.1kHz: 156 bytes 208bytes@64kbs (+1 if padding = 1) ++ // 48kHz: 144 bytes 192 ++ // ++ // MPEG 2 values are half the above. Need to go to a 128kbs frame ++ // to have room for the table of contents. not yet supported. ++ if (nVersion == 0 ) { ++ int tot; ++ #define XING64 ++ #ifdef XING64 ++ int framesize[3]={208,192,288}; ++ #else ++ int framesize[3]={156,144,216}; ++ #endif ++ if (SampIndex==3) { ++ fprintf(stderr,"illegal sampling frequency index\n"); ++ exit(-1); ++ } ++ TotalFrameSize= framesize[SampIndex]; ++ tot = (nZeroStreamSize+VBRHEADERSIZE); ++ #ifdef XING64 ++ tot += 20; /* extra 20 bytes for LAME & version string */ ++ #endif ++ if (TotalFrameSize < tot ) { ++ fprintf(stderr,"Xing VBR header problem...use -t\n"); ++ exit(-1); ++ } ++ } ++ if (nVersion == 1) { ++ fprintf(stderr,"MPEG 2 Xing VBR header not supported. use -t\n"); ++ exit(-1); ++ } ++ ++ ++ // Put empty bytes into the bitstream ++ for (i=0;i<TotalFrameSize;i++) ++ { ++ // Write a byte to the bitstream ++ putbits(pBs,0,8); ++ } ++ ++ // Success ++ return 0; ++ } ++ ++ ++ ++ /**************************************************************************** ++ * PutVbrTag: Write final VBR tag to the file ++ * Paramters: ++ * lpszFileName: filename of MP3 bit stream ++ * nVbrScale : encoder quality indicator (0..100) ++ **************************************************************************** ++ */ ++ int PutVbrTag(char* lpszFileName,int nVbrScale) ++ { ++ int i; ++ long lFileSize; ++ int nStreamIndex; ++ char abyte; ++ u_char btToc[NUMTOCENTRIES]; ++ FILE *fpStream; ++ char str1[80]; ++ ++ ++ if (nVbrNumFrames==0 || pVbrFrames==NULL) ++ return -1; ++ ++ // Open the bitstream again ++ fpStream=fopen(lpszFileName,"rb+"); ++ ++ // Assert stream is valid ++ if (fpStream==NULL) ++ return -1; ++ ++ // Clear stream buffer ++ memset(pbtStreamBuffer,0x00,sizeof(pbtStreamBuffer)); ++ ++ // Seek to end of file ++ fseek(fpStream,0,SEEK_END); ++ ++ // Get file size ++ lFileSize=ftell(fpStream); ++ ++ // Seek to first real frame ++ fseek(fpStream,(long)TotalFrameSize,SEEK_SET); ++ ++ // Read the header (first valid frame) ++ fread(pbtStreamBuffer,4,1,fpStream); ++ ++ /* the default VBR header. 48kbs layer III, no padding, no crc */ ++ /* but sampling freq, mode andy copyright/copy protection taken */ ++ /* from first valid frame */ ++ pbtStreamBuffer[0]=(u_char) 0xff; ++ pbtStreamBuffer[1]=(u_char) 0xfb; ++ abyte = pbtStreamBuffer[2] & (char) 0x0c; ++ #ifdef XING64 ++ pbtStreamBuffer[2]=(char) 0x50 | abyte; /* 64kbs frame */ ++ #else ++ pbtStreamBuffer[2]=(char) 0x30 | abyte; /* 48kbs frame */ ++ #endif ++ ++ // Seek to the beginning of the stream ++ fseek(fpStream,0,SEEK_SET); ++ ++ // Clear all TOC entries ++ memset(btToc,0,sizeof(btToc)); ++ ++ #if 0 ++ // Set unassigned toc value with its neighbor value ++ for (i=0;i<nVbrNumFrames;i++) ++ { ++ // Calculate percentage ++ int nPercent= i*(NUMTOCENTRIES-1)/nVbrNumFrames; ++ ++ // Calculate relative file postion, normalized to 0..255 ++ float fRelStreamPos=255*(float)pVbrFrames[i]/(float)lFileSize; ++ ++ // Just to be safe ++ if (nPercent>=NUMTOCENTRIES ) nPercent=NUMTOCENTRIES-1; ++ if (fRelStreamPos>255) fRelStreamPos=255; ++ ++ // Assign toc entry value ++ btToc[nPercent]=(u_char)fRelStreamPos; ++ } ++ #endif ++ for (i=1;i<NUMTOCENTRIES;i++) // Don't touch zero point... ++ { ++ // Calculate frame from given percentage ++ int frameNum=floor(0.01*i*nVbrNumFrames); ++ ++ // Calculate relative file postion, normalized to 0..256!(?) ++ float fRelStreamPos=256.0*(float)pVbrFrames[frameNum]/(float)lFileSize; ++ ++ // Just to be safe ++ if (fRelStreamPos>255) fRelStreamPos=255; ++ ++ // Assign toc entry value ++ btToc[i]=(u_char) fRelStreamPos; ++ } ++ ++ ++ ++ ++ // Fix up empty fields (if we have less than 100 frames) ++ for (i=1;i<NUMTOCENTRIES ;i++) ++ { ++ if (btToc[i]==0) ++ btToc[i]=btToc[i-1]; ++ } ++ ++ ++ // Start writing the tag after the zero frame ++ nStreamIndex=nZeroStreamSize; ++ ++ // Put Vbr tag ++ pbtStreamBuffer[nStreamIndex++]=VBRTag[0]; ++ pbtStreamBuffer[nStreamIndex++]=VBRTag[1]; ++ pbtStreamBuffer[nStreamIndex++]=VBRTag[2]; ++ pbtStreamBuffer[nStreamIndex++]=VBRTag[3]; ++ ++ // Put header flags ++ CreateI4(&pbtStreamBuffer[nStreamIndex],FRAMES_FLAG+BYTES_FLAG+TOC_FLAG+VBR_SCALE_FLAG); ++ nStreamIndex+=4; ++ ++ // Put Total Number of frames ++ CreateI4(&pbtStreamBuffer[nStreamIndex],nVbrNumFrames); ++ nStreamIndex+=4; ++ ++ // Put Total file size ++ CreateI4(&pbtStreamBuffer[nStreamIndex],(int)lFileSize); ++ nStreamIndex+=4; ++ ++ // Put TOC ++ memcpy(&pbtStreamBuffer[nStreamIndex],btToc,sizeof(btToc)); ++ nStreamIndex+=sizeof(btToc); ++ ++ // Put VBR SCALE ++ CreateI4(&pbtStreamBuffer[nStreamIndex],nVbrScale); ++ nStreamIndex+=4; ++ ++ // Put LAME id ++ sprintf(str1,"LAME%s",get_lame_version()); ++ strncpy((char *)&pbtStreamBuffer[nStreamIndex],str1,(size_t) 20); ++ nStreamIndex+=20; ++ ++ ++ #ifdef DEBUG_VBRTAG ++ { ++ VBRTAGDATA TestHeader; ++ GetVbrTag(&TestHeader,pbtStreamBuffer); ++ } ++ #endif ++ ++ // Put it all to disk again ++ if (fwrite(pbtStreamBuffer,TotalFrameSize,1,fpStream)!=1) ++ { ++ // Could not write header ++ return -1; ++ } ++ ++ // CLose bitstream ++ fclose(fpStream); ++ ++ // Save to delete the frame buffer ++ free(pVbrFrames); ++ pVbrFrames=NULL; ++ ++ return 0; // success ++ } ++ ++ /*-------------------------------------------------------------*/ ++ int SeekPoint(unsigned char TOC[NUMTOCENTRIES], int file_bytes, float percent) ++ { ++ // interpolate in TOC to get file seek point in bytes ++ int a, seekpoint; ++ float fa, fb, fx; ++ ++ ++ if( percent < 0.0f ) percent = 0.0f; ++ if( percent > 100.0f ) percent = 100.0f; ++ ++ a = (int)percent; ++ if( a > 99 ) a = 99; ++ fa = TOC[a]; ++ if( a < 99 ) { ++ fb = TOC[a+1]; ++ } ++ else { ++ fb = 256.0f; ++ } ++ ++ ++ fx = fa + (fb-fa)*(percent-a); ++ ++ seekpoint = (int)((1.0f/256.0f)*fx*file_bytes); ++ ++ ++ return seekpoint; ++ } ++ /*-------------------------------------------------------------*/ +diff -r -c -N encoder/VbrTag.h lame3.50/VbrTag.h +*** encoder/VbrTag.h Wed Dec 31 17:00:00 1969 +--- lame3.50/VbrTag.h Sun Oct 17 12:30:36 1999 +*************** +*** 0 **** +--- 1,83 ---- ++ /* ++ * Version numbering for LAME. ++ * ++ * Copyright (c) 1999 A.L. Faber ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License as published by ++ * the Free Software Foundation; either version 2, or (at your option) ++ * any later version. ++ * ++ * This program is distributed in the hope that it will be useful, ++ * but WITHOUT ANY WARRANTY; without even the implied warranty of ++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the ++ * GNU General Public License for more details. ++ * ++ * You should have received a copy of the GNU General Public License ++ * along with this program; see the file COPYING. If not, write to ++ * the Free Software Foundation, 675 Mass Ave, Cambridge, MA 02139, USA. ++ */ ++ ++ #ifndef VRBTAG_H_INCLUDED ++ #define VRBTAG_H_INCLUDED ++ ++ /* ----------------------------------------------------------- ++ * A Vbr header may be present in the ancillary ++ * data field of the first frame of an mp3 bitstream ++ * The Vbr header (optionally) contains ++ * frames total number of audio frames in the bitstream ++ * bytes total number of bytes in the bitstream ++ * toc table of contents ++ ++ * toc (table of contents) gives seek points ++ * for random access ++ * the ith entry determines the seek point for ++ * i-percent duration ++ * seek point in bytes = (toc[i]/256.0) * total_bitstream_bytes ++ * e.g. half duration seek point = (toc[50]/256.0) * total_bitstream_bytes ++ */ ++ ++ #include "l3bitstream.h" ++ ++ #define FRAMES_FLAG 0x0001 ++ #define BYTES_FLAG 0x0002 ++ #define TOC_FLAG 0x0004 ++ #define VBR_SCALE_FLAG 0x0008 ++ ++ #define NUMTOCENTRIES 100 ++ ++ #define FRAMES_AND_BYTES (FRAMES_FLAG | BYTES_FLAG) ++ ++ // structure to receive extracted header ++ // toc may be NULL ++ typedef struct ++ { ++ int h_id; // from MPEG header, 0=MPEG2, 1=MPEG1 ++ int samprate; // determined from MPEG header ++ int flags; // from Vbr header data ++ int frames; // total bit stream frames from Vbr header data ++ int bytes; // total bit stream bytes from Vbr header data ++ int vbr_scale; // encoded vbr scale from Vbr header data ++ u_char toc[NUMTOCENTRIES]; // may be NULL if toc not desired ++ } VBRTAGDATA; ++ ++ // 4 bytes for Header Tag ++ // 4 bytes for Header Flags ++ // 100 bytes for entry (NUMTOCENTRIES) ++ // 4 bytes for FRAME SIZE ++ // 4 bytes for STREAM_SIZE ++ // 4 bytes for VBR SCALE. a VBR quality indicator: 0=best 100=worst ++ // 20 bytes for LAME tag. for example, "LAME3.12 (beta 6)" ++ // ___________ ++ // 140 bytes ++ ++ #define VBRHEADERSIZE (NUMTOCENTRIES+4+4+4+4+4) ++ ++ int GetVbrTag(VBRTAGDATA *pTagData, unsigned char *buf); ++ int SeekPoint(unsigned char TOC[NUMTOCENTRIES], int file_bytes, float percent); ++ int InitVbrTag(Bit_stream_struc* pBs,int nVersion,int nMode, int SampIndex); ++ int PutVbrTag(char* lpszFileName,int nVbrScale); ++ void AddVbrFrame(int nStreamPos); ++ ++ ++ #endif +diff -r -c -N encoder/auenc lame3.50/auenc +*** encoder/auenc Wed Dec 31 17:00:00 1969 +--- lame3.50/auenc Fri Sep 10 09:52:07 1999 +*************** +*** 0 **** +--- 1,39 ---- ++ #!/bin/sh ++ # ++ # auenc -- version 0.1 ++ # ++ # A wrapper for lame to encode multiple files. By default, a .wav ++ # extension is removed and replaced by .mp3 . ++ # ++ # (C) 1999 Gerhard Wesp <gwesp@cosy.sbg.ac.at> under the GPL. ++ ++ # set the variables below according to your taste ++ LAME=/usr/bin/lame ++ LAME_OPTS="-S -h -v -V 0 -b 256" # high quality, silent operation ++ ++ if [ $# -lt 1 ] ; then ++ exec 1>&2 ++ cat << _EOF_ ++ usage: $0 [options] file... ++ options: ++ -d --delete: delete original file after successful encoding ++ _EOF_ ++ exit 1 ++ fi ++ ++ unset DELETE ++ case "$1" in ( -d | --delete ) ++ DELETE=1 ++ shift ;; ++ esac ++ ++ for f in $* ; do ++ if ! $LAME $LAME_OPTS "$f" `basename "$f" .wav`.mp3 ; then ++ exec 1>&2 ++ echo "encoding of $f failed, aborting..." ++ exit 1 ++ fi ++ if [ -n "$DELETE" ] ; then ++ rm -f "$f" ++ fi ++ done +diff -r -c -N encoder/common.c lame3.50/common.c +*** encoder/common.c Wed Jan 22 02:43:13 1997 +--- lame3.50/common.c Wed Dec 31 17:00:00 1969 +*************** +*** 1,1439 **** +- /********************************************************************** +- * ISO MPEG Audio Subgroup Software Simulation Group (1996) +- * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- * +- * $Id: common.c,v 1.2 1997/01/19 22:28:29 rowlands Exp $ +- * +- * $Log: common.c,v $ +- * Revision 1.2 1997/01/19 22:28:29 rowlands +- * Layer 3 bug fixes from Seymour Shlien +- * +- * Revision 1.1 1996/02/14 04:04:23 rowlands +- * Initial revision +- * +- * Received from Mike Coleman +- **********************************************************************/ +- /********************************************************************** +- * date programmers comment * +- * 2/25/91 Doulas Wong, start of version 1.0 records * +- * Davis Pan * +- * 5/10/91 W. Joseph Carter Created this file for all common * +- * functions and global variables. * +- * Ported to Macintosh and Unix. * +- * Added Jean-Georges Fritsch's * +- * "bitstream.c" package. * +- * Added routines to handle AIFF PCM * +- * sound files. * +- * Added "mem_alloc()" and "mem_free()" * +- * routines for memory allocation * +- * portability. * +- * Added routines to convert between * +- * Apple SANE extended floating point * +- * format and IEEE double precision * +- * floating point format. For AIFF. * +- * 02jul91 dpwe (Aware Inc) Moved allocation table input here; * +- * Tables read from subdir TABLES_PATH. * +- * Added some debug printout fns (Write*)* +- * 7/10/91 Earle Jennings replacement of the one float by FLOAT * +- * port to MsDos from MacIntosh version * +- * 8/ 5/91 Jean-Georges Fritsch fixed bug in open_bit_stream_r() * +- *10/ 1/91 S.I. Sudharsanan, Ported to IBM AIX platform. * +- * Don H. Lee, * +- * Peter W. Farrett * +- *10/3/91 Don H. Lee implemented CRC-16 error protection * +- * newly introduced functions are * +- * I_CRC_calc, II_CRC_calc and * +- * update_CRC. Additions and revisions * +- * are marked with dhl for clarity * +- *10/18/91 Jean-Georges Fritsch fixed bug in update_CRC(), * +- * II_CRC_calc() and I_CRC_calc() * +- * 2/11/92 W. Joseph Carter Ported new code to Macintosh. Most * +- * important fixes involved changing * +- * 16-bit ints to long or unsigned in * +- * bit alloc routines for quant of 65535 * +- * and passing proper function args. * +- * Removed "Other Joint Stereo" option * +- * and made bitrate be total channel * +- * bitrate, irrespective of the mode. * +- * Fixed many small bugs & reorganized. * +- * 3/20/92 Jean-Georges Fritsch fixed bug in start-of-frame search * +- * 6/15/92 Juan Pineda added refill_buffer(bs) "n" * +- * initialization * +- * 7/08/92 Susanne Ritscher MS-DOS, MSC6.0 port fixes * +- * 7/27/92 Mike Li (re-)Port to MS-DOS * +- * 8/19/92 Soren H. Nielsen Fixed bug in I_CRC_calc and in * +- * II_CRC_calc. Added function: new_ext * +- * for better MS-DOS compatability * +- * 3/10/93 Kevin Peterson changed aiff_read_headers to handle * +- * chunks in any order. now returns * +- * position of sound data in file. * +- * 3/31/93 Jens Spille changed IFF_* string compares to use * +- * strcmp() * +- * 5/30/93 Masahiro Iwadare removed the previous modification * +- * for UNIX. * +- * 8/27/93 Seymour Shlien, Fixes in Unix and MSDOS ports, * +- * Daniel Lauzon, and * +- * Bill Truerniet * +- *--------------------------------------------------------------------* +- * 8/24/93 Masahiro Iwadare Included IS modification in Layer III.* +- * Changed for 1 pass decoding. * +- * 9/07/93 Toshiyuki Ishino Integrated Layer III with Ver 3.9. * +- *--------------------------------------------------------------------* +- * 11/20/93 Masahiro Iwadare Integrated Layer III with Ver 4.0. * +- *--------------------------------------------------------------------* +- * 7/14/94 Juergen Koller rewind of bitbuffer added * +- * 6/12/95 Soeren H. Nielsen Bug fix in new_ext(). * +- * 7/11/95 Soeren H. Nielsen Changes for MPEG-2 LSF Layer I and II * +- *--------------------------------------------------------------------* +- * 8/02/95 mc@fivebats.com Added code to determine byte-order, * +- * fixes to AIFF routines. Modified Mac * +- * code to work with new Apple headers * +- **********************************************************************/ +- +- /*********************************************************************** +- * +- * Global Include Files +- * +- ***********************************************************************/ +- +- #include "common.h" +- #include <string.h> /* 1995-07-11 shn */ +- #include <ctype.h> +- #include <stdlib.h> +- +- /*********************************************************************** +- * +- * Global Variable Definitions +- * +- ***********************************************************************/ +- +- char *mode_names[4] = { "stereo", "j-stereo", "dual-ch", "single-ch" }; +- char *layer_names[3] = { "I", "II", "III" }; +- char *version_names[2] = { "MPEG-2 LSF", "MPEG-1" }; +- +- /* 1: MPEG-1, 0: MPEG-2 LSF, 1995-07-11 shn */ +- double s_freq[2][4] = {{22.05, 24, 16, 0}, {44.1, 48, 32, 0}}; +- +- /* 1: MPEG-1, 0: MPEG-2 LSF, 1995-07-11 shn */ +- int bitrate[2][3][15] = { +- {{0,32,48,56,64,80,96,112,128,144,160,176,192,224,256}, +- {0,8,16,24,32,40,48,56,64,80,96,112,128,144,160}, +- {0,8,16,24,32,40,48,56,64,80,96,112,128,144,160}}, +- {{0,32,64,96,128,160,192,224,256,288,320,352,384,416,448}, +- {0,32,48,56,64,80,96,112,128,160,192,224,256,320,384}, +- {0,32,40,48,56,64,80,96,112,128,160,192,224,256,320}} +- }; +- +- double FAR multiple[64] = { +- 2.00000000000000, 1.58740105196820, 1.25992104989487, +- 1.00000000000000, 0.79370052598410, 0.62996052494744, 0.50000000000000, +- 0.39685026299205, 0.31498026247372, 0.25000000000000, 0.19842513149602, +- 0.15749013123686, 0.12500000000000, 0.09921256574801, 0.07874506561843, +- 0.06250000000000, 0.04960628287401, 0.03937253280921, 0.03125000000000, +- 0.02480314143700, 0.01968626640461, 0.01562500000000, 0.01240157071850, +- 0.00984313320230, 0.00781250000000, 0.00620078535925, 0.00492156660115, +- 0.00390625000000, 0.00310039267963, 0.00246078330058, 0.00195312500000, +- 0.00155019633981, 0.00123039165029, 0.00097656250000, 0.00077509816991, +- 0.00061519582514, 0.00048828125000, 0.00038754908495, 0.00030759791257, +- 0.00024414062500, 0.00019377454248, 0.00015379895629, 0.00012207031250, +- 0.00009688727124, 0.00007689947814, 0.00006103515625, 0.00004844363562, +- 0.00003844973907, 0.00003051757813, 0.00002422181781, 0.00001922486954, +- 0.00001525878906, 0.00001211090890, 0.00000961243477, 0.00000762939453, +- 0.00000605545445, 0.00000480621738, 0.00000381469727, 0.00000302772723, +- 0.00000240310869, 0.00000190734863, 0.00000151386361, 0.00000120155435, +- 1E-20 +- }; +- +- enum byte_order NativeByteOrder = order_unknown; +- +- /*********************************************************************** +- * +- * Global Function Definitions +- * +- ***********************************************************************/ +- +- /* The system uses a variety of data files. By opening them via this +- function, we can accommodate various locations. */ +- +- FILE *OpenTableFile(name) +- char *name; +- { +- char fulname[80]; +- char *envdir; +- FILE *f; +- +- fulname[0] = '\0'; +- +- #ifdef TABLES_PATH +- strcpy(fulname, TABLES_PATH); /* default relative path for tables */ +- #endif /* TABLES_PATH */ /* (includes terminal path seperator */ +- +- #ifdef UNIX /* envir. variables for UNIX only */ +- { +- char *getenv(); +- +- envdir = getenv(MPEGTABENV); /* check for environment */ +- if(envdir != NULL) +- strcpy(fulname, envdir); +- strcat(fulname, PATH_SEPARATOR); /* add a "/" on the end */ +- } +- #endif /* UNIX */ +- +- strcat(fulname, name); +- if( (f=fopen(fulname,"r"))==NULL ) { +- fprintf(stderr,"OpenTable: could not find %s\n", fulname); +- +- #ifdef UNIX +- if(envdir != NULL) +- fprintf(stderr,"Check %s directory '%s'\n",MPEGTABENV, envdir); +- else +- fprintf(stderr,"Check local directory './%s' or setenv %s\n", +- TABLES_PATH, MPEGTABENV); +- #else /* not unix : no environment variables */ +- +- #ifdef TABLES_PATH +- fprintf(stderr,"Check local directory './%s'\n",TABLES_PATH); +- #endif /* TABLES_PATH */ +- +- #endif /* UNIX */ +- +- } +- return f; +- } +- +- /*********************************************************************** +- * +- * Read one of the data files ("alloc_*") specifying the bit allocation/ +- * quatization parameters for each subband in layer II encoding +- * +- **********************************************************************/ +- +- int read_bit_alloc(table, alloc) /* read in table, return # subbands */ +- int table; +- al_table *alloc; +- { +- unsigned int a, b, c, d, i, j; +- FILE *fp; +- char name[16], t[80]; +- int sblim; +- +- strcpy(name, "alloc_0"); +- +- switch (table) { +- case 0 : name[6] = '0'; break; +- case 1 : name[6] = '1'; break; +- case 2 : name[6] = '2'; break; +- case 3 : name[6] = '3'; break; +- case 4 : name[6] = '4'; break; /* LSF, added 1995-07-11 shn */ +- default : name[6] = '0'; +- } +- +- if (!(fp = OpenTableFile(name))) { +- printf("Please check bit allocation table %s\n", name); +- exit(1); +- } +- +- printf("using bit allocation table %s\n", name); +- +- fgets(t, 80, fp); +- sscanf(t, "%d\n", &sblim); +- while (!feof(fp)) { +- fgets(t, 80, fp); +- sscanf(t, "%d %d %d %d %d %d\n", &i, &j, &a, &b, &c, &d); +- (*alloc)[i][j].steps = a; +- (*alloc)[i][j].bits = b; +- (*alloc)[i][j].group = c; +- (*alloc)[i][j].quant = d; +- } +- fclose(fp); +- return sblim; +- } +- +- /*********************************************************************** +- * +- * Using the decoded info the appropriate possible quantization per +- * subband table is loaded +- * +- **********************************************************************/ +- +- int pick_table(fr_ps) /* choose table, load if necess, return # sb's */ +- frame_params *fr_ps; +- { +- int table, lay, ws, bsp, br_per_ch, sfrq; +- int sblim = fr_ps->sblimit; /* return current value if no load */ +- +- lay = fr_ps->header->lay - 1; +- bsp = fr_ps->header->bitrate_index; +- br_per_ch = bitrate[fr_ps->header->version][lay][bsp] / fr_ps->stereo; +- ws = fr_ps->header->sampling_frequency; +- sfrq = s_freq[fr_ps->header->version][ws]; +- /* decision rules refer to per-channel bitrates (kbits/sec/chan) */ +- if (fr_ps->header->version == MPEG_AUDIO_ID) { /* MPEG-1 */ +- if ((sfrq == 48 && br_per_ch >= 56) || +- (br_per_ch >= 56 && br_per_ch <= 80)) table = 0; +- else if (sfrq != 48 && br_per_ch >= 96) table = 1; +- else if (sfrq != 32 && br_per_ch <= 48) table = 2; +- else table = 3; +- } +- else { /* MPEG-2 LSF */ +- table = 4; +- } +- if (fr_ps->tab_num != table) { +- if (fr_ps->tab_num >= 0) +- mem_free((void **)&(fr_ps->alloc)); +- fr_ps->alloc = (al_table FAR *) mem_alloc(sizeof(al_table), +- "alloc"); +- sblim = read_bit_alloc(fr_ps->tab_num = table, fr_ps->alloc); +- } +- return sblim; +- } +- +- int js_bound(lay, m_ext) +- int lay, m_ext; +- { +- static int jsb_table[3][4] = { { 4, 8, 12, 16 }, { 4, 8, 12, 16}, +- { 0, 4, 8, 16} }; /* lay+m_e -> jsbound */ +- +- if(lay<1 || lay >3 || m_ext<0 || m_ext>3) { +- fprintf(stderr, "js_bound bad layer/modext (%d/%d)\n", lay, m_ext); +- exit(1); +- } +- return(jsb_table[lay-1][m_ext]); +- } +- +- void hdr_to_frps(fr_ps) /* interpret data in hdr str to fields in fr_ps */ +- frame_params *fr_ps; +- { +- layer *hdr = fr_ps->header; /* (or pass in as arg?) */ +- +- fr_ps->actual_mode = hdr->mode; +- fr_ps->stereo = (hdr->mode == MPG_MD_MONO) ? 1 : 2; +- if (hdr->lay == 2) fr_ps->sblimit = pick_table(fr_ps); +- else fr_ps->sblimit = SBLIMIT; +- if(hdr->mode == MPG_MD_JOINT_STEREO) +- fr_ps->jsbound = js_bound(hdr->lay, hdr->mode_ext); +- else +- fr_ps->jsbound = fr_ps->sblimit; +- /* alloc, tab_num set in pick_table */ +- } +- +- void WriteHdr(fr_ps, s) +- frame_params *fr_ps; +- FILE *s; +- { +- layer *info = fr_ps->header; +- +- fprintf(s, "HDR: s=FFF, id=%X, l=%X, ep=%s, br=%X, sf=%X, pd=%X, ", +- info->version, info->lay, ((info->error_protection) ? "on" : "off"), +- info->bitrate_index, info->sampling_frequency, info->padding); +- fprintf(s, "pr=%X, m=%X, js=%X, c=%X, o=%X, e=%X\n", +- info->extension, info->mode, info->mode_ext, +- info->copyright, info->original, info->emphasis); +- fprintf(s, "alg.=%s, layer=%s, tot bitrate=%d, sfrq=%.1f\n", +- version_names[info->version], layer_names[info->lay-1], +- bitrate[info->version][info->lay-1][info->bitrate_index], +- s_freq[info->version][info->sampling_frequency]); +- fprintf(s, "mode=%s, sblim=%d, jsbd=%d, ch=%d\n", +- mode_names[info->mode], fr_ps->sblimit, fr_ps->jsbound, fr_ps->stereo); +- fflush(s); +- } +- +- void WriteBitAlloc(bit_alloc, f_p, s) +- unsigned int bit_alloc[2][SBLIMIT]; +- frame_params *f_p; +- FILE *s; +- { +- int i,j; +- int st = f_p->stereo; +- int sbl = f_p->sblimit; +- int jsb = f_p->jsbound; +- +- fprintf(s, "BITA "); +- for(i=0; i<sbl; ++i) { +- if(i == jsb) fprintf(s,"-"); +- for(j=0; j<st; ++j) +- fprintf(s, "%1x", bit_alloc[j][i]); +- } +- fprintf(s, "\n"); fflush(s); +- } +- +- void WriteScale(bit_alloc, scfsi, scalar, fr_ps, s) +- unsigned int bit_alloc[2][SBLIMIT], scfsi[2][SBLIMIT], scalar[2][3][SBLIMIT]; +- frame_params *fr_ps; +- FILE *s; +- { +- int stereo = fr_ps->stereo; +- int sblimit = fr_ps->sblimit; +- int lay = fr_ps->header->lay; +- int i,j,k; +- +- if(lay == 2) { +- fprintf(s, "SFSI "); +- for (i=0;i<sblimit;i++) for (k=0;k<stereo;k++) +- if (bit_alloc[k][i]) fprintf(s,"%d",scfsi[k][i]); +- fprintf(s, "\nSCFs "); +- for (k=0;k<stereo;k++) { +- for (i=0;i<sblimit;i++) +- if (bit_alloc[k][i]) +- switch (scfsi[k][i]) { +- case 0: for (j=0;j<3;j++) +- fprintf(s,"%2d%c",scalar[k][j][i], +- (j==2)?';':'-'); +- break; +- case 1: +- case 3: fprintf(s,"%2d-",scalar[k][0][i]); +- fprintf(s,"%2d;",scalar[k][2][i]); +- break; +- case 2: fprintf(s,"%2d;",scalar[k][0][i]); +- } +- fprintf(s, "\n"); +- } +- } +- else{ /* lay == 1 */ +- fprintf(s, "SCFs "); +- for (i=0;i<sblimit;i++) for (k=0;k<stereo;k++) +- if (bit_alloc[k][i]) fprintf(s,"%2d;",scalar[k][0][i]); +- fprintf(s, "\n"); +- } +- } +- +- void WriteSamples(ch, sample, bit_alloc, fr_ps, s) +- int ch; +- unsigned int FAR sample[SBLIMIT]; +- unsigned int bit_alloc[SBLIMIT]; +- frame_params *fr_ps; +- FILE *s; +- { +- int i; +- int stereo = fr_ps->stereo; +- int sblimit = fr_ps->sblimit; +- +- fprintf(s, "SMPL "); +- for (i=0;i<sblimit;i++) +- if ( bit_alloc[i] != 0) +- fprintf(s, "%d:", sample[i]); +- if(ch==(stereo-1) ) fprintf(s, "\n"); +- else fprintf(s, "\t"); +- } +- +- int NumericQ(s) /* see if a string lookd like a numeric argument */ +- char *s; +- { +- char c; +- +- while( (c = *s++)!='\0' && isspace((int)c)) /* strip leading ws */ +- ; +- if( c == '+' || c == '-' ) +- c = *s++; /* perhaps skip leading + or - */ +- return isdigit((int)c); +- } +- +- int BitrateIndex(layr, bRate, version) /* convert bitrate in kbps to index */ +- int layr; /* 1 or 2 */ +- int bRate; /* legal rates from 32 to 448 */ +- int version; /* MPEG-1 or MPEG-2 LSF */ +- { +- int index = 0; +- int found = 0; +- +- while(!found && index<15) { +- if(bitrate[version][layr-1][index] == bRate) +- found = 1; +- else +- ++index; +- } +- if(found) +- return(index); +- else { +- fprintf(stderr, "BitrateIndex: %d (layer %d) is not a legal bitrate\n", +- bRate, layr); +- return(-1); /* Error! */ +- } +- } +- +- int SmpFrqIndex(sRate, version) /* convert samp frq in Hz to index */ +- long sRate; /* legal rates 16000, 22050, 24000, 32000, 44100, 48000 */ +- int *version; +- { +- if (sRate == 44100L) { +- *version = MPEG_AUDIO_ID; return(0); +- } +- else if (sRate == 48000L) { +- *version = MPEG_AUDIO_ID; return(1); +- } +- else if (sRate == 32000L) { +- *version = MPEG_AUDIO_ID; return(2); +- } +- else if (sRate == 24000L) { +- *version = MPEG_PHASE2_LSF; return(1); +- } +- else if (sRate == 22050L) { +- *version = MPEG_PHASE2_LSF; return(0); +- } +- else if (sRate == 16000L) { +- *version = MPEG_PHASE2_LSF; return(2); +- } +- else { +- fprintf(stderr, "SmpFrqIndex: %ld is not a legal sample rate\n", sRate); +- return(-1); /* Error! */ +- } +- } +- +- /******************************************************************************* +- * +- * Allocate number of bytes of memory equal to "block". +- * +- *******************************************************************************/ +- +- void FAR *mem_alloc(block, item) +- unsigned long block; +- char *item; +- { +- +- void *ptr; +- +- #ifdef MACINTOSH +- ptr = NewPtr(block); +- #endif +- +- #ifdef MSC60 +- /*ptr = (void FAR *) _fmalloc((unsigned int)block);*/ /* far memory, 92-07-08 sr */ +- ptr = (void FAR *) malloc((unsigned int)block); /* far memory, 93-08-24 ss */ +- #endif +- +- #if ! defined (MACINTOSH) && ! defined (MSC60) +- ptr = (void FAR *) malloc(block); +- #endif +- +- if (ptr != NULL){ +- #ifdef MSC60 +- _fmemset(ptr, 0, (unsigned int)block); /* far memory, 92-07-08 sr */ +- #else +- memset(ptr, 0, block); +- #endif +- } +- else{ +- printf("Unable to allocate %s\n", item); +- exit(0); +- } +- return(ptr); +- } +- +- +- /**************************************************************************** +- * +- * Free memory pointed to by "*ptr_addr". +- * +- *****************************************************************************/ +- +- void mem_free(ptr_addr) +- void **ptr_addr; +- { +- +- if (*ptr_addr != NULL){ +- #ifdef MACINTOSH +- DisposPtr(*ptr_addr); +- #else +- free(*ptr_addr); +- #endif +- *ptr_addr = NULL; +- } +- +- } +- +- /******************************************************************************* +- * +- * Check block of memory all equal to a single byte, else return FALSE +- * +- *******************************************************************************/ +- +- int memcheck(array, test, num) +- char *array; +- int test; /* but only tested as a char (bottom 8 bits) */ +- int num; +- { +- int i=0; +- +- while (array[i] == test && i<num) i++; +- if (i==num) return TRUE; +- else return FALSE; +- } +- +- /***************************************************************************** +- * +- * Routines to determine byte order and swap bytes +- * +- *****************************************************************************/ +- +- enum byte_order DetermineByteOrder() +- { +- char s[ sizeof(long) + 1 ]; +- union +- { +- long longval; +- char charval[ sizeof(long) ]; +- } probe; +- probe.longval = 0x41424344L; /* ABCD in ASCII */ +- strncpy( s, probe.charval, sizeof(long) ); +- s[ sizeof(long) ] = '\0'; +- /* fprintf( stderr, "byte order is %s\n", s ); */ +- if ( strcmp(s, "ABCD") == 0 ) +- return order_bigEndian; +- else +- if ( strcmp(s, "DCBA") == 0 ) +- return order_littleEndian; +- else +- return order_unknown; +- } +- +- void SwapBytesInWords( short *loc, int words ) +- { +- int i; +- short thisval; +- char *dst, *src; +- src = (char *) &thisval; +- for ( i = 0; i < words; i++ ) +- { +- thisval = *loc; +- dst = (char *) loc++; +- dst[0] = src[1]; +- dst[1] = src[0]; +- } +- } +- +- /***************************************************************************** +- * +- * Read Audio Interchange File Format (AIFF) headers. +- * +- *****************************************************************************/ +- +- int aiff_read_headers( FILE *file_ptr, IFF_AIFF *aiff_ptr ) +- { +- int chunkSize, subSize, sound_position; +- +- if ( fseek(file_ptr, 0, SEEK_SET) != 0 ) +- return -1; +- +- if ( Read32BitsHighLow(file_ptr) != IFF_ID_FORM ) +- return -1; +- +- chunkSize = Read32BitsHighLow( file_ptr ); +- +- if ( Read32BitsHighLow(file_ptr) != IFF_ID_AIFF ) +- return -1; +- +- sound_position = 0; +- while ( chunkSize > 0 ) +- { +- chunkSize -= 4; +- switch ( Read32BitsHighLow(file_ptr) ) +- { +- +- case IFF_ID_COMM: +- chunkSize -= subSize = Read32BitsHighLow( file_ptr ); +- aiff_ptr->numChannels = Read16BitsHighLow( file_ptr ); +- subSize -= 2; +- aiff_ptr->numSampleFrames = Read32BitsHighLow( file_ptr ); +- subSize -= 4; +- aiff_ptr->sampleSize = Read16BitsHighLow( file_ptr ); +- subSize -= 2; +- aiff_ptr->sampleRate = ReadIeeeExtendedHighLow( file_ptr ); +- subSize -= 10; +- while ( subSize > 0 ) +- { +- getc( file_ptr ); +- subSize -= 1; +- } +- break; +- +- case IFF_ID_SSND: +- chunkSize -= subSize = Read32BitsHighLow( file_ptr ); +- aiff_ptr->blkAlgn.offset = Read32BitsHighLow( file_ptr ); +- subSize -= 4; +- aiff_ptr->blkAlgn.blockSize = Read32BitsHighLow( file_ptr ); +- subSize -= 4; +- sound_position = ftell( file_ptr ) + aiff_ptr->blkAlgn.offset; +- if ( fseek(file_ptr, (long) subSize, SEEK_CUR) != 0 ) +- return -1; +- aiff_ptr->sampleType = IFF_ID_SSND; +- break; +- +- default: +- chunkSize -= subSize = Read32BitsHighLow( file_ptr ); +- while ( subSize > 0 ) +- { +- getc( file_ptr ); +- subSize -= 1; +- } +- break; +- } +- } +- return sound_position; +- } +- +- /***************************************************************************** +- * +- * Seek past some Audio Interchange File Format (AIFF) headers to sound data. +- * +- *****************************************************************************/ +- +- int aiff_seek_to_sound_data( FILE *file_ptr ) +- { +- if ( fseek(file_ptr, AIFF_FORM_HEADER_SIZE + AIFF_SSND_HEADER_SIZE, SEEK_SET) != 0 ) +- return(-1); +- return(0); +- } +- +- /******************************************************************************* +- * +- * Write Audio Interchange File Format (AIFF) headers. +- * +- *******************************************************************************/ +- +- int aiff_write_headers( FILE *file_ptr, IFF_AIFF *aiff_ptr ) +- { +- int chunkSize; +- int sampleBytes = (aiff_ptr->sampleSize / 8) + (aiff_ptr->sampleSize % 8 ? 1 : 0); +- +- if ( fseek(file_ptr, 0L, SEEK_SET) != 0 ) +- return -1; +- +- /* write FORM chunk */ +- chunkSize = 8 + 18 + 8 + aiff_ptr->numChannels * aiff_ptr->numSampleFrames * sampleBytes; +- Write32BitsHighLow( file_ptr, IFF_ID_FORM ); +- Write32BitsHighLow( file_ptr, chunkSize ); +- Write32BitsHighLow( file_ptr, IFF_ID_AIFF ); +- /* write COMM chunk */ +- Write32BitsHighLow( file_ptr, IFF_ID_COMM ); +- Write32BitsHighLow( file_ptr, 18 ); /* chunk size */ +- Write16BitsHighLow( file_ptr, aiff_ptr->numChannels ); +- Write32BitsHighLow( file_ptr, aiff_ptr->numSampleFrames ); +- Write16BitsHighLow( file_ptr, aiff_ptr->sampleSize ); +- WriteIeeeExtendedHighLow( file_ptr, aiff_ptr->sampleRate ); +- /* write SSND chunk header */ +- chunkSize = 8 + aiff_ptr->numChannels * aiff_ptr->numSampleFrames * sampleBytes; +- Write32BitsHighLow( file_ptr, IFF_ID_SSND ); +- Write32BitsHighLow( file_ptr, chunkSize ); +- Write32BitsHighLow( file_ptr, 0 ); /* offset */ +- Write32BitsHighLow( file_ptr, 0 ); /* block size */ +- return 0; +- } +- +- /***************************************************************************** +- * +- * bit_stream.c package +- * Author: Jean-Georges Fritsch, C-Cube Microsystems +- * +- *****************************************************************************/ +- +- /******************************************************************** +- This package provides functions to write (exclusive or read) +- information from (exclusive or to) the bit stream. +- +- If the bit stream is opened in read mode only the get functions are +- available. If the bit stream is opened in write mode only the put +- functions are available. +- ********************************************************************/ +- +- /*open_bit_stream_w(); open the device to write the bit stream into it */ +- /*open_bit_stream_r(); open the device to read the bit stream from it */ +- /*close_bit_stream(); close the device containing the bit stream */ +- /*alloc_buffer(); open and initialize the buffer; */ +- /*desalloc_buffer(); empty and close the buffer */ +- /*back_track_buffer(); goes back N bits in the buffer */ +- /*unsigned int get1bit(); read 1 bit from the bit stream */ +- /*unsigned long getbits(); read N bits from the bit stream */ +- /*unsigned long byte_ali_getbits(); read the next byte aligned N bits from*/ +- /* the bit stream */ +- /*unsigned long look_ahead(); grep the next N bits in the bit stream without*/ +- /* changing the buffer pointer */ +- /*put1bit(); write 1 bit from the bit stream */ +- /*put1bit(); write 1 bit from the bit stream */ +- /*putbits(); write N bits from the bit stream */ +- /*byte_ali_putbits(); write byte aligned the next N bits into the bit stream*/ +- /*unsigned long sstell(); return the current bit stream length (in bits) */ +- /*int end_bs(); return 1 if the end of bit stream reached otherwise 0 */ +- /*int seek_sync(); return 1 if a sync word was found in the bit stream */ +- /* otherwise returns 0 */ +- +- /* refill the buffer from the input device when the buffer becomes empty */ +- int refill_buffer(bs) +- Bit_stream_struc *bs; /* bit stream structure */ +- { +- register int i=bs->buf_size-2-bs->buf_byte_idx; +- register unsigned long n=1; +- register int index=0; +- char val[2]; +- +- while ((i>=0) && (!bs->eob)) { +- +- if (bs->format == BINARY) +- n = fread(&bs->buf[i--], sizeof(unsigned char), 1, bs->pt); +- +- else { +- while((index < 2) && n) { +- n = fread(&val[index], sizeof(char), 1, bs->pt); +- switch (val[index]) { +- case 0x30: +- case 0x31: +- case 0x32: +- case 0x33: +- case 0x34: +- case 0x35: +- case 0x36: +- case 0x37: +- case 0x38: +- case 0x39: +- case 0x41: +- case 0x42: +- case 0x43: +- case 0x44: +- case 0x45: +- case 0x46: +- index++; +- break; +- default: break; +- } +- } +- +- if (val[0] <= 0x39) bs->buf[i] = (val[0] - 0x30) << 4; +- else bs->buf[i] = (val[0] - 0x37) << 4; +- if (val[1] <= 0x39) bs->buf[i--] |= (val[1] - 0x30); +- else bs->buf[i--] |= (val[1] - 0x37); +- index = 0; +- } +- +- if (!n) { +- bs->eob= i+1; +- } +- +- } +- return 0; +- } +- +- static char *he = "0123456789ABCDEF"; +- +- /* empty the buffer to the output device when the buffer becomes full */ +- void empty_buffer(bs, minimum) +- Bit_stream_struc *bs; /* bit stream structure */ +- int minimum; /* end of the buffer to empty */ +- { +- register int i; +- +- #if BS_FORMAT == BINARY +- for (i=bs->buf_size-1;i>=minimum;i--) +- fwrite(&bs->buf[i], sizeof(unsigned char), 1, bs->pt); +- #else +- for (i=bs->buf_size-1;i>=minimum;i--) { +- char val[2]; +- val[0] = he[((bs->buf[i] >> 4) & 0x0F)]; +- val[1] = he[(bs->buf[i] & 0x0F)]; +- fwrite(val, sizeof(char), 2, bs->pt); +- } +- #endif +- fflush(bs->pt); /* NEW SS to assist in debugging*/ +- +- for (i=minimum-1; i>=0; i--) +- bs->buf[bs->buf_size - minimum + i] = bs->buf[i]; +- +- bs->buf_byte_idx = bs->buf_size -1 - minimum; +- bs->buf_bit_idx = 8; +- } +- +- /* open the device to write the bit stream into it */ +- void open_bit_stream_w(bs, bs_filenam, size) +- Bit_stream_struc *bs; /* bit stream structure */ +- char *bs_filenam; /* name of the bit stream file */ +- int size; /* size of the buffer */ +- { +- if ((bs->pt = fopen(bs_filenam, "wb")) == NULL) { +- printf("Could not create \"%s\".\n", bs_filenam); +- exit(1); +- } +- alloc_buffer(bs, size); +- bs->buf_byte_idx = size-1; +- bs->buf_bit_idx=8; +- bs->totbit=0; +- bs->mode = WRITE_MODE; +- bs->eob = FALSE; +- bs->eobs = FALSE; +- } +- +- /* open the device to read the bit stream from it */ +- void open_bit_stream_r(bs, bs_filenam, size) +- Bit_stream_struc *bs; /* bit stream structure */ +- char *bs_filenam; /* name of the bit stream file */ +- int size; /* size of the buffer */ +- { +- register unsigned long n; +- register unsigned char flag = 1; +- unsigned char val; +- +- if ((bs->pt = fopen(bs_filenam, "rb")) == NULL) { +- printf("Could not find \"%s\".\n", bs_filenam); +- exit(1); +- } +- +- do { +- n = fread(&val, sizeof(unsigned char), 1, bs->pt); +- switch (val) { +- case 0x30: +- case 0x31: +- case 0x32: +- case 0x33: +- case 0x34: +- case 0x35: +- case 0x36: +- case 0x37: +- case 0x38: +- case 0x39: +- case 0x41: +- case 0x42: +- case 0x43: +- case 0x44: +- case 0x45: +- case 0x46: +- case 0xa: /* \n */ +- case 0xd: /* cr */ +- case 0x1a: /* sub */ +- break; +- +- default: /* detection of an binary character */ +- flag--; +- break; +- } +- +- } while (flag & n); +- +- if (flag) { +- printf ("the bit stream file %s is an ASCII file\n", bs_filenam); +- bs->format = ASCII; +- } +- else { +- bs->format = BINARY; +- printf ("the bit stream file %s is a BINARY file\n", bs_filenam); +- } +- +- fclose(bs->pt); +- +- if ((bs->pt = fopen(bs_filenam, "rb")) == NULL) { +- printf("Could not find \"%s\".\n", bs_filenam); +- exit(1); +- } +- +- alloc_buffer(bs, size); +- bs->buf_byte_idx=0; +- bs->buf_bit_idx=0; +- bs->totbit=0; +- bs->mode = READ_MODE; +- bs->eob = FALSE; +- bs->eobs = FALSE; +- } +- +- /*close the device containing the bit stream after a read process*/ +- void close_bit_stream_r(bs) +- Bit_stream_struc *bs; /* bit stream structure */ +- { +- fclose(bs->pt); +- desalloc_buffer(bs); +- } +- +- /*close the device containing the bit stream after a write process*/ +- void close_bit_stream_w(bs) +- Bit_stream_struc *bs; /* bit stream structure */ +- { +- empty_buffer(bs, bs->buf_byte_idx); +- fclose(bs->pt); +- desalloc_buffer(bs); +- } +- +- /*open and initialize the buffer; */ +- void alloc_buffer(bs, size) +- Bit_stream_struc *bs; /* bit stream structure */ +- int size; +- { +- bs->buf = (unsigned char FAR *) mem_alloc(size*sizeof(unsigned +- char), "buffer"); +- bs->buf_size = size; +- } +- +- /*empty and close the buffer */ +- void desalloc_buffer(bs) +- Bit_stream_struc *bs; /* bit stream structure */ +- { +- free(bs->buf); +- } +- +- int putmask[9]={0x0, 0x1, 0x3, 0x7, 0xf, 0x1f, 0x3f, 0x7f, 0xff}; +- int clearmask[9]={0xff, 0xfe, 0xfc, 0xf8, 0xf0, 0xe0, 0xc0, 0x80, 0x0}; +- +- void back_track_buffer(bs, N) /* goes back N bits in the buffer */ +- Bit_stream_struc *bs; /* bit stream structure */ +- int N; +- { +- int tmp = N - (N/8)*8; +- register int i; +- +- bs->totbit -= N; +- for (i=bs->buf_byte_idx;i< bs->buf_byte_idx+N/8-1;i++) bs->buf[i] = 0; +- bs->buf_byte_idx += N/8; +- if ( (tmp + bs->buf_bit_idx) <= 8) { +- bs->buf_bit_idx += tmp; +- } +- else { +- bs->buf_byte_idx ++; +- bs->buf_bit_idx += (tmp - 8); +- } +- bs->buf[bs->buf_byte_idx] &= clearmask[bs->buf_bit_idx]; +- } +- +- int mask[8]={0x1, 0x2, 0x4, 0x8, 0x10, 0x20, 0x40, 0x80}; +- +- /*read 1 bit from the bit stream */ +- unsigned int get1bit(bs) +- Bit_stream_struc *bs; /* bit stream structure */ +- { +- unsigned int bit; +- register int i; +- +- bs->totbit++; +- +- if (!bs->buf_bit_idx) { +- bs->buf_bit_idx = 8; +- bs->buf_byte_idx--; +- if ((bs->buf_byte_idx < MINIMUM) || (bs->buf_byte_idx < bs->eob)) { +- if (bs->eob) +- bs->eobs = TRUE; +- else { +- for (i=bs->buf_byte_idx; i>=0;i--) +- bs->buf[bs->buf_size-1-bs->buf_byte_idx+i] = bs->buf[i]; +- refill_buffer(bs); +- bs->buf_byte_idx = bs->buf_size-1; +- } +- } +- } +- bit = bs->buf[bs->buf_byte_idx]&mask[bs->buf_bit_idx-1]; +- bit = bit >> (bs->buf_bit_idx-1); +- bs->buf_bit_idx--; +- return(bit); +- } +- +- /*write 1 bit from the bit stream */ +- void put1bit(bs, bit) +- Bit_stream_struc *bs; /* bit stream structure */ +- int bit; /* bit to write into the buffer */ +- { +- bs->totbit++; +- +- bs->buf[bs->buf_byte_idx] |= (bit&0x1) << (bs->buf_bit_idx-1); +- bs->buf_bit_idx--; +- if (!bs->buf_bit_idx) { +- bs->buf_bit_idx = 8; +- bs->buf_byte_idx--; +- if (bs->buf_byte_idx < 0) +- empty_buffer(bs, MINIMUM); +- bs->buf[bs->buf_byte_idx] = 0; +- } +- } +- +- /*look ahead for the next N bits from the bit stream */ +- unsigned long look_ahead(bs, N) +- Bit_stream_struc *bs; /* bit stream structure */ +- int N; /* number of bits to read from the bit stream */ +- { +- unsigned long val=0; +- register int j = N; +- register int k, tmp; +- register int bit_idx = bs->buf_bit_idx; +- register int byte_idx = bs->buf_byte_idx; +- +- if (N > MAX_LENGTH) +- printf("Cannot read or write more than %d bits at a time.\n", MAX_LENGTH); +- +- while (j > 0) { +- if (!bit_idx) { +- bit_idx = 8; +- byte_idx--; +- } +- k = MIN (j, bit_idx); +- tmp = bs->buf[byte_idx]&putmask[bit_idx]; +- tmp = tmp >> (bit_idx-k); +- val |= tmp << (j-k); +- bit_idx -= k; +- j -= k; +- } +- return(val); +- } +- +- /*read N bit from the bit stream */ +- unsigned long getbits(bs, N) +- Bit_stream_struc *bs; /* bit stream structure */ +- int N; /* number of bits to read from the bit stream */ +- { +- unsigned long val=0; +- register int i; +- register int j = N; +- register int k, tmp; +- +- if (N > MAX_LENGTH) +- printf("Cannot read or write more than %d bits at a time.\n", MAX_LENGTH); +- +- bs->totbit += N; +- while (j > 0) { +- if (!bs->buf_bit_idx) { +- bs->buf_bit_idx = 8; +- bs->buf_byte_idx--; +- if ((bs->buf_byte_idx < MINIMUM) || (bs->buf_byte_idx < bs->eob)) { +- if (bs->eob) +- bs->eobs = TRUE; +- else { +- for (i=bs->buf_byte_idx; i>=0;i--) +- bs->buf[bs->buf_size-1-bs->buf_byte_idx+i] = bs->buf[i]; +- refill_buffer(bs); +- bs->buf_byte_idx = bs->buf_size-1; +- } +- } +- } +- k = MIN (j, bs->buf_bit_idx); +- tmp = bs->buf[bs->buf_byte_idx]&putmask[bs->buf_bit_idx]; +- tmp = tmp >> (bs->buf_bit_idx-k); +- val |= tmp << (j-k); +- bs->buf_bit_idx -= k; +- j -= k; +- } +- return(val); +- } +- +- /*write N bits into the bit stream */ +- void putbits(bs, val, N) +- Bit_stream_struc *bs; /* bit stream structure */ +- unsigned int val; /* val to write into the buffer */ +- int N; /* number of bits of val */ +- { +- register int j = N; +- register int k, tmp; +- +- if (N > MAX_LENGTH) +- printf("Cannot read or write more than %d bits at a time.\n", MAX_LENGTH); +- +- bs->totbit += N; +- while (j > 0) { +- k = MIN(j, bs->buf_bit_idx); +- tmp = val >> (j-k); +- bs->buf[bs->buf_byte_idx] |= (tmp&putmask[k]) << (bs->buf_bit_idx-k); +- bs->buf_bit_idx -= k; +- if (!bs->buf_bit_idx) { +- bs->buf_bit_idx = 8; +- bs->buf_byte_idx--; +- if (bs->buf_byte_idx < 0) +- empty_buffer(bs, MINIMUM); +- bs->buf[bs->buf_byte_idx] = 0; +- } +- j -= k; +- } +- } +- +- /*write N bits byte aligned into the bit stream */ +- void byte_ali_putbits(bs, val, N) +- Bit_stream_struc *bs; /* bit stream structure */ +- unsigned int val; /* val to write into the buffer */ +- int N; /* number of bits of val */ +- { +- unsigned long aligning, sstell(); +- +- if (N > MAX_LENGTH) +- printf("Cannot read or write more than %d bits at a time.\n", MAX_LENGTH); +- aligning = sstell(bs)%8; +- if (aligning) +- putbits(bs, (unsigned int)0, (int)(8-aligning)); +- +- putbits(bs, val, N); +- } +- +- /*read the next bute aligned N bits from the bit stream */ +- unsigned long byte_ali_getbits(bs, N) +- Bit_stream_struc *bs; /* bit stream structure */ +- int N; /* number of bits of val */ +- { +- unsigned long aligning, sstell(); +- +- if (N > MAX_LENGTH) +- printf("Cannot read or write more than %d bits at a time.\n", MAX_LENGTH); +- aligning = sstell(bs)%8; +- if (aligning) +- getbits(bs, (int)(8-aligning)); +- +- return(getbits(bs, N)); +- } +- +- /*return the current bit stream length (in bits)*/ +- unsigned long sstell(bs) +- Bit_stream_struc *bs; /* bit stream structure */ +- { +- return(bs->totbit); +- } +- +- /*return the status of the bit stream*/ +- /* returns 1 if end of bit stream was reached */ +- /* returns 0 if end of bit stream was not reached */ +- int end_bs(bs) +- Bit_stream_struc *bs; /* bit stream structure */ +- { +- return(bs->eobs); +- } +- +- /*this function seeks for a byte aligned sync word in the bit stream and +- places the bit stream pointer right after the sync. +- This function returns 1 if the sync was found otherwise it returns 0 */ +- int seek_sync(bs, sync, N) +- Bit_stream_struc *bs; /* bit stream structure */ +- long sync; /* sync word maximum 32 bits */ +- int N; /* sync word length */ +- { +- #if defined(MACINTOSH) && !defined(__powerc) +- double pow(); +- #endif +- unsigned long aligning, stell(); +- unsigned long val; +- long maxi = (int)pow(2.0, (FLOAT)N) - 1; +- +- aligning = sstell(bs)%ALIGNING; +- if (aligning) +- getbits(bs, (int)(ALIGNING-aligning)); +- +- val = getbits(bs, N); +- while (((val&maxi) != sync) && (!end_bs(bs))) { +- val <<= ALIGNING; +- val |= getbits(bs, ALIGNING); +- } +- +- if (end_bs(bs)) return(0); +- else return(1); +- } +- /***************************************************************************** +- * +- * End of bit_stream.c package +- * +- *****************************************************************************/ +- +- /***************************************************************************** +- * +- * CRC error protection package +- * +- *****************************************************************************/ +- +- void I_CRC_calc(fr_ps, bit_alloc, crc) +- frame_params *fr_ps; +- unsigned int bit_alloc[2][SBLIMIT]; +- unsigned int *crc; +- { +- int i, k; +- layer *info = fr_ps->header; +- int stereo = fr_ps->stereo; +- int jsbound = fr_ps->jsbound; +- +- *crc = 0xffff; /* changed from '0' 92-08-11 shn */ +- update_CRC(info->bitrate_index, 4, crc); +- update_CRC(info->sampling_frequency, 2, crc); +- update_CRC(info->padding, 1, crc); +- update_CRC(info->extension, 1, crc); +- update_CRC(info->mode, 2, crc); +- update_CRC(info->mode_ext, 2, crc); +- update_CRC(info->copyright, 1, crc); +- update_CRC(info->original, 1, crc); +- update_CRC(info->emphasis, 2, crc); +- +- for (i=0;i<SBLIMIT;i++) +- for (k=0;k<((i<jsbound)?stereo:1);k++) +- update_CRC(bit_alloc[k][i], 4, crc); +- } +- +- void II_CRC_calc(fr_ps, bit_alloc, scfsi, crc) +- frame_params *fr_ps; +- unsigned int bit_alloc[2][SBLIMIT], scfsi[2][SBLIMIT]; +- unsigned int *crc; +- { +- int i, k; +- layer *info = fr_ps->header; +- int stereo = fr_ps->stereo; +- int sblimit = fr_ps->sblimit; +- int jsbound = fr_ps->jsbound; +- al_table *alloc = fr_ps->alloc; +- +- *crc = 0xffff; /* changed from '0' 92-08-11 shn */ +- update_CRC(info->bitrate_index, 4, crc); +- update_CRC(info->sampling_frequency, 2, crc); +- update_CRC(info->padding, 1, crc); +- update_CRC(info->extension, 1, crc); +- update_CRC(info->mode, 2, crc); +- update_CRC(info->mode_ext, 2, crc); +- update_CRC(info->copyright, 1, crc); +- update_CRC(info->original, 1, crc); +- update_CRC(info->emphasis, 2, crc); +- +- for (i=0;i<sblimit;i++) +- for (k=0;k<((i<jsbound)?stereo:1);k++) +- update_CRC(bit_alloc[k][i], (*alloc)[i][0].bits, crc); +- +- for (i=0;i<sblimit;i++) +- for (k=0;k<stereo;k++) +- if (bit_alloc[k][i]) +- update_CRC(scfsi[k][i], 2, crc); +- } +- +- void update_CRC(data, length, crc) +- unsigned int data, length, *crc; +- { +- unsigned int masking, carry; +- +- masking = 1 << length; +- +- while((masking >>= 1)){ +- carry = *crc & 0x8000; +- *crc <<= 1; +- if (!carry ^ !(data & masking)) +- *crc ^= CRC16_POLYNOMIAL; +- } +- *crc &= 0xffff; +- } +- +- /***************************************************************************** +- * +- * End of CRC error protection package +- * +- *****************************************************************************/ +- +- #ifdef MACINTOSH +- /***************************************************************************** +- * +- * Set Macintosh file attributes. +- * +- *****************************************************************************/ +- void set_mac_file_attr( char fileName[MAX_NAME_SIZE], short vRefNum, OSType creator, OSType fileType) +- { +- char pascal_fileName[MAX_NAME_SIZE]; +- FInfo fndrInfo; +- OSErr anErr; +- long dirID = 0L; +- +- CtoPstr(strcpy(pascal_fileName, fileName)); +- anErr = HGetFInfo(vRefNum, dirID, (ConstStr255Param) pascal_fileName, &fndrInfo); +- if ( anErr != noErr ) +- return; +- fndrInfo.fdCreator = creator; +- fndrInfo.fdType = fileType; +- anErr = HSetFInfo(vRefNum, dirID, (ConstStr255Param) pascal_fileName, &fndrInfo); +- } +- +- #endif +- +- +- #ifdef MS_DOS +- /* ------------------------------------------------------------------------ +- new_ext() +- Puts a new extension name on a file name <filename>. +- Removes the last extension name, if any. +- 1992-08-19, 1995-06-12 shn +- ------------------------------------------------------------------------ */ +- void +- new_ext(char *filename, char *extname, char *newname) +- { +- int found, dotpos; +- +- /* First, strip the extension */ +- dotpos=strlen(filename); found=0; +- do +- { +- switch (filename[dotpos]) +- { +- case '.' : found=1; break; +- case '\\': +- case '/' : +- case ':' : found=-1; break; +- default : dotpos--; if (dotpos<0) found=-1; break; +- } +- } while (found==0); +- if (found==-1) strcpy(newname,filename); +- if (found== 1) { strncpy(newname,filename,dotpos); newname[dotpos]='\0'; } +- strcat(newname,extname); +- } +- #endif +- +- +- +- #define BUFSIZE 4096 +- static unsigned long offset,totbit=0, buf_byte_idx=0; +- static unsigned int buf[BUFSIZE]; +- static unsigned int buf_bit_idx=8; +- +- /*return the current bit stream length (in bits)*/ +- unsigned long hsstell() +- { +- return(totbit); +- } +- +- /* int putmask[9]={0x0, 0x1, 0x3, 0x7, 0xf, 0x1f, 0x3f, 0x7f, 0xff}; */ +- extern int putmask[9]; +- +- /*read N bit from the bit stream */ +- unsigned long hgetbits(N) +- int N; /* number of bits to read from the bit stream */ +- { +- unsigned long val=0; +- register int j = N; +- register int k, tmp; +- +- /* +- if (N > MAX_LENGTH) +- printf("Cannot read or write more than %d bits at a time.\n", MAX_LENGTH); +- */ +- totbit += N; +- while (j > 0) { +- if (!buf_bit_idx) { +- buf_bit_idx = 8; +- buf_byte_idx++; +- if (buf_byte_idx > offset) +- { printf("Buffer overflow !!\n");exit(3); } +- } +- k = MIN (j, buf_bit_idx); +- tmp = buf[buf_byte_idx%BUFSIZE]&putmask[buf_bit_idx]; +- tmp = tmp >> (buf_bit_idx-k); +- val |= tmp << (j-k); +- buf_bit_idx -= k; +- j -= k; +- } +- return(val); +- } +- +- unsigned int hget1bit() +- { +- return(hgetbits(1)); +- } +- +- /*write N bits into the bit stream */ +- void hputbuf(val, N) +- unsigned int val; /* val to write into the buffer */ +- int N; /* number of bits of val */ +- { +- if (N != 8) { printf("Not Supported yet!!\n"); exit(-3); } +- buf[offset % BUFSIZE] = val; +- offset++; +- } +- +- void rewindNbits( N ) +- int N; +- { +- totbit -= N; +- buf_bit_idx += N; +- while( buf_bit_idx >= 8 ) +- { buf_bit_idx -= 8; +- buf_byte_idx--; +- } +- } +- +- void rewindNbytes( N ) +- int N; +- { +- totbit -= N*8; +- buf_byte_idx -= N; +- } +--- 0 ---- +diff -r -c -N encoder/common.h lame3.50/common.h +*** encoder/common.h Wed Jan 22 02:43:13 1997 +--- lame3.50/common.h Wed Dec 31 17:00:00 1969 +*************** +*** 1,491 **** +- /********************************************************************** +- * ISO MPEG Audio Subgroup Software Simulation Group (1996) +- * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- * +- * $Id: common.h,v 1.1 1996/02/14 04:04:23 rowlands Exp $ +- * +- * $Log: common.h,v $ +- * Revision 1.1 1996/02/14 04:04:23 rowlands +- * Initial revision +- * +- * Received from Mike Coleman +- **********************************************************************/ +- /********************************************************************** +- * date programmers comment * +- * 2/25/91 Doulas Wong, start of version 1.0 records * +- * Davis Pan * +- * 5/10/91 W. Joseph Carter Reorganized & renamed all ".h" files * +- * into "common.h" and "encoder.h". * +- * Ported to Macintosh and Unix. * +- * Added additional type definitions for * +- * AIFF, double/SANE and "bitstream.c". * +- * Added function prototypes for more * +- * rigorous type checking. * +- * 27jun91 dpwe (Aware) Added "alloc_*" defs & prototypes * +- * Defined new struct 'frame_params'. * +- * Changed info.stereo to info.mode_ext * +- * #define constants for mode types * +- * Prototype arguments if PROTO_ARGS * +- * 5/28/91 Earle Jennings added MS_DOS definition * +- * MsDos function prototype declarations * +- * 7/10/91 Earle Jennings added FLOAT definition as double * +- *10/ 3/91 Don H. Lee implemented CRC-16 error protection * +- * 2/11/92 W. Joseph Carter Ported new code to Macintosh. Most * +- * important fixes involved changing * +- * 16-bit ints to long or unsigned in * +- * bit alloc routines for quant of 65535 * +- * and passing proper function args. * +- * Removed "Other Joint Stereo" option * +- * and made bitrate be total channel * +- * bitrate, irrespective of the mode. * +- * Fixed many small bugs & reorganized. * +- * Modified some function prototypes. * +- * Changed BUFFER_SIZE back to 4096. * +- * 7/27/92 Michael Li (re-)Ported to MS-DOS * +- * 7/27/92 Masahiro Iwadare Ported to Convex * +- * 8/07/92 mc@tv.tek.com * +- * 8/10/92 Amit Gulati Ported to the AIX Platform (RS6000) * +- * AIFF string constants redefined * +- * 8/27/93 Seymour Shlien, Fixes in Unix and MSDOS ports, * +- * Daniel Lauzon, and * +- * Bill Truerniet * +- *--------------------------------------------------------------------* +- * 4/23/92 J. Pineda Added code for Layer III. * +- * 11/9/92 Amit Gulati Added defines for layerIII stereo * +- * modes. * +- * 8/24/93 Masahiro Iwadare Included IS modification in Layer III.* +- * Changed for 1 pass decoding. * +- * 9/07/93 Toshiyuki Ishino Integrated Layer III with Ver 3.9. * +- *--------------------------------------------------------------------* +- * 11/20/93 Masahiro Iwadare Integrated Layer III with Ver 4.0. * +- *--------------------------------------------------------------------* +- * 7/14/94 Juergen Koller Fix for HPUX an IRIX in AIFF-Strings * +- * 6/12/95 Soeren H. Nielsen Bug fix in new_ext(). * +- * 7/11/95 Soeren H. Nielsen Added defs. for MPEG-2 LSF * +- *--------------------------------------------------------------------* +- * 8/02/95 mc@fivebats.com Changes to AIFF stuff for portability * +- * to little-endian machines * +- **********************************************************************/ +- #ifndef COMMON_DOT_H +- #define COMMON_DOT_H +- /*********************************************************************** +- * +- * Global Conditional Compile Switches +- * +- ***********************************************************************/ +- +- #if 0 +- #define UNIX /* Unix conditional compile switch */ +- #define MACINTOSH /* Macintosh conditional compile switch */ +- #define MS_DOS /* IBM PC conditional compile switch */ +- #define MSC60 /* Compiled for MS_DOS with MSC v6.0 */ +- #define AIX /* AIX conditional compile switch */ +- #define CONVEX /* CONVEX conditional compile switch */ +- #endif +- +- #if defined(MSC60) +- #ifndef MS_DOS +- #define MS_DOS +- #endif +- #ifndef PROTO_ARGS +- #define PROTO_ARGS +- #endif +- #endif +- +- #ifdef UNIX +- #define TABLES_PATH "tables" /* to find data files */ +- /* name of environment variable holding path of table files */ +- #define MPEGTABENV "MPEGTABLES" +- #define PATH_SEPARATOR "/" /* how to build paths */ +- #endif /* UNIX */ +- +- #ifdef MACINTOSH +- #define TABLES_PATH ":tables:" /* where to find data files */ +- #endif /* MACINTOSH */ +- +- /* +- * Don't define FAR to far unless you're willing to clean up the +- * prototypes +- */ +- #define FAR /*far*/ +- +- #ifdef __STDC__ +- #ifndef PROTO_ARGS +- #define PROTO_ARGS +- #endif +- #endif +- +- #ifdef CONVEX +- #define SEEK_SET 0 +- #define SEEK_CUR 1 +- #define SEEK_END 2 +- #endif +- +- /* MS_DOS and VMS do not define TABLES_PATH, so OpenTableFile will default +- to finding the data files in the default directory */ +- +- /*********************************************************************** +- * +- * Global Include Files +- * +- ***********************************************************************/ +- +- #include <stdio.h> +- #include <string.h> +- #include <math.h> +- #include "portableio.h" +- #include "ieeefloat.h" +- +- #ifdef UNIX +- #include <unistd.h> +- #endif /* UNIX */ +- +- #ifdef __sgi +- #include <stdlib.h> +- #endif +- +- #ifdef MACINTOSH +- #include <stdlib.h> +- #include <console.h> +- #endif /* MACINTOSH */ +- +- #ifdef MS_DOS +- #include <stdlib.h> +- #ifdef MSC60 +- #include <memory.h> +- #else +- #include <alloc.h> +- #include <mem.h> +- #endif /* MSC60 */ +- #endif /* MS_DOS */ +- +- #ifndef EXIT_SUCCESS +- #define EXIT_SUCCESS 0 +- #endif +- +- #ifndef EXIT_FAILURE +- #define EXIT_FAILURE 1 +- #endif +- +- /*********************************************************************** +- * +- * Global Definitions +- * +- ***********************************************************************/ +- +- /* General Definitions */ +- +- #ifdef MS_DOS +- #define FLOAT double +- #else +- #define FLOAT float +- #endif +- #ifndef FALSE +- #define FALSE 0 +- #endif +- +- #ifndef TRUE +- #define TRUE 1 +- #endif +- +- #define NULL_CHAR '\0' +- +- #define MAX_U_32_NUM 0xFFFFFFFF +- #ifndef PI +- #define PI 3.14159265358979 +- #endif +- #define PI4 PI/4 +- #define PI64 PI/64 +- #define LN_TO_LOG10 0.2302585093 +- +- #define VOL_REF_NUM 0 +- #define MPEG_AUDIO_ID 1 +- #define MPEG_PHASE2_LSF 0 /* 1995-07-11 SHN */ +- #define MAC_WINDOW_SIZE 24 +- +- #define MONO 1 +- #define STEREO 2 +- #define BITS_IN_A_BYTE 8 +- #define WORD 16 +- #define MAX_NAME_SIZE 81 +- #define SBLIMIT 32 +- #define SSLIMIT 18 +- #define FFT_SIZE 1024 +- #define HAN_SIZE 512 +- #define SCALE_BLOCK 12 +- #define SCALE_RANGE 64 +- #define SCALE 32768 +- #define CRC16_POLYNOMIAL 0x8005 +- +- /* MPEG Header Definitions - Mode Values */ +- +- #define MPG_MD_STEREO 0 +- #define MPG_MD_JOINT_STEREO 1 +- #define MPG_MD_DUAL_CHANNEL 2 +- #define MPG_MD_MONO 3 +- +- /* Mode Extention */ +- +- #define MPG_MD_LR_LR 0 +- #define MPG_MD_LR_I 1 +- #define MPG_MD_MS_LR 2 +- #define MPG_MD_MS_I 3 +- +- +- /* "bit_stream.h" Definitions */ +- +- #define MINIMUM 4 /* Minimum size of the buffer in bytes */ +- #define MAX_LENGTH 32 /* Maximum length of word written or +- read from bit stream */ +- #define READ_MODE 0 +- #define WRITE_MODE 1 +- #define ALIGNING 8 +- #define BINARY 0 +- #define ASCII 1 +- +- #ifndef BS_FORMAT +- #define BS_FORMAT ASCII /* BINARY or ASCII = 2x bytes */ +- #endif +- +- #define BUFFER_SIZE 4096 +- +- #define MIN(A, B) ((A) < (B) ? (A) : (B)) +- #define MAX(A, B) ((A) > (B) ? (A) : (B)) +- +- /*********************************************************************** +- * +- * Global Type Definitions +- * +- ***********************************************************************/ +- +- /* Structure for Reading Layer II Allocation Tables from File */ +- +- typedef struct { +- unsigned int steps; +- unsigned int bits; +- unsigned int group; +- unsigned int quant; +- } sb_alloc, *alloc_ptr; +- +- typedef sb_alloc al_table[SBLIMIT][16]; +- +- /* Header Information Structure */ +- +- typedef struct { +- int version; +- int lay; +- int error_protection; +- int bitrate_index; +- int sampling_frequency; +- int padding; +- int extension; +- int mode; +- int mode_ext; +- int copyright; +- int original; +- int emphasis; +- } layer, *the_layer; +- +- /* Parent Structure Interpreting some Frame Parameters in Header */ +- +- typedef struct { +- layer *header; /* raw header information */ +- int actual_mode; /* when writing IS, may forget if 0 chs */ +- al_table *alloc; /* bit allocation table read in */ +- int tab_num; /* number of table as loaded */ +- int stereo; /* 1 for mono, 2 for stereo */ +- int jsbound; /* first band of joint stereo coding */ +- int sblimit; /* total number of sub bands */ +- } frame_params; +- +- +- /* AIFF Definitions */ +- +- #define IFF_ID_FORM 0x464f524d /* "FORM" */ +- #define IFF_ID_AIFF 0x41494646 /* "AIFF" */ +- #define IFF_ID_COMM 0x434f4d4d /* "COMM" */ +- #define IFF_ID_SSND 0x53534e44 /* "SSND" */ +- #define IFF_ID_MPEG 0x4d504547 /* "MPEG" */ +- +- #define AIFF_FORM_HEADER_SIZE 12 +- #define AIFF_SSND_HEADER_SIZE 16 +- +- +- typedef struct blockAlign_struct { +- unsigned long offset; +- unsigned long blockSize; +- } blockAlign; +- +- typedef struct IFF_AIFF_struct { +- short numChannels; +- unsigned long numSampleFrames; +- short sampleSize; +- double sampleRate; +- unsigned long sampleType; +- blockAlign blkAlgn; +- } IFF_AIFF; +- +- enum byte_order { order_unknown, order_bigEndian, order_littleEndian }; +- extern enum byte_order NativeByteOrder; +- +- /* "bit_stream.h" Type Definitions */ +- +- typedef struct bit_stream_struc { +- FILE *pt; /* pointer to bit stream device */ +- unsigned char *buf; /* bit stream buffer */ +- int buf_size; /* size of buffer (in number of bytes) */ +- long totbit; /* bit counter of bit stream */ +- int buf_byte_idx; /* pointer to top byte in buffer */ +- int buf_bit_idx; /* pointer to top bit of top byte in buffer */ +- int mode; /* bit stream open in read or write mode */ +- int eob; /* end of buffer index */ +- int eobs; /* end of bit stream flag */ +- char format; +- +- /* format of file in rd mode (BINARY/ASCII) */ +- } Bit_stream_struc; +- +- #include "l3side.h" +- +- /*********************************************************************** +- * +- * Global Variable External Declarations +- * +- ***********************************************************************/ +- +- extern char *mode_names[4]; +- extern char *layer_names[3]; +- extern char *version_names[2]; +- extern double s_freq[2][4]; +- extern int bitrate[2][3][15]; +- extern double FAR multiple[64]; +- +- /*********************************************************************** +- * +- * Global Function Prototype Declarations +- * +- ***********************************************************************/ +- +- /* The following functions are in the file "common.c" */ +- +- #ifdef PROTO_ARGS +- extern FILE *OpenTableFile(char*); +- extern int read_bit_alloc(int, al_table*); +- extern int pick_table(frame_params*); +- extern int js_bound(int, int); +- extern void hdr_to_frps(frame_params*); +- extern void WriteHdr(frame_params*, FILE*); +- extern void WriteBitAlloc(unsigned int[2][SBLIMIT], frame_params*, +- FILE*); +- extern void WriteScale(unsigned int[2][SBLIMIT], +- unsigned int[2][SBLIMIT], unsigned int[2][3][SBLIMIT], +- frame_params*, FILE*); +- extern void WriteSamples(int, unsigned int FAR [SBLIMIT], +- unsigned int[SBLIMIT], frame_params*, FILE*); +- extern int NumericQ(char*); +- extern int BitrateIndex(int, int, int); +- extern int SmpFrqIndex(long, int*); +- extern int memcheck(char*, int, int); +- extern void FAR *mem_alloc(unsigned long, char*); +- extern void mem_free(void**); +- extern void double_to_extended(double*, char[10]); +- extern void extended_to_double(char[10], double*); +- extern int aiff_read_headers(FILE*, IFF_AIFF*); +- extern int aiff_seek_to_sound_data(FILE*); +- extern int aiff_write_headers(FILE*, IFF_AIFF*); +- extern int refill_buffer(Bit_stream_struc*); +- extern void empty_buffer(Bit_stream_struc*, int); +- extern void open_bit_stream_w(Bit_stream_struc*, char*, int); +- extern void open_bit_stream_r(Bit_stream_struc*, char*, int); +- extern void close_bit_stream_r(Bit_stream_struc*); +- extern void close_bit_stream_w(Bit_stream_struc*); +- extern void alloc_buffer(Bit_stream_struc*, int); +- extern void desalloc_buffer(Bit_stream_struc*); +- extern void back_track_buffer(Bit_stream_struc*, int); +- extern unsigned int get1bit(Bit_stream_struc*); +- extern void put1bit(Bit_stream_struc*, int); +- extern unsigned long look_ahead(Bit_stream_struc*, int); +- extern unsigned long getbits(Bit_stream_struc*, int); +- extern void putbits(Bit_stream_struc*, unsigned int, int); +- extern void byte_ali_putbits(Bit_stream_struc*, unsigned int, int); +- extern unsigned long byte_ali_getbits(Bit_stream_struc*, int); +- extern unsigned long sstell(Bit_stream_struc*); +- extern int end_bs(Bit_stream_struc*); +- extern int seek_sync(Bit_stream_struc*, long, int); +- extern void I_CRC_calc(frame_params*, unsigned int[2][SBLIMIT], +- unsigned int*); +- extern void II_CRC_calc(frame_params*, unsigned int[2][SBLIMIT], +- unsigned int[2][SBLIMIT], unsigned int*); +- extern void update_CRC(unsigned int, unsigned int, unsigned int*); +- extern void read_absthr(FLOAT*, int); +- extern unsigned int hget1bit(); /* MI */ +- extern unsigned long hgetbits(int); +- extern unsigned long hsstell(); +- extern void hputbuf(unsigned int,int); +- extern enum byte_order DetermineByteOrder(); +- extern void SwapBytesInWords( short *loc, int words ); +- +- +- #ifdef MACINTOSH +- extern void set_mac_file_attr(char[MAX_NAME_SIZE], short, OSType, +- OSType); +- #endif +- #ifdef MS_DOS +- extern void new_ext(char *filename, char *extname, char *newname); +- #endif +- +- #else +- extern FILE *OpenTableFile(); +- extern int read_bit_alloc(); +- extern int pick_table(); +- extern int js_bound(); +- extern void hdr_to_frps(); +- extern void WriteHdr(); +- extern void WriteBitAlloc(); +- extern void WriteScale(); +- extern void WriteSamples(); +- extern int NumericQ(); +- extern int BitrateIndex(); +- extern int SmpFrqIndex(); +- extern int memcheck(); +- extern void FAR *mem_alloc(); +- extern void mem_free(); +- extern void double_to_extended(); +- extern void extended_to_double(); +- extern int aiff_read_headers(); +- extern int aiff_seek_to_sound_data(); +- extern int aiff_write_headers(); +- extern int refill_buffer(); +- extern void empty_buffer(); +- extern void open_bit_stream_w(); +- extern void open_bit_stream_r(); +- extern void close_bit_stream_r(); +- extern void close_bit_stream_w(); +- extern void alloc_buffer(); +- extern void desalloc_buffer(); +- extern void back_track_buffer(); +- extern unsigned int get1bit(); +- extern void put1bit(); +- extern unsigned long look_ahead(); +- extern unsigned long getbits(); +- extern void putbits(); +- extern void byte_ali_putbits(); +- extern unsigned long byte_ali_getbits(); +- extern unsigned long sstell(); +- extern int end_bs(); +- extern int seek_sync(); +- extern void I_CRC_calc(); +- extern void II_CRC_calc(); +- extern void update_CRC(); +- extern void read_absthr(); +- +- extern unsigned int hget1bit(); +- extern unsigned long hgetbits(); +- extern unsigned long hsstell(); +- extern void hputbuf(); +- +- #ifdef MS_DOS +- extern void new_ext(); +- #endif +- #endif +- #endif +--- 0 ---- +diff -r -c -N encoder/configure lame3.50/configure +*** encoder/configure Wed Jan 22 02:43:13 1997 +--- lame3.50/configure Wed Dec 31 17:00:00 1969 +*************** +*** 1,742 **** +- #!/bin/sh +- ############################################################################ +- ## ISO MPEG Audio Subgroup Software Simulation Group (1996) +- ## ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- ## +- ## $Id: configure,v 1.1 1996/02/14 04:06:24 rowlands Exp $ +- ## +- ## $Log: configure,v $ +- ## Revision 1.1 1996/02/14 04:06:24 rowlands +- ## Initial revision +- ## +- ## Received from Mike Coleman +- ## +- ############################################################################ +- +- # Guess values for system-dependent variables and create Makefiles. +- # Generated automatically using autoconf version 2.3 +- # Copyright (C) 1992, 1993, 1994 Free Software Foundation, Inc. +- # +- # This configure script is free software; the Free Software Foundation +- # gives unlimited permission to copy, distribute and modify it. +- +- # Defaults: +- ac_help= +- ac_default_prefix=/usr/local +- # Any additions from configure.in: +- +- # Initialize some variables set by options. +- # The variables have the same names as the options, with +- # dashes changed to underlines. +- build=NONE +- cache_file=./config.cache +- exec_prefix=NONE +- host=NONE +- no_create= +- nonopt=NONE +- no_recursion= +- prefix=NONE +- program_prefix=NONE +- program_suffix=NONE +- program_transform_name=s,x,x, +- silent= +- site= +- srcdir= +- target=NONE +- verbose= +- x_includes=NONE +- x_libraries=NONE +- +- # Initialize some other variables. +- subdirs= +- +- ac_prev= +- for ac_option +- do +- +- # If the previous option needs an argument, assign it. +- if test -n "$ac_prev"; then +- eval "$ac_prev=\$ac_option" +- ac_prev= +- continue +- fi +- +- case "$ac_option" in +- -*=*) ac_optarg=`echo "$ac_option" | sed 's/[-_a-zA-Z0-9]*=//'` ;; +- *) ac_optarg= ;; 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"$ac_site_file" +- fi +- done +- +- if test -r "$cache_file"; then +- echo "loading cache $cache_file" +- . $cache_file +- else +- echo "creating cache $cache_file" +- > $cache_file +- fi +- +- ac_ext=c +- # CFLAGS is not in ac_cpp because -g, -O, etc. are not valid cpp options. +- ac_cpp='$CPP $CPPFLAGS' +- ac_compile='${CC-cc} -c $CFLAGS $CPPFLAGS conftest.$ac_ext 1>&5 2>&5' +- ac_link='${CC-cc} -o conftest $CFLAGS $CPPFLAGS $LDFLAGS conftest.$ac_ext $LIBS 1>&5 2>&5' +- +- if (echo "testing\c"; echo 1,2,3) | grep c >/dev/null; then +- # Stardent Vistra SVR4 grep lacks -e, says ghazi@caip.rutgers.edu. +- if (echo -n testing; echo 1,2,3) | sed s/-n/xn/ | grep xn >/dev/null; then +- ac_n= ac_c=' +- ' ac_t=' ' +- else +- ac_n=-n ac_c= ac_t= +- fi +- else +- ac_n= ac_c='\c' ac_t= +- fi +- +- +- +- # Extract the first word of "gcc", so it can be a program name with args. +- set dummy gcc; ac_word=$2 +- echo $ac_n "checking for $ac_word""... $ac_c" 1>&6 +- if eval "test \"`echo '$''{'ac_cv_prog_CC'+set}'`\" = set"; then +- echo $ac_n "(cached) $ac_c" 1>&6 +- else +- if test -n "$CC"; then +- ac_cv_prog_CC="$CC" # Let the user override the test. +- else +- IFS="${IFS= }"; ac_save_ifs="$IFS"; IFS="${IFS}:" +- for ac_dir in $PATH; do +- test -z "$ac_dir" && ac_dir=. +- if test -f $ac_dir/$ac_word; then +- ac_cv_prog_CC="gcc" +- break +- fi +- done +- IFS="$ac_save_ifs" +- test -z "$ac_cv_prog_CC" && ac_cv_prog_CC="cc" +- fi +- fi +- CC="$ac_cv_prog_CC" +- if test -n "$CC"; then +- echo "$ac_t""$CC" 1>&6 +- else +- echo "$ac_t""no" 1>&6 +- fi +- +- +- echo $ac_n "checking whether we are using GNU C""... $ac_c" 1>&6 +- if eval "test \"`echo '$''{'ac_cv_prog_gcc'+set}'`\" = set"; then +- echo $ac_n "(cached) $ac_c" 1>&6 +- else +- cat > conftest.c <<EOF +- #ifdef __GNUC__ +- yes; +- #endif +- EOF +- if ${CC-cc} -E conftest.c 2>&5 | egrep yes >/dev/null 2>&1; then +- ac_cv_prog_gcc=yes +- else +- ac_cv_prog_gcc=no +- fi +- fi +- echo "$ac_t""$ac_cv_prog_gcc" 1>&6 +- if test $ac_cv_prog_gcc = yes; then +- GCC=yes +- if test "${CFLAGS+set}" != set; then +- echo $ac_n "checking whether ${CC-cc} accepts -g""... $ac_c" 1>&6 +- if eval "test \"`echo '$''{'ac_cv_prog_gcc_g'+set}'`\" = set"; then +- echo $ac_n "(cached) $ac_c" 1>&6 +- else +- echo 'void f(){}' > conftest.c +- if test -z "`${CC-cc} -g -c conftest.c 2>&1`"; then +- ac_cv_prog_gcc_g=yes +- else +- ac_cv_prog_gcc_g=no +- fi +- rm -f conftest* +- +- fi +- echo "$ac_t""$ac_cv_prog_gcc_g" 1>&6 +- if test $ac_cv_prog_gcc_g = yes; then +- CFLAGS="-g -O" +- else +- CFLAGS="-O" +- fi +- fi +- else +- GCC= +- test "${CFLAGS+set}" = set || CFLAGS="-g" +- fi +- +- +- MATHLIBS="-lm" +- echo $ac_n "checking for -lieee""... $ac_c" 1>&6 +- if eval "test \"`echo '$''{'ac_cv_lib_ieee'+set}'`\" = set"; then +- echo $ac_n "(cached) $ac_c" 1>&6 +- else +- ac_save_LIBS="$LIBS" +- LIBS="-lieee $LIBS" +- cat > conftest.$ac_ext <<EOF +- #line 485 "configure" +- #include "confdefs.h" +- +- int main() { return 0; } +- int t() { +- main() +- ; return 0; } +- EOF +- if eval $ac_link; then +- rm -rf conftest* +- eval "ac_cv_lib_ieee=yes" +- else +- rm -rf conftest* +- eval "ac_cv_lib_ieee=no" +- fi +- rm -f conftest* +- LIBS="$ac_save_LIBS" +- +- fi +- if eval "test \"`echo '$ac_cv_lib_'ieee`\" = yes"; then +- echo "$ac_t""yes" 1>&6 +- MATHLIBS="$MATHLIBS -lieee" +- else +- echo "$ac_t""no" 1>&6 +- fi +- +- +- +- NINTSW="" +- echo $ac_n "checking for nint() in -lm""... $ac_c" 1>&6 +- if eval "test \"`echo '$''{'ac_cv_lib_m'+set}'`\" = set"; then +- echo $ac_n "(cached) $ac_c" 1>&6 +- else +- ac_save_LIBS="$LIBS" +- LIBS="-lm $LIBS" +- cat > conftest.$ac_ext <<EOF +- #line 521 "configure" +- #include "confdefs.h" +- +- int main() { return 0; } +- int t() { +- nint() +- ; return 0; } +- EOF +- if eval $ac_link; then +- rm -rf conftest* +- eval "ac_cv_lib_m=yes" +- else +- rm -rf conftest* +- eval "ac_cv_lib_m=no" +- fi +- rm -f conftest* +- LIBS="$ac_save_LIBS" +- +- fi +- if eval "test \"`echo '$ac_cv_lib_'m`\" = yes"; then +- echo "$ac_t""yes" 1>&6 +- NINTSW="-DHAVE_NINT" +- else +- echo "$ac_t""no" 1>&6 +- fi +- +- +- +- +- trap '' 1 2 15 +- cat > confcache <<\EOF +- # This file is a shell script that caches the results of configure +- # tests run on this system so they can be shared between configure +- # scripts and configure runs. It is not useful on other systems. +- # If it contains results you don't want to keep, you may remove or edit it. +- # +- # By default, configure uses ./config.cache as the cache file, +- # creating it if it does not exist already. You can give configure +- # the --cache-file=FILE option to use a different cache file; that is +- # what configure does when it calls configure scripts in +- # subdirectories, so they share the cache. +- # Giving --cache-file=/dev/null disables caching, for debugging configure. +- # config.status only pays attention to the cache file if you give it the +- # --recheck option to rerun configure. +- # +- EOF +- # Ultrix sh set writes to stderr and can't be redirected directly, +- # and sets the high bit in the cache file unless we assign to the vars. +- (set) 2>&1 | +- sed -n "s/^\([a-zA-Z0-9_]*_cv_[a-zA-Z0-9_]*\)=\(.*\)/\1=\${\1='\2'}/p" \ +- >> confcache +- if cmp -s $cache_file confcache; then +- : +- else +- if test -w $cache_file; then +- echo "updating cache $cache_file" +- cat confcache > $cache_file +- else +- echo "not updating unwritable cache $cache_file" +- fi +- fi +- rm -f confcache +- +- trap 'rm -fr conftest* confdefs* core core.* *.core $ac_clean_files; exit 1' 1 2 15 +- +- test "x$prefix" = xNONE && prefix=$ac_default_prefix +- # Let make expand exec_prefix. +- test "x$exec_prefix" = xNONE && exec_prefix='${prefix}' +- +- # Any assignment to VPATH causes Sun make to only execute +- # the first set of double-colon rules, so remove it if not needed. +- # If there is a colon in the path, we need to keep it. +- if test "x$srcdir" = x.; then +- ac_vpsub='/^[ ]*VPATH[ ]*=[^:]*$/d' +- fi +- +- trap 'rm -f $CONFIG_STATUS conftest*; exit 1' 1 2 15 +- +- # Transform confdefs.h into DEFS. +- # Protect against shell expansion while executing Makefile rules. +- # Protect against Makefile macro expansion. +- cat > conftest.defs <<\EOF +- s%#define \([A-Za-z_][A-Za-z0-9_]*\) \(.*\)%-D\1=\2%g +- s%[ `~#$^&*(){}\\|;'"<>?]%\\&%g +- s%\[%\\&%g +- s%\]%\\&%g +- s%\$%$$%g +- EOF +- DEFS=`sed -f conftest.defs confdefs.h | tr '\012' ' '` +- rm -f conftest.defs +- +- +- # Without the "./", some shells look in PATH for config.status. +- : ${CONFIG_STATUS=./config.status} +- +- echo creating $CONFIG_STATUS +- rm -f $CONFIG_STATUS +- cat > $CONFIG_STATUS <<EOF +- #! /bin/sh +- # Generated automatically by configure. +- # Run this file to recreate the current configuration. +- # This directory was configured as follows, +- # on host `(hostname || uname -n) 2>/dev/null | sed 1q`: +- # +- # $0 $ac_configure_args +- # +- # Compiler output produced by configure, useful for debugging +- # configure, is in ./config.log if it exists. +- +- ac_cs_usage="Usage: $CONFIG_STATUS [--recheck] [--version] [--help]" +- for ac_option +- do +- case "\$ac_option" in +- -recheck | --recheck | --rechec | --reche | --rech | --rec | --re | --r) +- echo "running \${CONFIG_SHELL-/bin/sh} $0 $ac_configure_args --no-create --no-recursion" +- exec \${CONFIG_SHELL-/bin/sh} $0 $ac_configure_args --no-create --no-recursion ;; +- -version | --version | --versio | --versi | --vers | --ver | --ve | --v) +- echo "$CONFIG_STATUS generated by autoconf version 2.3" +- exit 0 ;; +- -help | --help | --hel | --he | --h) +- echo "\$ac_cs_usage"; exit 0 ;; +- *) echo "\$ac_cs_usage"; exit 1 ;; +- esac +- done +- +- ac_given_srcdir=$srcdir +- +- trap 'rm -fr `echo "Makefile" | sed "s/:[^ ]*//g"` conftest*; exit 1' 1 2 15 +- +- # Protect against being on the right side of a sed subst in config.status. +- sed 's/%@/@@/; s/@%/@@/; s/%g$/@g/; /@g$/s/[\\\\&%]/\\\\&/g; +- s/@@/%@/; s/@@/@%/; s/@g$/%g/' > conftest.subs <<\CEOF +- $ac_vpsub +- $extrasub +- s%@CFLAGS@%$CFLAGS%g +- s%@CPPFLAGS@%$CPPFLAGS%g +- s%@CXXFLAGS@%$CXXFLAGS%g +- s%@DEFS@%$DEFS%g +- s%@LDFLAGS@%$LDFLAGS%g +- s%@LIBS@%$LIBS%g +- s%@exec_prefix@%$exec_prefix%g +- s%@prefix@%$prefix%g +- s%@program_transform_name@%$program_transform_name%g +- s%@CC@%$CC%g +- s%@MATHLIBS@%$MATHLIBS%g +- s%@NINTSW@%$NINTSW%g +- +- CEOF +- EOF +- cat >> $CONFIG_STATUS <<EOF +- +- CONFIG_FILES=\${CONFIG_FILES-"Makefile"} +- EOF +- cat >> $CONFIG_STATUS <<\EOF +- for ac_file in .. $CONFIG_FILES; do if test "x$ac_file" != x..; then +- # Support "outfile[:infile]", defaulting infile="outfile.in". +- case "$ac_file" in +- *:*) ac_file_in=`echo "$ac_file"|sed 's%.*:%%'` +- ac_file=`echo "$ac_file"|sed 's%:.*%%'` ;; +- *) ac_file_in="${ac_file}.in" ;; +- esac +- +- # Adjust relative srcdir, etc. for subdirectories. +- +- # Remove last slash and all that follows it. Not all systems have dirname. +- ac_dir=`echo $ac_file|sed 's%/[^/][^/]*$%%'` +- if test "$ac_dir" != "$ac_file" && test "$ac_dir" != .; then +- # The file is in a subdirectory. +- test ! -d "$ac_dir" && mkdir "$ac_dir" +- ac_dir_suffix="/`echo $ac_dir|sed 's%^\./%%'`" +- # A "../" for each directory in $ac_dir_suffix. +- ac_dots=`echo $ac_dir_suffix|sed 's%/[^/]*%../%g'` +- else +- ac_dir_suffix= ac_dots= +- fi +- +- case "$ac_given_srcdir" in +- .) srcdir=. +- if test -z "$ac_dots"; then top_srcdir=. +- else top_srcdir=`echo $ac_dots|sed 's%/$%%'`; fi ;; +- /*) srcdir="$ac_given_srcdir$ac_dir_suffix"; top_srcdir="$ac_given_srcdir" ;; +- *) # Relative path. +- srcdir="$ac_dots$ac_given_srcdir$ac_dir_suffix" +- top_srcdir="$ac_dots$ac_given_srcdir" ;; +- esac +- +- echo creating "$ac_file" +- rm -f "$ac_file" +- configure_input="Generated automatically from `echo $ac_file_in|sed 's%.*/%%'` by configure." +- case "$ac_file" in +- *Makefile*) ac_comsub="1i\\ +- # $configure_input" ;; +- *) ac_comsub= ;; +- esac +- sed -e "$ac_comsub +- s%@configure_input@%$configure_input%g +- s%@srcdir@%$srcdir%g +- s%@top_srcdir@%$top_srcdir%g +- " -f conftest.subs $ac_given_srcdir/$ac_file_in > $ac_file +- fi; done +- rm -f conftest.subs +- +- +- +- exit 0 +- EOF +- chmod +x $CONFIG_STATUS +- rm -fr confdefs* $ac_clean_files +- test "$no_create" = yes || ${CONFIG_SHELL-/bin/sh} $CONFIG_STATUS || exit 1 +- +--- 0 ---- +diff -r -c -N encoder/doc/html/contributors.html lame3.50/doc/html/contributors.html +*** encoder/doc/html/contributors.html Wed Dec 31 17:00:00 1969 +--- lame3.50/doc/html/contributors.html Tue Sep 21 14:38:25 1999 +*************** +*** 0 **** +--- 1,108 ---- ++ <!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN"> ++ <!-- saved from url=(0051)http://internet.roadrunner.com/~mt/mp3/history.html --> ++ <HTML><HEAD><TITLE>Contributors ++ ++ ++ ++

++

Contributors

++
++
++

The following people contributed to the LAME development:

++

Lauri Ahonen
++ Sakari Ailus
++ Tero Auvinen
++ Alex Ballegooy
++ Bob Bell
++ Lionel Bonnet
++ Gabriel Bouvigne
++ Erik de Castro Lopo
++ Michael Cheng
++ Goran Dokic
++ James Droppo
++ Alvaro Martinez Echevarria
++ Joakim Enerstam
++ Albert Faber
++ Nils Faerber
++ Mikhail M. Fedotov
++ Joseph Fourier
++ Jani Frilander
++ Steffan Haeuser
++ John Hayward-Warburton
++ Robert Hegemann
++ Ben "Jacobs"
++ Iwasa Kazmi
++ Leonid A. Kulakov
++ Juha Laukala
++ Rafael Luebbert
++ Macik
++ Lars Magne Ingebrigtsen
++ Scott Manley
++ Chris Matrakidis
++ Greg Maxwell
++ Scott Miller
++ Dan Nelson
++ Mike Oliphant
++ André Osterhues
++ Johannes Overmann
++ Jan Peman
++ Jan Rafaj
++ Gertjan van Ratingen
++ Miguel Revilla Rodriguez
++ Conrad Sanderson
++ William Schelter
++ Sigbj�rn Skj�ret
++ Nathan Slingerland
++ Stephane Tavenard
++ Mark Taylor
++ Brad Threatt
++ Takehiro Tominaga
++ Warren Toomey
++ Martin Weghofer
++ Gerhard Wesp
++ Alfred Weyers

++

 

++

Original ISO contributors:

++

Bill Aspromonte
++ Shaun Astarabadi
++ R. Bittner
++ Karlheinz Brandenburg
++ W. Joseph Carter
++ Jack Chang
++ Mike Coleman
++ Johnathan Devine
++ Ernst Eberlein
++ Dan Ellis
++ Peter Farrett
++ Jean-Georges Fritsch
++ Vlad Fruchter
++ Hendrik Fuchs
++ Bernhard Grill
++ Amit Gulati
++ Munsi Haque
++ Chuck Hsiao
++ Toshiyuki Ishino
++ Masahiro Iwadare
++ Earl Jennings
++ James Johnston
++ Leon v.d. Kerkhof
++ Don Lee
++ Mike Li
++ Yu-Tang Lin
++ Soren Neilsen
++ Simao F. Campos Neto
++ Mark Paley
++ Davis Pan
++ Tan Ah Peng
++ Kevin Peterson
++ Juan Pineda
++ Ernst F. Schroeder
++ Peter Siebert
++ Jens Spille
++ Sam Stewart
++ Al Tabayoyon
++ Kathy Wang
++ Franz-Otto Witte
++ Douglas Wong

++ ++ +diff -r -c -N encoder/doc/html/history.html lame3.50/doc/html/history.html +*** encoder/doc/html/history.html Wed Dec 31 17:00:00 1969 +--- lame3.50/doc/html/history.html Tue Sep 21 14:38:26 1999 +*************** +*** 0 **** +--- 1,930 ---- ++ ++ ++ ++ ++ ++ Cstarware : Do you remember where you wanted to go yesterday? ++ ++ ++ ++
++

++ History

++ Starting with LAME 3.0: ++
red = features and bug fixes which effect quality ++
blue = features and bug fixes which effect speed ++
black = usability, portability, other ++

++ ++

++ ++

++ LAME 3.28beta  September 15 1999

++ ++
    ++
  • ++ Serious bug fixed in high frequency MDCT coefficients.  ++ Huffman coding was reversing the order of the count1 block quadruples.   ++ (Leonid A. Kulakov)
    • ++ ++
  • ++ nint() problems under Tru64 unix fixed and preprocessor variable HAVE_NINT  ++ removed.  (Bob Bell)
    • ++ ++
  • ++ Compiler warning fixes and code cleanup   (Sigbjørn Skjæret, ++ Lionel Bonnet)
    • ++ ++
  • ++ USAGE file now includes suggestions for downsampling.  For low ++ bitrate encodings, proper downsampling can give dramatically better results.  ++ (John Hayward-Warburton)
    • ++
++ ++

++ LAME 3.27beta  September 12 1999

++ ++
    ++
  • ++ Several bugs in encode.c and l3bitstream.c fixed by Lionel Bonnet.
    • ++ ++
  • ++ Bugs in new VBR (#define RH) formula for mono input file and mid/side encoding ++ fixed.
    • ++
++ ++

++ LAME 3.26beta  September 10 1999

++ ++
    ++
  • ++ The "-m m" option (mono .mp3 file) will automatically mix left and right ++ channels if the input file is stereo.  (Alfred Weyers)
    • ++ ++
  • ++ New quant_compare algorithm (method for deciding ++ which of two quantizations is better) enabled with -X4 (Greg Maxwell)
    • ++ ++
  • ++ New mid/side VBR bit allocation formula.  Mid ++ channel bits are set by the quality requirements, and then the side channel ++ uses a reduced number of bits (in a proportion coming from the fixed bitrate ++ code).  This might not be optimal, but it should be pretty good and ++ no one knows what the optimal solution should be.  (Greg Maxwell)
    • ++ ++
  • ++ New VBR (#define RH) tunings based on detailed listening ++ tests by Macik and Greg Maxwell.
    • ++ ++
  • ++ Sigbjørn Skjæret fixed several compiler warnings (which turned ++ out to be potential bugs)
    • ++ ++
  • ++ Takehiro Tominaga fixed a low bitrate bug in reduce_side()
    • ++ ++
  • ++ Alfred Weyers fixed some buffer overflows.
    • ++ ++
  • ++ New ATH (absolute threshold of hearing) formula replaces ++ buggy ISO code, and adds  analog silence treatment  (removal ++ of coefficients below below ATH).   These are turned on by default ++ but have not been fully tested.  (Robert Hegemann)
    • ++ ++
  • ++ Bug in short block spreading function fixed.  ++ (Robert Hegemann)
    • ++
++ ++

++ LAME 3.25beta  August 22 1999

++ ++
    ++
  • ++ Sigbjørn Skjæret fixed a zero byte malloc call.  This ++ bug  was introduced in 3.24 and causes problems on non Linux systems.
    • ++ ++
  • ++ Bit allocation routines would sometimes allocate more than 4095 bits to ++ one channel of one granule.  A couple of people reported problems ++ that might be caused by this, especially at higher bitrates.
    • ++ ++
  • ++ Nils Faerber updated the man page and fixed many of the compiler warnings.
    • ++ ++
     
++ ++

++ LAME 3.24beta   August 15 1999

++ ++
    ++
  • ++ This release contains the following new code (for developers) which is ++ disabled by default:
    • ++ ++
  • ++ Robert Hegemann:  Completely overhauled VBR code.  Now computes ++ exact number of bits required for the given qualty and then quantized with ++ the appropriate bitrate.
    • ++ ++
  • ++ Several new quantization quality measures.
    • ++
++ ++

++ LAME 3.23beta  August 8 1999

++ ++
    ++
  • ++ Very nice continuously updated VBR histogram display from Iwasa Kazmi.  ++ (disabled with --nohist).
    • ++ ++
  • ++ More modulerization work.  The encoding engine can now be compiled ++ into libmp3lame, but the interface is awkward.
    • ++ ++
  • ++ Bug fixed in FFT Hann window formula (Leonid A. Kulakov).
    • ++ ++
  • ++ New LAME logo on the download page.  Created by Chris Michalisles.
    • ++ ++
  • ++ Several VBR algorithm improvements from Robert Hegemann.  ++ New quantization noise metrics and VBR quality measure takes into account ++ mid/side encoding.  Should produce smaller files with the same quality, ++ especially when using jstereo.
    • ++
++ ++

++ LAME 3.22beta  July 27 1999

++ ++
    ++
  • ++ Downsampling (stereo to mono) bug with MPEG2 fixed.  (Mike Oliphant)
    • ++ ++
  • ++ Downsampling now merges L & R channels - before it only took the L ++ channel.
    • ++ ++
  • ++ More modularization and code cleanup from Albert Faber and myself.
    • ++ ++
  • ++ Input filesize limit removed for raw pcm input files.  For other file ++ types, LAME will still only read the first 2^32 samples, (27 hours of playing ++ time at 44.1kHz).
    • ++ ++
     
++ ++

++ LAME 3.21beta  July 26 1999

++ ++
    ++
  • ++ Correct Mid/Side masking thresholds for JSTEREO mode!  ++ This is enabled with -h.  It  makes LAME about 20% slower since ++ it computes psycho-acoustics for L,R Mid and Side channels.
    • ++ ++
  • ++ "Analog silence" threshold added.  Keeps VBR ++ from upping the bitrate during very quite passages.  (Robert.Hegemann)
    • ++ ++
  • ++ New VBR quality setting from Robert Hegemann.  ++ It is based on the idea that distortion at lower bit rates sounds worse ++ than at higher bitrates, and so the allowed distortion (VBR quality setting) ++ is proportional to the bitrate.  Because of this, default minimum ++ bitrate is now 32kbs.
    • ++ ++
  • ++ Expermental subblock gain code enabled with -Z.
    • ++ ++
  • ++ New "-r" option for raw pcm input files.  With -r, LAME will not do ++ any fseek()'s or look for wav and aiff headers on the input file.
    • ++ ++
  • ++ Bug fixes in mp3x (frame analyzer) for viewing frames near end of the file.
    • ++ ++
  • ++ Bug fixed to allow setting the sampling rate of raw pcm input files.
    • ++
++ ++

++ LAME 3.20beta  July 19 1999

++ ++
    ++
  • ++ Bug in get_audio.c fixed.  Libsndfile wrappers would not compile (Miguel ++ Revilla  Rodriguez)
    • ++ ++
  • ++ Nils Faerber found some unitialized variables and some wierd extranous ++ computations in filter_subband, now fixed.  This was causing seg faults ++ on some machines.
    • ++ ++
     
++ ++

++ LAME 3.19beta  July 18 1999

++ ++
    ++
  • ++ Oops!  Robert Hegemann immediatly found a bug ++ in the new  (old -Z option) quantization code.  calc_noise1 was ++ not returning tot_noise, so non ms-stereo frames were buggy.
    • ++ ++
     
++ ++

++ LAME 3.18beta  July 17 1999

++ ++
    ++
  • ++ Many psycho-acoustic bug fixes.  Dan Nelson ++ discovered a bug in MPEG2: For short blocks, the code assumes 42 partition ++ bands.  MPEG1 sometimes has less, MPEG2 can have more.  In MPEG1, ++ this bug would not have effected the output if your compiler initializes ++ static variables to 0 on creation.  In MPEG2 it leads to array out-of-bounds ++ access errors. Finally, there was a related bug in MPEG1/MPEG2, short & ++ long blocks where the energy above 16kHz was all added to partition band ++ 0. (the lowest frequeny partition band!)
    • ++ ++
  • ++ The -Z option (Gabriel Bouvigne's idea of using total ++ quantization noise to choose between two quantizations with the same value ++ of "over") is now the default.  I believe this helps remove the trilling ++ sound in Jan's testsignal4.wav.  The quality of testsignal2.wav and ++ testsignal4.wav are now better than Xing and getting closer to FhG.
    • ++ ++
  • ++ Bug fixes in frame & sample count for downsampling mode. (ben "jacobs")
    • ++ ++
  • ++ Patches to improve modulization.  (ben "jacobs")
    • ++
++ ++

++ LAME 3.17beta  July 11 1999

++ ++
    ++
  • ++ substantial code cleanup towards goal of making LAME more modular.
    • ++
++ ++

++ LAME 3.16beta  July 11 1999

++ ++
    ++
  • ++ New tunings of window switching, and better bit allocation ++ based on pe.  (Jan Rafaj.  improves both testsignal2.wav and ++ testsignal4.wav).
    • ++ ++
  • ++ Bug in mid/side quantization when side channel was ++ zero fixed.  (Albert Faber)
    • ++ ++
  • ++ Removed some extranous computations in l3psy.c (Robert Hegemann)
    • ++ ++
  • ++ More detailed timing status info, including hours display. (Sakari Ailus) ++ and percentage indicator (Conrad Sanderson).
    • ++ ++
  • ++ Window_subband and calc_noise1,calc_noise2 speedups.  ++ Quantize_xrpow speedup should be significant on non GNU/intel systems. ++ (Mike Cheng)
    • ++ ++
  • ++ Better initial guess for VBR bitrate.  Should ++ speed up VBR encoding.  (Gabriel Bouvigne)
    • ++ ++
  • ++ More advanced .wav header parsing.  fixes bugs involving click in ++ first frame. (Robert.Hegemann)
    • ++ ++
  • ++ Correct filesize and total frame computation when using LIBSNDFILE (ben ++ "jacobs")
    • ++ ++
  • ++ Click in last frame (buffering problem) when using libsndfile fixed.
    • ++ ++
  • ++ Audio I/O code overhauled.  There is now a uniform audio i/o interface ++ to libsndfile or the LAME built in wav/aiff routines.  All audio i/o ++ code localized to get_audio.c.
    • ++ ++
     
++ ++

++ LAME 3.15beta

++ ++
    ++
  • ++ times()/clock() problem fixed for non-unix OS.  (Ben "Jacobs")
    • ++ ++
  • ++ Fixed uninitialized pe[] when using fast mode.  (Ben "Jacobs")
    • ++
++ ++

++ LAME 3.13  June 24 1999

++ ++
    ++
  • ++ Patches for BeOS from Gertjan van Ratingen.
    • ++ ++
  • ++ Makefile info for OS/2 Warp 4.0  (from dink.org).
    • ++ ++
  • ++ Status display now based on wall clock time, not cpu time.
    • ++ ++
  • ++ mem_alloc no longer allocates twice as much memory as needed (Jan Peman).
    • ++
++ ++

++ 3.12pre9

++ ++
    ++
  • ++ Updated BLADEDLL code to handle recent changes (Albert Faber).
    • ++ ++
  • ++ Bug fixed in parsing options when not using GTK (Albert Faber).
    • ++ ++
  • ++ MPEG2 Layer III psycho acoustics now working.
    • ++ ++
  • ++ Improved huffman encoding Chris Matrakidis. (10% ++ faster).  I dont know how he finds these improvements!  LAME ++ with full quality now encodes faster than real time on my PII 266.
    • ++ ++
  • ++ Fixed time display when encoding takes more than 60 minutes.
    • ++
++ ++

++ 3.12pre8

++ ++
    ++
  • ++ New mid/side stereo ++ criterion.  LAME will use mid/side stereo only when the difference ++ between L & R masking thresholds (averaged over all scalefactors) is ++ less then 5db.  In several test samples it does a very good job mimicking ++ the FhG encoder.
    • ++ ++
  • ++ Bug in mid/side stereo fixed:  independent variation ++ of mid & side channel scalefactors disabled.  Because of the way ++ outer_loop is currently coded, when encoding mid/side coefficietns using ++ left/right thresholds, you have to vary the scalefactors simultaneously.
    • ++ ++
  • ++ Bug in side/mid energy ratio calculation fixed. (Thanks ++ to Robert Hegemann)
    • ++ ++
  • ++ Default mode is stereo (not jstereo) if bitrate is chosen as 192kbs or ++ higher.  Tero Auvinen first pointed out that FhG seems to think at ++ 160kbs, their encoder is so good it doesn't need jstereo tricks. Since ++ LAME is not as good as FhG, I am going to claim that 192kbs LAME is so ++ good it doens't need jstereo tricks, and thus it is disabled by default.
    • ++ ++
  • ++ WAV header parsing for big-endian machines, and automatic detection of ++ big-endian machines.  (Thanks to  Sigbjørn Skjæret).
    • ++ ++
  • ++ added 56 sample delay to sync LAME with FhG.
    • ++ ++
  • ++ MP3x (frame analyzer) can now handle MPEG2 streams.
    • ++
++ ++

++ 3.12pre7

++ ++
    ++
  • ++ MPEG2 layer III now working!  lower bit rates (down to 8kbs) and 3 ++ more sampling frequencies:  16000, 22050, 24000Hz. Quality is poor ++ - the psy-model does not yet work with these sampling frequencies.
    • ++ ++
  • ++ Fixed "ERROR: outer_loop(): huff_bits < 0." bug when using VBR.
    • ++ ++
  • ++ bash and sh scripts to run LAME on multiple files now included.  (from ++ Robert Hegemann and Gerhard Wesp respectively)
    • ++ ++
  • ++ bug fix in encoding times for longer files from  (Alvaro Martinez ++ Echevarria)
    • ++ ++
  • ++ yet another segfault in the frame analyzer fixed.
    • ++ ++
  • ++ ISO psy-model/bit allocation routines removed.  This allowed makeframe() ++ to be made much simpler, and most of the complicated buffering is now gone. ++ Eventually I would like the encoding engine to be a stand alone library.
    • ++
++ ++

++ 3.12pre6

++ ++
    ++
  • ++ Better VBR tuning.  Find minimum bitrate with ++ distortion less than the allows maximum.  A minimum bit rate is imposed ++ on frames with short blocks (where the measured distortion can not be trusted).   ++ A minimum frame bitrate can be specified with -b, default=64kbs.
    • ++ ++
  • ++ LIBSNDFILE support.  ++ With libsndfile, LAME can encode almost all sound formats.  Albert ++ Faber did the work for this, including getting libsndfile running under ++ win32.
    • ++ ++
  • ++ CRC checksum now working!   (Thanks to Johannes Overmann )
    • ++ ++
  • ++ frame analyzer will now work with mono .mp3 files
    • ++ ++
  • ++ more code tweeks from Jan Peman.
    • ++ ++
  • ++ Compaq-Alpha(Linux) fixes and speedups from Nils ++ Faerber.
    • ++ ++
  • ++ Faster bin_search_StepSize from Juha Laukala.
    • ++ ++
  • ++ Faster quantize() from Mike Cheng
    • ++ ++
  • ++ Faster quantize_xrpow() from Chris Matrakidis.  ++ xrpow_flag removed since this option is now on by default.
    • ++ ++
  • ++ Fixed .wav header parsing from Nils Faerber.
    • ++ ++
  • ++ Xing VBR frame info header code from Albert Faber.   "Xing" and ++ "LAME 3.12" embedded in first frame.
    • ++ ++
  • ++ Bug in VBR bit allocation based on "over" value fixed.
    • ++
++ ++

++ LAME 3.11  June 3 1999

++ ++
++
  • ++ Almost all warnings (-Wall) now fixed!  (Thanks to Jan Peman)
    • ++ ++
  • ++ More coding improvements from Gabriel Bouvigne and Warren Toomey.
    • ++ ++
  • ++ VBR  (variable bit rate).   Increases ++ bit rate for short blocks and for frames where the number of bands containing ++ audible distortion is greater than a given value.  Much tuning needs ++ to be done.
    • ++ ++
  • ++ Patch to remove all atan() calls from James Droppo.
    • ++
    ++ ++

    ++ LAME 3.10 May 30 1999

    ++ ++
      ++
    • ++ Fast mode (-f)  disables psycho-acoustic model ++ for real time encoding on older machines.  Thanks to Lauri Ahonen ++ who first sent a patch for this.
      • ++ ++
    • ++ New bit reservoir usage scheme to accommodate the ++ new pre-echo detection formulas.
      • ++ ++
    • ++ Tuning of AWS and ENER_AWS pre-echo formulas by Gabriel ++ Bouvigne and myself.  They work great!  now on by default.
      • ++ ++
    • ++ In jstereo, force blocktypes for left & right channels to be identical.  ++ FhG seems to do this.  It can be disabled with "-d".
      • ++ ++
    • ++ Patches to compile MP3x under win32 (Thanks to Albert Faber).
      • ++ ++
    • ++ bin_serach_stepsize limited to a quantizationStepSize ++ of  -210 through 45.
      • ++ ++
    • ++ outer_loop()  will now vary Mid & Side scalefactors ++ independently.  Can lead to better quantizations, but it is slower ++ (twice as many quantizations to look at).  Running with "-m f" does ++ not need this and will run at the old speed
      • ++ ++
    • ++ Bug in inner_loop would allow quantizations larger ++ than allowed.  (introduced in lame3.04, now fixed.)
      • ++ ++
    • ++ Updated HTML documentation from Gabriel Bouvigne.
      • ++ ++
    • ++ Unix  man page from William Schelter.
      • ++ ++
    • ++ numlines[] bug fixed.  (Thanks to Rafael Luebbert, ++ MPecker author).
      • ++ ++
    • ++ Quantization speed improvements from Chirs Matrakidis.
      • ++ ++
    • ++ When comparing quantizations with the same number ++ of bands with audible distortion, use the one with the largest scalefactors, ++ not the first one outer_loop happened to find.
      • ++ ++
    • ++ Improved defination of best quantization when using -f (fast mode).
      • ++ ++
    • ++ subblock code now working.  But no algorithm to choose subblock gains ++ yet.
      • ++ ++
    • ++ Linux now segfaults on floating point exceptions.  Should prevent ++ me from releasing binaries that crash on other operating systems.
      • ++
    ++ ++

    ++ May 22 1999

    ++ ++
      ++
    • ++ Version 3.04 released.
      • ++ ++
    • ++ Preliminary documentation from Gabriel Bouvigne.
      • ++ ++
    • ++ I wouldn't have thought it was possible, but now ++ there are even more speed improvements from Chris Matrakidis!  Removed ++ one FFT when using joint stereo, and many improvements in loop.c.
      • ++ ++
    • ++ "Fake" ms_stereo mode renamed "Force" ms_stereo since it forces mid/side ++ stereo on all frames.  For some music this is said to be a problem, ++ but for most music mode is probably better than the default jstereo because ++ it uses specialized mid/side channel masking thresholds.
      • ++ ++
    • ++ Small bugs in Force ms_stereo mode fixed.
      • ++ ++
    • ++ Compaq Alpha fixes from Nathan Slingerland.
      • ++ ++
    • ++ Some new experimental pre-echo detection formulas ++ in l3psy.c (#ifdef AWS and #ifdef ENER_AWS, both off by default.  ++ Thanks to Gabriel Bouvigne and Andre Osterhues)
      • ++ ++
    • ++ Several bugs in the syncing of data displayed by mp3x (the frame analyzer) ++ were fixed.
      • ++ ++
    • ++ highq (-h) option added.  This turns on things (just one so far) that ++ should sound better but slow down LAME.
      • ++
    ++ May 18 1999 ++
      ++
    • ++ Version 3.03 released.
      • ++ ++
    • ++ Faster (20%) & cleaner FFT (Thanks to Chris Matrakidis  ++ http://www.geocities.com/ResearchTriangle/8869/fft_summary.html)
      • ++ ++
    • ++ mods so it works with VC++ (Thanks to Gabriel Bouvigne, www.mp3tech.org)
      • ++ ++
    • ++ MP3s marked "original" by default  (Thanks to Gabriel Bouvigne, www.mp3tech.org)
      • ++ ++
    • ++ Can now be compiled into a BladeEnc compatible .DLL   (Thanks ++ to  Albert Faber, CDex author)
      • ++ ++
    • ++ Patches for "silent mode" and stdin/stdout  (Thanks to Lars Magne ++ Ingebrigtsen)
      • ++ ++
    • ++ Fixed rare bug: if a long_block is sandwiched between ++ two short_blocks, it must be changed to a short_block, but the short_block ++ ratios have not been computed in l3psy.c.  Now always compute short_block ++ ratios just in case.
      • ++ ++
    • ++ Fixed bug with initial quantize step size when many ++ coefficients are zero.  (Thanks to Martin Weghofer).
      • ++ ++
    • ++ Bug fixed in MP3x display of audible distortion.
      • ++ ++
    • ++ improved status display (Thanks to Lauri Ahonen).
      • ++
    ++ ++

    ++ May 12 1999

    ++ ++
      ++
    • ++ Version 3.02 released.
      • ++ ++
    • ++ encoder could use ms_stereo even if channel 0 and ++ 1 block types were different.  (Thanks to Jan Rafaj)
      • ++ ++
    • ++ added -k option to disable the 16kHz cutoff at 128kbs.  ++ This cutoff is never used at higher bitrates. (Thanks to Jan Rafaj)
      • ++ ++
    • ++ modified pe bit allocation formula to make sense ++ at bit rates other than 128kbs.
      • ++ ++
    • ++ fixed l3_xmin initialization problem which showed up under FreeBSD.  ++ (Thanks to Warren Toomey)
      • ++
    ++ May 11 1999 ++
      ++
    • ++ Version 3.01 released
      • ++ ++
    • ++ max_name_size increased to 300  (Thanks to Mike Oliphant)
      • ++ ++
    • ++ patch to allow seeks on input file (Thanks to Scott Manley)
      • ++ ++
    • ++ fixes for mono modes (Thanks to everyone who pointed this out)
      • ++ ++
    • ++ overflow in calc_noise2 fixed
      • ++ ++
    • ++ bit reservoir overflow when encoding lots of frames with all zeros  ++ (Thanks to Jani Frilander)
      • ++
    ++ ++


    May 10 1999 ++

      ++
    • ++ Version 3.0 released
      • ++ ++
    • ++ added GPSYCHO (developed by Mark Taylor)
      • ++ ++
    • ++ added MP3x (developed by Mark Taylor)
      • ++ ++
    • ++ LAME now maintained by Mark Taylor
      • ++
    ++ November 8 1998 ++
      ++
    • ++ Version 2.1f released
      • ++ ++
    • ++ 50% faster filter_subband() routine in encode.c contributed by James Droppo
      • ++
    ++ November 2 1998 ++
      ++
    • ++ Version 2.1e released.
      • ++ ++
    • ++ New command line switch -a auto-resamples a stereo input file to ++ mono.
      • ++ ++
    • ++ New command line switch -r resamples from 44.1khz to 32khz [this ++ switch doesn't work really well. Very tinny sounding output files. Has ++ to do with the way I do the resampling probably]
      • ++ ++
    • ++ Both of these were put into the ISO code in the encode.c file, and are ++ simply different ways of filling the input buffers from a file.
      • ++
    ++ October 31 1998 ++
      ++
    • ++ Version 2.1d released
      • ++ ++
    • ++ Fixed memory alloc in musicin.c (for l3_sb_sample)
      • ++ ++
    • ++ Added new command line switch (-x) to force swapping of byte order
      • ++ ++
    • ++ Cleaned up memory routines in l3psy.c. All the mem_alloc() and free() routines ++ where changed so that it was only done once and not every single ++ time the routine was called.
      • ++ ++
    • ++ Added a compile time switch -DTIMER that includes all timing info. It's ++ a switch for the time being until some other people have tested on their ++ system. Timing code has a tendency to do different things on different ++ platforms.
      • ++
    ++ October 18 1998 ++
      ++
    • ++ Version 2.1b released.
      • ++ ++
    • ++ Fixed up bug: all PCM files were being read as WAV.
      • ++ ++
    • ++ Played with the mem_alloc routine to fix crash under amigaos (just allocating ++ twice as much memory as needed). Might see if we can totally do without ++ this routine. Individual malloc()s where they are needed instead
      • ++ ++
    • ++ Put Jan Peman's quality switch back in. This reduces quality via the '-q ' ++ switch. Fun speedup which is mostly harmless if you're not concerned with ++ quality.
      • ++ ++
    • ++ Compiling with amiga-gcc works fine
      • ++
    ++ October 16 1998 ++
      ++
    • ++ Version 2.1a released. User input/output has been cleaned up a bit. WAV ++ file reading is there in a very rudimentary sense ie the program will recognize ++ the header and skip it, but not read it. The WAV file is assumed to be ++ 16bit stereo 44.1khz.
      • ++
    ++ October 6 1998 ++
      ++
    • ++ Version 2.1 released with all tables now incorporated into the exe. Thanks ++ to Lars Magne Ingebrigtseni(larsi@ifi.uio.no)
      • ++
    ++ October 4 1998In response to some concerns about the quality of ++ the encoder, I have rebuilt the encoder from scratch and carefully compared ++ output at all stages with the output of the unmodified ISO encoder. Version2.0 ++ of LAME is built from the ISO source code (dist10), and incorporates modifications ++ from myself and the 8hz effort. The output file from LAME v2.0 is identical ++ to the output of the ISO encoder for my test file.Since I do not have heaps ++ of time, I left the ISO AIFF file reader in the code, and did not incorporate ++ a WAV file reader.Added section on qualityOctober ++ 1 1998 ++
      ++
    • ++ Updated web page and released LAME v1.0
      • ++
    ++ Up to September 1998 ++
      Working on the 8hz source code... ++
        ++
      • ++ Patched the 8hz source code
        • ++ ++
      • ++ 45% faster than original source (on my freebsd p166).
        • ++ ++
          ++
        • ++ m1 - sped up the mdct.c and quantize() functions [MDCTD, MDCTD2, LOOPD]
          • ++ ++
        • ++ m2 - sped up the filter_subband routine using Stephane Tavenard's ++ work from musicin [FILTST]
          • ++ ++
        • ++ m2 - minor cleanup of window_subband [WINDST2]
          • ++ ++
        • ++ m2 - Cleaned up a few bits in l3psy.c. Replaced a sparse matrix multiply ++ with a hand configured unrolling [PSYD]
          • ++ ++
        • ++ m3 - (amiga only) Added in the asm FFT for m68k (based on sources from ++ Henryk ++ Richter and Stephane Tavenard)
          • ++ ++
        • ++ m4 - raw pcm support back in
          • ++ ++
        • ++ m5 - put in a byte-ordering switch for raw PCM reading (just in case)
          • ++ ++
        • ++ m6 - reworked the whole fft.c file. fft now 10-15% faster.
          • ++ ++
        • ++ m7 - totally new fft routine. exploits fact that this is a real->complex ++ fft. About twice as fast as previous fastest fft (in m6). (C fft routine ++ is faster than the asm one on an m68k!)
          • ++ ++
        • ++ m8
          • ++ ++
            ++
          • ++ - Now encodes from stdin. Use '-' as the input filename. Thanks to Brad ++ Threatt
            • ++ ++
          • ++ - Worked out that the 1024point FFT only ever uses the first 6 phi values, ++ and the first 465 energy values. Saves a bunch of calculations.
            • ++ ++
          • ++ - Added a speed-up/quality switch. Speed is increased but quality is decreased ++ slightly. ++ My ears are bad enough not to be able to notice the difference in quality ++ at low settings :). Setting '-q 1' improves speed by about 10%. '-q 100' ++ improves speed by about 26%. Enoding of my test track goes from 111s (at ++ default '-q 0') to 82s (at -q 100). Thanks to Jan Peman for this ++ tip.
            • ++
          ++ ++
        • ++ m9 - fixed an error in l3psy.c. numlines[] is overwritten with incorrect ++ data. Added a new variable numlines_s[] to fix this. Thanks again to Jan ++ Peman.
          • ++ ++
        • ++ m10 - Down to 106 seconds by selecting a few more compiler options. Also ++ added a pow20() function in l3loop.c to speed up (ever so slightly) calls ++ to pow(2.0, x)
          • ++ ++
        • ++ m11
          • ++ ++
            ++
          • ++ No speedups. Just cleaned up some bits of the code.
            • ++ ++
          • ++ Changed K&R prototyping to 'normal' format. Thanks to Steffan Haeuser ++ for his help here.
            • ++ ++
          • ++ Changed some C++ style comments to normal C comments in huffman.c
            • ++ ++
          • ++ Removed the #warning from psy_data.h (it was getting annoying!)
            • ++ ++
          • ++ Removed reference in bitstream.c to malloc.h. Is there a system left where ++ malloc.h hasn't been superceded by stdlib.h?
            • ++
          ++ ++
        • ++ In Progess:
          • ++ ++
            ++
          • ++ my PSYD hack for the spreading functions is only valid for 44.1khz - Should ++ really put in a "if freq = 44.1khz" switch for it. Someone might want to ++ extend the speedup for 48 and 32khz.
            • ++ ++
          • ++ Putting in Jan Peman's quantanf_init speedup.
            • ++
          ++
        ++
      ++
    ++ ++
    mike/cstar ++ ++ +diff -r -c -N encoder/doc/html/index.html lame3.50/doc/html/index.html +*** encoder/doc/html/index.html Wed Dec 31 17:00:00 1969 +--- lame3.50/doc/html/index.html Tue Sep 21 14:38:26 1999 +*************** +*** 0 **** +--- 1,54 ---- ++ ++ ++ ++ L.A.M.E. V3 Next Generation High-End MPEG Layer III Encoding ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++

    ++

    LAME V3.28
    ++     ++


    ++     Next Generation High-End MPEG Layer III Encoding
    ++     ++

    ++

    Open Source Project
    ++ http://www.sulaco.org/mp3

    ++

    ++

    ++

    ++

    ++

    ++
    ++
    ++   ++ ++     ++ ++ +diff -r -c -N encoder/doc/html/lame.css lame3.50/doc/html/lame.css +*** encoder/doc/html/lame.css Wed Dec 31 17:00:00 1969 +--- lame3.50/doc/html/lame.css Tue Sep 21 14:38:27 1999 +*************** +*** 0 **** +--- 1,24 ---- ++ BODY { ++ font-family: Arial, Helvetica, sans-serif; ++ position: absolute; ++ padding-left: 1cm; ++ margin-left: 1cm; ++ } ++ ++ h1, h2, h3 { ++ margin-left: -1cm; text-align: left ++ } ++ ++ TABLE { text-align: left; font-family: Arial, Helvetica, sans-serif;} ++ TH {background-color: #C0C0C0} ++ TD {font-family: Arial, Helvetica, sans-serif;} ++ ++ SMALL.TINY { font-size : xx-small } ++ SMALL.SCRIPTSIZE { font-size : xx-small } ++ SMALL.FOOTNOTESIZE { font-size : x-small } ++ SMALL.SMALL { font-size : small } ++ BIG.LARGE { font-size : large } ++ BIG.XLARGE { font-size : x-large } ++ BIG.XXLARGE { font-size : xx-large } ++ BIG.HUGE { font-size : xx-large } ++ BIG.XHUGE { font-size : xx-large } +diff -r -c -N encoder/doc/html/node5.html lame3.50/doc/html/node5.html +*** encoder/doc/html/node5.html Wed Dec 31 17:00:00 1969 +--- lame3.50/doc/html/node5.html Tue Sep 21 14:38:27 1999 +*************** +*** 0 **** +--- 1,51 ---- ++ ++ ++ ++ For the impatient ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++

    For the impatient

    ++ If you are new to audio compression, you should read section 1.1 ++ for an introduction about audio compression and MPEG Layer-3. ++

    If, however, you want to jump right into the business of sound compression, ++ then Section 1.2 will show you some prefabricated ++ command lines that will give you compressed audio streams right away. ++

    If you are an expert in audio coding already, the command line switch reference ++ page might come in handy. ++


    ++

    ++   ++ ++
    ++     ++ ++ +diff -r -c -N encoder/doc/html/node6.html lame3.50/doc/html/node6.html +*** encoder/doc/html/node6.html Wed Dec 31 17:00:00 1969 +--- lame3.50/doc/html/node6.html Tue Sep 21 14:38:27 1999 +*************** +*** 0 **** +--- 1,133 ---- ++ ++ ++ ++ Introduction ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ Subsections ++ ++ ++
    ++

    Introduction

    ++   There is a lot of confusion surrounding the terms ++  audio compression,  audio ++ encoding, and  audio decoding. This section ++ will give you an overview what audio coding (another one of these terms...) is ++ all about. ++

    ++

    The purpose of audio compression ++

    ++

    Up to the advent of audio compression, high-quality digital audio data took ++ a lot of hard disk space to store. Let us go through a short example. ++

    You want to, say, sample your favorite 1-minute song and store it on your ++ harddisk. Because you want CD quality, you sample at 44.1 kHz, stereo, ++ with 16 bits per sample. ++

    44100 Hz means that you have 44100 values per second coming in from your sound ++ card (or input file). Multiply that by two because you have two channels. Multiply ++ by another factor of two because you have two bytes per value (that's what 16 ++ bit means). The song will take up 44100 samples/s · 2 channels ++ · 2 bytes/sample · 60 s/min ~ 10 MBytes of storage space ++ on your harddisk. ++

    If you wanted to download that over the internet, given an average 28.8 modem, ++ it would take you (at least) 10000000 bytes · 8 bits/byte / (28800 ++ bits/s) · / (60 s/min) ~ 45 minutes ++

    Just to download one minute of music!
    ++

    Digital audio coding, which - in this context - is synonymously called digital ++ audio compression as well, is the art of minimizing storage space (or channel ++ bandwidth) requirements for audio data. Modern perceptual audio coding techniques ++ (like MPEG Layer-3) exploit the properties of the human ear (the perception ++ of sound) to achieve a size reduction by a factor of 12 with little or no perceptible ++ loss of quality. ++

    Therefore, such schemes are the key technology for high quality low bit-rate ++ applications, like soundtracks for CD-ROM games, solid-state sound memories, ++ Internet audio, digital audio broadcasting systems, and the like. ++

    ++

    The two parts of audio compression ++

    ++

    Audio compression really consists of two parts. The first part, called  encoding, ++ transforms the digital audio data that resides, say, in a WAVE file, into a ++ highly compressed form called  bitstream. To play ++ the bitstream on your soundcard, you need the second part, called  decoding. ++ Decoding takes the bitstream and re-expands it to a WAVE file. ++

    The program that effects the first part is called an audio encoder. ++ LAME is such an encoder . The program that does the second part is called ++ an audio decoder. One well-known MPEG Layer-3 decoder is X11Amp, ++ another mpg123. Both can be found on http://www.mp3tech.org . ++

    Compression ratios, bitrate and quality ++

    ++

    It has not been explicitly mentioned up to now: What you end up with after ++ encoding and decoding is not the same sound file anymore: All superflous information ++ has been squeezed out, so to say. It is not the same file, but it will ++ sound the same - more or less, depending on how much compression had ++ been performed on it. ++

    Generally speaking, the lower the compression ratio achieved, the better the ++ sound quality will be in the end - and vice versa. Table 1.1 ++ gives you an overview about quality achievable. ++

    Because compression ratio is a somewhat unwieldy measure, experts use the ++ term  bitrate when speaking of the strength of ++ compression. Bitrate denotes the average number of bits that one second of audio ++ data will take up in your compressed bitstream. Usually the units used will ++ be kbps, which is kBits/s, or 1000 bits/s. ++ To calculate the number of bytes per second of audio data, simply divide the ++ number of bits per second by eight. ++


    ++

        ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++
    Table 1.1: Bitrate versus sound quality
    BitrateBandwidthQuality comparable to or better than
    8 kBps2.5 kHzPOTS (telephone sound)
    16 kBps4.5 kHzshortwave radio
    32 kBps7.5 kHzAM radio
    64 kBps11 kHzFM radio
    128 kBps16 kHzCD
    ++
    ++
    ++     ++ ++ +diff -r -c -N encoder/doc/html/node7.html lame3.50/doc/html/node7.html +*** encoder/doc/html/node7.html Wed Dec 31 17:00:00 1969 +--- lame3.50/doc/html/node7.html Tue Sep 21 22:40:21 1999 +*************** +*** 0 **** +--- 1,36 ---- ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ Some examples ++ ++ ++ ++ ++ ++ ++ ++ ++

    ++ Some examples

    ++ ++
      ++
    • ++ Recommended usage: Encode a WAVE-file myfile.wav ++ to a bitrate of 128000 bits/s, writing to a plain bitstream myfile.mp3 ++ and using high quality mode:
      • ++ ++
      lame -h myfile.wav myfile.mp3 ++
    • ++ Encode a plain PCM stream (2-channel, 44.1 kHz) ++ to a plain 56 kBit/s Layer-3 stream, using the encoder as filter
      • ++ ++
      lame -r -s 44.1 -b 56 - -
    ++ ++ ++ +diff -r -c -N encoder/doc/html/switchs.html lame3.50/doc/html/switchs.html +*** encoder/doc/html/switchs.html Wed Dec 31 17:00:00 1969 +--- lame3.50/doc/html/switchs.html Tue Sep 21 14:38:28 1999 +*************** +*** 0 **** +--- 1,280 ---- ++ ++ ++ ++ Command line switch reference ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++

    Command line switch reference

    ++

    ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++
    switchparameter
    -adownmix stereo file to mono
    -bbitrate (32, 40, 48, 56, 64, 96, 112, 128, ++ 160, 192, 224, 256, 320)
    -ccopyright
    -ede-emphasis (n, 5, c)
    -f fast mode
    -gturn on MP3x graphical frame analyser (might not be ++ available on your release)
    -hhigh quality
    -k full bandwidth
    -mstereo mode (s, j, f, m)
    --nohistdisable histogram display
    -onon-original
    -perror protection
    -rinput file is raw pcm
    -ssampling frequency
    -Ssilent operation
    -tdisable VBR tag
    -VVBR quality setting (0...9)
    -xswapbytes
    ++
    ++

    ++
    -a    downmix  ++
    Mix the stereo input file to mono and encode as mono. This option is only ++ needed in the case of raw PCM stereo input (because LAME cannot determine ++ the number of channels in the input file).
    ++ To encode a stereo PCM input file as mono, use "lame -m s -a".
    ++ For WAV and AIFF input files, using "-m m" will always produce a mono .mp3 ++ file from both mono and stereo input.
    ++ The downmix is calculated as the sum of the left and right channel, attenuated ++ by 6 dB. ++

    ++
    ++ -b    bitrate  ++
    ++
    ++
    ++
    For MPEG1 (sampling frequencies of 32, 44.1 and 48kHz)
    ++ n = 32,40,48,56,64,80,96,112,128,160,192,224,256,320
    ++
    ++ For MPEG2 (sampling frequencies of 16, 22.05 and 24kHz)
    ++ n = 8,16,24,32,40,48,56,64,80,96,112,128,144,160
    ++
    ++ Default is 128kbs MPEG1, 80kbs MPEG2. When used with variable bitrate encodings ++ (VBR), -b specifies the minimum bitrate to use. This is useful to prevent ++ LAME VBR from using some very aggressive compression which can cause some ++ distortion due to small flaws in the psycho-acoustic model. ++
    ++

    ++
    ++ -c    copyright ++
    ++
    ++
    ++
    Mark the encoded file as being copyrighted. ++
    ++

    ++
    ++ -e n/5/c    de-emphasis ++
    ++
    ++
    ++

    ++ default=n (none) ++
    ++

    ++
    ++ -f    fast mode ++
    ++
    ++
    ++
    This switch forces the encoder to use the fast encoding mode. It will reduce ++ a lot the quality of the output. This might be useful for real-time processing ++ or streaming on slowest computers. ++

    ++
    ++ -h    high quality ++
    ++
    ++
    ++
    Enables specialized mid/side masking thresholds to be used in jstereo mode. ++ Will sound better but is 20% slower. It is only effective on joint stereo ++ encoding. ++

    ++
    ++ -k    full bandwidth ++
    ++
    ++
    ++
    Tells the encoder to use full bandwidth. By default, the encoder uses a ++ 16kHz bandwidth up to 64kbps per channel, and full bandwidth at higher bitrates. ++ Increasing the bandwidth from the default setting might produces ringing artefacts ++ at low bitrates. Use with care! ++

    ++
    ++ -m s/j/f/m    stereo ++ mode
    ++
    ++
    ++
    jstereo is the default mode for stereo files with VBR or fixed bitrates ++ of 160kbs or less. At higher fixed bitrates, the default is stereo.
    ++
    ++ stereo
    ++ In this mode, the encoder makes no use of potentially existing correlations ++ between the two input channels. It can, however, negotiate the bit demand ++ between both channel, i.e. give one channel more bits if the other contains ++ silence.
    ++
    ++ joint stereo
    ++ In this mode, the encoder will make use of a correlation between both channels. ++ The signal will be matrixed into a sum ("mid") and difference ("side") signal. ++ For quasi-mono signals, this will give a significant gain in encoding quality.
    ++ This mode does not destroy phase information like IS stereo that may be used ++ by other encoders and thus can be used to encode DOLBY ProLogic surround signals. ++
    ++
    ++ forced joint stereo
    ++ This mode will force MS joint stereo on all frames. It's slightly faster then ++ jstereo. But it should only be used if you are sure every frame of the input ++ file has very little stereo seperation.
    ++
    ++ mono
    ++ The input will be encoded as a mono signal. If it was a stereo signal, it ++ will be downsampled to mono. ++

    ++
    ++ --nohist    disable ++ histogram display
    ++
    ++
    ++
    By default, LAME will display a bitrate histogram while producing VBR mp3 ++ files. This will disable that feature.
    ++ Histogram display might not be available on your release. ++

    ++
    ++ -o    non-original ++
    ++
    ++
    ++
    Mark the encoded file as a copy. ++

    ++
    ++ -p    error protection
    ++
    ++
    ++
    Turn on CRC error protection. ++

    ++
    ++ -r    input file is raw pcm
    ++
    ++
    ++
    Assume the input file is raw pcm. Sampling rate and mono/stereo/jstereo ++ must be specified on the command line. Without -r, LAME will perform several ++ fseek()'s on the input file looking for WAV and AIFF headers.
    ++ Might not be available on your release. ++

    ++
    ++ -s 16/22.05/24/32/44.1/48    sampling ++ frequency
    ++
    ++
    ++
    Required only for raw PCM input files. Otherwise it will be determined ++ from the header. ++

    ++
    ++ -S    silent operation ++
    ++
    ++
    ++
    Don't print progress report. ++

    ++
    ++ -t    disable VBR tag
    ++
    ++
    ++
    Disable writing of the VBR Tag (only valid in VBR mode).
    ++ This tag in embedded in frame 0 of the MP3 file. It lets VBR aware players ++ correctly seek and compute playing times of VBR files. ++

    ++
    ++ -V 0...9    VBR quality setting
    ++
    ++
    ++
    Enable VBR (Variable BitRate) and specifies the value of VBR quality.
    ++ default=4
    ++ 0=highest quality. ++

    ++
    ++ -x    swapbytes ++
    ++
    ++
    ++
    Swap bytes in the input file. For sorting out little endian/big endian ++ type problems. If your encodings sounds like static, try this first. ++
    ++
    ++
      ++
    ++ ++ +diff -r -c -N encoder/doc/man/lame.1 lame3.50/doc/man/lame.1 +*** encoder/doc/man/lame.1 Wed Dec 31 17:00:00 1969 +--- lame3.50/doc/man/lame.1 Fri Sep 10 09:52:02 1999 +*************** +*** 0 **** +--- 1,138 ---- ++ .TH lame 1 "August 16, 1999" " " "LAME audio compressorMP3 " ++ .SH NAME ++ lame \- create mp3 audio files ++ .SH SYNOPSIS ++ .nf ++ lame [options] ++ .fi ++ .SH DESCRIPTION ++ .PP ++ Lame is a program which can be used to create compressed audio files. ++ (Lame aint MP3 encoder). These audio files can be played back by popular ++ mp3 players such as mpg123. To read from stdin, use "-" for . ++ To write to stdout, use a "-" for . ++ ++ .SH OPTIONS ++ .TP ++ .BI -m " mode" ++ (s)tereo, (j)oint, (f)orce or (m)ono (default j) ++ force = force mid/side stereo on all frames. Faster if you know mid/side ++ stereo is appropriate for the input file. ++ .TP ++ .BI -s " sample" " freq" ++ frequency of input file (khz) - default 44.1 ++ .TP ++ .BI -b " bitrate" ++ set the bitrate, default 128kbps ++ (for VBR, this sets the allowed minimum bitrate) ++ ++ .TP ++ .BI -v ++ use variable bitrate (still undergoing development). ++ .TP ++ .BI -V " n" ++ quality setting for VBR. Default n=4, 0=high quality,bigger files. ++ 9=smaller files ++ .TP ++ .BI -t ++ disable Xing VBR informational tag ++ .TP ++ .BI --nohist ++ disable VBR histogram display ++ ++ .TP ++ .BI -h ++ slower, but higher quality ++ .TP ++ .BI -f ++ fast mode (low quality) ++ .TP ++ .BI -k ++ disable sfb=21 cutoff ++ .TP ++ .BI -d ++ allow channels to have different blocktypes ++ ++ .TP ++ .BI -r ++ input is raw pcm ++ .TP ++ .BI -x ++ force byte-swapping of input ++ .TP ++ .BI -a ++ autoconvert from stereo to mono file for mono encoding ++ .TP ++ .BI -e " emp " ++ de-emphasis n/5/c (default n) ++ .TP ++ .BI -p ++ add CRC error protection (this is working, but takes up 16bits) ++ .TP ++ .BI -c ++ mark as copyright ++ .TP ++ .BI -o ++ mark as non-original (i.e. do not set the original bit) ++ .TP ++ .BI -S ++ don't print progress report, VBR histograms ++ ++ .TP ++ .BI -g ++ run graphical analysis on . can also be a .mp3 file. ++ ++ .SH EXAMPLES ++ .LP ++ Recommended way to create a fixed 128kbs mp3 file from a wav file ++ .IP ++ .B lame -h ++ .I foo.wav foo.mp3 ++ .LP ++ Copy track 7 from a regular audio CD, using cdparanoia ++ ++ .IP ++ .B cdparanoia 7 foo.wav ; ++ .B lame -h foo.wav foo.mp3 ++ ++ .LP ++ Copy track 7 from a regular audio CD with cdda2wav, where the cdrom device is ++ device /dev/cdrom to an mp3 file foo.mp3 ++ ++ .IP ++ .B cdda2wav \-O ++ .I wav ++ .B \-t 7 \-D ++ .I /dev/cdrom ++ .I \- ++ .B | lame ++ .I \- foo.mp3 ++ ++ .IP ++ ++ .LP ++ Streaming from mono 22.05kHz raw pcm data and using sox to ++ downsampling to 16kHz: ++ .IP ++ cat inputfile | sox -t raw -x -w -s -c 1 -r 22050 - ++ -t raw -x -w -s -c 1 -r 16000 - | lame -r -m m -b 24 -s 16 - - > output ++ .IP ++ ++ ++ ++ ++ .SH BUGS ++ .PP ++ Quality is much better than encoders based on the ISO routines, ++ but still not as good as the FhG encoder. ++ .SH SEE ALSO ++ mpg123(1) ++ .SH AUTHORS ++ .nf ++ LAME originally developed by Mike Cheng ++ (www.uq.net.au/~zzmcheng) ++ GPSYCHO psycho-acoustic model by Mark Taylor ++ (www.sulaco.org/mp3). ++ mpglib by Michael Hipp ++ Manual page by William Schelter ++ .f +diff -r -c -N encoder/encode.c lame3.50/encode.c +*** encoder/encode.c Wed Jan 22 02:43:14 1997 +--- lame3.50/encode.c Wed Dec 31 17:00:00 1969 +*************** +*** 1,1456 **** +- /********************************************************************** +- * ISO MPEG Audio Subgroup Software Simulation Group (1996) +- * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- * +- * $Id: encode.c,v 1.1 1996/02/14 04:04:23 rowlands Exp $ +- * +- * $Log: encode.c,v $ +- * Revision 1.1 1996/02/14 04:04:23 rowlands +- * Initial revision +- * +- * Received from Mike Coleman +- **********************************************************************/ +- /********************************************************************** +- * date programmers comment * +- * 3/01/91 Douglas Wong, start of version 1.1 records * +- * Davis Pan * +- * 3/06/91 Douglas Wong rename: setup.h to endef.h * +- * efilter to enfilter * +- * ewindow to enwindow * +- * integrated "quantizer", "scalefactor",* +- * and "transmission" files * +- * update routine "window_subband" * +- * 3/31/91 Bill Aspromonte replaced read_filter by * +- * create_an_filter * +- * 5/10/91 W. Joseph Carter Ported to Macintosh and Unix. * +- * Incorporated Jean-Georges Fritsch's * +- * "bitstream.c" package. * +- * Incorporated Bill Aspromonte's * +- * filterbank coefficient matrix * +- * calculation routines and added * +- * roundoff to coincide with specs. * +- * Modified to strictly adhere to * +- * encoded bitstream specs, including * +- * "Berlin changes". * +- * Modified PCM sound file handling to * +- * process all incoming samples and fill * +- * out last encoded frame with zeros * +- * (silence) if needed. * +- * Located and fixed numerous software * +- * bugs and table data errors. * +- * 19jun91 dpwe (Aware) moved "alloc_*" reader to common.c * +- * Globals sblimit, alloc replaced by new* +- * struct 'frame_params' passed as arg. * +- * Added JOINT STEREO coding, layers I,II* +- * Affects: *_bit_allocation, * +- * subband_quantization, encode_bit_alloc* +- * sample_encoding * +- * 6/10/91 Earle Jennings modified II_subband_quantization to * +- * resolve type cast problem for MS_DOS * +- * 6/11/91 Earle Jennings modified to avoid overflow on MS_DOS * +- * in routine filter_subband * +- * 7/10/91 Earle Jennings port to MsDos from MacIntosh version * +- * 8/ 8/91 Jens Spille Change for MS-C6.00 * +- *10/ 1/91 S.I. Sudharsanan, Ported to IBM AIX platform. * +- * Don H. Lee, * +- * Peter W. Farrett * +- *10/ 3/91 Don H. Lee implemented CRC-16 error protection * +- * newly introduced function encode_CRC * +- *11/ 8/91 Kathy Wang Documentation of code * +- * All variablenames are referred to * +- * with surrounding pound (#) signs * +- * 2/11/92 W. Joseph Carter Ported new code to Macintosh. Most * +- * important fixes involved changing * +- * 16-bit ints to long or unsigned in * +- * bit alloc routines for quant of 65535 * +- * and passing proper function args. * +- * Removed "Other Joint Stereo" option * +- * and made bitrate be total channel * +- * bitrate, irrespective of the mode. * +- * Fixed many small bugs & reorganized. * +- * 6/16/92 Shaun Astarabadi Changed I_scale_factor_calc() and * +- * II_scale_factor_calc() to use scale * +- * factor 0 thru 62 only and not to * +- * encode index 63 into the bit stream. * +- * 7/27/92 Mike Li (re-)Port to MS-DOS * +- * 9/22/92 jddevine@aware.com Fixed _scale_factor_calc() defs * +- * 3/31/93 Giogio Dimino changed II_a_bit_allocation() from: * +- * if( ad > ...) to if(ad >= ...) * +- * 8/05/93 TEST changed I_a_bit_allocation() from: * +- * if( ad > ...) to if(ad >= ...) * +- * 8/02/95 mc@fivebats.com Changed audio file reading code to * +- * read samples big-endian * +- *10/15/95 mc@fivebats.com Modified get_audio() for layer3-LSF * +- **********************************************************************/ +- +- #include "common.h" +- #include "encoder.h" +- +- #ifdef MS_DOS +- extern unsigned _stklen = 16384; +- #endif +- +- +- /*=======================================================================\ +- | | +- | This segment contains all the core routines of the encoder, | +- | except for the psychoacoustic models. | +- | | +- | The user can select either one of the two psychoacoustic | +- | models. Model I is a simple tonal and noise masking threshold | +- | generator, and Model II is a more sophisticated cochlear masking | +- | threshold generator. Model I is recommended for lower complexity | +- | applications whereas Model II gives better subjective quality at low | +- | bit rates. | +- | | +- | Layers I and II of mono, stereo, and joint stereo modes are supported.| +- | Routines associated with a given layer are prefixed by "I_" for layer | +- | 1 and "II_" for layer 2. | +- \=======================================================================*/ +- +- /************************************************************************ +- * +- * read_samples() +- * +- * PURPOSE: reads the PCM samples from a file to the buffer +- * +- * SEMANTICS: +- * Reads #samples_read# number of shorts from #musicin# filepointer +- * into #sample_buffer[]#. Returns the number of samples read. +- * +- ************************************************************************/ +- +- unsigned long read_samples(musicin, sample_buffer, num_samples, frame_size) +- FILE *musicin; +- short sample_buffer[2304]; +- unsigned long num_samples, frame_size; +- { +- unsigned long samples_read; +- static unsigned long samples_to_read; +- static char init = TRUE; +- +- if (init) { +- samples_to_read = num_samples; +- init = FALSE; +- } +- if (samples_to_read >= frame_size) +- samples_read = frame_size; +- else +- samples_read = samples_to_read; +- if ((samples_read = +- fread(sample_buffer, sizeof(short), (int)samples_read, musicin)) == 0) +- printf("Hit end of audio data\n"); +- /* +- Samples are big-endian. If this is a little-endian machine +- we must swap +- */ +- if ( NativeByteOrder == order_unknown ) +- { +- NativeByteOrder = DetermineByteOrder(); +- if ( NativeByteOrder == order_unknown ) +- { +- fprintf( stderr, "byte order not determined\n" ); +- exit( 1 ); +- } +- } +- if ( NativeByteOrder == order_littleEndian ) +- SwapBytesInWords( sample_buffer, samples_read ); +- +- samples_to_read -= samples_read; +- if (samples_read < frame_size && samples_read > 0) { +- printf("Insufficient PCM input for one frame - fillout with zeros\n"); +- for (; samples_read < frame_size; sample_buffer[samples_read++] = 0); +- samples_to_read = 0; +- } +- return(samples_read); +- } +- +- /************************************************************************ +- * +- * get_audio() +- * +- * PURPOSE: reads a frame of audio data from a file to the buffer, +- * aligns the data for future processing, and separates the +- * left and right channels +- * +- * +- ************************************************************************/ +- +- unsigned long get_audio( musicin, buffer, num_samples, stereo, info ) +- FILE *musicin; +- short FAR buffer[2][1152]; +- unsigned long num_samples; +- int stereo; +- layer *info; +- { +- int j; +- short insamp[2304]; +- unsigned long samples_read; +- int lay; +- lay = info->lay; +- +- if ( (lay == 3) && (info->version == 0) ) +- { +- if ( stereo == 2 ) +- { +- samples_read = read_samples( musicin, insamp, num_samples, +- (unsigned long) 1152 ); +- for ( j = 0; j < 576; j++ ) +- { +- buffer[0][j] = insamp[2 * j]; +- buffer[1][j] = insamp[2 * j + 1]; +- } +- } +- else +- { +- samples_read = read_samples( musicin, insamp, num_samples, +- (unsigned long) 576 ); +- for ( j = 0; j < 576; j++ ) +- { +- buffer[0][j] = insamp[j]; +- buffer[1][j] = 0; +- } +- } +- } +- else +- { +- if (lay == 1){ +- if(stereo == 2){ /* layer 1, stereo */ +- samples_read = read_samples(musicin, insamp, num_samples, +- (unsigned long) 768); +- for(j=0;j<448;j++) { +- if(j<64) { +- buffer[0][j] = buffer[0][j+384]; +- buffer[1][j] = buffer[1][j+384]; +- } +- else { +- buffer[0][j] = insamp[2*j-128]; +- buffer[1][j] = insamp[2*j-127]; +- } +- } +- } +- else { /* layer 1, mono */ +- samples_read = read_samples(musicin, insamp, num_samples, +- (unsigned long) 384); +- for(j=0;j<448;j++){ +- if(j<64) { +- buffer[0][j] = buffer[0][j+384]; +- buffer[1][j] = 0; +- } +- else { +- buffer[0][j] = insamp[j-64]; +- buffer[1][j] = 0; +- } +- } +- } +- } +- else { +- if(stereo == 2){ /* layer 2 (or 3), stereo */ +- samples_read = read_samples(musicin, insamp, num_samples, +- (unsigned long) 2304); +- for(j=0;j<1152;j++) { +- buffer[0][j] = insamp[2*j]; +- buffer[1][j] = insamp[2*j+1]; +- } +- } +- else { /* layer 2 (or 3), mono */ +- samples_read = read_samples(musicin, insamp, num_samples, +- (unsigned long) 1152); +- for(j=0;j<1152;j++){ +- buffer[0][j] = insamp[j]; +- buffer[1][j] = 0; +- } +- } +- } +- } +- return(samples_read); +- } +- +- /************************************************************************ +- * +- * read_ana_window() +- * +- * PURPOSE: Reads encoder window file "enwindow" into array #ana_win# +- * +- ************************************************************************/ +- +- void read_ana_window(ana_win) +- double FAR ana_win[HAN_SIZE]; +- { +- int i,j[4]; +- FILE *fp; +- double f[4]; +- char t[150]; +- +- if (!(fp = OpenTableFile("enwindow") ) ) { +- printf("Please check analysis window table 'enwindow'\n"); +- exit(1); +- } +- for (i=0;i<512;i+=4) { +- fgets(t, 150, fp); +- sscanf(t,"C[%d] = %lf C[%d] = %lf C[%d] = %lf C[%d] = %lf\n", +- j, f,j+1,f+1,j+2,f+2,j+3,f+3); +- if (i==j[0]) { +- ana_win[i] = f[0]; +- ana_win[i+1] = f[1]; +- ana_win[i+2] = f[2]; +- ana_win[i+3] = f[3]; +- } +- else { +- printf("Check index in analysis window table\n"); +- exit(1); +- } +- fgets(t,150,fp); +- } +- fclose(fp); +- } +- +- /************************************************************************ +- * +- * window_subband() +- * +- * PURPOSE: Overlapping window on PCM samples +- * +- * SEMANTICS: +- * 32 16-bit pcm samples are scaled to fractional 2's complement and +- * concatenated to the end of the window buffer #x#. The updated window +- * buffer #x# is then windowed by the analysis window #c# to produce the +- * windowed sample #z# +- * +- ************************************************************************/ +- +- void window_subband(buffer, z, k) +- short FAR **buffer; +- double FAR z[HAN_SIZE]; +- int k; +- { +- typedef double FAR XX[2][HAN_SIZE]; +- static XX FAR *x; +- int i, j; +- static off[2] = {0,0}; +- static char init = 0; +- static double FAR *c; +- if (!init) { +- c = (double FAR *) mem_alloc(sizeof(double) * HAN_SIZE, "window"); +- read_ana_window(c); +- x = (XX FAR *) mem_alloc(sizeof(XX),"x"); +- for (i=0;i<2;i++) +- for (j=0;j= 0) +- modf(filter[i][k]+0.5, &filter[i][k]); +- else +- modf(filter[i][k]-0.5, &filter[i][k]); +- filter[i][k] *= 1e-9; +- } +- } +- +- /************************************************************************ +- * +- * filter_subband() +- * +- * PURPOSE: Calculates the analysis filter bank coefficients +- * +- * SEMANTICS: +- * The windowed samples #z# is filtered by the digital filter matrix #m# +- * to produce the subband samples #s#. This done by first selectively +- * picking out values from the windowed samples, and then multiplying +- * them by the filter matrix, producing 32 subband samples. +- * +- ************************************************************************/ +- +- void filter_subband(z,s) +- double FAR z[HAN_SIZE], s[SBLIMIT]; +- { +- double y[64]; +- int i,j; +- static char init = 0; +- typedef double MM[SBLIMIT][64]; +- static MM FAR *m; +- #ifdef MS_DOS +- long SIZE_OF_MM; +- SIZE_OF_MM = SBLIMIT*64; +- SIZE_OF_MM *= 8; +- if (!init) { +- m = (MM FAR *) mem_alloc(SIZE_OF_MM, "filter"); +- create_ana_filter(*m); +- init = 1; +- } +- #else +- if (!init) { +- m = (MM FAR *) mem_alloc(sizeof(MM), "filter"); +- create_ana_filter(*m); +- init = 1; +- } +- #endif +- for (i=0;i<64;i++) for (j=0, y[i] = 0;j<8;j++) y[i] += z[i+64*j]; +- for (i=0;iheader; +- +- putbits(bs,0xfff,12); /* syncword 12 bits */ +- put1bit(bs,info->version); /* ID 1 bit */ +- putbits(bs,4-info->lay,2); /* layer 2 bits */ +- put1bit(bs,!info->error_protection); /* bit set => no err prot */ +- putbits(bs,info->bitrate_index,4); +- putbits(bs,info->sampling_frequency,2); +- put1bit(bs,info->padding); +- put1bit(bs,info->extension); /* private_bit */ +- putbits(bs,info->mode,2); +- putbits(bs,info->mode_ext,2); +- put1bit(bs,info->copyright); +- put1bit(bs,info->original); +- putbits(bs,info->emphasis,2); +- } +- +- /************************************************************************ +- * +- * mod() +- * +- * PURPOSE: Returns the absolute value of its argument +- * +- ************************************************************************/ +- +- double mod(a) +- double a; +- { +- return (a > 0) ? a : -a; +- } +- +- /************************************************************************ +- * +- * I_combine_LR (Layer I) +- * II_combine_LR (Layer II) +- * +- * PURPOSE:Combines left and right channels into a mono channel +- * +- * SEMANTICS: The average of left and right subband samples is put into +- * #joint_sample# +- * +- * Layer I and II differ in frame length and # subbands used +- * +- ************************************************************************/ +- +- void I_combine_LR(sb_sample, joint_sample) +- double FAR sb_sample[2][3][SCALE_BLOCK][SBLIMIT]; +- double FAR joint_sample[3][SCALE_BLOCK][SBLIMIT]; +- { /* make a filtered mono for joint stereo */ +- int sb, smp; +- +- for(sb = 0; sb s[i]) +- s[i] = mod(sb_sample[k][0][j][i]); +- +- for (i=0;i=0;j--) /* $A 6/16/92 */ +- if (s[i] <= multiple[j]) { +- scalar[k][0][i] = j; +- break; +- } +- } +- } +- +- /******************************** Layer II ******************************/ +- +- void II_scale_factor_calc(sb_sample,scalar,stereo,sblimit) +- double FAR sb_sample[][3][SCALE_BLOCK][SBLIMIT]; +- unsigned int scalar[][3][SBLIMIT]; +- int stereo,sblimit; +- { +- int i,j, k,t; +- double s[SBLIMIT]; +- +- for (k=0;k s[i]) +- s[i] = mod(sb_sample[k][t][j][i]); +- +- for (i=0;i=0;j--) /* $A 6/16/92 */ +- if (s[i] <= multiple[j]) { +- scalar[k][t][i] = j; +- break; +- } +- for (i=sblimit;istereo; +- int sblimit = fr_ps->sblimit; +- +- for (k=0;k scalar[k][j][i]) max = scalar[k][j][i]; +- for (i=sblimit;istereo; +- int sblimit = fr_ps->sblimit; +- +- for (k=0;kstereo; +- int sblimit = fr_ps->sblimit; +- int dscf[2]; +- int class[2],i,j,k; +- static int pattern[5][5] = {0x123, 0x122, 0x122, 0x133, 0x123, +- 0x113, 0x111, 0x111, 0x444, 0x113, +- 0x111, 0x111, 0x111, 0x333, 0x113, +- 0x222, 0x222, 0x222, 0x333, 0x123, +- 0x123, 0x122, 0x122, 0x133, 0x123}; +- +- for (k=0;k -3 && dscf[j] <0) class[j] = 1; +- else if (dscf[j] == 0) class[j] = 2; +- else if (dscf[j] > 0 && dscf[j] < 3) class[j] = 3; +- else class[j] = 4; +- } +- switch (pattern[class[0]][class[1]]) { +- case 0x123 : scfsi[k][i] = 0; +- break; +- case 0x122 : scfsi[k][i] = 3; +- scalar[k][2][i] = scalar[k][1][i]; +- break; +- case 0x133 : scfsi[k][i] = 3; +- scalar[k][1][i] = scalar[k][2][i]; +- break; +- case 0x113 : scfsi[k][i] = 1; +- scalar[k][1][i] = scalar[k][0][i]; +- break; +- case 0x111 : scfsi[k][i] = 2; +- scalar[k][1][i] = scalar[k][2][i] = scalar[k][0][i]; +- break; +- case 0x222 : scfsi[k][i] = 2; +- scalar[k][0][i] = scalar[k][2][i] = scalar[k][1][i]; +- break; +- case 0x333 : scfsi[k][i] = 2; +- scalar[k][0][i] = scalar[k][1][i] = scalar[k][2][i]; +- break; +- case 0x444 : scfsi[k][i] = 2; +- if (scalar[k][0][i] > scalar[k][2][i]) +- scalar[k][0][i] = scalar[k][2][i]; +- scalar[k][1][i] = scalar[k][2][i] = scalar[k][0][i]; +- } +- } +- } +- +- /************************************************************************ +- * +- * I_encode_scale (Layer I) +- * II_encode_scale (Layer II) +- * +- * PURPOSE:The encoded scalar factor information is arranged and +- * queued into the output fifo to be transmitted. +- * +- * For Layer II, the three scale factors associated with +- * a given subband and channel are transmitted in accordance +- * with the scfsi, which is transmitted first. +- * +- ************************************************************************/ +- +- void I_encode_scale(scalar, bit_alloc, fr_ps, bs) +- unsigned int scalar[2][3][SBLIMIT]; +- unsigned int bit_alloc[2][SBLIMIT]; +- frame_params *fr_ps; +- Bit_stream_struc *bs; +- { +- int stereo = fr_ps->stereo; +- int sblimit = fr_ps->sblimit; +- int i,j; +- +- for (i=0;istereo; +- int sblimit = fr_ps->sblimit; +- int jsbound = fr_ps->jsbound; +- int i,j,k; +- +- for (i=0;istereo; +- int sblimit = fr_ps->sblimit; +- int jsbound = fr_ps->jsbound; +- int req_bits = 0; +- +- /* initial b_anc (header) allocation bits */ +- req_bits = 32 + 4 * ( (jsbound * stereo) + (SBLIMIT-jsbound) ); +- +- for(i=0; i= NOISY_MIN_MNR) +- break; /* we found enough bits */ +- if(stereo == 2 && i >= jsbound) /* check other JS channel */ +- for(;k<14; ++k) +- if( (-perm_smr[1-j][i] + snr[k]) >= NOISY_MIN_MNR) break; +- if(k>0) req_bits += (k+1)*SCALE_BLOCK + 6*((i>=jsbound)?stereo:1); +- } +- return req_bits; +- } +- +- /***************************** Layer II ********************************/ +- +- int II_bits_for_nonoise(perm_smr, scfsi, fr_ps) +- double FAR perm_smr[2][SBLIMIT]; +- unsigned int scfsi[2][SBLIMIT]; +- frame_params *fr_ps; +- { +- int sb,ch,ba; +- int stereo = fr_ps->stereo; +- int sblimit = fr_ps->sblimit; +- int jsbound = fr_ps->jsbound; +- al_table *alloc = fr_ps->alloc; +- int req_bits = 0, bbal = 0, berr = 0, banc = 32; +- int maxAlloc, sel_bits, sc_bits, smp_bits; +- static int sfsPerScfsi[] = { 3,2,1,2 }; /* lookup # sfs per scfsi */ +- +- /* added 92-08-11 shn */ +- if (fr_ps->header->error_protection) berr=16; else berr=0; +- +- for (sb=0; sb0)?1:0)]) +- >= NOISY_MIN_MNR) +- break; /* we found enough bits */ +- if(stereo == 2 && sb >= jsbound) /* check other JS channel */ +- for(;ba0)?1:0)]) +- >= NOISY_MIN_MNR) +- break; +- if(ba>0) { +- smp_bits = SCALE_BLOCK * ((*alloc)[sb][ba].group * (*alloc)[sb][ba].bits); +- /* scale factor bits required for subband */ +- sel_bits = 2; +- sc_bits = 6 * sfsPerScfsi[scfsi[ch][sb]]; +- if(stereo == 2 && sb >= jsbound) { +- /* each new js sb has L+R scfsis */ +- sel_bits += 2; +- sc_bits += 6 * sfsPerScfsi[scfsi[1-ch][sb]]; +- } +- req_bits += smp_bits+sel_bits+sc_bits; +- } +- } +- return req_bits; +- } +- +- /************************************************************************ +- * +- * I_main_bit_allocation (Layer I) +- * II_main_bit_allocation (Layer II) +- * +- * PURPOSE:For joint stereo mode, determines which of the 4 joint +- * stereo modes is needed. Then calls *_a_bit_allocation(), which +- * allocates bits for each of the subbands until there are no more bits +- * left, or the MNR is at the noise/no_noise threshold. +- * +- * SEMANTICS: +- * +- * For joint stereo mode, joint stereo is changed to stereo if +- * there are enough bits to encode stereo at or better than the +- * no-noise threshold (NOISY_MIN_MNR). Otherwise, the system +- * iteratively allocates less bits by using joint stereo until one +- * of the following occurs: +- * - there are no more noisy subbands (MNR >= NOISY_MIN_MNR) +- * - mode_ext has been reduced to 0, which means that all but the +- * lowest 4 subbands have been converted from stereo to joint +- * stereo, and no more subbands may be converted +- * +- * This function calls *_bits_for_nonoise() and *_a_bit_allocation(). +- * +- ************************************************************************/ +- +- void I_main_bit_allocation(perm_smr, bit_alloc, adb, fr_ps) +- double FAR perm_smr[2][SBLIMIT]; +- unsigned int bit_alloc[2][SBLIMIT]; +- int *adb; +- frame_params *fr_ps; +- { +- int noisy_sbs; +- int mode, mode_ext, lay, i; +- int rq_db, av_db = *adb; +- static int init = 0; +- +- if(init == 0) { +- /* rearrange snr for layer I */ +- snr[2] = snr[3]; +- for (i=3;i<16;i++) snr[i] = snr[i+2]; +- init = 1; +- } +- +- if((mode = fr_ps->actual_mode) == MPG_MD_JOINT_STEREO) { +- fr_ps->header->mode = MPG_MD_STEREO; +- fr_ps->header->mode_ext = 0; +- fr_ps->jsbound = fr_ps->sblimit; +- if(rq_db = I_bits_for_nonoise(perm_smr, fr_ps) > *adb) { +- fr_ps->header->mode = MPG_MD_JOINT_STEREO; +- mode_ext = 4; /* 3 is least severe reduction */ +- lay = fr_ps->header->lay; +- do { +- --mode_ext; +- fr_ps->jsbound = js_bound(lay, mode_ext); +- rq_db = I_bits_for_nonoise(perm_smr, fr_ps); +- } while( (rq_db > *adb) && (mode_ext > 0)); +- fr_ps->header->mode_ext = mode_ext; +- } /* well we either eliminated noisy sbs or mode_ext == 0 */ +- } +- noisy_sbs = I_a_bit_allocation(perm_smr, bit_alloc, adb, fr_ps); +- } +- +- /***************************** Layer II ********************************/ +- +- void II_main_bit_allocation(perm_smr, scfsi, bit_alloc, adb, fr_ps) +- double FAR perm_smr[2][SBLIMIT]; +- unsigned int scfsi[2][SBLIMIT]; +- unsigned int bit_alloc[2][SBLIMIT]; +- int *adb; +- frame_params *fr_ps; +- { +- int noisy_sbs, nn; +- int mode, mode_ext, lay; +- int rq_db, av_db = *adb; +- +- if((mode = fr_ps->actual_mode) == MPG_MD_JOINT_STEREO) { +- fr_ps->header->mode = MPG_MD_STEREO; +- fr_ps->header->mode_ext = 0; +- fr_ps->jsbound = fr_ps->sblimit; +- if((rq_db=II_bits_for_nonoise(perm_smr, scfsi, fr_ps)) > *adb) { +- fr_ps->header->mode = MPG_MD_JOINT_STEREO; +- mode_ext = 4; /* 3 is least severe reduction */ +- lay = fr_ps->header->lay; +- do { +- --mode_ext; +- fr_ps->jsbound = js_bound(lay, mode_ext); +- rq_db = II_bits_for_nonoise(perm_smr, scfsi, fr_ps); +- } while( (rq_db > *adb) && (mode_ext > 0)); +- fr_ps->header->mode_ext = mode_ext; +- } /* well we either eliminated noisy sbs or mode_ext == 0 */ +- } +- noisy_sbs = II_a_bit_allocation(perm_smr, scfsi, bit_alloc, adb, fr_ps); +- } +- +- /************************************************************************ +- * +- * I_a_bit_allocation (Layer I) +- * II_a_bit_allocation (Layer II) +- * +- * PURPOSE:Adds bits to the subbands with the lowest mask-to-noise +- * ratios, until the maximum number of bits for the subband has +- * been allocated. +- * +- * SEMANTICS: +- * 1. Find the subband and channel with the smallest MNR (#min_sb#, +- * and #min_ch#) +- * 2. Calculate the increase in bits needed if we increase the bit +- * allocation to the next higher level +- * 3. If there are enough bits available for increasing the resolution +- * in #min_sb#, #min_ch#, and the subband has not yet reached its +- * maximum allocation, update the bit allocation, MNR, and bits +- available accordingly +- * 4. Repeat until there are no more bits left, or no more available +- * subbands. (A subband is still available until the maximum +- * number of bits for the subband has been allocated, or there +- * aren't enough bits to go to the next higher resolution in the +- subband.) +- * +- ************************************************************************/ +- +- int I_a_bit_allocation(perm_smr, bit_alloc, adb, fr_ps) /* return noisy sbs */ +- double FAR perm_smr[2][SBLIMIT]; +- unsigned int bit_alloc[2][SBLIMIT]; +- int *adb; +- frame_params *fr_ps; +- { +- int i, k, smpl_bits, scale_bits, min_sb, min_ch, oth_ch; +- int bspl, bscf, ad, noisy_sbs, done = 0, bbal ; +- double mnr[2][SBLIMIT], small; +- char used[2][SBLIMIT]; +- int stereo = fr_ps->stereo; +- int sblimit = fr_ps->sblimit; +- int jsbound = fr_ps->jsbound; +- al_table *alloc = fr_ps->alloc; +- static char init= 0; +- static int banc=32, berr=0; +- +- if (!init) { +- init = 1; +- if (fr_ps->header->error_protection) berr = 16; /* added 92-08-11 shn */ +- } +- bbal = 4 * ( (jsbound * stereo) + (SBLIMIT-jsbound) ); +- *adb -= bbal + berr + banc; +- ad= *adb; +- +- for (i=0;i mnr[k][i]) { +- small = mnr[k][i]; +- min_sb = i; min_ch = k; +- } +- if(min_sb > -1) { /* there was something to find */ +- /* first step of bit allocation is biggest */ +- if (used[min_ch][min_sb]) { smpl_bits = SCALE_BLOCK; scale_bits = 0; } +- else { smpl_bits = 24; scale_bits = 6; } +- if(min_sb >= jsbound) scale_bits *= stereo; +- +- /* check to see enough bits were available for */ +- /* increasing resolution in the minimum band */ +- +- if (ad >= bspl + bscf + scale_bits + smpl_bits) { +- bspl += smpl_bits; /* bit for subband sample */ +- bscf += scale_bits; /* bit for scale factor */ +- bit_alloc[min_ch][min_sb]++; +- used[min_ch][min_sb] = 1; /* subband has bits */ +- mnr[min_ch][min_sb] = -perm_smr[min_ch][min_sb] +- + snr[bit_alloc[min_ch][min_sb]]; +- /* Check if subband has been fully allocated max bits */ +- if (bit_alloc[min_ch][min_sb] == 14 ) used[min_ch][min_sb] = 2; +- } +- else /* no room to improve this band */ +- used[min_ch][min_sb] = 2; /* for allocation anymore */ +- if(stereo == 2 && min_sb >= jsbound) { +- oth_ch = 1-min_ch; /* joint-st : fix other ch */ +- bit_alloc[oth_ch][min_sb] = bit_alloc[min_ch][min_sb]; +- used[oth_ch][min_sb] = used[min_ch][min_sb]; +- mnr[oth_ch][min_sb] = -perm_smr[oth_ch][min_sb] +- + snr[bit_alloc[oth_ch][min_sb]]; +- } +- } +- } while(min_sb>-1); /* i.e. still some sub-bands to find */ +- +- /* Calculate the number of bits left, add on to pointed var */ +- ad -= bspl+bscf; +- *adb = ad; +- +- /* see how many channels are noisy */ +- noisy_sbs = 0; small = mnr[0][0]; +- for(k=0; k mnr[k][i]) small = mnr[k][i]; +- } +- } +- return noisy_sbs; +- } +- +- /***************************** Layer II ********************************/ +- +- int II_a_bit_allocation(perm_smr, scfsi, bit_alloc, adb, fr_ps) +- double FAR perm_smr[2][SBLIMIT]; +- unsigned int scfsi[2][SBLIMIT]; +- unsigned int bit_alloc[2][SBLIMIT]; +- int *adb; +- frame_params *fr_ps; +- { +- int i, min_ch, min_sb, oth_ch, k, increment, scale, seli, ba; +- int bspl, bscf, bsel, ad, noisy_sbs, bbal=0; +- double mnr[2][SBLIMIT], small; +- char used[2][SBLIMIT]; +- int stereo = fr_ps->stereo; +- int sblimit = fr_ps->sblimit; +- int jsbound = fr_ps->jsbound; +- al_table *alloc = fr_ps->alloc; +- static char init= 0; +- static int banc=32, berr=0; +- static int sfsPerScfsi[] = { 3,2,1,2 }; /* lookup # sfs per scfsi */ +- +- if (!init) { +- init = 1; +- if (fr_ps->header->error_protection) berr=16; /* added 92-08-11 shn */ +- } +- for (i=0; i mnr[k][i]) { +- small = mnr[k][i]; +- min_sb = i; min_ch = k; +- } +- if(min_sb > -1) { /* there was something to find */ +- /* find increase in bit allocation in subband [min] */ +- increment = SCALE_BLOCK * ((*alloc)[min_sb][bit_alloc[min_ch][min_sb]+1].group * +- (*alloc)[min_sb][bit_alloc[min_ch][min_sb]+1].bits); +- if (used[min_ch][min_sb]) +- increment -= SCALE_BLOCK * ((*alloc)[min_sb][bit_alloc[min_ch][min_sb]].group* +- (*alloc)[min_sb][bit_alloc[min_ch][min_sb]].bits); +- +- /* scale factor bits required for subband [min] */ +- oth_ch = 1 - min_ch; /* above js bound, need both chans */ +- if (used[min_ch][min_sb]) scale = seli = 0; +- else { /* this channel had no bits or scfs before */ +- seli = 2; +- scale = 6 * sfsPerScfsi[scfsi[min_ch][min_sb]]; +- if(stereo == 2 && min_sb >= jsbound) { +- /* each new js sb has L+R scfsis */ +- seli += 2; +- scale += 6 * sfsPerScfsi[scfsi[oth_ch][min_sb]]; +- } +- } +- /* check to see enough bits were available for */ +- /* increasing resolution in the minimum band */ +- if (ad >= bspl + bscf + bsel + seli + scale + increment) { +- ba = ++bit_alloc[min_ch][min_sb]; /* next up alloc */ +- bspl += increment; /* bits for subband sample */ +- bscf += scale; /* bits for scale factor */ +- bsel += seli; /* bits for scfsi code */ +- used[min_ch][min_sb] = 1; /* subband has bits */ +- mnr[min_ch][min_sb] = -perm_smr[min_ch][min_sb] + +- snr[(*alloc)[min_sb][ba].quant+1]; +- /* Check if subband has been fully allocated max bits */ +- if (ba >= (1<<(*alloc)[min_sb][0].bits)-1) used[min_ch][min_sb] = 2; +- } +- else used[min_ch][min_sb] = 2; /* can't increase this alloc */ +- if(min_sb >= jsbound && stereo == 2) { +- /* above jsbound, alloc applies L+R */ +- ba = bit_alloc[oth_ch][min_sb] = bit_alloc[min_ch][min_sb]; +- used[oth_ch][min_sb] = used[min_ch][min_sb]; +- mnr[oth_ch][min_sb] = -perm_smr[oth_ch][min_sb] + +- snr[(*alloc)[min_sb][ba].quant+1]; +- } +- } +- } while(min_sb > -1); /* until could find no channel */ +- /* Calculate the number of bits left */ +- ad -= bspl+bscf+bsel; *adb = ad; +- for (i=sblimit;i mnr[k][i]) small = mnr[k][i]; +- if(mnr[k][i] < NOISY_MIN_MNR) ++noisy_sbs; /* noise is not masked */ +- +- } +- } +- return noisy_sbs; +- } +- +- /************************************************************************ +- * +- * I_subband_quantization (Layer I) +- * II_subband_quantization (Layer II) +- * +- * PURPOSE:Quantizes subband samples to appropriate number of bits +- * +- * SEMANTICS: Subband samples are divided by their scalefactors, which +- makes the quantization more efficient. The scaled samples are +- * quantized by the function a*x+b, where a and b are functions of +- * the number of quantization levels. The result is then truncated +- * to the appropriate number of bits and the MSB is inverted. +- * +- * Note that for fractional 2's complement, inverting the MSB for a +- negative number x is equivalent to adding 1 to it. +- * +- ************************************************************************/ +- +- static double a[17] = { +- 0.750000000, 0.625000000, 0.875000000, 0.562500000, 0.937500000, +- 0.968750000, 0.984375000, 0.992187500, 0.996093750, 0.998046875, +- 0.999023438, 0.999511719, 0.999755859, 0.999877930, 0.999938965, +- 0.999969482, 0.999984741 }; +- +- static double b[17] = { +- -0.250000000, -0.375000000, -0.125000000, -0.437500000, -0.062500000, +- -0.031250000, -0.015625000, -0.007812500, -0.003906250, -0.001953125, +- -0.000976563, -0.000488281, -0.000244141, -0.000122070, -0.000061035, +- -0.000030518, -0.000015259 }; +- +- void I_subband_quantization(scalar, sb_samples, j_scale, j_samps, +- bit_alloc, sbband, fr_ps) +- unsigned int scalar[2][3][SBLIMIT]; +- double FAR sb_samples[2][3][SCALE_BLOCK][SBLIMIT]; +- unsigned int j_scale[3][SBLIMIT]; +- double FAR j_samps[3][SCALE_BLOCK][SBLIMIT]; /* L+R for j-stereo if necess */ +- unsigned int bit_alloc[2][SBLIMIT]; +- unsigned int FAR sbband[2][3][SCALE_BLOCK][SBLIMIT]; +- frame_params *fr_ps; +- { +- int i, j, k, n, sig; +- int stereo = fr_ps->stereo; +- int sblimit = fr_ps->sblimit; +- int jsbound = fr_ps->jsbound; +- double d; +- static char init = 0; +- +- if (!init) { +- init = 1; +- /* rearrange quantization coef to correspond to layer I table */ +- a[1] = a[2]; b[1] = b[2]; +- for (i=2;i<15;i++) { a[i] = a[i+2]; b[i] = b[i+2]; } +- } +- for (j=0;j=jsbound) +- /* use the joint data passed in */ +- d = j_samps[0][j][i] / multiple[j_scale[0][i]]; +- else +- d = sb_samples[k][0][j][i] / multiple[scalar[k][0][i]]; +- /* scale and quantize floating point sample */ +- n = bit_alloc[k][i]; +- d = d * a[n-1] + b[n-1]; +- /* extract MSB N-1 bits from the floating point sample */ +- if (d >= 0) sig = 1; +- else { sig = 0; d += 1.0; } +- sbband[k][0][j][i] = (unsigned int) (d * (double) (1L<stereo; +- int sblimit = fr_ps->sblimit; +- int jsbound = fr_ps->jsbound; +- unsigned int stps; +- double d; +- al_table *alloc = fr_ps->alloc; +- +- for (s=0;s<3;s++) +- for (j=0;j=jsbound) /* use j-stereo samples */ +- d = j_samps[s][j][i] / multiple[j_scale[s][i]]; +- else +- d = sb_samples[k][s][j][i] / multiple[scalar[k][s][i]]; +- if (mod(d) > 1.0) +- printf("Not scaled properly %d %d %d %d\n",k,s,j,i); +- qnt = (*alloc)[i][bit_alloc[k][i]].quant; +- d = d * a[qnt] + b[qnt]; +- /* extract MSB N-1 bits from the floating point sample */ +- if (d >= 0) sig = 1; +- else { sig = 0; d += 1.0; } +- n = 0; +- #ifndef MS_DOS +- stps = (*alloc)[i][bit_alloc[k][i]].steps; +- while ((1L<stereo; +- int sblimit = fr_ps->sblimit; +- int jsbound = fr_ps->jsbound; +- +- for (i=0;istereo; +- int sblimit = fr_ps->sblimit; +- int jsbound = fr_ps->jsbound; +- al_table *alloc = fr_ps->alloc; +- +- for (i=0;istereo; +- int sblimit = fr_ps->sblimit; +- int jsbound = fr_ps->jsbound; +- +- for(j=0;jstereo; +- int sblimit = fr_ps->sblimit; +- int jsbound = fr_ps->jsbound; +- al_table *alloc = fr_ps->alloc; +- +- for (s=0;s<3;s++) +- for (j=0;jgr[gr].ch[ch]) ; ++ block_type = cod_info->block_type; ++ ++ /* ++ Compensate for inversion in the analysis filter ++ */ ++ for ( band = 1; band < 32; band+=2 ) ++ for ( k = 1; k < 18; k+=2 ) ++ sb_sample[ch][gr+1][k][band] *= -1.0; ++ ++ /* ++ Perform imdct of 18 previous subband samples ++ + 18 current subband samples ++ */ ++ for ( band = 0; band < 32; band++ ) ++ { ++ for ( k = 0; k < 18; k++ ) ++ { ++ mdct_in[k] = sb_sample[ch][ gr ][k][band]; ++ mdct_in[k+18] = sb_sample[ch][gr+1][k][band]; ++ } ++ #ifdef ALLOW_MIXED ++ if ( cod_info->mixed_block_flag && (band < 2) ) ++ block_type = 0; ++ #endif ++ if (block_type==2) ++ mdct_short(mdct_in, &mdct_enc[gr][ch][band][0], block_type ); ++ else ++ mdct_long( mdct_in, &mdct_enc[gr][ch][band][0], block_type ); ++ } ++ ++ /* ++ Perform aliasing reduction butterfly ++ on long blocks ++ */ ++ if ( block_type != 2 ) ++ for ( band = 0; band < 31; band++ ) ++ for ( k = 0; k < 8; k++ ) ++ { ++ bu = mdct_enc[gr][ch][band][17-k] * cs[k] + mdct_enc[gr][ch][band+1][k] * ca[k]; ++ bd = mdct_enc[gr][ch][band+1][k] * cs[k] - mdct_enc[gr][ch][band][17-k] * ca[k]; ++ mdct_enc[gr][ch][band][17-k] = bu; ++ mdct_enc[gr][ch][band+1][k] = bd; ++ } ++ ++ } ++ ++ /* ++ Save latest granule's subband samples to be used in ++ the next mdct call ++ */ ++ for ( ch = 0; ch < stereo; ch++ ) ++ for ( j = 0; j < 18; j++ ) ++ for ( band = 0; band < 32; band++ ) ++ sb_sample[ch][0][j][band] = sb_sample[ch][mode_gr][j][band]; ++ } ++ ++ void mdct_init(void) ++ { ++ int k,i,m,N; ++ ++ /* type 0 */ ++ for ( i = 0; i < 36; i++ ) ++ win[0][i] = sin( PI/36 * (i + 0.5) ); ++ /* type 1*/ ++ for ( i = 0; i < 18; i++ ) ++ win[1][i] = sin( PI/36 * (i + 0.5) ); ++ for ( i = 18; i < 24; i++ ) ++ win[1][i] = 1.0; ++ for ( i = 24; i < 30; i++ ) ++ win[1][i] = sin( PI/12 * ( i + 0.5 - 18) ); ++ for ( i = 30; i < 36; i++ ) ++ win[1][i] = 0.0; ++ /* type 3*/ ++ for ( i = 0; i < 6; i++ ) ++ win[3][i] = 0.0; ++ for ( i = 6; i < 12; i++ ) ++ win[3][i] = sin( PI/12 * (i + 0.5 - 6) ); ++ for ( i = 12; i < 18; i++ ) ++ win[3][i] = 1.0; ++ for ( i = 18; i < 36; i++ ) ++ win[3][i] = sin( PI/36 * (i + 0.5) ); ++ ++ N = 12; ++ for ( m = 0; m < N / 2; m++ ) ++ for ( k = 0; k < N; k++ ) ++ cos_s[m][k] = cos( (PI /(2 * N)) * (2 * k + 1 + N / 2) * ++ (2 * m + 1) ) / (N / 4) * sin( PI/12 * (k + 0.5) ); ++ ++ N = 36; ++ for ( m = 0; m < N / 2; m++ ) ++ { ++ for ( k = 0; k < 9; k++ ) ++ { ++ cos_l[m][k] = cos( (PI / (2 * N)) * (2 * k + 1 + N / 2) * ++ (2 * m + 1) ) / (N / 4); ++ cos_l[m][9+k] = cos( (PI / (2 * N)) * (2 * (18 + k) + 1 + N / 2) * ++ (2 * m + 1) ) / (N / 4); ++ } ++ } ++ } ++ ++ void mdct_short( double *in, double *out, int block_type) ++ { ++ int l,m,k,N; ++ double sum; ++ N = 12; ++ for ( l = 0; l < 3; l++ ) ++ { ++ for ( m = 0; m < N / 2; m++ ) ++ { ++ for ( sum = 0.0, k = 0; k < N; k++ ) ++ sum += in[k + 6 * l + 6] * cos_s[m][k]; ++ out[ 3 * m + l] = sum; ++ } ++ } ++ } ++ ++ void mdct_long (double *in, double *out, int block_type) { ++ int k; ++ double sum; ++ static double fin[18]; ++ static double *cos_l0; ++ double s0, s1, s2, s3, s4, s5; ++ double *finp; ++ ++ int all[] = {0,2,3,5,6,8,9,11,12,14,15,17,-1}; ++ int i,j=0; ++ for (k=0;k<9;k++) ++ { ++ fin[k] = win[block_type][k] * in[k] - win[block_type][17-k] * in[17-k]; ++ fin[9+k] = win[block_type][18+k] * in[18+k] + win[block_type][35-k] * in[35-k]; ++ } ++ ++ /* 0, 2, 3, 5, 6, 8, 9, 11, 12, 14, 15, 17 */ ++ while ((i = all[j++])>=0) { ++ cos_l0= &cos_l[i][0]; ++ finp = &fin[0]; ++ sum = ( *finp++ ) * *cos_l0; ++ sum += ( *finp++ ) * *++cos_l0; ++ sum += ( *finp++ ) * *++cos_l0; ++ sum += ( *finp++ ) * *++cos_l0; ++ sum += ( *finp++ ) * *++cos_l0; ++ sum += ( *finp++ ) * *++cos_l0; ++ sum += ( *finp++ ) * *++cos_l0; ++ sum += ( *finp++ ) * *++cos_l0; ++ sum += ( *finp++ ) * *++cos_l0; ++ sum += ( *finp++ ) * *++cos_l0; ++ sum += ( *finp++ ) * *++cos_l0; ++ sum += ( *finp++ ) * *++cos_l0; ++ sum += ( *finp++ ) * *++cos_l0; ++ sum += ( *finp++ ) * *++cos_l0; ++ sum += ( *finp++ ) * *++cos_l0; ++ sum += ( *finp++ ) * *++cos_l0; ++ sum += ( *finp++ ) * *++cos_l0; ++ sum += ( *finp++ ) * *++cos_l0; ++ out[i]=sum; ++ } ++ ++ ++ s0 = fin[0] + fin[ 5] + fin[15]; ++ s1 = fin[1] + fin[ 4] + fin[16]; ++ s2 = fin[2] + fin[ 3] + fin[17]; ++ s3 = fin[6] - fin[ 9] + fin[14]; ++ s4 = fin[7] - fin[10] + fin[13]; ++ s5 = fin[8] - fin[11] + fin[12]; ++ ++ /* 1 */ ++ cos_l0 = & cos_l[1][0]; ++ sum = s0 * *cos_l0; /* mfc=3 2*/ ++ sum += s1 * *++cos_l0; /* mfc=9 1*/ ++ sum += s2 * *++cos_l0; /* mfc=15 0*/ ++ sum += s3 * *(cos_l0+=4); /* mfc=21 6*/ ++ sum += s4 * *++cos_l0; /* mfc=27 7*/ ++ sum += s5 * *++cos_l0;; /* mfc = 28 8*/ ++ out[1]=sum; ++ ++ /* 7 */ ++ cos_l0 = &cos_l[7][0]; ++ sum = s0 * *cos_l0; ++ sum += s1 * *++cos_l0; ++ sum += s2 * *++cos_l0; ++ sum += s3 * *(cos_l0+=4); ++ sum += s4 * *++cos_l0; ++ sum += s5 * *++cos_l0; ++ out[7]=sum; ++ ++ /* 10 */ ++ cos_l0= &cos_l[10][0]; ++ sum = s0 * *cos_l0; /* 33 */ ++ sum += s1 * *++cos_l0; /* 9 */ ++ sum += s2 * *++cos_l0; /* 21 */ ++ sum += s3 * *(cos_l0+=4); /* 3 */ ++ sum += s4 * *++cos_l0; /* 27 */ ++ sum += s5 * *++cos_l0; /* 15 */ ++ out[10]=sum; ++ ++ /* 16 */ ++ cos_l0 = &cos_l[16][0]; ++ sum = s0 * *cos_l0; /* 21 */ ++ sum += s1 * *++cos_l0; /* 27 */ ++ sum += s2 * *++cos_l0; /* 33 */ ++ sum += s3 * *(cos_l0+=4); /* 15 */ ++ sum += s4 * *++cos_l0; /* 9 */ ++ sum += s5 * *++cos_l0; /* 3 */ ++ out[16]=sum; ++ ++ s0 = s0 - s1 + s5; ++ s2 = s2 - s3 - s4; ++ ++ /* 4 */ ++ sum = s0 * cos_l[4][0]; ++ sum += s2* cos_l[4][2]; ++ out[4]=sum; ++ ++ /* 13 */ ++ sum = s0 * cos_l[13][0]; ++ sum += s2 * cos_l[13][2]; ++ out[13]=sum; ++ ++ } +diff -r -c -N encoder/fastmdct.h lame3.50/fastmdct.h +*** encoder/fastmdct.h Wed Dec 31 17:00:00 1969 +--- lame3.50/fastmdct.h Mon Oct 25 12:00:08 1999 +*************** +*** 0 **** +--- 1,25 ---- ++ /********************************************************************** ++ * ISO MPEG Audio Subgroup Software Simulation Group (1996) ++ * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension ++ * ++ * $Id: mdct.h,v 1.1 1996/02/14 04:04:23 rowlands Exp $ ++ * ++ * $Log: mdct.h,v $ ++ * Revision 1.1 1996/02/14 04:04:23 rowlands ++ * Initial revision ++ * ++ * Received from Mike Coleman ++ **********************************************************************/ ++ ++ #ifndef MDCT_DOT_H ++ #define MDCT_DOT_H ++ void mdct(double *in, double *out, int block_type); ++ void inv_mdct(double *in, double *out, int block_type); ++ ++ typedef double D32_18[SBLIMIT][18]; ++ typedef double L3SBS[2][3][18][SBLIMIT]; /* [gr][ch] */ ++ ++ void mdct_sub(L3SBS sb_sample, double (*mdct_freq)[2][576], int stereo, III_side_info_t *l3_side, int mode_gr ); ++ void mdct_sub_dec(double (*mdct_freq)[2][576], double inv_mdct_dec[3][2][18][32], int stereo, III_side_info_t *l3_side); ++ void delay(double (*xr)[2][576], int stereo); ++ #endif +diff -r -c -N encoder/fft.c lame3.50/fft.c +*** encoder/fft.c Wed Dec 31 17:00:00 1969 +--- lame3.50/fft.c Mon Nov 1 17:45:44 1999 +*************** +*** 0 **** +--- 1,259 ---- ++ /* ++ ** FFT and FHT routines ++ ** Copyright 1988, 1993; Ron Mayer ++ ** ++ ** fht(fz,n); ++ ** Does a hartley transform of "n" points in the array "fz". ++ ** ++ ** NOTE: This routine uses at least 2 patented algorithms, and may be ++ ** under the restrictions of a bunch of different organizations. ++ ** Although I wrote it completely myself; it is kind of a derivative ++ ** of a routine I once authored and released under the GPL, so it ++ ** may fall under the free software foundation's restrictions; ++ ** it was worked on as a Stanford Univ project, so they claim ++ ** some rights to it; it was further optimized at work here, so ++ ** I think this company claims parts of it. The patents are ++ ** held by R. Bracewell (the FHT algorithm) and O. Buneman (the ++ ** trig generator), both at Stanford Univ. ++ ** If it were up to me, I'd say go do whatever you want with it; ++ ** but it would be polite to give credit to the following people ++ ** if you use this anywhere: ++ ** Euler - probable inventor of the fourier transform. ++ ** Gauss - probable inventor of the FFT. ++ ** Hartley - probable inventor of the hartley transform. ++ ** Buneman - for a really cool trig generator ++ ** Mayer(me) - for authoring this particular version and ++ ** including all the optimizations in one package. ++ ** Thanks, ++ ** Ron Mayer; mayer@acuson.com ++ ** ++ */ ++ ++ #include ++ #include "util.h" ++ ++ static FLOAT costab[20]= ++ { ++ .00000000000000000000000000000000000000000000000000, ++ .70710678118654752440084436210484903928483593768847, ++ .92387953251128675612818318939678828682241662586364, ++ .98078528040323044912618223613423903697393373089333, ++ .99518472667219688624483695310947992157547486872985, ++ .99879545620517239271477160475910069444320361470461, ++ .99969881869620422011576564966617219685006108125772, ++ .99992470183914454092164649119638322435060646880221, ++ .99998117528260114265699043772856771617391725094433, ++ .99999529380957617151158012570011989955298763362218, ++ .99999882345170190992902571017152601904826792288976, ++ .99999970586288221916022821773876567711626389934930, ++ .99999992646571785114473148070738785694820115568892, ++ .99999998161642929380834691540290971450507605124278, ++ .99999999540410731289097193313960614895889430318945, ++ .99999999885102682756267330779455410840053741619428 ++ }; ++ static FLOAT sintab[20]= ++ { ++ 1.0000000000000000000000000000000000000000000000000, ++ .70710678118654752440084436210484903928483593768846, ++ .38268343236508977172845998403039886676134456248561, ++ .19509032201612826784828486847702224092769161775195, ++ .09801714032956060199419556388864184586113667316749, ++ .04906767432741801425495497694268265831474536302574, ++ .02454122852291228803173452945928292506546611923944, ++ .01227153828571992607940826195100321214037231959176, ++ .00613588464915447535964023459037258091705788631738, ++ .00306795676296597627014536549091984251894461021344, ++ .00153398018628476561230369715026407907995486457522, ++ .00076699031874270452693856835794857664314091945205, ++ .00038349518757139558907246168118138126339502603495, ++ .00019174759731070330743990956198900093346887403385, ++ .00009587379909597734587051721097647635118706561284, ++ .00004793689960306688454900399049465887274686668768 ++ }; ++ ++ /* This is a simplified version for n an even power of 2 */ ++ ++ static void fht(FLOAT *fz, int n) ++ { ++ int i,k,k1,k2,k3,k4,kx; ++ FLOAT *fi,*fn,*gi; ++ FLOAT t_c,t_s; ++ ++ for (k1=1,k2=0;k1>1; (!((k2^=k)&k)); k>>=1); ++ if (k1>k2) ++ { ++ a=fz[k1];fz[k1]=fz[k2];fz[k2]=a; ++ } ++ } ++ for (fi=fz,fn=fz+n;fi> 1; ++ fi = fz; ++ gi = fi + kx; ++ fn = fz + n; ++ do ++ { ++ FLOAT g0,f0,f1,g1,f2,g2,f3,g3; ++ f1 = fi[0 ] - fi[k1]; ++ f0 = fi[0 ] + fi[k1]; ++ f3 = fi[k2] - fi[k3]; ++ f2 = fi[k2] + fi[k3]; ++ fi[k2] = f0 - f2; ++ fi[0 ] = f0 + f2; ++ fi[k3] = f1 - f3; ++ fi[k1] = f1 + f3; ++ g1 = gi[0 ] - gi[k1]; ++ g0 = gi[0 ] + gi[k1]; ++ g3 = SQRT2 * gi[k3]; ++ g2 = SQRT2 * gi[k2]; ++ gi[k2] = g0 - g2; ++ gi[0 ] = g0 + g2; ++ gi[k3] = g1 - g3; ++ gi[k1] = g1 + g3; ++ gi += k4; ++ fi += k4; ++ } while (finGranules <= MAX_GRANULES ); + assert( frameInfo->nChannels <= MAX_CHANNELS ); +--- 57,67 ---- + See formatBitstream.h for more information about the data + structures and the bitstream syntax. + */ ++ static int elements, forwardFrameLength, forwardSILength; + void + BF_BitstreamFrame( BF_FrameData *frameInfo, BF_FrameResults *results ) + { +! /* int elements, forwardFrameLength, forwardSILength; */ + + assert( frameInfo->nGranules <= MAX_GRANULES ); + assert( frameInfo->nChannels <= MAX_CHANNELS ); +*************** +*** 87,108 **** + void + BF_FlushBitstream( BF_FrameData *frameInfo, BF_FrameResults *results ) + { +! int elements, forwardFrameLength, forwardSILength; + + /* get ptr to bit writing function */ + PutBits = frameInfo->putbits; + assert( PutBits ); + +- elements = side_queue_elements( &forwardFrameLength, &forwardSILength ); +- + if ( elements ) + { +! int bitsRemaining = forwardFrameLength - forwardSILength; +! int wordsRemaining = bitsRemaining / 32; +! while ( wordsRemaining-- ) +! WriteMainDataBits( 0, 32, results ); +! WriteMainDataBits( 0, (bitsRemaining % 32), results ); + } + + results->mainDataLength = forwardFrameLength - forwardSILength; + results->SILength = forwardSILength; +--- 96,117 ---- + void + BF_FlushBitstream( BF_FrameData *frameInfo, BF_FrameResults *results ) + { +! /* int elements, forwardFrameLength, forwardSILength; */ + + /* get ptr to bit writing function */ + PutBits = frameInfo->putbits; + assert( PutBits ); + + if ( elements ) + { +! int bitsRemaining = forwardFrameLength - forwardSILength; +! int wordsRemaining = bitsRemaining / 32; +! while ( wordsRemaining-- ) { +! WriteMainDataBits( 0, 32, results ); +! } +! WriteMainDataBits( 0, (bitsRemaining % 32), results ); + } ++ + + results->mainDataLength = forwardFrameLength - forwardSILength; + results->SILength = forwardSILength; +*************** +*** 131,137 **** + return bits; + } + +! + /* + The following is all private to this file + */ +--- 140,146 ---- + return bits; + } + +! + /* + The following is all private to this file + */ +*************** +*** 148,155 **** + BF_PartHolder *spectrumSIPH[MAX_GRANULES][MAX_CHANNELS]; + } MYSideInfo; + +! static MYSideInfo *get_side_info(); +! static int write_side_info(); + typedef int (*PartWriteFcnPtr)( BF_BitstreamPart *part, BF_FrameResults *results ); + + +--- 157,164 ---- + BF_PartHolder *spectrumSIPH[MAX_GRANULES][MAX_CHANNELS]; + } MYSideInfo; + +! static MYSideInfo *get_side_info(void); +! static int write_side_info(void); + typedef int (*PartWriteFcnPtr)( BF_BitstreamPart *part, BF_FrameResults *results ); + + +*************** +*** 246,254 **** + assert( (BitCount + BitsRemaining) == ThisFrameSize ); + } + +! + static int +! write_side_info() + { + MYSideInfo *si; + int bits, ch, gr; +--- 255,263 ---- + assert( (BitCount + BitsRemaining) == ThisFrameSize ); + } + +! + static int +! write_side_info(void) + { + MYSideInfo *si; + int bits, ch, gr; +*************** +*** 284,290 **** + side_queue_elements( int *frameLength, int *SILength ) + { + int elements = 0; +! side_info_link *l = side_queue_head; + + *frameLength = 0; + *SILength = 0; +--- 293,299 ---- + side_queue_elements( int *frameLength, int *SILength ) + { + int elements = 0; +! side_info_link *l; + + *frameLength = 0; + *SILength = 0; +*************** +*** 302,308 **** + store_side_info( BF_FrameData *info ) + { + int ch, gr; +! side_info_link *l = NULL; + /* obtain a side_info_link to store info */ + side_info_link *f = side_queue_free; + int bits = 0; +--- 311,317 ---- + store_side_info( BF_FrameData *info ) + { + int ch, gr; +! side_info_link *l; + /* obtain a side_info_link to store info */ + side_info_link *f = side_queue_free; + int bits = 0; +*************** +*** 377,383 **** + } + + static MYSideInfo* +! get_side_info() + { + side_info_link *f = side_queue_free; + side_info_link *l = side_queue_head; +--- 386,392 ---- + } + + static MYSideInfo* +! get_side_info(void) + { + side_info_link *f = side_queue_free; + side_info_link *l = side_queue_head; +*************** +*** 404,410 **** + } + + static void +! free_side_queues() + { + side_info_link *l, *next; + +--- 413,419 ---- + } + + static void +! free_side_queues(void) + { + side_info_link *l, *next; + +*************** +*** 445,456 **** + */ + BF_PartHolder *BF_newPartHolder( int max_elements ) + { +! BF_PartHolder *newPH = calloc( 1, sizeof(BF_PartHolder) ); + assert( newPH ); + newPH->max_elements = max_elements; +! newPH->part = calloc( 1, sizeof(BF_BitstreamPart) ); + assert( newPH->part ); +! newPH->part->element = calloc( max_elements, sizeof(BF_BitstreamElement) ); + assert( newPH->part->element ); + newPH->part->nrEntries = 0; + return newPH; +--- 454,465 ---- + */ + BF_PartHolder *BF_newPartHolder( int max_elements ) + { +! BF_PartHolder *newPH = (BF_PartHolder*) calloc( 1, sizeof(BF_PartHolder) ); + assert( newPH ); + newPH->max_elements = max_elements; +! newPH->part = (BF_BitstreamPart*) calloc( 1, sizeof(BF_BitstreamPart) ); + assert( newPH->part ); +! newPH->part->element = (BF_BitstreamElement*) calloc( max_elements, sizeof(BF_BitstreamElement) ); + assert( newPH->part->element ); + newPH->part->nrEntries = 0; + return newPH; +diff -r -c -N encoder/formatBitstream.h lame3.50/formatBitstream.h +*** encoder/formatBitstream.h Wed Jan 22 02:43:15 1997 +--- lame3.50/formatBitstream.h Tue Sep 28 19:37:14 1999 +*************** +*** 97,145 **** + int nextBackPtr; + } BF_FrameResults; + +- /* +- The following is a shorthand bitstream syntax for +- the type of bitstream this package will create. +- The bitstream has headers and side information that +- are placed at appropriate sections to allow framing. +- The main data is placed where it fits in a manner +- similar to layer3, which means that main data for a +- frame may be written to the bitstream before the +- frame's header and side information is written. +- +- BitstreamFrame() +- { +- Header(); +- FrameSI(); +- +- for ( ch ) +- ChannelSI(); +- +- for ( gr ) +- for ( ch ) +- SpectrumSI(); +- +- MainData(); +- } +- +- MainData() +- { +- for ( gr ) +- for ( ch ) +- { +- Scalefactors(); +- CodedData(); +- UserSpectrum(); +- } +- UserFrameData(); +- } +- +- */ + + /* + public functions in formatBitstream.c + */ + + /* count the bits in a BitstreamPart */ + int BF_PartLength( BF_BitstreamPart *part ); + +--- 97,111 ---- + int nextBackPtr; + } BF_FrameResults; + + + /* + public functions in formatBitstream.c + */ + ++ ++ /* Initialize the bitstream */ ++ void InitFormatBitStream(void); ++ + /* count the bits in a BitstreamPart */ + int BF_PartLength( BF_BitstreamPart *part ); + +*************** +*** 149,154 **** +--- 115,122 ---- + /* write any remaining frames to the bitstream, padding with zeros */ + void BF_FlushBitstream( BF_FrameData *frameInfo, BF_FrameResults *results ); + ++ ++ + typedef struct BF_PartHolder + { + int max_elements; +*************** +*** 162,165 **** + BF_PartHolder *BF_NewHolderFromBitstreamPart( BF_BitstreamPart *thePart ); + BF_PartHolder *BF_LoadHolderFromBitstreamPart( BF_PartHolder *theHolder, BF_BitstreamPart *thePart ); + BF_PartHolder *BF_freePartHolder( BF_PartHolder *thePH ); +! #endif +--- 130,134 ---- + BF_PartHolder *BF_NewHolderFromBitstreamPart( BF_BitstreamPart *thePart ); + BF_PartHolder *BF_LoadHolderFromBitstreamPart( BF_PartHolder *theHolder, BF_BitstreamPart *thePart ); + BF_PartHolder *BF_freePartHolder( BF_PartHolder *thePH ); +! +! #endif /* #ifndef _FORMAT_BITSTREAM_H */ +diff -r -c -N encoder/get_audio.c lame3.50/get_audio.c +*** encoder/get_audio.c Wed Dec 31 17:00:00 1969 +--- lame3.50/get_audio.c Thu Oct 28 21:45:33 1999 +*************** +*** 0 **** +--- 1,806 ---- ++ #include "util.h" ++ #include "get_audio.h" ++ #include "globalflags.h" ++ #include "lame.h" ++ #ifdef HAVEGTK ++ #include "gtkanal.h" ++ #include ++ #endif ++ ++ static FILE *musicin=NULL; /* input file pointer */ ++ static unsigned long num_samples; ++ static int samp_freq; ++ static int input_bitrate; ++ static int num_channels; ++ ++ int read_samples_pcm( short sample_buffer[2304],int frame_size, int samples_to_read); ++ int read_samples_mp3(FILE *musicin,short int mpg123pcm[2][1152],int num_chan); ++ ++ /************************************************************************ ++ * ++ * get_audio() ++ * ++ * PURPOSE: reads a frame of audio data from a file to the buffer, ++ * aligns the data for future processing, and separates the ++ * left and right channels ++ * ++ * ++ ************************************************************************/ ++ int get_audio(short buffer[2][1152],int stereo, layer* info) ++ { ++ ++ int j; ++ short insamp[2304]; ++ int samples_read; ++ int framesize,samples_to_read; ++ static unsigned long num_samples_read; ++ unsigned long remaining; ++ if (frameNum==0) { ++ num_samples_read=0; ++ num_samples= GetSndSamples(); ++ } ++ ++ remaining=num_samples-num_samples_read; ++ framesize = (info->version ==0) ? 576 : 1152; ++ samples_to_read = remaining > framesize ? framesize : remaining; ++ if (samples_to_read<0) samples_to_read=0; ++ ++ if (input_format==sf_mp3) { ++ if ((stereo==2) || (autoconvert) ) ++ samples_read=read_samples_mp3(musicin,buffer,2); ++ else ++ samples_read=read_samples_mp3(musicin,buffer,1); ++ }else{ ++ /* MPEG 1 */ ++ if(stereo == 2) { ++ samples_read = read_samples_pcm(insamp, 2*framesize,2*samples_to_read); ++ samples_read /=2; ++ for(j=0;j0) { ++ /* set adj=0 for 1152 samples, adj=576 for 576 samples */ ++ adj=0; ++ if (out==576) adj=576; ++ ++ /* create a 1 frame lag so MP3 analysis syncs with WAV analysis */ ++ /* Compensate for the decoder_delay. add DECDELAY, where ++ * decoder_delay + DECDELAY = 1152 (or 576), so data is delayed ++ * by exactly one frame */ ++ /* if out=0, last frame of data is in mpg123pcm */ ++ for ( ch = 0; ch < stereo; ch++ ) { ++ for ( j = 0; j < DECDELAY-adj; j++ ) ++ pinfo->pcmdata2[ch][j] = pinfo->pcmdata2[ch][j+1152-adj]; ++ for ( j = 0; j < 1152-adj; j++ ) ++ pinfo->pcmdata2[ch][j+DECDELAY-adj] = mpg123pcm[ch][j]; ++ } ++ } ++ ++ pinfo->frameNum123 = frameNum-1; ++ pinfo->frameNum = frameNum; ++ } ++ #endif ++ return out; ++ } ++ ++ ++ ++ ++ ++ ++ #ifdef LIBSNDFILE ++ /* ++ ** Copyright (C) 1999 Albert Faber ++ ** ++ ** This program is free software; you can redistribute it and/or modify ++ ** it under the terms of the GNU General Public License as published by ++ ** the Free Software Foundation; either version 2 of the License, or ++ ** (at your option) any later version. ++ ** ++ ** This program is distributed in the hope that it will be useful, ++ ** but WITHOUT ANY WARRANTY; without even the implied warranty of ++ ** MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the ++ ** GNU General Public License for more details. ++ ** ++ ** You should have received a copy of the GNU General Public License ++ ** along with this program; if not, write to the Free Software ++ ** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. ++ */ ++ ++ #include ++ ++ /* External references */ ++ ++ static SNDFILE* gs_pSndFileIn=NULL; ++ static SF_INFO gs_wfInfo; ++ ++ ++ unsigned long GetSndSamples(void) ++ { ++ return gs_wfInfo.samples; ++ } ++ int GetSndSampleRate(void) ++ { ++ return gs_wfInfo.samplerate; ++ } ++ ++ int GetSndChannels(void) ++ { ++ return gs_wfInfo.channels; ++ } ++ ++ void CloseSndFile(void) ++ { ++ if (input_format==sf_mp3) { ++ if (fclose(musicin) != 0){ ++ fprintf(stderr, "Could not close audio input file\n"); ++ exit(2); ++ } ++ }else{ ++ if (gs_pSndFileIn) ++ { ++ if (sf_close(gs_pSndFileIn) !=0) ++ { ++ fprintf(stderr, "Could not close sound file \n"); ++ exit(2); ++ } ++ } ++ } ++ } ++ ++ ++ void InitSndFile() ++ { ++ gs_wfInfo.samples=MAX_U_32_NUM; ++ gs_wfInfo.channels=0; ++ gs_wfInfo.samplerate=0; ++ input_bitrate=0; ++ } ++ ++ ++ void OpenSndFile(const char* lpszFileName, layer *info,int default_samp, ++ int default_channels) ++ { ++ input_bitrate=0; ++ if (input_format==sf_mp3) { ++ if ((musicin = fopen(lpszFileName, "rb")) == NULL) { ++ fprintf(stderr, "Could not find \"%s\".\n", lpszFileName); ++ exit(1); ++ } ++ if (-1==lame_decode_initfile(musicin,&num_channels,&samp_freq,&input_bitrate)) { ++ fprintf(stderr,"Error reading headers in mp3 input file %s.\n", lpszFileName); ++ exit(1); ++ } ++ gs_wfInfo.samples=MAX_U_32_NUM; ++ gs_wfInfo.channels=num_channels; ++ gs_wfInfo.samplerate=samp_freq; ++ ++ } else { ++ ++ // Try to open the sound file ++ /* set some defaults incase input is raw PCM */ ++ gs_wfInfo.seekable=(input_format!=sf_raw); /* if user specified -r, set to not seekable */ ++ gs_wfInfo.samplerate=default_samp; ++ gs_wfInfo.pcmbitwidth=16; ++ gs_wfInfo.channels=default_channels; ++ if (DetermineByteOrder()==order_littleEndian) ++ gs_wfInfo.format=SF_FORMAT_RAW_LE; ++ else ++ gs_wfInfo.format=SF_FORMAT_RAW_BE; ++ ++ gs_pSndFileIn=sf_open_read(lpszFileName,&gs_wfInfo); ++ ++ // Check result ++ if (gs_pSndFileIn==NULL) ++ { ++ sf_perror(gs_pSndFileIn); ++ fprintf(stderr, "Could not open sound file \"%s\".\n", lpszFileName); ++ exit(1); ++ } ++ ++ if ((gs_wfInfo.format==SF_FORMAT_RAW_LE) || ++ (gs_wfInfo.format==SF_FORMAT_RAW_BE)) ++ input_format=sf_raw; ++ ++ #ifdef _DEBUG_SND_FILE ++ printf("\n\nSF_INFO structure\n"); ++ printf("samplerate :%d\n",gs_wfInfo.samplerate); ++ printf("samples :%d\n",gs_wfInfo.samples); ++ printf("channels :%d\n",gs_wfInfo.channels); ++ printf("pcmbitwidth :%d\n",gs_wfInfo.pcmbitwidth); ++ printf("format :"); ++ ++ if (gs_wfInfo.format&SF_FORMAT_WAV) printf("Microsoft WAV format (big endian)"); ++ if (gs_wfInfo.format&SF_FORMAT_AIFF) printf("Apple/SGI AIFF format (little endian)."); ++ if (gs_wfInfo.format&SF_FORMAT_AU) printf("Sun/NeXT AU format (big endian)."); ++ if (gs_wfInfo.format&SF_FORMAT_AULE) printf("DEC AU format (little endian)."); ++ if (gs_wfInfo.format&SF_FORMAT_PCM) printf("PCM data in 8, 16, 24 or 32 bits."); ++ if (gs_wfInfo.format&SF_FORMAT_FLOAT) printf("32 bit Intel x86 floats."); ++ if (gs_wfInfo.format&SF_FORMAT_ULAW) printf("U-Law encoded."); ++ if (gs_wfInfo.format&SF_FORMAT_ALAW) printf("A-Law encoded."); ++ if (gs_wfInfo.format&SF_FORMAT_MS_ADPCM) printf("Microsoft ADPCM."); ++ if (gs_wfInfo.format&SF_FORMAT_IMA_ADPCM) printf("IMA ADPCM."); ++ ++ printf("\n"); ++ printf("pcmbitwidth :%d\n",gs_wfInfo.pcmbitwidth); ++ printf("sections :%d\n",gs_wfInfo.sections); ++ printf("seekable :\n",gs_wfInfo.seekable); ++ #endif ++ } ++ } ++ ++ ++ /************************************************************************ ++ * ++ * read_samples() ++ * ++ * PURPOSE: reads the PCM samples from a file to the buffer ++ * ++ * SEMANTICS: ++ * Reads #samples_read# number of shorts from #musicin# filepointer ++ * into #sample_buffer[]#. Returns the number of samples read. ++ * ++ ************************************************************************/ ++ ++ int read_samples_pcm(short sample_buffer[2304],int frame_size,int samples_to_read) ++ { ++ int samples_read; ++ int rcode; ++ ++ samples_read=sf_read_short(gs_pSndFileIn,sample_buffer,samples_to_read); ++ ++ rcode = samples_read; ++ if (samples_read < frame_size) ++ { ++ /* printf("Insufficient PCM input for one frame - fillout with zeros\n"); */ ++ if (samples_read<0) samples_read=0; ++ for (; samples_read < frame_size; sample_buffer[samples_read++] = 0); ++ } ++ return(rcode); ++ } ++ ++ ++ #else /* ifdef LIBSNDFILE */ ++ ++ /************************************************************************ ++ ************************************************************************ ++ ************************************************************************ ++ ************************************************************************ ++ ************************************************************************ ++ ************************************************************************ ++ * ++ * OLD ISO/LAME routines follow. Used if you dont have LIBSNDFILE ++ * or for stdin/stdout support ++ * ++ ************************************************************************ ++ ************************************************************************ ++ ************************************************************************ ++ ************************************************************************ ++ ************************************************************************ ++ ************************************************************************/ ++ ++ /* Replacement for forward fseek(,,SEEK_CUR), because fseek() fails on pipes */ ++ int fskip(FILE *sf,long num_bytes,int dummy) ++ { ++ char data[1024]; ++ int nskip = 0; ++ while (num_bytes > 0) { ++ nskip = (num_bytes>1024) ? 1024 : num_bytes; ++ num_bytes -= fread(data,(size_t)1,(size_t)nskip,sf); ++ } ++ /* return 0 if last read was successful */ ++ return num_bytes; ++ } ++ ++ ++ ++ void CloseSndFile(void) ++ { ++ if (fclose(musicin) != 0){ ++ fprintf(stderr, "Could not close audio input file\n"); ++ exit(2); ++ } ++ } ++ ++ ++ unsigned long GetSndSamples(void) ++ { ++ return num_samples; ++ } ++ int GetSndSampleRate(void) ++ { ++ return samp_freq; ++ } ++ ++ int GetSndChannels(void) ++ { ++ return num_channels; ++ } ++ ++ void InitSndFile() ++ { ++ num_samples=MAX_U_32_NUM; ++ samp_freq=0; ++ num_channels = 0; ++ input_bitrate=0; ++ } ++ ++ ++ ++ void OpenSndFile(const char* inPath, layer *info,int default_samp, ++ int default_channels) ++ { ++ void parse_file_header(FILE *sf); ++ /* set the defaults from info incase we cannot determine them from file */ ++ num_samples=MAX_U_32_NUM; ++ samp_freq=default_samp; ++ num_channels = default_channels; ++ ++ if (!strcmp(inPath, "-")) { ++ /* Read from standard input. */ ++ #ifdef __EMX__ ++ _fsetmode(stdin,"b"); ++ #elif (defined __BORLANDC__) ++ setmode(_fileno(stdin), O_BINARY); ++ #elif (defined __CYGWIN__) ++ setmode(fileno(stdin), _O_BINARY); ++ #elif (defined _WIN32) ++ _setmode(_fileno(stdin), _O_BINARY); ++ #endif ++ musicin = stdin; ++ } else { ++ if ((musicin = fopen(inPath, "rb")) == NULL) { ++ fprintf(stderr, "Could not find \"%s\".\n", inPath); ++ exit(1); ++ } ++ } ++ ++ input_bitrate=0; ++ if (input_format==sf_mp3) { ++ if (-1==lame_decode_initfile(musicin,&num_channels,&samp_freq,&input_bitrate)) { ++ fprintf(stderr,"Error reading headers in mp3 input file %s.\n", inPath); ++ exit(1); ++ } ++ }else{ ++ if (input_format != sf_raw) { ++ parse_file_header(musicin); ++ } ++ ++ ++ if (input_format==sf_raw) { ++ /* assume raw PCM */ ++ fprintf(stderr, "Assuming raw pcm input file"); ++ if (swapbytes==TRUE) ++ fprintf(stderr, " : Forcing byte-swapping\n"); ++ else ++ fprintf(stderr, "\n"); ++ } ++ } ++ } ++ ++ ++ /************************************************************************ ++ * ++ * read_samples() ++ * ++ * PURPOSE: reads the PCM samples from a file to the buffer ++ * ++ * SEMANTICS: ++ * Reads #samples_read# number of shorts from #musicin# filepointer ++ * into #sample_buffer[]#. Returns the number of samples read. ++ * ++ ************************************************************************/ ++ ++ int read_samples_pcm(short sample_buffer[2304], int frame_size,int samples_to_read) ++ { ++ int samples_read; ++ int rcode; ++ int iswav=(input_format==sf_wave); ++ ++ if ((samples_read = ++ fread(sample_buffer, sizeof(short), samples_to_read, musicin)) == 0) ++ { /* printf("Hit end of audio data\n");*/ } ++ /* ++ Samples are big-endian. If this is a little-endian machine ++ we must swap ++ */ ++ if ( NativeByteOrder == order_unknown ) ++ { ++ NativeByteOrder = DetermineByteOrder(); ++ if ( NativeByteOrder == order_unknown ) ++ { ++ fprintf( stderr, "byte order not determined\n" ); ++ exit( 1 ); ++ } ++ } ++ if (!iswav && ( NativeByteOrder == order_littleEndian )) ++ SwapBytesInWords( sample_buffer, samples_read ); ++ ++ if (iswav && ( NativeByteOrder == order_bigEndian )) ++ SwapBytesInWords( sample_buffer, samples_read ); ++ ++ if (swapbytes==TRUE) ++ SwapBytesInWords( sample_buffer, samples_read ); ++ ++ ++ rcode=samples_read; ++ if (samples_read < frame_size) { ++ if (samples_read<0) samples_read=0; ++ /* printf("Insufficient PCM input for one frame - fillout with zeros\n"); */ ++ for (; samples_read < frame_size; sample_buffer[samples_read++] = 0); ++ } ++ return(rcode); ++ } ++ ++ ++ ++ #define WAV_ID_RIFF 0x52494646 /* "RIFF" */ ++ #define WAV_ID_WAVE 0x57415645 /* "WAVE" */ ++ #define WAV_ID_FMT 0x666d7420 /* "fmt " */ ++ #define WAV_ID_DATA 0x64617461 /* "data" */ ++ ++ typedef struct fmt_chunk_data_struct { ++ short format_tag; /* Format category */ ++ u_short channels; /* Number of channels */ ++ u_long samples_per_sec; /* Sampling rate */ ++ u_long avg_bytes_per_sec; /* For buffer estimation */ ++ u_short block_align; /* Data block size */ ++ u_short bits_per_sample; /* for PCM data, anyway... */ ++ } fmt_chunk_data; ++ ++ ++ ++ ++ ++ ++ /************************************************************************ ++ * ++ * wave_check ++ * ++ * PURPOSE: Checks Wave header information to make sure it is valid. ++ * Exits if not. ++ * ++ ************************************************************************/ ++ ++ static void ++ wave_check(char *file_name, fmt_chunk_data *wave_info) ++ { ++ if (wave_info->bits_per_sample != 16) { ++ fprintf(stderr, "%d-bit sample-size is not supported!\n", ++ wave_info->bits_per_sample); ++ exit(1); ++ } ++ } ++ ++ ++ /***************************************************************************** ++ * ++ * Read Microsoft Wave headers ++ * ++ * By the time we get here the first 32-bits of the file have already been ++ * read, and we're pretty sure that we're looking at a WAV file. ++ * ++ *****************************************************************************/ ++ ++ static int ++ parse_wave_header(FILE *sf) ++ { ++ fmt_chunk_data wave_info; ++ int is_wav = 0; ++ long data_length = 0, file_length = 0, subSize = 0; ++ int loop_sanity = 0; ++ ++ memset(&wave_info, 0, sizeof(wave_info)); ++ ++ file_length = Read32BitsHighLow(sf); ++ ++ if (Read32BitsHighLow(sf) != WAV_ID_WAVE) ++ return 0; ++ ++ for (loop_sanity = 0; loop_sanity < 20; ++loop_sanity) { ++ u_int type = Read32BitsHighLow(sf); ++ ++ if (type == WAV_ID_FMT) { ++ subSize = Read32BitsLowHigh(sf); ++ if (subSize < 16) { ++ /*fprintf(stderr, ++ "'fmt' chunk too short (only %ld bytes)!", subSize); */ ++ return 0; ++ } ++ ++ wave_info.format_tag = Read16BitsLowHigh(sf); ++ subSize -= 2; ++ wave_info.channels = Read16BitsLowHigh(sf); ++ subSize -= 2; ++ wave_info.samples_per_sec = Read32BitsLowHigh(sf); ++ subSize -= 4; ++ wave_info.avg_bytes_per_sec = Read32BitsLowHigh(sf); ++ subSize -= 4; ++ wave_info.block_align = Read16BitsLowHigh(sf); ++ subSize -= 2; ++ wave_info.bits_per_sample = Read16BitsLowHigh(sf); ++ subSize -= 2; ++ ++ /* fprintf(stderr, " skipping %d bytes\n", subSize); */ ++ ++ if (subSize > 0) { ++ if (fskip(sf, (long)subSize, SEEK_CUR) != 0 ) ++ return 0; ++ }; ++ ++ } else if (type == WAV_ID_DATA) { ++ subSize = Read32BitsLowHigh(sf); ++ data_length = subSize; ++ is_wav = 1; ++ /* We've found the audio data. Read no further! */ ++ break; ++ ++ } else { ++ subSize = Read32BitsLowHigh(sf); ++ if (fskip(sf, (long) subSize, SEEK_CUR) != 0 ) return 0; ++ } ++ } ++ ++ if (is_wav) { ++ /* make sure the header is sane */ ++ wave_check("name", &wave_info); ++ ++ num_channels = wave_info.channels; ++ samp_freq = wave_info.samples_per_sec; ++ num_samples = data_length / (wave_info.channels * wave_info.bits_per_sample / 8); ++ } ++ return is_wav; ++ } ++ ++ ++ ++ /************************************************************************ ++ * ++ * aiff_check ++ * ++ * PURPOSE: Checks AIFF header information to make sure it is valid. ++ * Exits if not. ++ * ++ ************************************************************************/ ++ ++ static void ++ aiff_check2(const char *file_name, IFF_AIFF *pcm_aiff_data) ++ { ++ if (pcm_aiff_data->sampleType != IFF_ID_SSND) { ++ fprintf(stderr, "Sound data is not PCM in \"%s\".\n", file_name); ++ exit(1); ++ } ++ ++ if (pcm_aiff_data->sampleSize != sizeof(short) * BITS_IN_A_BYTE) { ++ fprintf(stderr, "Sound data is not %d bits in \"%s\".\n", ++ (unsigned int) sizeof(short) * BITS_IN_A_BYTE, file_name); ++ exit(1); ++ } ++ ++ if (pcm_aiff_data->numChannels != 1 && ++ pcm_aiff_data->numChannels != 2) { ++ fprintf(stderr, "Sound data is not mono or stereo in \"%s\".\n", ++ file_name); ++ exit(1); ++ } ++ ++ if (pcm_aiff_data->blkAlgn.blockSize != 0) { ++ fprintf(stderr, "Block size is not %d bytes in \"%s\".\n", ++ 0, file_name); ++ exit(1); ++ } ++ ++ if (pcm_aiff_data->blkAlgn.offset != 0) { ++ fprintf(stderr, "Block offset is not %d bytes in \"%s\".\n", ++ 0, file_name); ++ exit(1); ++ } ++ } ++ ++ /***************************************************************************** ++ * ++ * Read Audio Interchange File Format (AIFF) headers. ++ * ++ * By the time we get here the first 32-bits of the file have already been ++ * read, and we're pretty sure that we're looking at an AIFF file. ++ * ++ *****************************************************************************/ ++ ++ static int ++ parse_aiff_header(FILE *sf) ++ { ++ int is_aiff = 0; ++ long chunkSize = 0, subSize = 0; ++ IFF_AIFF aiff_info; ++ ++ memset(&aiff_info, 0, sizeof(aiff_info)); ++ chunkSize = Read32BitsHighLow(sf); ++ ++ if ( Read32BitsHighLow(sf) != IFF_ID_AIFF ) ++ return 0; ++ ++ while ( chunkSize > 0 ) ++ { ++ u_int type = 0; ++ chunkSize -= 4; ++ ++ type = Read32BitsHighLow(sf); ++ ++ /* fprintf(stderr, ++ "found chunk type %08x '%4.4s'\n", type, (char*)&type); */ ++ ++ /* don't use a switch here to make it easier to use 'break' for SSND */ ++ if (type == IFF_ID_COMM) { ++ subSize = Read32BitsHighLow(sf); ++ chunkSize -= subSize; ++ ++ aiff_info.numChannels = Read16BitsHighLow(sf); ++ subSize -= 2; ++ aiff_info.numSampleFrames = Read32BitsHighLow(sf); ++ subSize -= 4; ++ aiff_info.sampleSize = Read16BitsHighLow(sf); ++ subSize -= 2; ++ aiff_info.sampleRate = ReadIeeeExtendedHighLow(sf); ++ subSize -= 10; ++ ++ if (fskip(sf, (long) subSize, SEEK_CUR) != 0 ) ++ return 0; ++ ++ } else if (type == IFF_ID_SSND) { ++ subSize = Read32BitsHighLow(sf); ++ chunkSize -= subSize; ++ ++ aiff_info.blkAlgn.offset = Read32BitsHighLow(sf); ++ subSize -= 4; ++ aiff_info.blkAlgn.blockSize = Read32BitsHighLow(sf); ++ subSize -= 4; ++ ++ if (fskip(sf, aiff_info.blkAlgn.offset, SEEK_CUR) != 0 ) ++ return 0; ++ ++ aiff_info.sampleType = IFF_ID_SSND; ++ is_aiff = 1; ++ ++ /* We've found the audio data. Read no further! */ ++ break; ++ ++ } else { ++ subSize = Read32BitsHighLow(sf); ++ chunkSize -= subSize; ++ ++ if (fskip(sf, (long) subSize, SEEK_CUR) != 0 ) ++ return 0; ++ } ++ } ++ ++ /* fprintf(stderr, "Parsed AIFF %d\n", is_aiff); */ ++ if (is_aiff) { ++ /* make sure the header is sane */ ++ aiff_check2("name", &aiff_info); ++ num_channels = aiff_info.numChannels; ++ samp_freq = aiff_info.sampleRate; ++ num_samples = aiff_info.numSampleFrames; ++ } ++ return is_aiff; ++ } ++ ++ ++ ++ /************************************************************************ ++ * ++ * parse_file_header ++ * ++ * PURPOSE: Read the header from a bytestream. Try to determine whether ++ * it's a WAV file or AIFF without rewinding, since rewind ++ * doesn't work on pipes and there's a good chance we're reading ++ * from stdin (otherwise we'd probably be using libsndfile). ++ * ++ * When this function returns, the file offset will be positioned at the ++ * beginning of the sound data. ++ * ++ ************************************************************************/ ++ ++ void parse_file_header(FILE *sf) ++ { ++ u_int type = 0; ++ type = Read32BitsHighLow(sf); ++ ++ /* fprintf(stderr, ++ "First word of input stream: %08x '%4.4s'\n", type, (char*) &type); */ ++ ++ input_format = sf_raw; ++ ++ if (type == WAV_ID_RIFF) { ++ /* It's probably a WAV file */ ++ if (parse_wave_header(sf)) { ++ input_format = sf_wave; ++ } ++ ++ } else if (type == IFF_ID_FORM) { ++ /* It's probably an AIFF file */ ++ if (parse_aiff_header(sf)) { ++ input_format = sf_aiff; ++ } ++ } ++ if (input_format==sf_raw) { ++ /* ++ ** Assume it's raw PCM. Since the audio data is assumed to begin ++ ** at byte zero, this will unfortunately require seeking. ++ */ ++ if (fseek(sf, 0L, SEEK_SET) != 0) { ++ /* ignore errors */ ++ // fprintf(stderr, "can't seek to beginning of file\n"); ++ //exit(1); ++ } ++ input_format = sf_raw; ++ } ++ } ++ #endif /* LIBSNDFILE */ ++ ++ +diff -r -c -N encoder/get_audio.h lame3.50/get_audio.h +*** encoder/get_audio.h Wed Dec 31 17:00:00 1969 +--- lame3.50/get_audio.h Wed Oct 20 22:01:38 1999 +*************** +*** 0 **** +--- 1,94 ---- ++ /* ++ ** Copyright (C) 1999 Albert Faber ++ ** ++ ** This program is free software; you can redistribute it and/or modify ++ ** it under the terms of the GNU General Public License as published by ++ ** the Free Software Foundation; either version 2 of the License, or ++ ** (at your option) any later version. ++ ** ++ ** This program is distributed in the hope that it will be useful, ++ ** but WITHOUT ANY WARRANTY; without even the implied warranty of ++ ** MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the ++ ** GNU General Public License for more details. ++ ** ++ ** You should have received a copy of the GNU General Public License ++ ** along with this program; if not, write to the Free Software ++ ** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. ++ */ ++ ++ #ifndef GET_AUDIO_H_INCLUDED ++ #define GET_AUDIO_H_INCLUDED ++ ++ ++ #ifndef _BLADEDLL ++ ++ /* AIFF Definitions */ ++ ++ #define IFF_ID_FORM 0x464f524d /* "FORM" */ ++ #define IFF_ID_AIFF 0x41494646 /* "AIFF" */ ++ #define IFF_ID_COMM 0x434f4d4d /* "COMM" */ ++ #define IFF_ID_SSND 0x53534e44 /* "SSND" */ ++ #define IFF_ID_MPEG 0x4d504547 /* "MPEG" */ ++ ++ ++ ++ ++ void InitSndFile(void); ++ void CloseSndFile(void); ++ void OpenSndFile(const char* lpszFileName,layer *info,int default_samp, int ++ default_chan); ++ unsigned long GetSndSamples(void); ++ int GetSndSampleRate(void); ++ int GetSndChannels(void); ++ int GetSndBitrate(void); ++ ++ ++ int get_audio(short buffer[2][1152],int stereo,layer* info); ++ ++ ++ ++ #ifdef LIBSNDFILE ++ /* INCLUDE the sound library header file */ ++ #ifdef _MSC_VER ++ /* one byte alignment for WIN32 platforms */ ++ #pragma pack(push,1) ++ #include "./libsndfile/src/sndfile.h" ++ #pragma pack(pop,1) ++ #else ++ #include "sndfile.h" ++ #endif ++ ++ ++ #else ++ /***************************************************************** ++ * LAME/ISO built in audio file I/O routines ++ *******************************************************************/ ++ #include "portableio.h" ++ #include "ieeefloat.h" ++ ++ ++ typedef struct blockAlign_struct { ++ unsigned long offset; ++ unsigned long blockSize; ++ } blockAlign; ++ ++ typedef struct IFF_AIFF_struct { ++ short numChannels; ++ unsigned long numSampleFrames; ++ short sampleSize; ++ double sampleRate; ++ unsigned long sampleType; ++ blockAlign blkAlgn; ++ } IFF_AIFF; ++ ++ extern int aiff_read_headers(FILE*, IFF_AIFF*); ++ extern int aiff_seek_to_sound_data(FILE*); ++ extern int aiff_write_headers(FILE*, IFF_AIFF*); ++ extern int parse_wavheader(void); ++ extern int parse_aiff(const char fn[],layer *info); ++ extern void aiff_check(const char*, IFF_AIFF*, int*); ++ ++ ++ #endif // ifndef _LIBSNDDLL ++ #endif // ifndef _BLADEDLL ++ #endif // ifndef GET_AUDIO_H_INCLUDED +diff -r -c -N encoder/globalflags.h lame3.50/globalflags.h +*** encoder/globalflags.h Wed Dec 31 17:00:00 1969 +--- lame3.50/globalflags.h Tue Oct 12 15:15:42 1999 +*************** +*** 0 **** +--- 1,39 ---- ++ #ifndef GLOBAL_FLAGS_H_INCLUDED ++ #define GLOBAL_FLAGS_H_INCLUDED ++ ++ typedef enum sound_file_format_e { ++ sf_unknown, sf_wave, sf_aiff, sf_mp3, sf_raw ++ } sound_file_format; ++ ++ /*********************************************************************** ++ * ++ * Global Variable External Declarations ++ * ++ ***********************************************************************/ ++ ++ extern int allow_diff_short; ++ extern int ATHonly; ++ extern int autoconvert; ++ extern int experimentalX; ++ extern int experimentalY; ++ extern int experimentalZ; ++ extern int force_ms; ++ extern int fast_mode; ++ extern long int frameNum; ++ extern int gtkflag; ++ extern int g_bWriteVbrTag; ++ extern int gpsycho; ++ extern int highq; ++ extern sound_file_format input_format; ++ extern int lame_nowrite; ++ extern double resample_ratio; /* if > 0, = input_samp/output_samp */ ++ extern int sfb21; ++ extern int silent; ++ extern int swapbytes; ++ extern long totalframes; /* frames: 0..totalframes-1 */ ++ extern int VBR; ++ extern int VBR_q; ++ extern int VBR_min_bitrate; ++ extern int VBR_max_bitrate; ++ extern int voice_mode; ++ #endif +diff -r -c -N encoder/gpkplotting.c lame3.50/gpkplotting.c +*** encoder/gpkplotting.c Wed Dec 31 17:00:00 1969 +--- lame3.50/gpkplotting.c Sun Oct 24 21:49:24 1999 +*************** +*** 0 **** +--- 1,282 ---- ++ #ifdef HAVEGTK ++ #include "gpkplotting.h" ++ #include "string.h" ++ ++ static gint num_plotwindows = 0; ++ static gint max_plotwindows = 10; ++ static GdkPixmap *pixmaps[10]; ++ static GtkWidget *pixmapboxes[10]; ++ ++ ++ ++ ++ /* compute a gdkcolor */ ++ void setcolor(GtkWidget *widget, GdkColor *color, gint red,gint green,gint blue) ++ { ++ ++ /* colors in GdkColor are taken from 0 to 65535, not 0 to 255. */ ++ color->red = red * (65535/255); ++ color->green = green * (65535/255); ++ color->blue = blue * (65535/255); ++ color->pixel = (gulong)(color->red*65536 + color->green*256 + color->blue); ++ /* find closest in colormap, if needed */ ++ gdk_color_alloc(gtk_widget_get_colormap(widget),color); ++ } ++ ++ ++ void gpk_redraw(GdkPixmap *pixmap, GtkWidget *pixmapbox) ++ { ++ /* redraw the entire pixmap */ ++ gdk_draw_pixmap(pixmapbox->window, ++ pixmapbox->style->fg_gc[GTK_WIDGET_STATE (pixmapbox)], ++ pixmap,0,0,0,0, ++ pixmapbox->allocation.width, ++ pixmapbox->allocation.height); ++ } ++ ++ ++ static GdkPixmap **findpixmap(GtkWidget *widget) ++ { ++ int i; ++ for (i=0; i=num_plotwindows) { ++ g_print("findpixmap(): bad argument widget \n"); ++ return NULL; ++ } ++ return &pixmaps[i]; ++ } ++ ++ void gpk_graph_draw(GtkWidget *widget, /* plot on this widged */ ++ int n, /* number of data points */ ++ gdouble *xcord, gdouble *ycord, /* data */ ++ gdouble xmn,gdouble ymn, /* coordinates of corners */ ++ gdouble xmx,gdouble ymx, ++ int clear, /* clear old plot first */ ++ char *title, /* add a title (only if clear=1) */ ++ GdkColor *color) ++ { ++ GdkPixmap **ppixmap; ++ GdkPoint *points; ++ int i; ++ gint16 width,height; ++ GdkFont *fixed_font; ++ GdkGC *gc; ++ ++ gc = gdk_gc_new(widget->window); ++ gdk_gc_set_foreground(gc, color); ++ ++ ++ ++ if ((ppixmap=findpixmap(widget))) { ++ width = widget->allocation.width; ++ height = widget->allocation.height; ++ ++ ++ if (clear) { ++ /* white background */ ++ gdk_draw_rectangle (*ppixmap, ++ widget->style->white_gc, ++ TRUE,0, 0,width,height); ++ /* title */ ++ #ifndef _WIN32 ++ fixed_font = gdk_font_load ("-misc-fixed-medium-r-*-*-*-100-*-*-*-*-*-*"); ++ #else ++ fixed_font = gdk_font_load ("-misc-fixed-large-r-*-*-*-100-*-*-*-*-*-*"); ++ #endif ++ ++ gdk_draw_text (*ppixmap,fixed_font, ++ widget->style->fg_gc[GTK_WIDGET_STATE (widget)], ++ 0,10,title,strlen(title)); ++ } ++ ++ ++ points = g_malloc(n*sizeof(GdkPoint)); ++ for (i=0; iwindow); ++ gdk_gc_set_foreground(gc, color); ++ ++ ++ if ((ppixmap=findpixmap(widget))) { ++ width = widget->allocation.width; ++ height = widget->allocation.height; ++ ++ ++ for (i=0; i<2 ; i++) { ++ points[i].x =.5+ ((xcord[i]-xmn)*(width-1)/(xmx-xmn)); ++ points[i].y =.5+ ((ycord[i]-ymx)*(height-1)/(ymn-ymx)); ++ } ++ width=points[1].x-points[0].x + 1; ++ height=points[1].y-points[0].y + 1; ++ gdk_draw_rectangle(*ppixmap,gc,TRUE, ++ points[0].x,points[0].y,width,height); ++ gpk_redraw(*ppixmap,widget); ++ } ++ gdk_gc_destroy(gc); ++ } ++ ++ ++ ++ void gpk_bargraph_draw(GtkWidget *widget, /* plot on this widged */ ++ int n, /* number of data points */ ++ gdouble *xcord, gdouble *ycord, /* data */ ++ gdouble xmn,gdouble ymn, /* coordinates of corners */ ++ gdouble xmx,gdouble ymx, ++ int clear, /* clear old plot first */ ++ char *title, /* add a title (only if clear=1) */ ++ int barwidth, /* bar width. 0=compute based on window size */ ++ GdkColor *color) ++ { ++ GdkPixmap **ppixmap; ++ GdkPoint points[2]; ++ int i; ++ gint16 width,height,x,y,barheight; ++ GdkFont *fixed_font; ++ GdkGC *gc; ++ ++ ++ gc = gdk_gc_new(widget->window); ++ gdk_gc_set_foreground(gc, color); ++ ++ ++ if ((ppixmap=findpixmap(widget))) { ++ width = widget->allocation.width; ++ height = widget->allocation.height; ++ ++ ++ if (clear) { ++ /* white background */ ++ gdk_draw_rectangle (*ppixmap, ++ widget->style->white_gc, ++ TRUE,0, 0,width,height); ++ /* title */ ++ #ifndef _WIN32 ++ fixed_font = gdk_font_load ("-misc-fixed-medium-r-*-*-*-100-*-*-*-*-*-*"); ++ #else ++ fixed_font = gdk_font_load ("-misc-fixed-large-r-*-*-*-100-*-*-*-*-*-*"); ++ #endif ++ ++ gdk_draw_text (*ppixmap,fixed_font, ++ widget->style->fg_gc[GTK_WIDGET_STATE (widget)], ++ 0,10,title,strlen(title)); ++ } ++ ++ ++ for (i=0; i 5 ? 5 : barwidth; ++ barwidth = barwidth < 1 ? 1 : barwidth; ++ barheight = height-1 - y; ++ /* gdk_draw_lines(*ppixmap,gc,points,2); */ ++ gdk_draw_rectangle(*ppixmap,gc,TRUE,x,y,barwidth,barheight); ++ ++ } ++ gpk_redraw(*ppixmap,widget); ++ } ++ gdk_gc_destroy(gc); ++ } ++ ++ ++ ++ ++ ++ /* Create a new backing pixmap of the appropriate size */ ++ static gint ++ configure_event (GtkWidget *widget, GdkEventConfigure *event, gpointer data) ++ { ++ GdkPixmap **ppixmap; ++ if ((ppixmap=findpixmap(widget))){ ++ if (*ppixmap) gdk_pixmap_unref(*ppixmap); ++ *ppixmap = gdk_pixmap_new(widget->window, ++ widget->allocation.width, ++ widget->allocation.height, ++ -1); ++ gdk_draw_rectangle (*ppixmap, ++ widget->style->white_gc, ++ TRUE, ++ 0, 0, ++ widget->allocation.width, ++ widget->allocation.height); ++ } ++ return TRUE; ++ } ++ ++ ++ ++ /* Redraw the screen from the backing pixmap */ ++ static gint ++ expose_event (GtkWidget *widget, GdkEventExpose *event, gpointer data) ++ { ++ GdkPixmap **ppixmap; ++ if ((ppixmap=findpixmap(widget))){ ++ gdk_draw_pixmap(widget->window, ++ widget->style->fg_gc[GTK_WIDGET_STATE (widget)], ++ *ppixmap, ++ event->area.x, event->area.y, ++ event->area.x, event->area.y, ++ event->area.width, event->area.height); ++ } ++ ++ return FALSE; ++ } ++ ++ ++ ++ ++ ++ GtkWidget *gpk_plot_new(int width, int height) ++ { ++ GtkWidget *pixmapbox; ++ ++ pixmapbox = gtk_drawing_area_new(); ++ gtk_drawing_area_size(GTK_DRAWING_AREA(pixmapbox),width,height); ++ gtk_signal_connect (GTK_OBJECT (pixmapbox), "expose_event", ++ (GtkSignalFunc) expose_event, NULL); ++ gtk_signal_connect (GTK_OBJECT(pixmapbox),"configure_event", ++ (GtkSignalFunc) configure_event, NULL); ++ gtk_widget_set_events (pixmapbox, GDK_EXPOSURE_MASK); ++ ++ if (num_plotwindows < max_plotwindows) { ++ pixmapboxes[num_plotwindows] = pixmapbox; ++ pixmaps[num_plotwindows] = NULL; ++ num_plotwindows ++; ++ } else { ++ g_print("gtk_plotarea_new(): exceeded maximum of 10 plotarea windows\n"); ++ } ++ ++ return pixmapbox; ++ } ++ ++ ++ #endif +diff -r -c -N encoder/gpkplotting.h lame3.50/gpkplotting.h +*** encoder/gpkplotting.h Wed Dec 31 17:00:00 1969 +--- lame3.50/gpkplotting.h Fri Sep 10 09:52:00 1999 +*************** +*** 0 **** +--- 1,27 ---- ++ #include ++ ++ /* allocate a graphing widget */ ++ GtkWidget *gpk_plot_new(int width,int height); ++ ++ /* graph a function in the graphing widged */ ++ void gpk_graph_draw(GtkWidget *widget, ++ int n, gdouble *xcord, gdouble *ycord, ++ gdouble xmn, gdouble ymn,gdouble xmx,gdouble ymx, ++ int clear, char * title,GdkColor *color); ++ ++ /* draw a rectangle in the graphing widget */ ++ void gpk_rectangle_draw(GtkWidget *widget, /* plot on this widged */ ++ gdouble xcord[2], gdouble ycord[2], /* corners */ ++ gdouble xmn,gdouble ymn, /* coordinates of corners */ ++ gdouble xmx,gdouble ymx, ++ GdkColor *color); /* color to use */ ++ ++ /* make a bar graph in the graphing widged */ ++ void gpk_bargraph_draw(GtkWidget *widget, ++ int n, gdouble *xcord, gdouble *ycord, ++ gdouble xmn, gdouble ymn,gdouble xmx,gdouble ymx, ++ int clear, char * title, int bwidth,GdkColor *color); ++ ++ /* set forground color */ ++ void setcolor(GtkWidget *widget, GdkColor *color,int red,int green,int blue); ++ +diff -r -c -N encoder/gtkanal.c lame3.50/gtkanal.c +*** encoder/gtkanal.c Wed Dec 31 17:00:00 1969 +--- lame3.50/gtkanal.c Wed Oct 20 21:29:46 1999 +*************** +*** 0 **** +--- 1,1213 ---- ++ #ifdef HAVEGTK ++ #include ++ #include ++ #include "gpkplotting.h" ++ #include "globalflags.h" ++ #include "util.h" ++ #include "encoder.h" ++ #include "psymodel.h" ++ #include "gtkanal.h" ++ #include "version.h" ++ #include "lame.h" ++ ++ ++ extern char inPath[MAX_NAME_SIZE]; ++ extern int makeframe(void); ++ ++ /* global variables for the state of the system */ ++ static gint idle_keepgoing; /* processing of frames is ON */ ++ static gint idle_count_max; /* number of frames to process before plotting */ ++ static gint idle_count; /* pause & plot when idle_count=idel_count_max */ ++ static gint idle_finish=0; /* process all frames, finish mp3 file and exit */ ++ static gint idle_back = 0; /* set when we are displaying the old data */ ++ static int mp3done = 0; /* last frame has been read */ ++ static GtkWidget *frameprogress; /* progress bar */ ++ static GtkWidget *framecounter; /* progress counter */ ++ ++ /* main window */ ++ GtkWidget *window; ++ /* Backing pixmap for drawing areas */ ++ GtkWidget *pcmbox; /* PCM data plotted here */ ++ GtkWidget *winbox; /* mpg123 synthesis data plotted here */ ++ GtkWidget *enerbox[2]; /* spectrum, gr=0,1 plotted here */ ++ GtkWidget *mdctbox[2]; /* mdct coefficients gr=0,1 plotted here */ ++ GtkWidget *headerbox; /* mpg123 header info shown here */ ++ ++ plotting_data *pinfo,*pplot; ++ plotting_data Pinfo[NUMPINFO]; ++ ++ struct gtkinfostruct { ++ int filetype; /* input file type 0=WAV, 1=MP3 */ ++ int msflag; /* toggle between L&R vs M&S PCM data display */ ++ int chflag; /* toggle between L & R channels */ ++ int kbflag; /* toggle between wave # and barks */ ++ int flag123; /* show mpg123 frame info, OR ISO encoder frame info */ ++ double avebits; /* running average bits per frame */ ++ int approxbits; /* (approx) bits per frame */ ++ int maxbits; /* max bits per frame used so far*/ ++ int totemph; /* total of frames with de-emphasis */ ++ int totms; /* total frames with ms_stereo */ ++ int totis; /* total frames with i_stereo */ ++ int totshort; /* total granules with short blocks */ ++ int totmix; /* total granules with mixed blocks */ ++ int pupdate; /* plot while processing, or only when needed */ ++ } gtkinfo; ++ ++ ++ ++ /********************************************************************** ++ * read one frame and encode it ++ **********************************************************************/ ++ int gtkmakeframe() ++ { ++ int iread; ++ static int mpglag; ++ static short int Buffer[2][1152]; ++ int ch,j,adj; ++ int mp3count,mp3out; ++ short mpg123pcm[2][1152]; ++ char mp3buffer[LAME_MAXMP3BUFFER]; ++ ++ iread=lame_readframe(Buffer); ++ /* even if iread=0, get_audio hit EOF and returned Buffer=all 0's. call ++ * makeframe() one last time to flush all buffers */ ++ if (input_format == sf_mp3) frameNum++; ++ else { ++ if (frameNum==0) { ++ mpglag=1; ++ lame_decode_init(); ++ } ++ mp3count=lame_encode(Buffer,mp3buffer); /* encode frame */ ++ mp3out=lame_decode(mp3buffer,mp3count,mpg123pcm); /* re-synthesis to pcm */ ++ if (mp3out>0) { ++ /* decoded output is for frame pinfo->frameNum123 */ ++ pinfo->frameNum123=pinfo->frameNum-mpglag; ++ adj = 1152-pinfo->framesize; ++ for ( ch = 0; ch < pinfo->stereo; ch++ ) { ++ for ( j = 0; j < DECDELAY-adj; j++ ) ++ pinfo->pcmdata2[ch][j] = pinfo->pcmdata2[ch][j+1152-adj]; ++ for ( j = 0; j < 1152-adj; j++ ) { ++ pinfo->pcmdata2[ch][j+DECDELAY-adj] = mpg123pcm[ch][j]; ++ } ++ } ++ }else{ ++ if (mpglag == MAXMPGLAG) { ++ fprintf(stderr,"READ_AHEAD set too low - not enough frame buffering.\n"); ++ fprintf(stderr,"MP3x display of input and output PCM data out of sync.\n"); ++ } ++ else mpglag++; ++ pinfo->frameNum123=-1; /* no frame output */ ++ } ++ } ++ return iread; ++ } ++ ++ ++ void plot_frame(void) ++ { ++ int i,j,n,ch,gr; ++ gdouble *xcord,*ycord; ++ gdouble xmx,xmn,ymx,ymn; ++ double *data,*data2,*data3; ++ char title2[80]; ++ char *title; ++ plotting_data *pplot1; ++ plotting_data *pplot2; ++ ++ double en,samp; ++ static int firstcall=1; ++ static GdkColor *barcolor,*color,*grcolor[2]; ++ static GdkColor yellow,gray,cyan,magenta,orange,pink,red,green,blue,black,oncolor,offcolor; ++ int blocktype[2][2]; ++ int headbits; ++ int mode_gr = 2; ++ ++ /* find the frame where mpg123 produced output coming from input frame ++ * pinfo. i.e.: out_frame + out_frame_lag = input_frame */ ++ for (i=1; i<=MAXMPGLAG; i++ ) { ++ if ((pplot-i)->frameNum123 == pplot->frameNum ) { ++ pplot2 = pplot-i; ++ break; ++ } ++ } ++ if (i > MAXMPGLAG) { ++ fprintf(stderr,"input/output pcm syncing problem. should not happen!\n"); ++ pplot2=pplot-1; ++ } ++ ++ ++ /* however, the PCM data is delayed by 528 samples in the encoder filterbanks. ++ * We added another 1152-528 delay to this so the PCM data is *exactly* one ++ * frame behind the header & MDCT information */ ++ pplot1 =pplot2 +1; /* back one frame for header info, MDCT */ ++ ++ /* allocate these GC's only once */ ++ if (firstcall) { ++ firstcall=0; ++ /* grcolor[0] = &magenta; */ ++ grcolor[0] = &blue; ++ grcolor[1] = &green; ++ barcolor = &gray; ++ ++ setcolor(headerbox,&oncolor,255,0,0); ++ setcolor(headerbox,&offcolor,175,175,175); ++ setcolor(pcmbox,&red,255,0,0); ++ setcolor(pcmbox,&pink,255,0,255); ++ setcolor(pcmbox,&magenta,255,0,100); ++ setcolor(pcmbox,&orange,255,127,0); ++ setcolor(pcmbox,&cyan,0,255,255); ++ setcolor(pcmbox,&green,0,255,0); ++ setcolor(pcmbox,&blue,0,0,255); ++ setcolor(pcmbox,&black,0,0,0); ++ setcolor(pcmbox,&gray,100,100,100); ++ setcolor(pcmbox,&yellow,255,255,0); ++ ++ } ++ ++ /******************************************************************* ++ * frame header info ++ *******************************************************************/ ++ if (pplot1->sampfreq) ++ samp=pplot1->sampfreq; ++ else samp=1; ++ ++ headbits = 32 + ((pplot1->stereo==2) ? 256 : 136); ++ gtkinfo.approxbits = (pplot1->bitrate*1000*1152.0/samp) - headbits; ++ /*font = gdk_font_load ("-misc-fixed-medium-r-*-*-*-100-*-*-*-*-*-*");*/ ++ sprintf(title2,"%3.1fkHz %ikbs ",samp/1000,pplot1->bitrate); ++ gtk_text_freeze (GTK_TEXT(headerbox)); ++ gtk_text_backward_delete(GTK_TEXT(headerbox), ++ gtk_text_get_length(GTK_TEXT(headerbox))); ++ gtk_text_set_point(GTK_TEXT(headerbox),0); ++ gtk_text_insert(GTK_TEXT(headerbox),NULL,&oncolor,NULL,title2, -1); ++ title = " mono "; ++ if (2==pplot1->stereo) title = pplot1->js ? " jstereo " : " stereo "; ++ gtk_text_insert (GTK_TEXT(headerbox), NULL, &oncolor, NULL,title, -1); ++ color = pplot1->ms_stereo ? &oncolor : &offcolor ; ++ gtk_text_insert (GTK_TEXT(headerbox), NULL, color, NULL,"ms_stereo ", -1); ++ color = pplot1->i_stereo ? &oncolor : &offcolor ; ++ gtk_text_insert (GTK_TEXT(headerbox), NULL, color, NULL,"i_stereo ", -1); ++ color = pplot1->emph ? &oncolor : &offcolor ; ++ gtk_text_insert (GTK_TEXT(headerbox), NULL, color, NULL,"emph ", -1); ++ color = pplot1->scfsi ? &oncolor : &offcolor ; ++ gtk_text_insert (GTK_TEXT(headerbox), NULL, color, NULL,"scfsi ", -1); ++ if (gtkinfo.filetype) ++ sprintf(title2," mdb=%i %i/NA",pplot1->maindata,pplot1->totbits); ++ else ++ sprintf(title2," mdb=%i %i/%i", ++ pplot1->maindata,pplot1->totbits,pplot->resvsize); ++ gtk_text_insert (GTK_TEXT(headerbox), NULL, &oncolor, NULL,title2, -1); ++ gtk_text_thaw (GTK_TEXT(headerbox)); ++ ++ ++ ch = gtkinfo.chflag; ++ /******************************************************************* ++ * block type ++ *******************************************************************/ ++ for (gr = 0 ; gr < mode_gr ; gr ++) ++ if (gtkinfo.flag123) ++ blocktype[gr][ch]=pplot1->mpg123blocktype[gr][ch]; ++ else blocktype[gr][ch]=pplot->blocktype[gr][ch]; ++ ++ ++ /******************************************************************* ++ * draw the PCM data * ++ *******************************************************************/ ++ n = 1600; /* PCM frame + FFT window: 224 + 1152 + 224 */ ++ xcord = g_malloc(n*sizeof(gdouble)); ++ ycord = g_malloc(n*sizeof(gdouble)); ++ ++ ++ if (gtkinfo.msflag) ++ title=ch ? "Side Channel" : "Mid Channel"; ++ else ++ title=ch ? "Right Channel" : "Left Channel"; ++ ++ ymn = -32767 ; ++ ymx = 32767; ++ xmn = 0; ++ xmx = 1600-1; ++ ++ /* 0 ... 224 draw in black, connecting to 224 pixel ++ * 1375 .. 1599 draw in black connecting to 1375 pixel ++ * 224 ... 1375 MP3 frame. draw in blue ++ */ ++ ++ /* draw the title */ ++ gpk_graph_draw(pcmbox,0,xcord,ycord,xmn,ymn,xmx,ymx,1,title, ++ &black); ++ ++ ++ /* draw some hash marks dividing the frames */ ++ ycord[0] = ymx*.8; ycord[1] = ymn*.8; ++ for (gr=0 ; gr<=2; gr++) { ++ xcord[0] = 223.5 + gr*576; xcord[1] = 223.5 +gr*576; ++ gpk_rectangle_draw(pcmbox,xcord,ycord,xmn,ymn,xmx,ymx,&yellow); ++ } ++ for (gr = 0 ; gr < mode_gr ; gr++) { ++ if (blocktype[gr][ch]==2) ++ for (i=1 ; i<=2; i++) { ++ xcord[0] = 223.5+gr*576 + i*192; ++ xcord[1] = 223.5+gr*576 + i*192; ++ gpk_rectangle_draw(pcmbox,xcord,ycord,xmn,ymn,xmx,ymx,&yellow); ++ } ++ } ++ /* bars representing FFT windows */ ++ xcord[0] = 0; ycord[0] = ymn+3000; ++ xcord[1] = 1024-1; ycord[1] = ymn+1000; ++ gpk_rectangle_draw(pcmbox,xcord,ycord,xmn,ymn,xmx,ymx,grcolor[0]); ++ xcord[0] = 576; ycord[0] = ymn+2000; ++ xcord[1] = 576+1024-1; ycord[1] = ymn; ++ gpk_rectangle_draw(pcmbox,xcord,ycord,xmn,ymn,xmx,ymx,grcolor[1]); ++ ++ ++ /* plot PCM data */ ++ for (i=0; ipcmdata[0][i]-pplot->pcmdata[1][i]) : ++ .5*(pplot->pcmdata[0][i]+pplot->pcmdata[1][i]); ++ else ++ ycord[i]=pplot->pcmdata[ch][i]; ++ } ++ ++ /* skip plot if we are doing an mp3 file */ ++ if (!gtkinfo.filetype) { ++ n = 224; /* number of points on end of blue part */ ++ /* data left of frame */ ++ gpk_graph_draw(pcmbox,n+1,xcord,ycord,xmn,ymn,xmx,ymx,0,title,&black); ++ /* data right of frame */ ++ gpk_graph_draw(pcmbox,n+1,&xcord[1152+n-1],&ycord[1152+n-1], ++ xmn,ymn,xmx,ymx,0,title,&black); ++ /* the actual frame */ ++ gpk_graph_draw(pcmbox,1152,&xcord[n],&ycord[n],xmn,ymn,xmx,ymx,0,title,&black); ++ } ++ ++ ++ /*******************************************************************/ ++ /* draw the PCM re-synthesis data */ ++ /*******************************************************************/ ++ n = 1152; ++ ++ sprintf(title2,"Re-synthesis ms_ratio=%3.2f %3.2f", ++ pplot->ms_ratio[0],pplot->ms_ratio[1]); ++ title=title2; ++ ++ ++ ymn = -32767 ; ++ ymx = 32767; ++ xmn = 0; ++ xmx = 1600-1; ++ gpk_graph_draw(winbox,0,xcord,ycord, ++ xmn,ymn,xmx,ymx,1,title,&black); ++ /* draw some hash marks dividing the frames */ ++ ycord[0] = ymx*.8; ycord[1] = ymn*.8; ++ for (gr=0 ; gr<=2; gr++) { ++ xcord[0] = 223.5 + gr*576; xcord[1] = 223.5 +gr*576; ++ gpk_rectangle_draw(winbox,xcord,ycord,xmn,ymn,xmx,ymx,&yellow); ++ } ++ for (gr = 0 ; gr < 2 ; gr++) { ++ if (blocktype[gr][ch]==2) ++ for (i=1 ; i<=2; i++) { ++ xcord[0] = 223.5+gr*576 + i*192; ++ xcord[1] = 223.5+gr*576 + i*192; ++ gpk_rectangle_draw(winbox,xcord,ycord,xmn,ymn,xmx,ymx,&yellow); ++ } ++ } ++ ++ ++ ++ n = 224; ++ for (j=1152-n,i=0; i<=n; i++,j++) { ++ xcord[i] = i; ++ if (gtkinfo.msflag) ++ ycord[i] = ch ? .5*(pplot1->pcmdata2[0][j]- ++ pplot1->pcmdata2[1][j]) : ++ .5*(pplot1->pcmdata2[0][j]+pplot1->pcmdata2[1][j]); ++ else ++ ycord[i]=pplot1->pcmdata2[ch][j]; ++ } ++ gpk_graph_draw(winbox,n+1,xcord,ycord, ++ xmn,ymn,xmx,ymx,0,title,&black); ++ ++ n = 1152; ++ for (i=0; ipcmdata2[0][i]-pplot2->pcmdata2[1][i]) : ++ .5*(pplot2->pcmdata2[0][i]+pplot2->pcmdata2[1][i]); ++ else ++ ycord[i]=pplot2->pcmdata2[ch][i]; ++ } ++ gpk_graph_draw(winbox,n,xcord,ycord, ++ xmn,ymn,xmx,ymx,0,title,&black); ++ ++ ++ ++ ++ ++ /*******************************************************************/ ++ /* draw the MDCT energy spectrum */ ++ /*******************************************************************/ ++ for (gr = 0 ; gr < mode_gr ; gr ++) { ++ int bits; ++ char *blockname=""; ++ char label[80],label2[80]; ++ switch (blocktype[gr][ch]) { ++ case 0: blockname = "normal"; break; ++ case 1: blockname = "start"; break; ++ case 2: blockname = "short"; break; ++ case 3: blockname = "end"; break; ++ } ++ strcpy(label,blockname); ++ if (pplot1->mixed[gr][ch]) strcat(label,"(mixed)"); ++ if (blocktype[gr][ch]==2) { ++ sprintf(label2, ++ "%i %i%i%i", ++ pplot1->scalefac_scale[gr][ch], ++ pplot1->sub_gain[gr][ch][0], ++ pplot1->sub_gain[gr][ch][1], ++ pplot1->sub_gain[gr][ch][2]); ++ strcat(label,label2); ++ }else{ ++ sprintf(label2,"%i",pplot1->scalefac_scale[gr][ch]); ++ strcat(label,label2); ++ } ++ ++ ++ ++ ++ n = 576; ++ if (gtkinfo.flag123) { ++ data = pplot1->mpg123xr[gr][0]; ++ data2 = pplot1->mpg123xr[gr][1]; ++ }else{ ++ data = pplot->xr[gr][0]; ++ data2 = pplot->xr[gr][1]; ++ } ++ ++ ++ ++ ymn=9e20; ++ ymx=-9e20; ++ for (i=0; i ymx) ? ycord[i] : ymx; ++ ymn=(ycord[i] < ymn) ? ycord[i] : ymn; ++ } ++ /* print the min/max ++ sprintf(title2,"MDCT%1i %5.2f %5.2f bits=%i",gr,ymn,ymx, ++ pplot1->mainbits[gr][ch]); ++ */ ++ if (gtkinfo.flag123) bits=pplot1->mainbits[gr][ch]; ++ else bits=pplot->LAMEmainbits[gr][ch]; ++ ++ ++ sprintf(title2,"MDCT%1i(%s) bits=%i q=%i ",gr,label,bits, ++ pplot1->qss[gr][ch]-210); ++ ++ xmn = 1; ++ xmx = n; ++ ymn=0; ++ ymx=11; ++ gpk_bargraph_draw(mdctbox[gr],n,xcord,ycord, ++ xmn,ymn,xmx,ymx,1,title2,0,barcolor); ++ } ++ ++ ++ ++ ++ /******************************************************************* ++ * draw the psy model energy spectrum (k space) ++ * l3psy.c computes pe, en, thm for THIS granule. ++ * To encode, ISO uses (bug): en,thm from last granule. ++ * pe from this granule ++ * gpsycho model corrects this ++ *******************************************************************/ ++ if (gtkinfo.msflag || gtkinfo.filetype) { ++ /* erase everything */ ++ gpk_bargraph_draw(enerbox[0],0,xcord,ycord, ++ xmn,ymn,xmx,ymx,1," ",0,barcolor); ++ gpk_bargraph_draw(enerbox[1],0,xcord,ycord, ++ xmn,ymn,xmx,ymx,1," ",0,barcolor); ++ } ++ else if (gtkinfo.kbflag){ ++ for (gr = 0 ; gr < mode_gr ; gr ++) { ++ n = HBLKSIZE; /* only show half the spectrum */ ++ ++ data = &pplot->energy[gr][ch][0]; ++ ++ ymn=9e20; ++ ymx=-9e20; ++ for (i=0; i ymx) ? ycord[i] : ymx; ++ ymn=(ycord[i] < ymn) ? ycord[i] : ymn; ++ } ++ for (en=0 , j=0; jenergy[gr][ch][j]; ++ ++ sprintf(title2,"FFT%1i pe=%4.1fK en=%5.2e ",gr, ++ pplot->pe[gr][ch]/1000,en); ++ ++ ymn = 3; ++ ymx = 15; ++ xmn = 1; ++ xmx = n; ++ gpk_bargraph_draw(enerbox[gr],n,xcord,ycord, ++ xmn,ymn,xmx,ymx,1,title2,0,barcolor); ++ ++ } ++ }else{ ++ /******************************************************************* ++ * draw the psy model energy spectrum (scalefactor bands) ++ * ISO model (because of bugs) uses: ++ * pe,en,thm,energy delayed 1 granule ++ *******************************************************************/ ++ for (gr = 0 ; gr < mode_gr ; gr ++) { ++ ++ if (blocktype[gr][ch]==2) { ++ n = 3*SBMAX_s; ++ data = &pplot->en_s[gr][ch][0]; ++ data2 = &pplot->thr_s[gr][ch][0]; ++ data3 = &pplot->xfsf_s[gr][ch][0]; ++ } else { ++ n = SBMAX_l; ++ data = &pplot->en[gr][ch][0]; ++ data2 = &pplot->thr[gr][ch][0]; ++ data3 = &pplot->xfsf[gr][ch][0]; ++ } ++ ymn=9e20; ++ ymx=-9e20; ++ for (i=0; i ymx) ? ycord[i] : ymx; ++ ymn=(ycord[i] < ymn) ? ycord[i] : ymn; ++ } ++ ++ ++ ++ /* en = max energy difference amoung the 3 short FFTs for this granule */ ++ en = pplot->ers[gr][ch]; ++ sprintf(title2,"FFT%1i pe=%4.1fK/%3.1f n=%i/%3.1f/%3.1f/%3.1f",gr, ++ pplot->pe[gr][ch]/1000,en,pplot->over[gr][ch], ++ pplot->max_noise[gr][ch], ++ pplot->over_noise[gr][ch], ++ pplot->tot_noise[gr][ch]); ++ ++ ++ ymn = 3; ++ ymx = 15; ++ xmn = 1; ++ xmx = n+1; /* a little extra because of the bar thickness */ ++ gpk_bargraph_draw(enerbox[gr],n,xcord,ycord, ++ xmn,ymn,xmx,ymx,1,title2,0,barcolor); ++ ++ ++ for (i=0; i ymx) ? ycord[i] : ymx; ++ ymn=(ycord[i] < ymn) ? ycord[i] : ymn; ++ } ++ gpk_bargraph_draw(enerbox[gr],n,xcord,ycord, ++ xmn,ymn,xmx,ymx,0,title2,3,&red); ++ ++ ++ for (i=0; i ymx) ? ycord[i] : ymx; ++ ymn=(ycord[i] < ymn) ? ycord[i] : ymn; ++ } ++ gpk_bargraph_draw(enerbox[gr],n,xcord,ycord, ++ xmn,ymn,xmx,ymx,0,title2,3,grcolor[gr]); ++ } ++ } ++ } ++ ++ ++ ++ static void update_progress(void) ++ { ++ char label[80]; ++ sprintf(label,"Frame:%4i/%4i %6.2fs", ++ pplot->frameNum,(int)totalframes-1, pplot->frametime); ++ gtk_progress_set_value (GTK_PROGRESS (frameprogress), (gdouble) pplot->frameNum); ++ gtk_label_set_text(GTK_LABEL(framecounter),label); ++ } ++ ++ ++ ++ static void analyze(void) ++ { ++ if ( idle_keepgoing) { ++ idle_count = 0; ++ idle_count_max=0; ++ idle_keepgoing=0; ++ idle_finish=0; ++ } ++ plot_frame(); ++ update_progress(); ++ } ++ ++ static void plotclick( GtkWidget *widget, gpointer data ) ++ { analyze(); } ++ ++ ++ ++ ++ static int frameadv1( GtkWidget *widget, gpointer data ) ++ { ++ int i; ++ if (idle_keepgoing ){ ++ if (idle_back) { ++ /* frame displayed is the old frame. to advance, just swap in new frame */ ++ idle_back--; ++ pplot = &Pinfo[READ_AHEAD+idle_back]; ++ }else{ ++ /* advance the frame by reading in a new frame */ ++ pplot = &Pinfo[READ_AHEAD]; ++ if (mp3done) { ++ /* dont try to read any more frames, and quit if "finish MP3" was selected */ ++ if (idle_finish) gtk_main_quit(); ++ idle_count_max=0; ++ } else { ++ /* read in the next frame */ ++ for (i=NUMPINFO-1 ; i>0 ; i--) ++ memcpy(&Pinfo[i],&Pinfo[i-1],sizeof(plotting_data)); ++ pinfo = &Pinfo[0]; ++ pinfo->num_samples = gtkmakeframe(); ++ ++ if (pinfo->sampfreq) ++ pinfo->frametime = (pinfo->frameNum)*1152.0/pinfo->sampfreq; ++ else pinfo->frametime=0; ++ ++ /* eof? */ ++ if (!pinfo->num_samples) { ++ if (idle_finish) gtk_main_quit(); ++ } ++ ++ pinfo->totbits = 0; ++ { int gr,ch; ++ for (gr = 0 ; gr < 2 ; gr ++) ++ for (ch = 0 ; ch < 2 ; ch ++) { ++ gtkinfo.totshort += (pinfo->mpg123blocktype[gr][ch]==2); ++ gtkinfo.totmix += !(pinfo->mixed[gr][ch]==0); ++ pinfo->totbits += pinfo->mainbits[gr][ch]; ++ } ++ } ++ if (pinfo->frameNum > 0) /* start averaging at second frame */ ++ gtkinfo.avebits = (gtkinfo.avebits*((pinfo->frameNum)-1) ++ + pinfo->totbits ) /(pinfo->frameNum); ++ ++ gtkinfo.maxbits=MAX(gtkinfo.maxbits,pinfo->totbits); ++ gtkinfo.totemph += !(pinfo->emph==0); ++ gtkinfo.totms += !(pinfo->ms_stereo==0); ++ gtkinfo.totis += !(pinfo->i_stereo==0); ++ ++ if (pplot->frameNum >= totalframes-1) mp3done=1; ++ } ++ } ++ ++ idle_count++; ++ if (gtkinfo.pupdate) plot_frame(); ++ update_progress(); ++ if ((idle_count>=idle_count_max) && (! idle_finish)) analyze(); ++ } ++ return 1; ++ } ++ ++ ++ static void frameadv( GtkWidget *widget, gpointer data ) ++ { ++ int adv; ++ ++ if (!strcmp((char *) data,"-1")) { ++ /* ignore if we've already gone back as far as possible */ ++ if (pplot->frameNum==0 || (idle_back==NUMBACK)) return; ++ idle_back++; ++ pplot = &Pinfo[READ_AHEAD+idle_back]; ++ analyze(); ++ return; ++ } ++ ++ ++ adv = 1; ++ if (!strcmp((char *) data,"1")) adv = 1; ++ if (!strcmp((char *) data,"10")) adv = 10; ++ if (!strcmp((char *) data,"100")) adv = 100; ++ if (!strcmp((char *) data,"finish")) idle_finish = 1; ++ ++ ++ if (idle_keepgoing) { ++ /* already running - que up additional frame advance requests */ ++ idle_count_max += adv; ++ } ++ else { ++ /* turn on idleing */ ++ idle_count_max = adv; ++ idle_count = 0; ++ idle_keepgoing = 1; ++ } ++ } ++ ++ ++ ++ ++ /* another callback */ ++ static void delete_event( GtkWidget *widget, ++ GdkEvent *event, ++ gpointer data ) ++ { ++ gtk_main_quit (); ++ } ++ ++ ++ ++ ++ ++ ++ ++ static void channel_option (GtkWidget *widget, gpointer data) ++ { ++ int option; ++ option = (int) data; ++ switch (option) { ++ case 1: ++ gtkinfo.msflag=0; ++ gtkinfo.chflag=0; ++ break; ++ case 2: ++ gtkinfo.msflag=0; ++ gtkinfo.chflag=1; ++ break; ++ case 3: ++ gtkinfo.msflag=1; ++ gtkinfo.chflag=0; ++ break; ++ case 4: ++ gtkinfo.msflag=1; ++ gtkinfo.chflag=1; ++ } ++ analyze(); ++ } ++ static void spec_option (GtkWidget *widget, gpointer data) ++ { ++ int option; ++ option = (int) data; ++ switch (option) { ++ case 1: ++ gtkinfo.kbflag=0; ++ break; ++ case 2: ++ gtkinfo.kbflag=1; ++ break; ++ case 3: ++ gtkinfo.flag123=0; ++ break; ++ case 4: ++ gtkinfo.flag123=1; ++ break; ++ case 5: ++ gtkinfo.pupdate=1; ++ break; ++ case 6: ++ gtkinfo.pupdate=0; ++ break; ++ } ++ analyze(); ++ } ++ ++ ++ ++ static void text_window (GtkWidget *widget, gpointer data) ++ { ++ int option; ++ GtkWidget *hbox,*vbox,*button,*box; ++ GtkWidget *textwindow,*vscrollbar; ++ char text[80]; ++ ++ option = (int) data; ++ ++ textwindow = gtk_window_new(GTK_WINDOW_DIALOG); ++ gtk_signal_connect_object (GTK_OBJECT (window), "delete_event", ++ GTK_SIGNAL_FUNC(gtk_widget_destroy), ++ GTK_OBJECT (textwindow)); ++ ++ gtk_container_set_border_width (GTK_CONTAINER (textwindow), 0); ++ vbox = gtk_vbox_new(FALSE,0); ++ hbox = gtk_hbox_new(FALSE,0); ++ ++ button = gtk_button_new_with_label ("close"); ++ gtk_signal_connect_object (GTK_OBJECT (button), "clicked", ++ GTK_SIGNAL_FUNC(gtk_widget_destroy), ++ GTK_OBJECT (textwindow)); ++ ++ box = gtk_text_new (NULL, NULL); ++ gtk_text_set_editable (GTK_TEXT (box), FALSE); ++ vscrollbar = gtk_vscrollbar_new (GTK_TEXT(box)->vadj); ++ ++ ++ switch (option) { ++ case 0: ++ gtk_window_set_title (GTK_WINDOW (textwindow), "Documentation"); ++ gtk_widget_set_usize(box,450,500); ++ gtk_text_set_word_wrap(GTK_TEXT(box),TRUE); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL, ++ "Frame header information: "\ ++ "First the bitrate, sampling frequency and mono, stereo or jstereo "\ ++ "indicators are displayed . If the bitstream is jstereo, then mid/side "\ ++ "stereo or intensity stereo may be on (indicated in red). If "\ ++ "de-emphasis is used, this is also indicated in red. The mdb value is "\ ++ "main_data_begin. The encoded data starts this many bytes *before* the "\ ++ "frame header. A large value of mdb means the bitstream has saved some "\ ++ "bits into the reservoir, which it may allocate for some future frame. "\ ++ "The two numbers after mdb are the size (in bits) used to encode the "\ ++ "MDCT coefficients for this frame, followed byt the size of the bit "\ ++ "resevoir before encoding this frame. The maximum frame size and a "\ ++ "running average are given in the Stats pull down menu. A large "\ ++ "maximum frame size indicates the bitstream has made use of the bit "\ ++ "reservoir. \n\n",-1); ++ ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL, ++ "PCM data (top graph): "\ ++ "The PCM data is plotted in black. The layer3 frame is divided into 2 "\ ++ "granules of 576 samples (marked with yellow vertical lines). In the "\ ++ "case of normal, start and stop blocks, the MDCT coefficients for each "\ ++ "granule are computed using a 1152 sample window centered over the "\ ++ "granule. In the case of short blocks, the granule is further divided "\ ++ "into 3 blocks of 192 samples (also marked with yellow vertical lines)."\ ++ "The MDCT coefficients for these blocks are computed using 384 sample "\ ++ "windows centered over the 192 sample window. (This info not available "\ ++ "when analyzing .mp3 files.) For the psycho-acoustic model, a windowed "\ ++ "FFT is computed for each granule. The range of these windows "\ ++ "is denoted by the blue and green bars.\n\n",-1); ++ ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL, ++ "PCM re-synthesis data (second graph): "\ ++ "Same as the PCM window described above. The data displayed is the "\ ++ "result of encoding and then decoding the original sample. \n\n",-1); ++ ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL, ++ "MDCT windows: "\ ++ "Shows the energy in the MDCT spectrum for granule 0 (left window) "\ ++ "and granule 1 (right window). The text also shows the blocktype "\ ++ "used, the number of bits used to encode the coefficients and the "\ ++ "number of extra bits allocated from the reservoir. The MDCT pull down "\ ++ "window will toggle between the original unquantized MDCT coefficients "\ ++ "and the compressed (quantized) coefficients.\n\n",-1); ++ ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL, ++ "FFT window: "\ ++ "The gray bars show the energy in the FFT spectrum used by the "\ ++ "psycho-acoustic model. Granule 0 is in the left window, granule 1 in "\ ++ "the right window. The green and blue bars show how much distortion is "\ ++ "allowable, as computed by the psycho-acoustic model. The red bars show "\ ++ "the actual distortion after encoding. There is one FFT for each "\ ++ "granule, computed with a 1024 Hann window centered over the "\ ++ "appropriate granule. (the range of this 1024 sample window is shown "\ ++ "by the blue and green bars in the PCM data window). The Spectrum pull "\ ++ "down window will toggle between showing the energy in equally spaced "\ ++ "frequency domain and the scale factor bands used by layer3. Finally, "\ ++ "the perceptual entropy, total energy and number of scalefactor bands "\ ++ "with audible distortion is shown. (This info not available when "\ ++ "analyzing .mp3 files.)",-1); ++ ++ break; ++ case 1: ++ /* Set the about box information */ ++ gtk_window_set_title (GTK_WINDOW (textwindow), "About"); ++ gtk_widget_set_usize(box,350,260); ++ ++ sprintf(text,"LAME version %s \nwww.sulaco.org/mp3\n\n",get_lame_version()); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ ++ sprintf(text,"psycho-acoustic model: GPSYCHO version %s\n",get_psy_version()); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ ++ sprintf(text,"frame analyzer: MP3x version %s\n\n",get_mp3x_version()); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL, ++ "decoder: mpg123/mpglib .59q \nMichael Hipp (www.mpg123.de)\n\n",-1); ++ ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL, ++ "Encoder, decoder & psy-models based on ISO\ndemonstration source. ",-1); ++ break; ++ ++ case 2: ++ gtk_window_set_title (GTK_WINDOW (textwindow), "Statistics"); ++ gtk_widget_set_usize(box,350,260); ++ sprintf(text,"frames processed so far: %i \n",Pinfo[0].frameNum+1); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ sprintf(text,"granules processed so far: %i \n\n",4*(Pinfo[0].frameNum+1)); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ sprintf(text,"mean bits/frame (approximate): %i\n", ++ gtkinfo.approxbits); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ sprintf(text,"mean bits/frame (from LAME): %i\n", ++ 4*Pinfo[0].mean_bits); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ sprintf(text,"bitsize of largest frame: %i \n",gtkinfo.maxbits); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ sprintf(text,"average bits/frame: %3.1f \n\n",gtkinfo.avebits); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ sprintf(text,"ms_stereo frames: %i \n",gtkinfo.totms); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ sprintf(text,"i_stereo frames: %i \n",gtkinfo.totis); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ sprintf(text,"de-emphasis frames: %i \n",gtkinfo.totemph); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ sprintf(text,"short block granules: %i \n",gtkinfo.totshort); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ sprintf(text,"mixed block granules: %i \n",gtkinfo.totmix); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ break; ++ } ++ ++ ++ ++ gtk_widget_show (vscrollbar); ++ gtk_widget_show (box); ++ gtk_widget_show (vbox); ++ gtk_widget_show (hbox); ++ gtk_widget_show (button); ++ ++ gtk_box_pack_start (GTK_BOX(hbox), box, FALSE, TRUE, 0); ++ gtk_box_pack_start(GTK_BOX(hbox), vscrollbar, FALSE, FALSE, 0); ++ gtk_box_pack_start (GTK_BOX (vbox), hbox, FALSE, TRUE, 0); ++ gtk_box_pack_end (GTK_BOX (vbox), button, FALSE, TRUE, 0); ++ gtk_container_add (GTK_CONTAINER (textwindow), vbox); ++ gtk_widget_show(textwindow); ++ ++ } ++ ++ ++ ++ ++ /* #include */ ++ ++ ++ /* This is the GtkItemFactoryEntry structure used to generate new menus. ++ Item 1: The menu path. The letter after the underscore indicates an ++ accelerator key once the menu is open. ++ Item 2: The accelerator key for the entry ++ Item 3: The callback function. ++ Item 4: The callback action. This changes the parameters with ++ which the function is called. The default is 0. ++ Item 5: The item type, used to define what kind of an item it is. ++ Here are the possible values: ++ ++ NULL -> "" ++ "" -> "" ++ "" -> create a title item ++ "<Item>" -> create a simple item ++ "<CheckItem>" -> create a check item ++ "<ToggleItem>" -> create a toggle item ++ "<RadioItem>" -> create a radio item ++ <path> -> path of a radio item to link against ++ "<Separator>" -> create a separator ++ "<Branch>" -> create an item to hold sub items ++ "<LastBranch>" -> create a right justified branch ++ */ ++ ++ static GtkItemFactoryEntry menu_items[] = { ++ {"/_File", NULL, NULL, 0, "<Branch>"}, ++ /* ++ {"/File/_New", "<control>N", print_hello, 0, NULL}, ++ {"/File/_Open", "<control>O", print_hello, 0, NULL}, ++ {"/File/_Save", "<control>S", print_hello, 0, NULL}, ++ {"/File/Save _As", NULL, NULL, 0, NULL}, ++ {"/File/sep1", NULL, NULL, 0, "<Separator>"}, ++ {"/File/Quit", "<control>Q", gtk_main_quit, 0, NULL}, ++ */ ++ {"/File/_Quit", "<control>Q", delete_event, 0, NULL}, ++ ++ {"/_Plotting", NULL, NULL, 0, "<Branch>"}, ++ {"/Plotting/_While advancing" , NULL, spec_option, 5, NULL}, ++ {"/Plotting/_After advancing", NULL, spec_option, 6, NULL}, ++ ++ {"/_Channel", NULL, NULL, 0, "<Branch>"}, ++ {"/Channel/show _Left" , NULL, channel_option, 1, NULL}, ++ {"/Channel/show _Right", NULL, channel_option, 2, NULL}, ++ {"/Channel/show _Mid" , NULL, channel_option, 3, NULL}, ++ {"/Channel/show _Side", NULL, channel_option, 4, NULL}, ++ ++ {"/_Spectrum", NULL, NULL, 0, "<Branch>"}, ++ {"/Spectrum/_Scalefactor bands", NULL, spec_option, 1, NULL}, ++ {"/Spectrum/_Wave number", NULL, spec_option, 2, NULL}, ++ ++ {"/_MDCT", NULL, NULL, 0, "<Branch>"}, ++ {"/MDCT/_Original", NULL, spec_option, 3, NULL}, ++ {"/MDCT/_Compressed", NULL, spec_option, 4, NULL}, ++ ++ {"/_Stats", NULL, NULL, 0, "<Branch>"}, ++ {"/Stats/_Show", NULL, text_window, 2, NULL}, ++ ++ {"/_Help", NULL, NULL, 0, "<LastBranch>"}, ++ {"/_Help/_Documentation", NULL, text_window, 0, NULL}, ++ {"/_Help/_About", NULL, text_window, 1, NULL}, ++ }; ++ ++ ++ static void get_main_menu(GtkWidget *window, GtkWidget ** menubar) { ++ int nmenu_items = sizeof(menu_items) / sizeof(menu_items[0]); ++ GtkItemFactory *item_factory; ++ GtkAccelGroup *accel_group; ++ ++ accel_group = gtk_accel_group_new(); ++ ++ /* This function initializes the item factory. ++ Param 1: The type of menu - can be GTK_TYPE_MENU_BAR, GTK_TYPE_MENU, ++ or GTK_TYPE_OPTION_MENU. ++ Param 2: The path of the menu. ++ Param 3: A pointer to a gtk_accel_group. The item factory sets up ++ the accelerator table while generating menus. ++ */ ++ ++ item_factory = gtk_item_factory_new(GTK_TYPE_MENU_BAR, "<main>", ++ accel_group); ++ ++ /* This function generates the menu items. Pass the item factory, ++ the number of items in the array, the array itself, and any ++ callback data for the the menu items. */ ++ gtk_item_factory_create_items(item_factory, nmenu_items, menu_items, NULL); ++ ++ /* Attach the new accelerator group to the window. */ ++ gtk_accel_group_attach (accel_group, GTK_OBJECT (window)); ++ ++ if (menubar) ++ /* Finally, return the actual menu bar created by the item factory. */ ++ *menubar = gtk_item_factory_get_widget(item_factory, "<main>"); ++ } ++ ++ ++ ++ ++ int gtkcontrol(void) ++ { ++ /* GtkWidget is the storage type for widgets */ ++ GtkWidget *button; ++ GtkAdjustment *adj; ++ GtkWidget *mbox; /* main box */ ++ GtkWidget *box1; /* frame control buttons go */ ++ GtkWidget *box2; /* frame counters */ ++ GtkWidget *box3; /* frame header info */ ++ GtkWidget *table; /* table for all the plotting areas */ ++ GtkWidget *menubar; ++ ++ gint tableops,graphx,graphy; ++ char frameinfo[80]; ++ ++ graphx = 500; /* minimum allowed size of pixmap */ ++ graphy = 100; ++ ++ ++ /* set some global defaults/variables */ ++ gtkinfo.filetype = (input_format == sf_mp3); ++ gtkinfo.msflag=0; ++ gtkinfo.chflag=0; ++ gtkinfo.kbflag=0; ++ gtkinfo.flag123 = (input_format == sf_mp3); /* MP3 file=use mpg123 output */ ++ gtkinfo.pupdate=0; ++ gtkinfo.avebits = 0; ++ gtkinfo.maxbits = 0; ++ gtkinfo.approxbits = 0; ++ gtkinfo.totemph = 0; ++ gtkinfo.totms = 0; ++ gtkinfo.totis = 0; ++ gtkinfo.totshort = 0; ++ gtkinfo.totmix = 0; ++ ++ totalframes=Max(totalframes,READ_AHEAD+1); ++ ++ memset((char *) Pinfo, 0, sizeof(Pinfo)); ++ pplot = &Pinfo[READ_AHEAD]; ++ ++ strcpy(frameinfo,"MP3x: "); ++ strncat(frameinfo,inPath,70); ++ ++ window = gtk_window_new (GTK_WINDOW_TOPLEVEL); ++ gtk_window_set_title (GTK_WINDOW (window), frameinfo); ++ gtk_signal_connect (GTK_OBJECT (window), "delete_event", ++ GTK_SIGNAL_FUNC (delete_event), NULL); ++ gtk_container_set_border_width (GTK_CONTAINER (window), 0); ++ ++ ++ mbox = gtk_vbox_new(FALSE, 0); ++ ++ ++ /* layout of mbox */ ++ box1 = gtk_hbox_new(FALSE, 0); ++ box2 = gtk_hbox_new(FALSE, 0); ++ box3 = gtk_hbox_new(FALSE, 0); ++ table = gtk_table_new (4, 2, FALSE); ++ tableops = GTK_FILL | GTK_EXPAND | GTK_SHRINK; ++ get_main_menu(window, &menubar); ++ ++ gtk_box_pack_start(GTK_BOX(mbox), menubar, FALSE, TRUE, 0); ++ gtk_box_pack_end (GTK_BOX (mbox), box1, FALSE, TRUE, 0); ++ gtk_box_pack_end(GTK_BOX (mbox),box2, FALSE, TRUE, 0); ++ gtk_box_pack_start(GTK_BOX (mbox),box3, FALSE, TRUE, 0); ++ gtk_box_pack_start (GTK_BOX (mbox), table, TRUE, TRUE, 0); ++ gtk_container_add (GTK_CONTAINER (window), mbox); ++ ++ ++ /*********************************************************************/ ++ /* stuff in box3 frame header info */ ++ /*********************************************************************/ ++ /* ++ headerbox = gtk_label_new(" "); ++ gtk_label_set_justify(GTK_LABEL(headerbox),GTK_JUSTIFY_LEFT); ++ */ ++ headerbox = gtk_text_new (NULL, NULL); ++ gtk_text_set_editable (GTK_TEXT (headerbox), FALSE); ++ gtk_widget_set_usize(headerbox,200,20); ++ gtk_widget_show (headerbox); ++ gtk_box_pack_start(GTK_BOX (box3),headerbox, TRUE, TRUE, 0); ++ ++ ++ ++ /*********************************************************************/ ++ /* stuff in box2 frame counters */ ++ /*********************************************************************/ ++ framecounter = gtk_label_new(""); ++ gtk_widget_show(framecounter); ++ gtk_box_pack_start(GTK_BOX (box2),framecounter, FALSE, TRUE, 0); ++ ++ adj = (GtkAdjustment *) gtk_adjustment_new (0, 0,(gint) totalframes-1, 0, 0, 0); ++ frameprogress = gtk_progress_bar_new_with_adjustment (adj); ++ /* Set the format of the string that can be displayed in the ++ * trough of the progress bar: ++ * %p - percentage ++ * %v - value ++ * %l - lower range value ++ * %u - upper range value */ ++ gtk_progress_set_format_string (GTK_PROGRESS (frameprogress), ++ "%p%%"); ++ gtk_progress_set_value (GTK_PROGRESS (frameprogress), (gdouble) 0); ++ gtk_progress_set_show_text (GTK_PROGRESS (frameprogress),TRUE); ++ gtk_widget_show (frameprogress); ++ gtk_box_pack_end (GTK_BOX (box2), frameprogress, FALSE, TRUE, 0); ++ ++ ++ ++ /*********************************************************************/ ++ /* stuff in box1 buttons along bottom */ ++ /*********************************************************************/ ++ button = gtk_button_new_with_label ("-1"); ++ gtk_signal_connect (GTK_OBJECT (button), "clicked", ++ GTK_SIGNAL_FUNC (frameadv), (gpointer) "-1"); ++ gtk_box_pack_start(GTK_BOX(box1), button, TRUE, TRUE, 0); ++ gtk_widget_show(button); ++ ++ button = gtk_button_new_with_label ("+1"); ++ gtk_signal_connect (GTK_OBJECT (button), "clicked", ++ GTK_SIGNAL_FUNC (frameadv), (gpointer) "1"); ++ gtk_box_pack_start(GTK_BOX(box1), button, TRUE, TRUE, 0); ++ gtk_widget_show(button); ++ ++ button = gtk_button_new_with_label ("+10"); ++ gtk_signal_connect (GTK_OBJECT (button), "clicked", ++ GTK_SIGNAL_FUNC (frameadv), (gpointer) "10"); ++ gtk_box_pack_start(GTK_BOX(box1), button, TRUE, TRUE, 0); ++ gtk_widget_show(button); ++ ++ button = gtk_button_new_with_label ("+100"); ++ gtk_signal_connect (GTK_OBJECT (button), "clicked", ++ GTK_SIGNAL_FUNC (frameadv), (gpointer) "100"); ++ gtk_box_pack_start(GTK_BOX(box1), button, TRUE, TRUE, 0); ++ gtk_widget_show(button); ++ ++ button = gtk_button_new_with_label ("finish mp3"); ++ gtk_signal_connect (GTK_OBJECT (button), "clicked", ++ GTK_SIGNAL_FUNC (frameadv), (gpointer) "finish"); ++ gtk_box_pack_start(GTK_BOX(box1), button, TRUE, TRUE, 0); ++ gtk_widget_show(button); ++ ++ button = gtk_button_new_with_label ("stop/plot"); ++ gtk_signal_connect (GTK_OBJECT (button), "clicked", ++ GTK_SIGNAL_FUNC (plotclick), NULL); ++ gtk_box_pack_start(GTK_BOX(box1), button, TRUE, TRUE, 0); ++ gtk_widget_show(button); ++ ++ ++ /*********************************************************************/ ++ /* stuff in table. all the plotting windows */ ++ /*********************************************************************/ ++ pcmbox = gpk_plot_new(graphx,graphy); ++ gtk_table_attach (GTK_TABLE(table),pcmbox,0,2,0,1,tableops,tableops,2,2 ); ++ gtk_widget_show (pcmbox); ++ ++ winbox = gpk_plot_new(graphy,graphy); ++ gtk_table_attach(GTK_TABLE(table),winbox,0,2,1,2,tableops,tableops,2,2); ++ gtk_widget_show (winbox); ++ ++ ++ mdctbox[0] = gpk_plot_new(graphy,graphy); ++ gtk_table_attach(GTK_TABLE(table),mdctbox[0],0,1,2,3,tableops,tableops,2,2); ++ gtk_widget_show (mdctbox[0]); ++ ++ mdctbox[1] = gpk_plot_new(graphy,graphy); ++ gtk_table_attach (GTK_TABLE(table),mdctbox[1],1,2,2,3,tableops,tableops,2,2); ++ gtk_widget_show (mdctbox[1]); ++ ++ enerbox[0] = gpk_plot_new(graphy,graphy); ++ gtk_table_attach(GTK_TABLE(table),enerbox[0],0,1,3,4,tableops,tableops,2,2); ++ gtk_widget_show (enerbox[0]); ++ ++ enerbox[1] = gpk_plot_new(graphy,graphy); ++ gtk_table_attach (GTK_TABLE(table),enerbox[1],1,2,3,4,tableops,tableops,2,2); ++ gtk_widget_show (enerbox[1]); ++ ++ ++ ++ ++ gtk_idle_add((GtkFunction) frameadv1, NULL); ++ gtk_widget_show(menubar); ++ gtk_widget_show(box2); ++ gtk_widget_show(box3); ++ gtk_widget_show(table); ++ gtk_widget_show(box1); ++ gtk_widget_show (mbox); ++ gtk_widget_show (window); /* show smallest allowed window */ ++ ++ /* make window bigger. */ ++ /* now the user will be able to shrink it, if desired */ ++ /* gtk_widget_set_usize(mbox,500,500); */ ++ /* gtk_widget_show (window); */ /* show smallest allowed window */ ++ ++ ++ ++ idle_keepgoing=1; /* processing of frames is ON */ ++ idle_count_max=READ_AHEAD+1; /* number of frames to process before plotting */ ++ idle_count=0; /* pause & plot when idle_count=idle_count_max */ ++ ++ ++ gtk_main (); ++ if (!mp3done) exit(2); ++ return(0); ++ } ++ ++ #endif ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ +diff -r -c -N encoder/gtkanal.h lame3.50/gtkanal.h +*** encoder/gtkanal.h Wed Dec 31 17:00:00 1969 +--- lame3.50/gtkanal.h Thu Oct 14 22:38:16 1999 +*************** +*** 0 **** +--- 1,90 ---- ++ /* ++ layerIII enc->dec delay: 1056 (observed) ++ layerII enc->dec dealy: 480 (observed) ++ ++ ++ polyphase 256-16 (dec or enc) ++ mdct 256+32 (9*32) (dec or enc) ++ total: 512+16 ++ ++ ++ SYNC_FLUSH calculation: ++ layerIII enc delay: 528 ++ extra bits to display full fft window: 224 ++ total: 752 ++ ++ layerII sync_flush: should be 464 ++ layerIII sync_flush: should be 752 ++ ++ ++ Also we would like to remove mpg123 decoder delay with ++ DECDELAY of -528 (not possible) ++ so delay by a enough to make the total delay 1 whole frame (1152-528) ++ */ ++ ++ ++ ++ #define ENCDELAY 528 ++ #define DECDELAY (1152-528) ++ /* include the 224 data from the FFT window */ ++ #define WINDELAY (224 + ENCDELAY) ++ ++ #ifdef HAVEGTK ++ ++ #define READ_AHEAD 10 /* number of frames to read ahead */ ++ #define MAXMPGLAG READ_AHEAD /* if the mpg123 lag becomes bigger than this we have to stop */ ++ #define NUMBACK 6 /* number of frames we can back up */ ++ #define NUMPINFO (NUMBACK+READ_AHEAD+1) ++ ++ #define BLKSIZE 1024 ++ #define HBLKSIZE 513 ++ #define SBMAX_l 21 ++ #define SBMAX_s 12 ++ ++ ++ ++ typedef struct { ++ int frameNum; /* current frame number */ ++ int frameNum123; ++ int num_samples; /* number of pcm samples read for this frame */ ++ double frametime; /* starting time of frame, in seconds */ ++ double pcmdata[2][1152+WINDELAY]; ++ double pcmdata2[2][1152+DECDELAY]; ++ double energy[2][4][BLKSIZE]; ++ double xr[2][2][576]; ++ double mpg123xr[2][2][576]; ++ double pe[2][4]; ++ double ms_ratio[2]; ++ ++ double thr[2][4][SBMAX_l]; ++ double en[2][4][SBMAX_l]; ++ double thr_s[2][4][3*SBMAX_s]; ++ double en_s[2][4][3*SBMAX_s]; ++ double ers[2][4]; ++ ++ double xfsf[2][2][SBMAX_l]; ++ double xfsf_s[2][2][3*SBMAX_s]; ++ ++ int qss[2][2]; ++ int sub_gain[2][2][3]; ++ int over[2][2]; ++ double tot_noise[2][2]; ++ double max_noise[2][2]; ++ double over_noise[2][2]; ++ int blocktype[2][2]; ++ int scalefac_scale[2][2]; ++ int mpg123blocktype[2][2]; ++ int mixed[2][2]; ++ int mainbits[2][2]; ++ int LAMEmainbits[2][2]; ++ int framesize,stereo,js,ms_stereo,i_stereo,emph,bitrate,sampfreq,maindata; ++ int scfsi,mean_bits,resvsize; ++ int totbits; ++ } plotting_data; ++ ++ ++ int gtkcontrol(void); ++ extern plotting_data *pinfo; ++ extern int gtkflag; ++ #endif ++ +diff -r -c -N encoder/huffman.c lame3.50/huffman.c +*** encoder/huffman.c Wed Jan 22 02:43:15 1997 +--- lame3.50/huffman.c Wed Dec 31 17:00:00 1969 +*************** +*** 1,380 **** +- /********************************************************************** +- * ISO MPEG Audio Subgroup Software Simulation Group (1996) +- * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- * +- * $Id: huffman.c,v 1.1 1996/02/14 04:04:23 rowlands Exp $ +- * +- * $Log: huffman.c,v $ +- * Revision 1.1 1996/02/14 04:04:23 rowlands +- * Initial revision +- * +- * Received from Mike Coleman +- **********************************************************************/ +- /********************************************************************** +- * changes made since last update: * +- * date programmers comment * +- *27.2.92 F.O.Witte (ITT Intermetall) * +- * email: otto.witte@itt-sc.de * +- * tel: ++49 (761)517-125 * +- * fax: ++49 (761)517-880 * +- *12.6.92 J. Pineda Added sign bit to decoder. * +- * 08/24/93 M. Iwadare Changed for 1 pass decoding. * +- *--------------------------------------------------------------------* +- * 7/14/94 Juergen Koller Bug fixes in Layer III code * +- *********************************************************************/ +- #include <stdlib.h> +- #include "common.h" +- #include "huffman.h" +- +- HUFFBITS dmask = 1 << (sizeof(HUFFBITS)*8-1); +- unsigned int hs = sizeof(HUFFBITS)*8; +- +- struct huffcodetab ht[HTN]; /* array of all huffcodtable headers */ +- /* 0..31 Huffman code table 0..31 */ +- /* 32,33 count1-tables */ +- +- /* read the huffman encode table */ +- int read_huffcodetab(fi) +- FILE *fi; +- { +- +- char line[100],command[40],huffdata[40]; +- unsigned int t,i,j,k,nn,x,y,n=0; +- unsigned int xl, yl, len; +- HUFFBITS h; +- int hsize; +- +- hsize = sizeof(HUFFBITS)*8; +- do { +- fgets(line,99,fi); +- } while ((line[0] == '#') || (line[0] < ' ') ); +- +- do { +- while ((line[0]=='#') || (line[0] < ' ')) { +- fgets(line,99,fi); +- } +- +- sscanf(line,"%s %s %u %u %u",command,ht[n].tablename, +- &xl,&yl,&ht[n].linbits); +- if (strcmp(command,".end")==0) +- return n; +- else if (strcmp(command,".table")!=0) { +- fprintf(stderr,"huffman table %u data corrupted\n",n); +- return -1; +- } +- ht[n].linmax = (1<<ht[n].linbits)-1; +- +- sscanf(ht[n].tablename,"%u",&nn); +- if (nn != n) { +- fprintf(stderr,"wrong table number %u\n",n); +- return(-2); +- } +- +- ht[n].xlen = xl; +- ht[n].ylen = yl; +- +- do { +- fgets(line,99,fi); +- } while ((line[0] == '#') || (line[0] < ' ')); +- +- sscanf(line,"%s %u",command,&t); +- if (strcmp(command,".reference")==0) { +- ht[n].ref = t; +- ht[n].table = ht[t].table; +- ht[n].hlen = ht[t].hlen; +- if ( (xl != ht[t].xlen) || +- (yl != ht[t].ylen) ) { +- fprintf(stderr,"wrong table %u reference\n",n); +- return (-3); +- }; +- do { +- fgets(line,99,fi); +- } while ((line[0] == '#') || (line[0] < ' ') ); +- } +- else { +- ht[n].ref = -1; +- ht[n].table=(HUFFBITS *) calloc(xl*yl,sizeof(HUFFBITS)); +- if (ht[n].table == NULL) { +- fprintf(stderr,"unsufficient heap error\n"); +- return (-4); +- } +- ht[n].hlen=(unsigned char *) calloc(xl*yl,sizeof(unsigned char)); +- if (ht[n].hlen == NULL) { +- fprintf(stderr,"unsufficient heap error\n"); +- return (-4); +- } +- for (i=0; i<xl; i++) { +- for (j=0;j<yl; j++) { +- if (xl>1) +- sscanf(line,"%u %u %u %s",&x, &y, &len,huffdata); +- else +- sscanf(line,"%u %u %s",&x,&len,huffdata); +- h=0;k=0; +- while (huffdata[k]) { +- h <<= 1; +- if (huffdata[k] == '1') +- h++; +- else if (huffdata[k] != '0'){ +- fprintf(stderr,"huffman-table %u bit error\n",n); +- return (-5); +- }; +- k++; +- }; +- if (k != len) { +- fprintf(stderr, +- "warning: wrong codelen in table %u, pos [%2u][%2u]\n", +- n,i,j); +- }; +- ht[n].table[i*xl+j] = h; +- ht[n].hlen[i*xl+j] = (unsigned char) len; +- do { +- fgets(line,99,fi); +- } while ((line[0] == '#') || (line[0] < ' ')); +- } +- } +- } +- n++; +- } while (1); +- } +- +- /* read the huffman decoder table */ +- int read_decoder_table(fi) +- FILE *fi; +- { +- int n,i,nn,t; +- unsigned int v0,v1; +- char command[100],line[100]; +- for (n=0;n<HTN;n++) { +- /* .table number treelen xlen ylen linbits */ +- do { +- fgets(line,99,fi); +- } while ((line[0] == '#') || (line[0] < ' ')); +- +- sscanf(line,"%s %s %u %u %u %u",command,ht[n].tablename, +- &ht[n].treelen, &ht[n].xlen, &ht[n].ylen, &ht[n].linbits); +- if (strcmp(command,".end")==0) +- return n; +- else if (strcmp(command,".table")!=0) { +- fprintf(stderr,"huffman table %u data corrupted\n",n); +- return -1; +- } +- ht[n].linmax = (1<<ht[n].linbits)-1; +- +- sscanf(ht[n].tablename,"%u",&nn); +- if (nn != n) { +- fprintf(stderr,"wrong table number %u\n",n); +- return(-2); +- } +- do { +- fgets(line,99,fi); +- } while ((line[0] == '#') || (line[0] < ' ')); +- +- sscanf(line,"%s %u",command,&t); +- if (strcmp(command,".reference")==0) { +- ht[n].ref = t; +- ht[n].val = ht[t].val; +- ht[n].treelen = ht[t].treelen; +- if ( (ht[n].xlen != ht[t].xlen) || +- (ht[n].ylen != ht[t].ylen) ) { +- fprintf(stderr,"wrong table %u reference\n",n); +- return (-3); +- }; +- while ((line[0] == '#') || (line[0] < ' ') ) { +- fgets(line,99,fi); +- } +- } +- else if (strcmp(command,".treedata")==0) { +- ht[n].ref = -1; +- if ( ht[n].treelen ) +- { +- ht[n].val = (unsigned char (*)[2]) +- calloc(2*(ht[n].treelen),sizeof(unsigned char)); +- if (ht[n].val == NULL) { +- fprintf(stderr, "heaperror at table %d\n",n); +- exit (-10); +- } +- } +- else +- ht[n].val = NULL; +- for (i=0;i<ht[n].treelen; i++) { +- fscanf(fi,"%x %x",&v0, &v1); +- ht[n].val[i][0]=(unsigned char)v0; +- ht[n].val[i][1]=(unsigned char)v1; +- } +- fgets(line,99,fi); /* read the rest of the line */ +- } +- else { +- fprintf(stderr,"huffman decodertable error at table %d\n",n); +- } +- } +- return n; +- } +- +- +- /* do the huffman coding, */ +- /* note! for counta,countb - the 4 bit value is passed in y, set x to 0 */ +- /* return value: 0-no error, 1 decode error */ +- void huffman_coder( x, y, h, bs) +- unsigned int x; /* x-value */ +- unsigned int y; /* y-value */ +- struct huffcodetab *h; /* pointer to huffman code record */ +- Bit_stream_struc *bs; /* pointer to open write bitstream */ +- { +- HUFFBITS huffbits; /* data left aligned */ +- HUFFBITS linbitsX; +- HUFFBITS linbitsY; +- unsigned int len; +- unsigned int xl1 = h->xlen-1; +- unsigned int yl1 = h->ylen-1; +- linbitsX = 0; +- linbitsY = 0; +- if (h->table == NULL) return; +- if (((x < xl1) || (xl1==0)) && (y < yl1)) { +- huffbits = h->table[x*(h->xlen)+y]; +- len = h->hlen[x*(h->xlen)+y]; +- putbits(bs,huffbits,len); +- return; +- } +- else if (x >= xl1) { +- linbitsX = x-xl1; +- if (linbitsX > h->linmax) { +- fprintf(stderr,"warning: Huffman X table overflow\n"); +- linbitsX= h->linmax; +- }; +- if (y >= yl1) { +- huffbits = h->table[(h->ylen)*(h->xlen)-1]; +- len = h->hlen[(h->ylen)*(h->xlen)-1]; +- putbits(bs,huffbits,len); +- linbitsY = y-yl1; +- if (linbitsY > h->linmax) { +- fprintf(stderr,"warning: Huffman Y table overflow\n"); +- linbitsY = h->linmax; +- }; +- if (h->linbits) { +- putbits(bs,linbitsX,h->linbits); +- putbits(bs,linbitsY,h->linbits); +- } +- } +- else { /* x>= h->xlen, y<h->ylen */ +- huffbits = h->table[(h->ylen)*xl1+y]; +- len = h->hlen[(h->ylen)*xl1+y]; +- putbits(bs,huffbits,len); +- if (h->linbits) { +- putbits(bs,linbitsX,h->linbits); +- } +- } +- } +- else { /* ((x < h->xlen) && (y>=h->ylen)) */ +- huffbits = h->table[(h->ylen)*x+yl1]; +- len = h->hlen[(h->ylen)*x+yl1]; +- putbits(bs,huffbits,len); +- linbitsY = y-yl1; +- if (linbitsY > h->linmax) { +- fprintf(stderr,"warning: Huffman Y table overflow\n"); +- linbitsY = h->linmax; +- }; +- if (h->linbits) { +- putbits(bs,linbitsY,h->linbits); +- } +- } +- } +- +- /* do the huffman-decoding */ +- /* note! for counta,countb -the 4 bit value is returned in y, discard x */ +- int huffman_decoder(h, x, y, v, w) +- struct huffcodetab *h; /* pointer to huffman code record */ +- /* unsigned */ int *x; /* returns decoded x value */ +- /* unsigned */ int *y; /* returns decoded y value */ +- int *v; +- int *w; +- { +- HUFFBITS level; +- int point = 0; +- int error = 1; +- level = dmask; +- if (h->val == NULL) return 2; +- +- /* table 0 needs no bits */ +- if ( h->treelen == 0) +- { *x = *y = 0; +- return 0; +- } +- +- +- /* Lookup in Huffman table. */ +- +- do { +- if (h->val[point][0]==0) { /*end of tree*/ +- *x = h->val[point][1] >> 4; +- *y = h->val[point][1] & 0xf; +- +- error = 0; +- break; +- } +- if (hget1bit()) { +- while (h->val[point][1] >= MXOFF) point += h->val[point][1]; +- point += h->val[point][1]; +- } +- else { +- while (h->val[point][0] >= MXOFF) point += h->val[point][0]; +- point += h->val[point][0]; +- } +- level >>= 1; +- } while (level || (point < ht->treelen) ); +- +- /* Check for error. */ +- +- if (error) { /* set x and y to a medium value as a simple concealment */ +- printf("Illegal Huffman code in data.\n"); +- *x = (h->xlen-1 << 1); +- *y = (h->ylen-1 << 1); +- } +- +- /* Process sign encodings for quadruples tables. */ +- +- if (h->tablename[0] == '3' +- && (h->tablename[1] == '2' || h->tablename[1] == '3')) { +- *v = (*y>>3) & 1; +- *w = (*y>>2) & 1; +- *x = (*y>>1) & 1; +- *y = *y & 1; +- +- /* v, w, x and y are reversed in the bitstream. +- switch them around to make test bistream work. */ +- +- /* {int i=*v; *v=*y; *y=i; i=*w; *w=*x; *x=i;} MI */ +- +- if (*v) +- if (hget1bit() == 1) *v = -*v; +- if (*w) +- if (hget1bit() == 1) *w = -*w; +- if (*x) +- if (hget1bit() == 1) *x = -*x; +- if (*y) +- if (hget1bit() == 1) *y = -*y; +- } +- +- /* Process sign and escape encodings for dual tables. */ +- +- else { +- +- /* x and y are reversed in the test bitstream. +- Reverse x and y here to make test bitstream work. */ +- +- /* removed 11/11/92 -ag +- {int i=*x; *x=*y; *y=i;} +- */ +- if (h->linbits) +- if ((h->xlen-1) == *x) +- *x += hgetbits(h->linbits); +- if (*x) +- if (hget1bit() == 1) *x = -*x; +- if (h->linbits) +- if ((h->ylen-1) == *y) +- *y += hgetbits(h->linbits); +- if (*y) +- if (hget1bit() == 1) *y = -*y; +- } +- +- return error; +- } +--- 0 ---- +diff -r -c -N encoder/huffman.h lame3.50/huffman.h +*** encoder/huffman.h Wed Jan 22 02:43:15 1997 +--- lame3.50/huffman.h Tue Oct 5 18:07:40 1999 +*************** +*** 1,60 **** +- /********************************************************************** +- * ISO MPEG Audio Subgroup Software Simulation Group (1996) +- * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- * +- * $Id: huffman.h,v 1.1 1996/02/14 04:04:23 rowlands Exp $ +- * +- * $Log: huffman.h,v $ +- * Revision 1.1 1996/02/14 04:04:23 rowlands +- * Initial revision +- * +- * Received from Mike Coleman +- **********************************************************************/ +- /********************************************************************** +- * date programmers comment * +- * 27.2.92 F.O.Witte (ITT Intermetall) * +- * 8/24/93 M. Iwadare Changed for 1 pass decoding. * +- * 7/14/94 J. Koller useless 'typedef' before huffcodetab * +- * removed * +- *********************************************************************/ +- + #define HUFFBITS unsigned long int + #define HTN 34 + #define MXOFF 250 + + struct huffcodetab { +- char tablename[3]; /*string, containing table_description */ + unsigned int xlen; /*max. x-index+ */ + unsigned int ylen; /*max. y-index+ */ + unsigned int linbits; /*number of linbits */ + unsigned int linmax; /*max number to be stored in linbits */ +- int ref; /*a positive value indicates a reference*/ + HUFFBITS *table; /*pointer to array[xlen][ylen] */ + unsigned char *hlen; /*pointer to array[xlen][ylen] */ +- unsigned char(*val)[2];/*decoder tree */ +- unsigned int treelen; /*length of decoder tree */ + }; + + extern struct huffcodetab ht[HTN];/* global memory block */ + /* array of all huffcodtable headers */ + /* 0..31 Huffman code table 0..31 */ + /* 32,33 count1-tables */ +- #ifdef PROTO_ARGS +- +- extern int read_huffcodetab(FILE *); +- extern int read_decoder_table(FILE *); +- +- extern void huffman_coder(unsigned int, unsigned int, +- struct huffcodetab *, Bit_stream_struc *); +- +- extern int huffman_decoder(struct huffcodetab *, +- /* unsigned */ int *, /* unsigned */ int*, int*, int*); +- +- #else +- +- extern int read_huffcodetab(); +- extern int read_decoder_table(); +- extern void huffman_coder(); +- extern int huffman_decoder(); + +- #endif +--- 1,18 ---- +diff -r -c -N encoder/id3tag.c lame3.50/id3tag.c +*** encoder/id3tag.c Wed Dec 31 17:00:00 1969 +--- lame3.50/id3tag.c Tue Sep 21 21:25:57 1999 +*************** +*** 0 **** +--- 1,69 ---- ++ /* ++ * functions for writing ID3 tags in LAME ++ * ++ * text functions stolen from mp3info by Ricardo Cerqueira <rmc@rccn.net> ++ * adapted for LAME by Conrad Sanderson <c.sanderson@me.gu.edu.au> ++ * ++ * ++ */ ++ ++ #include <stdio.h> ++ #include <string.h> ++ #include "id3tag.h" ++ ++ /* ++ * If "string" is shorter than "length", pad it with ' ' (spaces) ++ */ ++ ++ static void id3_pad(char *string, int length) { ++ int l; l=strlen(string); ++ ++ while(l<length) { string[l] = ' '; l++; } ++ string[l]='\0'; ++ } ++ ++ ++ /* ++ * initialize temporary fields ++ */ ++ ++ void id3_inittag(ID3TAGDATA *tag) { ++ strcpy( tag->title, ""); ++ strcpy( tag->artist, ""); ++ strcpy( tag->album, ""); ++ strcpy( tag->year, ""); ++ strcpy( tag->comment, ""); ++ ++ tag->valid = 0; // not ready for writing ++ } ++ ++ /* ++ * build an ID3 tag from temporary fields ++ */ ++ ++ void id3_buildtag(ID3TAGDATA *tag) { ++ strcpy(tag->tagtext,"TAG"); ++ ++ id3_pad( tag->title, 30); strncat( tag->tagtext, tag->title,30); ++ id3_pad( tag->artist, 30); strncat( tag->tagtext, tag->artist,30); ++ id3_pad( tag->album, 30); strncat( tag->tagtext, tag->album,30); ++ id3_pad( tag->year, 4); strncat( tag->tagtext, tag->year,4); ++ id3_pad( tag->comment, 30); strncat( tag->tagtext, tag->comment,30); ++ ++ strncat( tag->tagtext, "�", 1); ++ tag->valid = 1; // ready for writing ++ } ++ ++ /* ++ * write ID3 tag ++ */ ++ ++ int id3_writetag(char* filename, ID3TAGDATA *tag) { ++ FILE* f; ++ if( ! tag->valid ) return -1; ++ ++ f=fopen(filename,"rb+"); if(!f) return -1; ++ ++ fseek(f,0,SEEK_END); fwrite(tag->tagtext,1,128,f); ++ fclose(f); return 0; ++ } +diff -r -c -N encoder/id3tag.h lame3.50/id3tag.h +*** encoder/id3tag.h Wed Dec 31 17:00:00 1969 +--- lame3.50/id3tag.h Wed Sep 22 00:23:12 1999 +*************** +*** 0 **** +--- 1,20 ---- ++ #ifndef ID3TAG_H_INCLUDED ++ #define ID3TAG_H_INCLUDED ++ ++ typedef struct ++ { ++ int used; ++ int valid; ++ char title[31]; ++ char artist[31]; ++ char album[31]; ++ char year[5]; ++ char comment[31]; ++ char tagtext[128]; ++ ++ } ID3TAGDATA; ++ ++ void id3_inittag(ID3TAGDATA *tag); ++ void id3_buildtag(ID3TAGDATA *tag); ++ int id3_writetag(char* filename, ID3TAGDATA *tag); ++ #endif +diff -r -c -N encoder/ieeefloat.c lame3.50/ieeefloat.c +*** encoder/ieeefloat.c Wed Jan 22 02:43:17 1997 +--- lame3.50/ieeefloat.c Mon Oct 25 12:22:30 1999 +*************** +*** 82,89 **** + + + defdouble +! ConvertFromIeeeSingle(bytes) +! char* bytes; + { + defdouble f; + long mantissa, expon; +--- 82,88 ---- + + + defdouble +! ConvertFromIeeeSingle(char* bytes) + { + defdouble f; + long mantissa, expon; +*************** +*** 106,116 **** + else { + if (expon == 0) { /* Denormalized number */ + mantissa = (bits & 0x7fffff); +! f = ldexp((defdouble)mantissa, expon - SEXP_OFFSET - SEXP_POSITION + 1); + } + else { /* Normalized number */ + mantissa = (bits & 0x7fffff) + 0x800000; /* Insert hidden bit */ +! f = ldexp((defdouble)mantissa, expon - SEXP_OFFSET - SEXP_POSITION); + } + } + } +--- 105,115 ---- + else { + if (expon == 0) { /* Denormalized number */ + mantissa = (bits & 0x7fffff); +! f = ldexp((defdouble) mantissa, (int) (expon - SEXP_OFFSET - SEXP_POSITION + 1)); + } + else { /* Normalized number */ + mantissa = (bits & 0x7fffff) + 0x800000; /* Insert hidden bit */ +! f = ldexp((defdouble) mantissa, (int) (expon - SEXP_OFFSET - SEXP_POSITION)); + } + } + } +*************** +*** 126,134 **** + + + void +! ConvertToIeeeSingle(num, bytes) +! defdouble num; +! char* bytes; + { + long sign; + register long bits; +--- 125,131 ---- + + + void +! ConvertToIeeeSingle(defdouble num, char* bytes) + { + long sign; + register long bits; +*************** +*** 195,202 **** + + + defdouble +! ConvertFromIeeeDouble(bytes) +! char* bytes; + { + defdouble f; + long mantissa, expon; +--- 192,198 ---- + + + defdouble +! ConvertFromIeeeDouble(char* bytes) + { + defdouble f; + long mantissa, expon; +*************** +*** 223,235 **** + else { + if (expon == 0) { /* Denormalized number */ + mantissa = (first & 0x000FFFFF); +! f = ldexp((defdouble)mantissa, expon - DEXP_OFFSET - DEXP_POSITION + 1); +! f += ldexp(UnsignedToFloat(second), expon - DEXP_OFFSET - DEXP_POSITION + 1 - 32); + } + else { /* Normalized number */ + mantissa = (first & 0x000FFFFF) + 0x00100000; /* Insert hidden bit */ +! f = ldexp((defdouble)mantissa, expon - DEXP_OFFSET - DEXP_POSITION); +! f += ldexp(UnsignedToFloat(second), expon - DEXP_OFFSET - DEXP_POSITION - 32); + } + } + } +--- 219,231 ---- + else { + if (expon == 0) { /* Denormalized number */ + mantissa = (first & 0x000FFFFF); +! f = ldexp((defdouble) mantissa, (int) (expon - DEXP_OFFSET - DEXP_POSITION + 1)); +! f += ldexp(UnsignedToFloat(second), (int) (expon - DEXP_OFFSET - DEXP_POSITION + 1 - 32)); + } + else { /* Normalized number */ + mantissa = (first & 0x000FFFFF) + 0x00100000; /* Insert hidden bit */ +! f = ldexp((defdouble) mantissa, (int) (expon - DEXP_OFFSET - DEXP_POSITION)); +! f += ldexp(UnsignedToFloat(second), (int) (expon - DEXP_OFFSET - DEXP_POSITION - 32)); + } + } + } +*************** +*** 245,253 **** + + + void +! ConvertToIeeeDouble(num, bytes) +! defdouble num; +! char *bytes; + { + long sign; + long first, second; +--- 241,247 ---- + + + void +! ConvertToIeeeDouble(defdouble num, char *bytes) + { + long sign; + long first, second; +*************** +*** 326,333 **** + ****************************************************************/ + + defdouble +! ConvertFromIeeeExtended(bytes) +! char* bytes; + { + defdouble f; + long expon; +--- 320,326 ---- + ****************************************************************/ + + defdouble +! ConvertFromIeeeExtended(char* bytes) + { + defdouble f; + long expon; +*************** +*** 359,366 **** + } + else { + expon -= 16383; +! f = ldexp(UnsignedToFloat(hiMant), expon-=31); +! f += ldexp(UnsignedToFloat(loMant), expon-=32); + } + } + +--- 352,359 ---- + } + else { + expon -= 16383; +! f = ldexp(UnsignedToFloat(hiMant), (int) (expon -= 31)); +! f += ldexp(UnsignedToFloat(loMant), (int) (expon -= 32)); + } + } + +*************** +*** 375,383 **** + + + void +! ConvertToIeeeExtended(num, bytes) +! defdouble num; +! char *bytes; + { + int sign; + int expon; +--- 368,374 ---- + + + void +! ConvertToIeeeExtended(defdouble num, char *bytes) + { + int sign; + int expon; +*************** +*** 467,474 **** + + + int +! GetHexValue(x) +! register int x; + { + x &= 0x7F; + +--- 458,464 ---- + + + int +! GetHexValue(register int x) + { + x &= 0x7F; + +*************** +*** 486,493 **** + + + void +! Hex2Bytes(hex, bytes) +! register char *hex, *bytes; + { + for ( ; *hex; hex += 2) { + *bytes++ = (GetHexValue(hex[0]) << 4) | GetHexValue(hex[1]); +--- 476,482 ---- + + + void +! Hex2Bytes(register char *hex, register char *bytes) + { + for ( ; *hex; hex += 2) { + *bytes++ = (GetHexValue(hex[0]) << 4) | GetHexValue(hex[1]); +*************** +*** 498,505 **** + + + int +! GetHexSymbol(x) +! register int x; + { + x &= 0xF; + if (x <= 9) +--- 487,493 ---- + + + int +! GetHexSymbol(register int x) + { + x &= 0xF; + if (x <= 9) +*************** +*** 511,519 **** + + + void +! Bytes2Hex(bytes, hex, nBytes) +! register char *bytes, *hex; +! register int nBytes; + { + for ( ; nBytes--; bytes++) { + *hex++ = GetHexSymbol(*bytes >> 4); +--- 499,505 ---- + + + void +! Bytes2Hex(register char *bytes, register char *hex, register int nBytes) + { + for ( ; nBytes--; bytes++) { + *hex++ = GetHexSymbol(*bytes >> 4); +*************** +*** 524,532 **** + + + void +! MaybeSwapBytes(bytes, nBytes) +! char* bytes; +! int nBytes; + { + #ifdef LITTLE_ENDIAN + register char *p, *q, t; +--- 510,516 ---- + + + void +! MaybeSwapBytes(char* bytes, int nBytes) + { + #ifdef LITTLE_ENDIAN + register char *p, *q, t; +*************** +*** 543,550 **** + + + float +! MachineIEEESingle(bytes) +! char* bytes; + { + float t; + MaybeSwapBytes(bytes, 4); +--- 527,533 ---- + + + float +! MachineIEEESingle(char* bytes) + { + float t; + MaybeSwapBytes(bytes, 4); +*************** +*** 555,562 **** + + + Double +! MachineIEEEDouble(bytes) +! char* bytes; + { + Double t; + MaybeSwapBytes(bytes, 8); +--- 538,544 ---- + + + Double +! MachineIEEEDouble(char* bytes) + { + Double t; + MaybeSwapBytes(bytes, 8); +*************** +*** 567,574 **** + + + void +! TestFromIeeeSingle(hex) +! char *hex; + { + defdouble f; + union SParts p; +--- 549,555 ---- + + + void +! TestFromIeeeSingle(char *hex) + { + defdouble f; + union SParts p; +*************** +*** 589,596 **** + + + void +! TestToIeeeSingle(f) +! defdouble f; + { + union SParts p; + char bytes[4]; +--- 570,576 ---- + + + void +! TestToIeeeSingle(defdouble f) + { + union SParts p; + char bytes[4]; +*************** +*** 613,620 **** + + + void +! TestFromIeeeDouble(hex) +! char *hex; + { + defdouble f; + union DParts p; +--- 593,599 ---- + + + void +! TestFromIeeeDouble(char *hex) + { + defdouble f; + union DParts p; +*************** +*** 636,643 **** + } + + void +! TestToIeeeDouble(f) +! defdouble f; + { + union DParts p; + char bytes[8]; +--- 615,621 ---- + } + + void +! TestToIeeeDouble(defdouble f) + { + union DParts p; + char bytes[8]; +*************** +*** 663,670 **** + + + void +! TestFromIeeeExtended(hex) +! char *hex; + { + defdouble f; + union EParts p; +--- 641,647 ---- + + + void +! TestFromIeeeExtended(char *hex) + { + defdouble f; + union EParts p; +*************** +*** 701,708 **** + + + void +! TestToIeeeExtended(f) +! defdouble f; + { + char bytes[12]; + char hex[24+1]; +--- 678,684 ---- + + + void +! TestToIeeeExtended(defdouble f) + { + char bytes[12]; + char hex[24+1]; +*************** +*** 736,751 **** + + #include <signal.h> + +! void SignalFPE(i, j) +! int i; +! void (*j)(); + { + printf("[Floating Point Interrupt Caught.]\n", i, j); + signal(SIGFPE, SignalFPE); + } + + void +! main() + { + long d[3]; + char bytes[12]; +--- 712,725 ---- + + #include <signal.h> + +! void SignalFPE(int i, void (*j)()) + { + printf("[Floating Point Interrupt Caught.]\n", i, j); + signal(SIGFPE, SignalFPE); + } + + void +! main(void) + { + long d[3]; + char bytes[12]; +diff -r -c -N encoder/l3bitstream.c lame3.50/l3bitstream.c +*** encoder/l3bitstream.c Wed Jan 22 02:43:15 1997 +--- lame3.50/l3bitstream.c Thu Oct 28 21:52:53 1999 +*************** +*** 21,29 **** + + #include <stdlib.h> + #include "l3bitstream.h" /* the public interface */ +! #include "l3psy.h" +! #include "mdct.h" +! #include "loop.h" + #include "formatBitstream.h" + #include "huffman.h" + #include <assert.h> +--- 21,29 ---- + + #include <stdlib.h> + #include "l3bitstream.h" /* the public interface */ +! #include "psymodel.h" +! #include "fastmdct.h" +! #include "quantize.h" + #include "formatBitstream.h" + #include "huffman.h" + #include <assert.h> +*************** +*** 75,96 **** + char *ancillary, + int ancillary_bits ) + { +! int gr, ch, i, mode_gr; + fr_ps = in_fr_ps; + bs = in_bs; + stereo = fr_ps->stereo; + mode_gr = (fr_ps->header->version == 1) ? 2 : 1; +! + if ( frameData == NULL ) + { +! frameData = calloc( 1, sizeof(*frameData) ); + assert( frameData ); + } + if ( frameResults == NULL ) + { +! frameResults = calloc( 1, sizeof(*frameData) ); + assert( frameData ); + } + if ( !PartHoldersInitialized ) + { + headerPH = BF_newPartHolder( 12 ); +--- 75,107 ---- + char *ancillary, + int ancillary_bits ) + { +! int gr, ch, mode_gr; +! static BF_FrameData real_frameData; +! static BF_FrameResults real_frameResults; +! +! + fr_ps = in_fr_ps; + bs = in_bs; + stereo = fr_ps->stereo; + mode_gr = (fr_ps->header->version == 1) ? 2 : 1; +! +! frameData = &real_frameData; +! frameResults = &real_frameResults; +! +! +! #if 0 + if ( frameData == NULL ) + { +! frameData = (BF_FrameData*) calloc( 1, sizeof(BF_FrameData) ); + assert( frameData ); + } + if ( frameResults == NULL ) + { +! frameResults = (BF_FrameResults*) calloc( 1, sizeof(BF_FrameResults) ); + assert( frameData ); + } ++ #endif ++ + if ( !PartHoldersInitialized ) + { + headerPH = BF_newPartHolder( 12 ); +*************** +*** 111,126 **** + PartHoldersInitialized = 1; + } + +! #if 1 + for ( gr = 0; gr < mode_gr; gr++ ) + for ( ch = 0; ch < stereo; ch++ ) + { + int *pi = &l3_enc[gr][ch][0]; +! double *pr = &xr[gr][ch][0]; +! for ( i = 0; i < 576; i++, pr++, pi++ ) +! { +! if ( (*pr < 0) && (*pi > 0) ) +! *pi *= -1; + } + } + #endif +--- 122,143 ---- + PartHoldersInitialized = 1; + } + +! #if 0 +! /* this makes more sense in loop.c */ + for ( gr = 0; gr < mode_gr; gr++ ) + for ( ch = 0; ch < stereo; ch++ ) + { + int *pi = &l3_enc[gr][ch][0]; +! int i; +! for ( i = 0; i < 576; i++) { +! double pr = xr[gr][ch][i]; +! /* ms stereo fix. if ms_stereo, l3_enc will be M/S values */ +! if (in_fr_ps->header->mode_ext == 2) { +! pr = ch ? (xr[gr][0][i]-xr[gr][1][i]) : (xr[gr][0][i]+xr[gr][1][i]); +! /* pr /= sqrt(2.0); */ +! } +! +! if ( (pr < 0) && (pi[i] > 0) ) pi[i] *= -1; + } + } + #endif +*************** +*** 129,134 **** +--- 146,153 ---- + encodeMainData( l3_enc, l3_side, scalefac ); + write_ancillary_data( ancillary, ancillary_bits ); + ++ ++ + if ( l3_side->resvDrain ) + drain_into_ancillary_data( l3_side->resvDrain ); + /* +*************** +*** 159,168 **** + + /* we set this here -- it will be tested in the next loops iteration */ + l3_side->main_data_begin = frameResults->nextBackPtr; + } + + void +! III_FlushBitstream() + { + assert( PartHoldersInitialized ); + BF_FlushBitstream( frameData, frameResults ); +--- 178,188 ---- + + /* we set this here -- it will be tested in the next loops iteration */ + l3_side->main_data_begin = frameResults->nextBackPtr; ++ + } + + void +! III_FlushBitstream(void) + { + assert( PartHoldersInitialized ); + BF_FlushBitstream( frameData, frameResults ); +*************** +*** 179,184 **** +--- 199,207 ---- + int i, gr, ch, sfb, window, mode_gr; + layer *info = fr_ps->header; + ++ ++ ++ + if ( info->version == 1 ) + mode_gr = 2; + else +*************** +*** 206,211 **** +--- 229,235 ---- + + if ( (gi->window_switching_flag == 1) && (gi->block_type == 2) ) + { ++ #ifdef ALLOW_MIXED + if ( gi->mixed_block_flag ) + { + for ( sfb = 0; sfb < 8; sfb++ ) +*************** +*** 221,226 **** +--- 245,251 ---- + + } + else ++ #endif + { + for ( sfb = 0; sfb < 6; sfb++ ) + for ( window = 0; window < 3; window++ ) +*************** +*** 262,272 **** + gr_info *gi = &(si->gr[gr].ch[ch].tt); + int *ix = &l3_enc[gr][ch][0]; + int sfb_partition; +- + assert( gi->sfb_partition_table ); + + if ( (gi->window_switching_flag == 1) && (gi->block_type == 2) ) + { + if ( gi->mixed_block_flag ) + { + sfb_partition = 0; +--- 287,297 ---- + gr_info *gi = &(si->gr[gr].ch[ch].tt); + int *ix = &l3_enc[gr][ch][0]; + int sfb_partition; + assert( gi->sfb_partition_table ); + + if ( (gi->window_switching_flag == 1) && (gi->block_type == 2) ) + { ++ #ifdef ALLOW_MIXED + if ( gi->mixed_block_flag ) + { + sfb_partition = 0; +*************** +*** 283,288 **** +--- 308,314 ---- + } + } + else ++ #endif + { + for ( sfb = 0, sfb_partition = 0; sfb_partition < 4; sfb_partition++ ) + { +*************** +*** 304,315 **** + *pph = BF_addEntry( *pph, scalefac->l[gr][ch][sfb], slen ); + } + } + Huffmancodebits( &codedDataPH[gr][ch], ix, gi ); + } /* for ch */ + } + } /* main_data */ + +! static unsigned int crc = 0; + + static int encodeSideInfo( III_side_info_t *si ) + { +--- 330,358 ---- + *pph = BF_addEntry( *pph, scalefac->l[gr][ch][sfb], slen ); + } + } ++ ++ ++ + Huffmancodebits( &codedDataPH[gr][ch], ix, gi ); + } /* for ch */ + } + } /* main_data */ + +! static unsigned int crc = 0; /* (jo) current crc */ +! +! /* (jo) this wrapper function for BF_addEntry() updates also the crc */ +! static BF_PartHolder *CRC_BF_addEntry( BF_PartHolder *thePH, uint32 value, uint16 length ) +! { +! unsigned int bit = 1 << length; +! +! while((bit >>= 1)){ +! crc <<= 1; +! if (!(crc & 0x10000) ^ !(value & bit)) +! crc ^= CRC16_POLYNOMIAL; +! } +! crc &= 0xffff; +! return BF_addEntry(thePH, value, length); +! } + + static int encodeSideInfo( III_side_info_t *si ) + { +*************** +*** 317,345 **** + layer *info = fr_ps->header; + + mode_gr = (info->version == 1) ? 2 : 1; + + headerPH->part->nrEntries = 0; + headerPH = BF_addEntry( headerPH, 0xfff, 12 ); + headerPH = BF_addEntry( headerPH, info->version, 1 ); + headerPH = BF_addEntry( headerPH, 4 - info->lay, 2 ); + headerPH = BF_addEntry( headerPH, !info->error_protection, 1 ); +! headerPH = BF_addEntry( headerPH, info->bitrate_index, 4 ); +! headerPH = BF_addEntry( headerPH, info->sampling_frequency, 2 ); +! headerPH = BF_addEntry( headerPH, info->padding, 1 ); +! headerPH = BF_addEntry( headerPH, info->extension, 1 ); +! headerPH = BF_addEntry( headerPH, info->mode, 2 ); +! headerPH = BF_addEntry( headerPH, info->mode_ext, 2 ); +! headerPH = BF_addEntry( headerPH, info->copyright, 1 ); +! headerPH = BF_addEntry( headerPH, info->original, 1 ); +! headerPH = BF_addEntry( headerPH, info->emphasis, 2 ); + + bits_sent = 32; +! +! if ( fr_ps->header->error_protection ) +! { +! headerPH = BF_addEntry( headerPH, crc, 16 ); +! bits_sent += 16; +! } + + frameSIPH->part->nrEntries = 0; + +--- 360,387 ---- + layer *info = fr_ps->header; + + mode_gr = (info->version == 1) ? 2 : 1; ++ ++ crc = 0xffff; /* (jo) init crc16 for error_protection */ + + headerPH->part->nrEntries = 0; + headerPH = BF_addEntry( headerPH, 0xfff, 12 ); + headerPH = BF_addEntry( headerPH, info->version, 1 ); + headerPH = BF_addEntry( headerPH, 4 - info->lay, 2 ); + headerPH = BF_addEntry( headerPH, !info->error_protection, 1 ); +! /* (jo) from now on call the CRC_BF_addEntry() wrapper to update crc */ +! headerPH = CRC_BF_addEntry( headerPH, info->bitrate_index, 4 ); +! headerPH = CRC_BF_addEntry( headerPH, info->sampling_frequency, 2 ); +! headerPH = CRC_BF_addEntry( headerPH, info->padding, 1 ); +! headerPH = CRC_BF_addEntry( headerPH, info->extension, 1 ); +! headerPH = CRC_BF_addEntry( headerPH, info->mode, 2 ); +! headerPH = CRC_BF_addEntry( headerPH, info->mode_ext, 2 ); +! headerPH = CRC_BF_addEntry( headerPH, info->copyright, 1 ); +! headerPH = CRC_BF_addEntry( headerPH, info->original, 1 ); +! headerPH = CRC_BF_addEntry( headerPH, info->emphasis, 2 ); + + bits_sent = 32; +! +! /* (jo) see below for BF_addEntry( headerPH, crc, 16 ); */ + + frameSIPH->part->nrEntries = 0; + +*************** +*** 352,369 **** + + if ( info->version == 1 ) + { /* MPEG1 */ +! frameSIPH = BF_addEntry( frameSIPH, si->main_data_begin, 9 ); + + if ( stereo == 2 ) +! frameSIPH = BF_addEntry( frameSIPH, si->private_bits, 3 ); + else +! frameSIPH = BF_addEntry( frameSIPH, si->private_bits, 5 ); + + for ( ch = 0; ch < stereo; ch++ ) + for ( scfsi_band = 0; scfsi_band < 4; scfsi_band++ ) + { + BF_PartHolder **pph = &channelSIPH[ch]; +! *pph = BF_addEntry( *pph, si->scfsi[ch][scfsi_band], 1 ); + } + + for ( gr = 0; gr < 2; gr++ ) +--- 394,411 ---- + + if ( info->version == 1 ) + { /* MPEG1 */ +! frameSIPH = CRC_BF_addEntry( frameSIPH, si->main_data_begin, 9 ); + + if ( stereo == 2 ) +! frameSIPH = CRC_BF_addEntry( frameSIPH, si->private_bits, 3 ); + else +! frameSIPH = CRC_BF_addEntry( frameSIPH, si->private_bits, 5 ); + + for ( ch = 0; ch < stereo; ch++ ) + for ( scfsi_band = 0; scfsi_band < 4; scfsi_band++ ) + { + BF_PartHolder **pph = &channelSIPH[ch]; +! *pph = CRC_BF_addEntry( *pph, si->scfsi[ch][scfsi_band], 1 ); + } + + for ( gr = 0; gr < 2; gr++ ) +*************** +*** 371,405 **** + { + BF_PartHolder **pph = &spectrumSIPH[gr][ch]; + gr_info *gi = &(si->gr[gr].ch[ch].tt); +! *pph = BF_addEntry( *pph, gi->part2_3_length, 12 ); +! *pph = BF_addEntry( *pph, gi->big_values, 9 ); +! *pph = BF_addEntry( *pph, gi->global_gain, 8 ); +! *pph = BF_addEntry( *pph, gi->scalefac_compress, 4 ); +! *pph = BF_addEntry( *pph, gi->window_switching_flag, 1 ); + + if ( gi->window_switching_flag ) + { +! *pph = BF_addEntry( *pph, gi->block_type, 2 ); +! *pph = BF_addEntry( *pph, gi->mixed_block_flag, 1 ); + + for ( region = 0; region < 2; region++ ) +! *pph = BF_addEntry( *pph, gi->table_select[region], 5 ); + for ( window = 0; window < 3; window++ ) +! *pph = BF_addEntry( *pph, gi->subblock_gain[window], 3 ); + } + else + { + assert( gi->block_type == 0 ); + for ( region = 0; region < 3; region++ ) +! *pph = BF_addEntry( *pph, gi->table_select[region], 5 ); + +! *pph = BF_addEntry( *pph, gi->region0_count, 4 ); +! *pph = BF_addEntry( *pph, gi->region1_count, 3 ); + } + +! *pph = BF_addEntry( *pph, gi->preflag, 1 ); +! *pph = BF_addEntry( *pph, gi->scalefac_scale, 1 ); +! *pph = BF_addEntry( *pph, gi->count1table_select, 1 ); + } + + if ( stereo == 2 ) +--- 413,447 ---- + { + BF_PartHolder **pph = &spectrumSIPH[gr][ch]; + gr_info *gi = &(si->gr[gr].ch[ch].tt); +! *pph = CRC_BF_addEntry( *pph, gi->part2_3_length, 12 ); +! *pph = CRC_BF_addEntry( *pph, gi->big_values, 9 ); +! *pph = CRC_BF_addEntry( *pph, gi->global_gain, 8 ); +! *pph = CRC_BF_addEntry( *pph, gi->scalefac_compress, 4 ); +! *pph = CRC_BF_addEntry( *pph, gi->window_switching_flag, 1 ); + + if ( gi->window_switching_flag ) + { +! *pph = CRC_BF_addEntry( *pph, gi->block_type, 2 ); +! *pph = CRC_BF_addEntry( *pph, gi->mixed_block_flag, 1 ); + + for ( region = 0; region < 2; region++ ) +! *pph = CRC_BF_addEntry( *pph, gi->table_select[region], 5 ); + for ( window = 0; window < 3; window++ ) +! *pph = CRC_BF_addEntry( *pph, gi->subblock_gain[window], 3 ); + } + else + { + assert( gi->block_type == 0 ); + for ( region = 0; region < 3; region++ ) +! *pph = CRC_BF_addEntry( *pph, gi->table_select[region], 5 ); + +! *pph = CRC_BF_addEntry( *pph, gi->region0_count, 4 ); +! *pph = CRC_BF_addEntry( *pph, gi->region1_count, 3 ); + } + +! *pph = CRC_BF_addEntry( *pph, gi->preflag, 1 ); +! *pph = CRC_BF_addEntry( *pph, gi->scalefac_scale, 1 ); +! *pph = CRC_BF_addEntry( *pph, gi->count1table_select, 1 ); + } + + if ( stereo == 2 ) +*************** +*** 409,459 **** + } + else + { /* MPEG2 */ +! frameSIPH = BF_addEntry( frameSIPH, si->main_data_begin, 8 ); + + if ( stereo == 2 ) +! frameSIPH = BF_addEntry( frameSIPH, si->private_bits, 2 ); + else +! frameSIPH = BF_addEntry( frameSIPH, si->private_bits, 1 ); + + gr = 0; + for ( ch = 0; ch < stereo; ch++ ) + { + BF_PartHolder **pph = &spectrumSIPH[gr][ch]; + gr_info *gi = &(si->gr[gr].ch[ch].tt); +! *pph = BF_addEntry( *pph, gi->part2_3_length, 12 ); +! *pph = BF_addEntry( *pph, gi->big_values, 9 ); +! *pph = BF_addEntry( *pph, gi->global_gain, 8 ); +! *pph = BF_addEntry( *pph, gi->scalefac_compress, 9 ); +! *pph = BF_addEntry( *pph, gi->window_switching_flag, 1 ); + + if ( gi->window_switching_flag ) + { +! *pph = BF_addEntry( *pph, gi->block_type, 2 ); +! *pph = BF_addEntry( *pph, gi->mixed_block_flag, 1 ); + + for ( region = 0; region < 2; region++ ) +! *pph = BF_addEntry( *pph, gi->table_select[region], 5 ); + for ( window = 0; window < 3; window++ ) +! *pph = BF_addEntry( *pph, gi->subblock_gain[window], 3 ); + } + else + { + for ( region = 0; region < 3; region++ ) +! *pph = BF_addEntry( *pph, gi->table_select[region], 5 ); + +! *pph = BF_addEntry( *pph, gi->region0_count, 4 ); +! *pph = BF_addEntry( *pph, gi->region1_count, 3 ); + } + +! *pph = BF_addEntry( *pph, gi->scalefac_scale, 1 ); +! *pph = BF_addEntry( *pph, gi->count1table_select, 1 ); + } + if ( stereo == 2 ) + bits_sent += 136; + else + bits_sent += 72; + } + return bits_sent; + } + +--- 451,508 ---- + } + else + { /* MPEG2 */ +! frameSIPH = CRC_BF_addEntry( frameSIPH, si->main_data_begin, 8 ); + + if ( stereo == 2 ) +! frameSIPH = CRC_BF_addEntry( frameSIPH, si->private_bits, 2 ); + else +! frameSIPH = CRC_BF_addEntry( frameSIPH, si->private_bits, 1 ); + + gr = 0; + for ( ch = 0; ch < stereo; ch++ ) + { + BF_PartHolder **pph = &spectrumSIPH[gr][ch]; + gr_info *gi = &(si->gr[gr].ch[ch].tt); +! *pph = CRC_BF_addEntry( *pph, gi->part2_3_length, 12 ); +! *pph = CRC_BF_addEntry( *pph, gi->big_values, 9 ); +! *pph = CRC_BF_addEntry( *pph, gi->global_gain, 8 ); +! *pph = CRC_BF_addEntry( *pph, gi->scalefac_compress, 9 ); +! *pph = CRC_BF_addEntry( *pph, gi->window_switching_flag, 1 ); + + if ( gi->window_switching_flag ) + { +! *pph = CRC_BF_addEntry( *pph, gi->block_type, 2 ); +! *pph = CRC_BF_addEntry( *pph, gi->mixed_block_flag, 1 ); + + for ( region = 0; region < 2; region++ ) +! *pph = CRC_BF_addEntry( *pph, gi->table_select[region], 5 ); + for ( window = 0; window < 3; window++ ) +! *pph = CRC_BF_addEntry( *pph, gi->subblock_gain[window], 3 ); + } + else + { + for ( region = 0; region < 3; region++ ) +! *pph = CRC_BF_addEntry( *pph, gi->table_select[region], 5 ); + +! *pph = CRC_BF_addEntry( *pph, gi->region0_count, 4 ); +! *pph = CRC_BF_addEntry( *pph, gi->region1_count, 3 ); + } + +! *pph = CRC_BF_addEntry( *pph, gi->scalefac_scale, 1 ); +! *pph = CRC_BF_addEntry( *pph, gi->count1table_select, 1 ); + } + if ( stereo == 2 ) + bits_sent += 136; + else + bits_sent += 72; + } ++ ++ if ( fr_ps->header->error_protection ) ++ { /* (jo) error_protection: add crc16 information to header */ ++ headerPH = BF_addEntry( headerPH, crc, 16 ); ++ bits_sent += 16; ++ } ++ + return bits_sent; + } + +*************** +*** 524,535 **** + int i, bigvalues, count1End; + int v, w, x, y, bits, cbits, xbits, stuffingBits; + unsigned int code, ext; +! struct huffcodetab *h; +! int bvbits, c1bits, tablezeros, r0, r1, r2, rt, *pr; + int bitsWritten = 0; +! int idx = 0; +! tablezeros = 0; +! r0 = r1 = r2 = 0; + + /* 1: Write the bigvalues */ + bigvalues = gi->big_values * 2; +--- 573,581 ---- + int i, bigvalues, count1End; + int v, w, x, y, bits, cbits, xbits, stuffingBits; + unsigned int code, ext; +! int bvbits, c1bits; + int bitsWritten = 0; +! + + /* 1: Write the bigvalues */ + bigvalues = gi->big_values * 2; +*************** +*** 569,576 **** + { + x = (*ix_s)[line][window]; + y = (*ix_s)[line + 1][window]; +- assert( idx < 576 ); +- assert( idx >= 0 ); + bits = HuffmanCode( tableindex, x, y, &code, &ext, &cbits, &xbits ); + *pph = BF_addEntry( *pph, code, cbits ); + *pph = BF_addEntry( *pph, ext, xbits ); +--- 615,620 ---- +*************** +*** 580,585 **** +--- 624,630 ---- + } + } + else ++ #ifdef ALLOW_MIXED + if ( gi->mixed_block_flag && gi->block_type == 2 ) + { /* Mixed blocks long, short */ + int sfb, window, line, start, end; +*************** +*** 625,630 **** +--- 670,676 ---- + + } + else ++ #endif + { /* Long blocks */ + int *scalefac = &sfBandIndex[fr_ps->header->sampling_frequency + (fr_ps->header->version * 3)].l[0]; + unsigned scalefac_index = 100; +*************** +*** 642,649 **** + scalefac_index += gi->region1_count + 1; + assert( scalefac_index < 23 ); + region2Start = scalefac[ scalefac_index ]; +! assert( region1Start == gi->address1 ); + } + for ( i = 0; i < bigvalues; i += 2 ) + { + unsigned tableindex = 100; +--- 688,696 ---- + scalefac_index += gi->region1_count + 1; + assert( scalefac_index < 23 ); + region2Start = scalefac[ scalefac_index ]; +! // assert( region1Start == gi->address1 ); + } ++ + for ( i = 0; i < bigvalues; i += 2 ) + { + unsigned tableindex = 100; +*************** +*** 651,686 **** + if ( i < region1Start ) + { + tableindex = gi->table_select[0]; +- pr = &r0; + } + else + if ( i < region2Start ) + { + tableindex = gi->table_select[1]; +- pr = &r1; + } + else + { + tableindex = gi->table_select[2]; +- pr = &r2; + } + assert( tableindex < 32 ); +- h = &ht[ tableindex ]; + /* get huffman code */ + x = ix[i]; + y = ix[i + 1]; + if ( tableindex ) + { + bits = HuffmanCode( tableindex, x, y, &code, &ext, &cbits, &xbits ); + *pph = BF_addEntry( *pph, code, cbits ); + *pph = BF_addEntry( *pph, ext, xbits ); +! bitsWritten += rt = bits; +! *pr += rt; +! } +! else +! { +! tablezeros += 1; +! *pr = 0; + } + } + } +--- 698,724 ---- + if ( i < region1Start ) + { + tableindex = gi->table_select[0]; + } + else + if ( i < region2Start ) + { + tableindex = gi->table_select[1]; + } + else + { + tableindex = gi->table_select[2]; + } + assert( tableindex < 32 ); + /* get huffman code */ + x = ix[i]; + y = ix[i + 1]; + if ( tableindex ) + { ++ + bits = HuffmanCode( tableindex, x, y, &code, &ext, &cbits, &xbits ); + *pph = BF_addEntry( *pph, code, cbits ); + *pph = BF_addEntry( *pph, ext, xbits ); +! bitsWritten += bits; + } + } + } +*************** +*** 689,695 **** + + /* 2: Write count1 area */ + assert( (gi->count1table_select < 2) ); +! h = &ht[gi->count1table_select + 32]; + count1End = bigvalues + (gi->count1 * 4); + assert( count1End <= 576 ); + for ( i = bigvalues; i < count1End; i += 4 ) +--- 727,733 ---- + + /* 2: Write count1 area */ + assert( (gi->count1table_select < 2) ); +! // h = &ht[gi->count1table_select + 32]; + count1End = bigvalues + (gi->count1 * 4); + assert( count1End <= 576 ); + for ( i = bigvalues; i < count1End; i += 4 ) +*************** +*** 698,712 **** + w = ix[i+1]; + x = ix[i+2]; + y = ix[i+3]; +! bitsWritten += L3_huffman_coder_count1( pph, h, v, w, x, y ); + } + c1bits = bitsWritten - bvbits; + if ( (stuffingBits = gi->part2_3_length - gi->part2_length - bitsWritten) ) + { + int stuffingWords = stuffingBits / 32; + int remainingBits = stuffingBits % 32; +- assert( stuffingBits > 0 ); + + /* + Due to the nature of the Huffman code + tables, we will pad with ones +--- 736,756 ---- + w = ix[i+1]; + x = ix[i+2]; + y = ix[i+3]; +! // bitsWritten += L3_huffman_coder_count1( pph, h, v, w, x, y ); +! bitsWritten += L3_huffman_coder_count1( pph, &ht[gi->count1table_select + 32], v, w, x, y ); + } + c1bits = bitsWritten - bvbits; ++ #ifdef DEBUG ++ fprintf(stderr,"## Huffman bits written (%02d + %02d), part2_length = %d, part2_3_length = %d ##\n", ++ bvbits, c1bits, gi->part2_length, gi->part2_3_length ); ++ #endif + if ( (stuffingBits = gi->part2_3_length - gi->part2_length - bitsWritten) ) + { + int stuffingWords = stuffingBits / 32; + int remainingBits = stuffingBits % 32; + ++ printf("opps - adding stuffing bits = %i ... \n",stuffingBits); ++ assert( stuffingBits > 0 ); + /* + Due to the nature of the Huffman code + tables, we will pad with ones +*************** +*** 719,725 **** + } + assert( bitsWritten == gi->part2_3_length - gi->part2_length ); + #ifdef DEBUG +! printf( "#### %d Huffman bits written (%02d + %02d), part2_length = %d, part2_3_length = %d, %d stuffing ####\n", + bitsWritten, bvbits, c1bits, gi->part2_length, gi->part2_3_length, stuffingBits ); + #endif + } +--- 763,769 ---- + } + assert( bitsWritten == gi->part2_3_length - gi->part2_length ); + #ifdef DEBUG +! fprintf(stderr, "## %d Huffman bits written (%02d + %02d), part2_length = %d, part2_3_length = %d, %d stuffing ##\n", + bitsWritten, bvbits, c1bits, gi->part2_length, gi->part2_3_length, stuffingBits ); + #endif + } +*************** +*** 746,755 **** + signx = abs_and_sign( &x ); + signy = abs_and_sign( &y ); + +! p = v + (w << 1) + (x << 2) + (y << 3); + huffbits = h->table[p]; + len = h->hlen[ p ]; +! *pph = BF_addEntry( *pph, huffbits, len ); + totalBits += len; + if ( v ) + { +--- 790,801 ---- + signx = abs_and_sign( &x ); + signy = abs_and_sign( &y ); + +! // bug fix from Leonid A. Kulakov 9/1999: +! p = (v << 3) + (w << 2) + (x << 1) + y; +! + huffbits = h->table[p]; + len = h->hlen[ p ]; +! *pph = BF_addEntry(*pph, (uint32) huffbits, len); + totalBits += len; + if ( v ) + { +*************** +*** 781,787 **** + int + HuffmanCode( int table_select, int x, int y, unsigned int *code, unsigned int *ext, int *cbits, int *xbits ) + { +! unsigned signx, signy, linbitsx, linbitsy, linbits, xlen, ylen, idx; + struct huffcodetab *h; + + *cbits = 0; +--- 827,833 ---- + int + HuffmanCode( int table_select, int x, int y, unsigned int *code, unsigned int *ext, int *cbits, int *xbits ) + { +! unsigned signx, signy, linbitsx, linbitsy, linbits, ylen, idx; + struct huffcodetab *h; + + *cbits = 0; +*************** +*** 795,801 **** + signx = abs_and_sign( &x ); + signy = abs_and_sign( &y ); + h = &(ht[table_select]); +- xlen = h->xlen; + ylen = h->ylen; + linbits = h->linbits; + linbitsx = linbitsy = 0; +--- 841,846 ---- +diff -r -c -N encoder/l3bitstream.h lame3.50/l3bitstream.h +*** encoder/l3bitstream.h Wed Jan 22 02:43:15 1997 +--- lame3.50/l3bitstream.h Tue Oct 12 14:56:11 1999 +*************** +*** 14,20 **** + #ifndef L3_BITSTREAM_H + #define L3_BITSTREAM_H + +! #include "common.h" + #include "encoder.h" + + void III_format_bitstream( int bitsPerFrame, +--- 14,20 ---- + #ifndef L3_BITSTREAM_H + #define L3_BITSTREAM_H + +! #include "util.h" + #include "encoder.h" + + void III_format_bitstream( int bitsPerFrame, +*************** +*** 28,34 **** + int anc_bits ); + + int HuffmanCode( int table_select, int x, int y, unsigned *code, unsigned int *extword, int *codebits, int *extbits ); +! void III_FlushBitstream(); + + int abs_and_sign( int *x ); /* returns signx and changes *x to abs(*x) */ + +--- 28,34 ---- + int anc_bits ); + + int HuffmanCode( int table_select, int x, int y, unsigned *code, unsigned int *extword, int *codebits, int *extbits ); +! void III_FlushBitstream(void); + + int abs_and_sign( int *x ); /* returns signx and changes *x to abs(*x) */ + +diff -r -c -N encoder/l3psy.c lame3.50/l3psy.c +*** encoder/l3psy.c Wed Jan 22 02:43:15 1997 +--- lame3.50/l3psy.c Wed Dec 31 17:00:00 1969 +*************** +*** 1,1033 **** +- /********************************************************************** +- * ISO MPEG Audio Subgroup Software Simulation Group (1996) +- * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- * +- * $Id: l3psy.c,v 1.2 1997/01/19 22:28:29 rowlands Exp $ +- * +- * $Log: l3psy.c,v $ +- * Revision 1.2 1997/01/19 22:28:29 rowlands +- * Layer 3 bug fixes from Seymour Shlien +- * +- * Revision 1.1 1996/02/14 04:04:23 rowlands +- * Initial revision +- * +- * Received from Mike Coleman +- **********************************************************************/ +- /********************************************************************** +- * date programmers comment * +- * 2/25/91 Davis Pan start of version 1.0 records * +- * 5/10/91 W. Joseph Carter Ported to Macintosh and Unix. * +- * 7/10/91 Earle Jennings Ported to MsDos. * +- * replace of floats with FLOAT * +- * 2/11/92 W. Joseph Carter Fixed mem_alloc() arg for "absthr". * +- * 3/16/92 Masahiro Iwadare Modification for Layer III * +- * 17/4/93 Masahiro Iwadare Updated for IS Modification * +- **********************************************************************/ +- +- #include "common.h" +- #include "encoder.h" +- #include "l3psy.h" +- #include "l3side.h" +- #include <assert.h> +- +- #define maximum(x,y) ( (x>y) ? x : y ) +- #define minimum(x,y) ( (x<y) ? x : y ) +- +- void L3para_read( double sfreq, int numlines[CBANDS], int partition_l[HBLKSIZE], +- double minval[CBANDS], double qthr_l[CBANDS], double norm_l[CBANDS], +- double s3_l[CBANDS][CBANDS], int partition_s[HBLKSIZE_s], double qthr_s[CBANDS_s], +- double norm_s[CBANDS_s], double SNR_s[CBANDS_s], +- int cbw_l[SBMAX_l], int bu_l[SBMAX_l], int bo_l[SBMAX_l], +- double w1_l[SBMAX_l], double w2_l[SBMAX_l], +- int cbw_s[SBMAX_s], int bu_s[SBMAX_s], int bo_s[SBMAX_s], +- double w1_s[SBMAX_s], double w2_s[SBMAX_s] ); +- +- void L3psycho_anal( short int *buffer, short int savebuf[1344], int chn, int lay, FLOAT snr32[32], +- double sfreq, double ratio_d[21], double ratio_ds[12][3], +- double *pe, gr_info *cod_info ) +- { +- static double ratio[2][21]; +- static double ratio_s[2][12][3]; +- int blocktype; +- unsigned int b, i, j, k; +- double r_prime, phi_prime; /* not FLOAT */ +- FLOAT freq_mult, bval_lo, min_thres, sum_energy; +- double tb, temp1,temp2,temp3; +- +- /* nint(); Layer III */ +- double thr[CBANDS]; +- +- /* The static variables "r", "phi_sav", "new", "old" and "oldest" have */ +- /* to be remembered for the unpredictability measure. For "r" and */ +- /* "phi_sav", the first index from the left is the channel select and */ +- /* the second index is the "age" of the data. */ +- +- +- static FLOAT window_s[BLKSIZE_s] ; +- static int new = 0, old = 1, oldest = 0; +- static int init = 0, flush, sync_flush, syncsize, sfreq_idx; +- static double cw[HBLKSIZE], eb[CBANDS]; +- static double ctb[CBANDS]; +- static double SNR_l[CBANDS], SNR_s[CBANDS_s]; +- static int init_L3; +- static double minval[CBANDS],qthr_l[CBANDS],norm_l[CBANDS]; +- static double qthr_s[CBANDS_s],norm_s[CBANDS_s]; +- static double nb_1[2][CBANDS], nb_2[2][CBANDS]; +- static double s3_l[CBANDS][CBANDS]; /* s3_s[CBANDS_s][CBANDS_s]; */ +- +- /* Scale Factor Bands */ +- static int cbw_l[SBMAX_l],bu_l[SBMAX_l],bo_l[SBMAX_l] ; +- static int cbw_s[SBMAX_s],bu_s[SBMAX_s],bo_s[SBMAX_s] ; +- static double w1_l[SBMAX_l], w2_l[SBMAX_l]; +- static double w1_s[SBMAX_s], w2_s[SBMAX_s]; +- static double en[SBMAX_l], thm[SBMAX_l] ; +- static int blocktype_old[2] ; +- int sb,sblock; +- static int partition_l[HBLKSIZE],partition_s[HBLKSIZE_s]; +- +- +- /* The following static variables are constants. */ +- +- static double nmt = 5.5; +- +- static FLOAT crit_band[27] = {0, 100, 200, 300, 400, 510, 630, 770, +- 920, 1080, 1270,1480,1720,2000,2320, 2700, +- 3150, 3700, 4400,5300,6400,7700,9500,12000, +- 15500,25000,30000}; +- +- static FLOAT bmax[27] = {20.0, 20.0, 20.0, 20.0, 20.0, 17.0, 15.0, +- 10.0, 7.0, 4.4, 4.5, 4.5, 4.5, 4.5, +- 4.5, 4.5, 4.5, 4.5, 4.5, 4.5, 4.5, +- 4.5, 4.5, 4.5, 3.5, 3.5, 3.5}; +- +- /* The following pointer variables point to large areas of memory */ +- /* dynamically allocated by the mem_alloc() function. Dynamic memory */ +- /* allocation is used in order to avoid stack frame or data area */ +- /* overflow errors that otherwise would have occurred at compile time */ +- /* on the Macintosh computer. */ +- +- FLOAT *grouped_c, *grouped_e, *nb, *cb, *ecb, *bc; +- FLOAT *wsamp_r, *wsamp_i, *phi, *energy; +- static FLOAT energy_s[3][256]; +- static FLOAT phi_s[3][256] ; /* 256 samples not 129 */ +- FLOAT *c, *fthr; +- F32 *snrtmp; +- +- static int *numlines ; +- static int *partition; +- static FLOAT *cbval, *rnorm; +- static FLOAT *window; +- static FLOAT *absthr; +- static double *tmn; +- static FCB *s; +- static FHBLK *lthr; +- static F2HBLK *r, *phi_sav; +- +- /* These dynamic memory allocations simulate "automatic" variables */ +- /* placed on the stack. For each mem_alloc() call here, there must be */ +- /* a corresponding mem_free() call at the end of this function. */ +- +- grouped_c = (FLOAT *) mem_alloc(sizeof(FCB), "grouped_c"); +- grouped_e = (FLOAT *) mem_alloc(sizeof(FCB), "grouped_e"); +- nb = (FLOAT *) mem_alloc(sizeof(FCB), "nb"); +- cb = (FLOAT *) mem_alloc(sizeof(FCB), "cb"); +- ecb = (FLOAT *) mem_alloc(sizeof(FCB), "ecb"); +- bc = (FLOAT *) mem_alloc(sizeof(FCB), "bc"); +- wsamp_r = (FLOAT *) mem_alloc(sizeof(FBLK), "wsamp_r"); +- wsamp_i = (FLOAT *) mem_alloc(sizeof(FBLK), "wsamp_i"); +- phi = (FLOAT *) mem_alloc(sizeof(FBLK), "phi"); +- energy = (FLOAT *) mem_alloc(sizeof(FBLK), "energy"); +- c = (FLOAT *) mem_alloc(sizeof(FHBLK), "c"); +- fthr = (FLOAT *) mem_alloc(sizeof(FHBLK), "fthr"); +- snrtmp = (F32 *) mem_alloc(sizeof(F2_32), "snrtmp"); +- +- assert( lay == 3 ); +- if(init==0){ +- +- /* These dynamic memory allocations simulate "static" variables placed */ +- /* in the data space. Each mem_alloc() call here occurs only once at */ +- /* initialization time. The mem_free() function must not be called. */ +- numlines = (int *) mem_alloc(sizeof(ICB), "numlines"); +- partition = (int *) mem_alloc(sizeof(IHBLK), "partition"); +- cbval = (FLOAT *) mem_alloc(sizeof(FCB), "cbval"); +- rnorm = (FLOAT *) mem_alloc(sizeof(FCB), "rnorm"); +- window = (FLOAT *) mem_alloc(sizeof(FBLK), "window"); +- absthr = (FLOAT *) mem_alloc(sizeof(FHBLK), "absthr"); +- tmn = (double *) mem_alloc(sizeof(DCB), "tmn"); +- s = (FCB *) mem_alloc(sizeof(FCBCB), "s"); +- lthr = (FHBLK *) mem_alloc(sizeof(F2HBLK), "lthr"); +- r = (F2HBLK *) mem_alloc(sizeof(F22HBLK), "r"); +- phi_sav = (F2HBLK *) mem_alloc(sizeof(F22HBLK), "phi_sav"); +- +- /*#if 0 */ +- i = sfreq + 0.5; +- switch(i){ +- case 32000: sfreq_idx = 0; break; +- case 44100: sfreq_idx = 1; break; +- case 48000: sfreq_idx = 2; break; +- default: printf("error, invalid sampling frequency: %d Hz\n",i); +- exit(-1); +- } +- printf("absthr[][] sampling frequency index: %d\n",sfreq_idx); +- read_absthr(absthr, sfreq_idx); +- switch(lay){ +- case 1: sync_flush=576; flush=384; syncsize=1024; break; +- case 2: sync_flush=480; flush=576; syncsize=1056; break; +- case 3: sync_flush=768; flush=576; syncsize=1344; break; +- default: printf("Bad lay value:(%d)",lay); exit(-1); break; +- } +- /* #endif */ +- +- /* calculate HANN window coefficients */ +- /* for(i=0;i<BLKSIZE;i++) window[i] =0.5*(1-cos(2.0*PI*i/(BLKSIZE-1.0)));*/ +- for(i=0;i<BLKSIZE;i++) window[i] =0.5*(1-cos(2.0*PI*(i-0.5)/BLKSIZE)); +- for(i=0;i<BLKSIZE_s;i++)window_s[i]=0.5*(1-cos(2.0*PI*(i-0.5)/BLKSIZE_s)); +- /* reset states used in unpredictability measure */ +- for(i=0;i<HBLKSIZE;i++){ +- r[0][0][i]=r[1][0][i]=r[0][1][i]=r[1][1][i]=0; +- phi_sav[0][0][i]=phi_sav[1][0][i]=0; +- phi_sav[0][1][i]=phi_sav[1][1][i]=0; +- lthr[0][i] = 60802371420160.0; +- lthr[1][i] = 60802371420160.0; +- } +- /***************************************************************************** +- * Initialization: Compute the following constants for use later * +- * partition[HBLKSIZE] = the partition number associated with each * +- * frequency line * +- * cbval[CBANDS] = the center (average) bark value of each * +- * partition * +- * numlines[CBANDS] = the number of frequency lines in each partition * +- * tmn[CBANDS] = tone masking noise * +- *****************************************************************************/ +- /* compute fft frequency multiplicand */ +- freq_mult = sfreq/BLKSIZE; +- +- /* calculate fft frequency, then bval of each line (use fthr[] as tmp storage)*/ +- for(i=0;i<HBLKSIZE;i++){ +- temp1 = i*freq_mult; +- j = 1; +- while(temp1>crit_band[j])j++; +- fthr[i]=j-1+(temp1-crit_band[j-1])/(crit_band[j]-crit_band[j-1]); +- } +- partition[0] = 0; +- /* temp2 is the counter of the number of frequency lines in each partition */ +- temp2 = 1; +- cbval[0]=fthr[0]; +- bval_lo=fthr[0]; +- for(i=1;i<HBLKSIZE;i++){ +- if((fthr[i]-bval_lo)>0.33){ +- partition[i]=partition[i-1]+1; +- cbval[partition[i-1]] = cbval[partition[i-1]]/temp2; +- cbval[partition[i]] = fthr[i]; +- bval_lo = fthr[i]; +- numlines[partition[i-1]] = temp2; +- temp2 = 1; +- } +- else { +- partition[i]=partition[i-1]; +- cbval[partition[i]] += fthr[i]; +- temp2++; +- } +- } +- numlines[partition[i-1]] = temp2; +- cbval[partition[i-1]] = cbval[partition[i-1]]/temp2; +- +- /************************************************************************ +- * Now compute the spreading function, s[j][i], the value of the spread-* +- * ing function, centered at band j, for band i, store for later use * +- ************************************************************************/ +- for(j=0;j<CBANDS;j++){ +- for(i=0;i<CBANDS;i++){ +- temp1 = (cbval[i] - cbval[j])*1.05; +- if(temp1>=0.5 && temp1<=2.5){ +- temp2 = temp1 - 0.5; +- temp2 = 8.0 * (temp2*temp2 - 2.0 * temp2); +- } +- else temp2 = 0; +- temp1 += 0.474; +- temp3 = 15.811389+7.5*temp1-17.5*sqrt((double) (1.0+temp1*temp1)); +- if(temp3 <= -100) s[i][j] = 0; +- else { +- temp3 = (temp2 + temp3)*LN_TO_LOG10; +- s[i][j] = exp(temp3); +- } +- } +- } +- +- /* Calculate Tone Masking Noise values */ +- for(j=0;j<CBANDS;j++){ +- temp1 = 15.5 + cbval[j]; +- tmn[j] = (temp1>24.5) ? temp1 : 24.5; +- /* Calculate normalization factors for the net spreading functions */ +- rnorm[j] = 0; +- for(i=0;i<CBANDS;i++){ +- rnorm[j] += s[j][i]; +- } +- } +- init++; +- } +- +- /************************* End of Initialization *****************************/ +- switch(lay) { +- case 1: +- case 2: +- for ( i=0; i<lay; i++) +- { +- /***************************************************************************** +- * Net offset is 480 samples (1056-576) for layer 2; this is because one must* +- * stagger input data by 256 samples to synchronize psychoacoustic model with* +- * filter bank outputs, then stagger so that center of 1024 FFT window lines * +- * up with center of 576 "new" audio samples. * +- * * +- * For layer 1, the input data still needs to be staggered by 256 samples, * +- * then it must be staggered again so that the 384 "new" samples are centered* +- * in the 1024 FFT window. The net offset is then 576 and you need 448 "new"* +- * samples for each iteration to keep the 384 samples of interest centered * +- *****************************************************************************/ +- for (j=0; j<syncsize; j++) +- { +- if (j < (sync_flush) ) +- savebuf[j] = savebuf[j+flush]; +- else +- savebuf[j] = *buffer++; +- +- /**window data with HANN window***********************************************/ +- if (j<BLKSIZE) +- { +- wsamp_r[j] = window[j]*((FLOAT) savebuf[j]); +- wsamp_i[j] = 0; +- } +- } +- /**Compute FFT****************************************************************/ +- fft(wsamp_r,wsamp_i,energy,phi,1024); +- /***************************************************************************** +- * calculate the unpredictability measure, given energy[f] and phi[f] * +- *****************************************************************************/ +- for(j=0; j<HBLKSIZE; j++){ +- r_prime = 2.0 * r[chn][old][j] - r[chn][oldest][j]; +- phi_prime = 2.0 * phi_sav[chn][old][j] - phi_sav[chn][oldest][j]; +- r[chn][new][j] = sqrt((double) energy[j]); +- phi_sav[chn][new][j] = phi[j]; +- temp1 = r[chn][new][j] * cos((double) phi[j]) +- - r_prime * cos(phi_prime); +- temp2=r[chn][new][j] * sin((double) phi[j]) +- - r_prime * sin(phi_prime); +- temp3=r[chn][new][j] + fabs(r_prime); +- if(temp3 != 0)c[j]=sqrt(temp1*temp1+temp2*temp2)/temp3; +- else c[j] = 0; +- } +- /*only update data "age" pointers after you are done with the second channel */ +- /*for layer 1 computations, for the layer 2 double computations, the pointers*/ +- /*are reset automatically on the second pass */ +- if(lay==2 || chn==1){ +- if(new==0){new = 1; oldest = 1;} +- else {new = 0; oldest = 0;} +- if(old==0)old = 1; else old = 0; +- } +- /***************************************************************************** +- * Calculate the grouped, energy-weighted, unpredictability measure, * +- * grouped_c[], and the grouped energy. grouped_e[] * +- *****************************************************************************/ +- for(j=1;j<CBANDS;j++){ +- grouped_e[j] = 0; +- grouped_c[j] = 0; +- } +- grouped_e[0] = energy[0]; +- grouped_c[0] = energy[0]*c[0]; +- for(j=1;j<HBLKSIZE;j++){ +- grouped_e[partition[j]] += energy[j]; +- grouped_c[partition[j]] += energy[j]*c[j]; +- } +- /***************************************************************************** +- * convolve the grouped energy-weighted unpredictability measure * +- * and the grouped energy with the spreading function, s[j][k] * +- *****************************************************************************/ +- for(j=0;j<CBANDS;j++){ +- ecb[j] = 0; +- cb[j] = 0; +- for(k=0;k<CBANDS;k++){ +- if(s[j][k] != 0.0){ +- ecb[j] += s[j][k]*grouped_e[k]; +- cb[j] += s[j][k]*grouped_c[k]; +- } +- } +- if(ecb[j] !=0)cb[j] = cb[j]/ecb[j]; +- else cb[j] = 0; +- } +- /***************************************************************************** +- * Calculate the required SNR for each of the frequency partitions * +- * this whole section can be accomplished by a table lookup * +- *****************************************************************************/ +- for(j=0;j<CBANDS;j++){ +- if(cb[j]<.05)cb[j]=0.05; +- else if(cb[j]>.5)cb[j]=0.5; +- /* tb = -0.434294482*log((double) cb[j])-0.301029996; */ +- tb = -0.43 *log((double) cb[j]) - 0.29 ; +- if(tb<0.0) tb=0.0; else if(tb>1.0) tb=1.0; +- bc[j] = tmn[j]*tb + nmt*(1.0-tb); +- k = cbval[j] + 0.5; +- bc[j] = (bc[j] > bmax[k]) ? bc[j] : bmax[k]; +- bc[j] = exp((double) -bc[j]*LN_TO_LOG10); +- } +- /***************************************************************************** +- * Calculate the permissible noise energy level in each of the frequency * +- * partitions. Include absolute threshold and pre-echo controls * +- * this whole section can be accomplished by a table lookup * +- *****************************************************************************/ +- for(j=0;j<CBANDS;j++) +- if(rnorm[j] && numlines[j]) +- nb[j] = ecb[j]*bc[j]/(rnorm[j]*numlines[j]); +- else nb[j] = 0; +- for(j=0;j<HBLKSIZE;j++){ +- temp1=nb[partition[j]]; /* preliminary threshold */ +- temp1=(temp1>absthr[j])?temp1:absthr[j]; +- /*do not use pre-echo control for layer 2 because it may do bad things to the*/ +- /* MUSICAM bit allocation algorithm */ +- if(lay==1){ +- fthr[j] = (temp1 < lthr[chn][j]) ? temp1 : lthr[chn][j]; +- temp2 = temp1 * 0.00316; +- fthr[j] = (temp2 > fthr[j]) ? temp2 : fthr[j]; +- } +- else fthr[j] = temp1; +- lthr[chn][j] = LXMIN*temp1; +- } +- /***************************************************************************** +- * Translate the 512 threshold values to the 32 filter bands of the coder * +- *****************************************************************************/ +- for(j=0;j<193;j += 16){ +- min_thres = fthr[j]; +- sum_energy = energy[j]; +- for(k=1;k<17;k++){ +- if(min_thres>fthr[j+k]) min_thres = fthr[j+k]; +- sum_energy += energy[j+k]; +- } +- snrtmp[i][j/16] = sum_energy/(min_thres * 17.0); +- snrtmp[i][j/16] = 4.342944819 * log((double)snrtmp[i][j/16]); +- } +- for(j=208;j<(HBLKSIZE-1);j += 16){ +- min_thres = 0.0; +- sum_energy = 0.0; +- for(k=0;k<17;k++){ +- min_thres += fthr[j+k]; +- sum_energy += energy[j+k]; +- } +- snrtmp[i][j/16] = sum_energy/min_thres; +- snrtmp[i][j/16] = 4.342944819 * log((double)snrtmp[i][j/16]); +- } +- /***************************************************************************** +- * End of Psychoacuostic calculation loop * +- *****************************************************************************/ +- } +- for(i=0; i<32; i++){ +- if(lay==2) /* if(lay==2 && chn==2) MI */ +- snr32[i]=(snrtmp[0][i]>snrtmp[1][i])?snrtmp[0][i]:snrtmp[1][i]; +- else snr32[i]=snrtmp[0][i]; +- } +- break; +- +- /*************************************************************************/ +- /** LAYER 3 */ +- /*************************************************************************/ +- +- case 3: +- if ( init_L3 == 0 ) +- { +- L3para_read( sfreq,numlines,partition_l,minval,qthr_l,norm_l,s3_l, +- partition_s,qthr_s,norm_s,SNR_s, +- cbw_l,bu_l,bo_l,w1_l,w2_l, cbw_s,bu_s,bo_s,w1_s,w2_s ); +- init_L3 ++ ; +- } +- +- for ( j = 0; j < 21; j++ ) +- ratio_d[j] = ratio[chn][j]; +- for ( j = 0; j < 12; j++ ) +- for ( i = 0; i < 3; i++ ) +- ratio_ds[j][i] = ratio_s[chn][j][i]; +- +- if ( chn == 0 ) +- if ( new == 0 ) +- { +- new = 1; +- old = 0; +- oldest = 1; +- } +- else +- { +- new = 0; +- old = 1; +- oldest = 0; +- } +- +- +- /********************************************************************** +- * Delay signal by sync_flush=768 samples * +- **********************************************************************/ +- +- for ( j = 0; j < sync_flush; j++ ) /* for long window samples */ +- savebuf[j] = savebuf[j+flush]; +- +- for ( j = sync_flush; j < syncsize; j++ ) +- savebuf[j] = *buffer++; +- +- for ( j = 0; j < BLKSIZE; j++ ) +- { /**window data with HANN window**/ +- wsamp_r[j] = window[j] * savebuf[j]; +- wsamp_i[j] = 0.0; +- } +- +- +- /********************************************************************** +- * compute unpredicatability of first six spectral lines * +- **********************************************************************/ +- +- fft( wsamp_r, wsamp_i, energy, phi, 1024 ); /**long FFT**/ +- for ( j = 0; j < 6; j++ ) +- { /* calculate unpredictability measure cw */ +- r_prime = 2.0 * r[chn][old][j] - r[chn][oldest][j]; +- phi_prime = 2.0 * phi_sav[chn][old][j]-phi_sav[chn][oldest][j]; +- r[chn][new][j] = sqrt((double) energy[j]); +- phi_sav[chn][new][j] = phi[j]; +- temp1 = r[chn][new][j] * cos((double) phi[j]) +- - r_prime * cos(phi_prime); +- temp2 = r[chn][new][j] * sin((double) phi[j]) +- - r_prime * sin(phi_prime); +- temp3=r[chn][new][j] + fabs(r_prime); +- +- if ( temp3 != 0.0 ) +- cw[j] = sqrt( temp1*temp1+temp2*temp2 ) / temp3; +- else +- cw[j] = 0; +- } +- +- +- /********************************************************************** +- * compute unpredicatibility of next 200 spectral lines * +- **********************************************************************/ +- for ( sblock = 0; sblock < 3; sblock++ ) +- { /**window data with HANN window**/ +- for ( j = 0, k = 128 * (2 + sblock); j < 256; j++, k++ ) +- { +- wsamp_r[j] = window_s[j]* savebuf[k]; +- wsamp_i[j] = 0.0; +- } /* short FFT*/ +- +- fft( wsamp_r, wsamp_i, &energy_s[sblock][0], &phi_s[sblock][0], 256 ); +- } +- +- sblock = 1; +- +- for ( j = 6; j < 206; j += 4 ) +- {/* calculate unpredictability measure cw */ +- double r2, phi2, temp1, temp2, temp3; +- +- k = (j+2) / 4; +- r_prime = 2.0 * sqrt((double) energy_s[0][k]) +- - sqrt((double) energy_s[2][k]); +- phi_prime = 2.0 * phi_s[0][k] - phi_s[2][k]; +- r2 = sqrt((double) energy_s[1][k]); +- phi2 = phi_s[1][k]; +- temp1 = r2 * cos( phi2 ) - r_prime * cos( phi_prime ); +- temp2 = r2 * sin( phi2 ) - r_prime * sin( phi_prime ); +- temp3 = r2 + fabs( r_prime ); +- if ( temp3 != 0.0 ) +- cw[j] = sqrt( temp1 * temp1 + temp2 * temp2 ) / temp3; +- else +- cw[j] = 0.0; +- cw[j+1] = cw[j+2] = cw[j+3] = cw[j]; +- } +- +- +- /********************************************************************** +- * Set unpredicatiblility of remaining spectral lines to 0.4 * +- **********************************************************************/ +- for ( j = 206; j < HBLKSIZE; j++ ) +- cw[j] = 0.4; +- +- +- +- /********************************************************************** +- * Calculate the energy and the unpredictability in the threshold * +- * calculation partitions * +- **********************************************************************/ +- +- for ( b = 0; b < CBANDS; b++ ) +- { +- eb[b] = 0.0; +- cb[b] = 0.0; +- } +- for ( j = 0; j < HBLKSIZE; j++ ) +- { +- int tp = partition_l[j]; +- if ( tp >= 0 ) +- { +- eb[tp] += energy[j]; +- cb[tp] += cw[j] * energy[j]; +- } +- } +- +- +- /********************************************************************** +- * convolve the partitioned energy and unpredictability * +- * with the spreading function, s3_l[b][k] * +- ******************************************************************** */ +- +- for ( b = 0; b < CBANDS; b++ ) +- { +- ecb[b] = 0.0; +- ctb[b] = 0.0; +- } +- for ( b = 0;b < CBANDS; b++ ) +- { +- for ( k = 0; k < CBANDS; k++ ) +- { +- ecb[b] += s3_l[b][k] * eb[k]; /* sprdngf for Layer III */ +- ctb[b] += s3_l[b][k] * cb[k]; +- } +- } +- +- /* calculate the tonality of each threshold calculation partition */ +- /* calculate the SNR in each threshhold calculation partition */ +- +- for ( b = 0; b < CBANDS; b++ ) +- { +- double cbb,tbb; +- if (ecb[b] != 0.0 ) +- { +- cbb = ctb[b]/ecb[b]; +- if (cbb <0.01) cbb = 0.01; +- cbb = log( cbb); +- } +- else +- cbb = 0.0 ; +- tbb = -0.299 - 0.43*cbb; /* conv1=-0.299, conv2=-0.43 */ +- tbb = minimum( 1.0, maximum( 0.0, tbb) ) ; /* 0<tbb<1 */ +- SNR_l[b] = maximum( minval[b], 29.0*tbb+6.0*(1.0-tbb) ); +- } /* TMN=29.0,NMT=6.0 for all calculation partitions */ +- +- for ( b = 0; b < CBANDS; b++ ) /* calculate the threshold for each partition */ +- nb[b] = ecb[b] * norm_l[b] * exp( -SNR_l[b] * LN_TO_LOG10 ); +- +- for ( b = 0; b < CBANDS; b++ ) +- { /* pre-echo control */ +- double temp_1; /* BUG of IS */ +- temp_1 = minimum( nb[b], minimum(2.0*nb_1[chn][b],16.0*nb_2[chn][b]) ); +- thr[b] = maximum( qthr_l[b], temp_1 );/* rpelev=2.0, rpelev2=16.0 */ +- nb_2[chn][b] = nb_1[chn][b]; +- nb_1[chn][b] = nb[b]; +- } +- +- +- *pe = 0.0; /* calculate percetual entropy */ +- for ( b = 0; b < CBANDS; b++ ) +- { +- double tp ; +- tp = minimum( 0.0, log((thr[b]+1.0) / (eb[b]+1.0) ) ) ; /*not log*/ +- *pe -= numlines[b] * tp ; +- } /* thr[b] -> thr[b]+1.0 : for non sound portition */ +- +- #define switch_pe 1800 +- blocktype = NORM_TYPE; +- +- +- if ( *pe < switch_pe ) +- { /* no attack : use long blocks */ +- switch( blocktype_old[chn] ) +- { +- case NORM_TYPE: +- case STOP_TYPE: +- blocktype = NORM_TYPE; +- break; +- +- case SHORT_TYPE: +- blocktype = STOP_TYPE; +- break; +- +- case START_TYPE: +- fprintf( stderr, "Error in block selecting\n" ); +- abort(); +- break; /* problem */ +- } +- +- /* threshold calculation (part 2) */ +- for ( sb = 0; sb < SBMAX_l; sb++ ) +- { +- en[sb] = w1_l[sb] * eb[bu_l[sb]] + w2_l[sb] * eb[bo_l[sb]]; +- thm[sb] = w1_l[sb] *thr[bu_l[sb]] + w2_l[sb] * thr[bo_l[sb]]; +- for ( b = bu_l[sb]+1; b < bo_l[sb]; b++ ) +- { +- en[sb] += eb[b]; +- thm[sb] += thr[b]; +- } +- if ( en[sb] != 0.0 ) +- ratio[chn][sb] = thm[sb]/en[sb]; +- else +- ratio[chn][sb] = 0.0; +- } +- } +- else +- { +- /* attack : use short blocks */ +- blocktype = SHORT_TYPE; +- +- if ( blocktype_old[chn] == NORM_TYPE ) +- blocktype_old[chn] = START_TYPE; +- if ( blocktype_old[chn] == STOP_TYPE ) +- blocktype_old[chn] = SHORT_TYPE ; +- +- /* threshold calculation for short blocks */ +- +- for ( sblock = 0; sblock < 3; sblock++ ) +- { +- for ( b = 0; b < CBANDS_s; b++ ) +- { +- eb[b] = 0.0; +- ecb[b] = 0.0; +- } +- for ( j = 0; j < HBLKSIZE_s; j++ ) +- eb[partition_s[j]] += energy_s[sblock][j]; +- for ( b = 0; b < CBANDS_s; b++ ) +- for ( k = 0; k < CBANDS_s; k++ ) +- ecb[b] += s3_l[b][k] * eb[k]; +- for ( b = 0; b < CBANDS_s; b++ ) +- { +- nb[b] = ecb[b] * norm_l[b] * exp( (double) SNR_s[b] * LN_TO_LOG10 ); +- thr[b] = maximum (qthr_s[b],nb[b]); +- } +- for ( sb = 0; sb < SBMAX_s; sb++ ) +- { +- en[sb] = w1_s[sb] * eb[bu_s[sb]] + w2_s[sb] * eb[bo_s[sb]]; +- thm[sb] = w1_s[sb] *thr[bu_s[sb]] + w2_s[sb] * thr[bo_s[sb]]; +- for ( b = bu_s[sb]+1; b < bo_s[sb]; b++ ) +- { +- en[sb] += eb[b]; +- thm[sb] += thr[b]; +- } +- if ( en[sb] != 0.0 ) +- ratio_s[chn][sb][sblock] = thm[sb]/en[sb]; +- else +- ratio_s[chn][sb][sblock] = 0.0; +- } +- } +- } +- +- cod_info->block_type = blocktype_old[chn]; +- blocktype_old[chn] = blocktype; +- +- if ( cod_info->block_type == NORM_TYPE ) +- cod_info->window_switching_flag = 0; +- else +- cod_info->window_switching_flag = 1; +- cod_info->mixed_block_flag = 0; +- break; +- +- +- default: +- printf("error, invalid MPEG/audio coding layer: %d\n",lay); +- } +- +- /* These mem_free() calls must correspond with the mem_alloc() calls */ +- /* used at the beginning of this function to simulate "automatic" */ +- /* variables placed on the stack. */ +- +- mem_free((void **) &grouped_c); +- mem_free((void **) &grouped_e); +- mem_free((void **) &nb); +- mem_free((void **) &cb); +- mem_free((void **) &ecb); +- mem_free((void **) &bc); +- mem_free((void **) &wsamp_r); +- mem_free((void **) &wsamp_i); +- mem_free((void **) &phi); +- mem_free((void **) &energy); +- mem_free((void **) &c); +- mem_free((void **) &fthr); +- mem_free((void **) &snrtmp); +- } +- #ifdef DEBUG +- #undef DEBUG +- #endif +- +- +- void L3para_read(double sfreq, int *numlines, int *partition_l, double *minval, double *qthr_l, double *norm_l, double (*s3_l)[63], int *partition_s, double *qthr_s, double *norm_s, double *SNR, int *cbw_l, int *bu_l, int *bo_l, double *w1_l, double *w2_l, int *cbw_s, int *bu_s, int *bo_s, double *w1_s, double *w2_s) +- { +- double freq_tp; +- static double bval_l[CBANDS], bval_s[CBANDS]; +- int cbmax, cbmax_tp; +- static double s3_s[CBANDS][CBANDS]; +- +- FILE *fin; +- char tp[256]; +- int sbmax ; +- int i,j,k,k2,loop, part_max ; +- +- fin = OpenTableFile( "psy_data" ); +- if (fin == NULL) +- exit( 1 ); +- +- /* Read long block data */ +- +- for(loop=0;loop<6;loop++) +- { +- fscanf(fin,"freq = %lf partition = %d\n",&freq_tp,&cbmax_tp); +- cbmax_tp++; +- #ifdef DEBUG +- printf("freq = %f partition = %d\n",freq_tp,cbmax); +- #endif +- +- if (sfreq == freq_tp ) +- { +- cbmax = cbmax_tp; +- for(i=0,k2=0;i<cbmax_tp;i++) +- { +- fscanf(fin, +- "No=%d #lines=%d minval=%lf qthr=%lf norm=%lf bval=%lf\n", +- &j,&numlines[i],&minval[i],&qthr_l[i],&norm_l[i],&bval_l[i]); +- if (j!=i) +- { printf("please check \"psy_data\""); +- exit(-1); +- } +- for(k=0;k<numlines[i];k++) +- partition_l[k2++] = i ; +- #ifdef DEBUG +- printf("No=%2d #lines=%2d minval=%4.1f qthr=%8.3f norm=%5.3f bval=%8.3f\n", +- i,numlines[i],minval[i],qthr_l[i],norm_l[i],bval_l[i] ); +- #endif +- } +- } +- else +- { +- for(j=0;j<cbmax_tp;j++) +- { +- fgets(tp,255,fin); +- sscanf(tp,"No=%d %s\n",&i,tp); +- if (j!=i) +- { printf("please check \"psy_data.\"\n"); +- exit(-1); +- } +- } +- } +- } +- +- /************************************************************************ +- * Now compute the spreading function, s[j][i], the value of the spread-* +- * ing function, centered at band j, for band i, store for later use * +- ************************************************************************/ +- #ifdef DEBUG +- printf("freq = %f\n",sfreq); +- #endif +- part_max = cbmax ; +- for(i=0;i<part_max;i++) +- { +- double tempx,x,tempy,temp; +- for(j=0;j<part_max;j++) +- { +- tempx = (bval_l[i] - bval_l[j])*1.05; +- if (j>=i) tempx = (bval_l[i] - bval_l[j])*3.0; +- else tempx = (bval_l[i] - bval_l[j])*1.5; +- /* if (j>=i) tempx = (bval_l[j] - bval_l[i])*3.0; +- else tempx = (bval_l[j] - bval_l[i])*1.5; */ +- if(tempx>=0.5 && tempx<=2.5) +- { +- temp = tempx - 0.5; +- x = 8.0 * (temp*temp - 2.0 * temp); +- } +- else x = 0.0; +- tempx += 0.474; +- tempy = 15.811389 + 7.5*tempx - 17.5*sqrt(1.0+tempx*tempx); +- if (tempy <= -60.0) s3_l[i][j] = 0.0; +- else s3_l[i][j] = exp( (x + tempy)*LN_TO_LOG10 ); +- #ifdef DEBUG_S3 +- printf("s[%d][%d]=%f\n",i,j,s3_l[i][j]); +- #endif +- +- #ifdef DEBUGP +- printf("j=%d i=%d tempy=%f s[i][j]=%f \n",i,j,tempy,s[i][j]); +- minval[i] = bmax[j-1]; +- printf("minval[%d] = %f, j-1=%d %f\n",i, minval[i] , j,fthr[i]) ; +- #endif +- } +- } +- +- +- /* Read short block data */ +- +- for(loop=0;loop<6;loop++) +- { +- fscanf(fin,"freq = %lf partition = %d\n",&freq_tp,&cbmax_tp); +- cbmax_tp++; +- #ifdef DEBUG +- printf("freq = %f partition = %d\n",freq_tp,cbmax); +- #endif +- +- if (sfreq == freq_tp ) +- { +- cbmax = cbmax_tp; +- for(i=0,k2=0;i<cbmax_tp;i++) +- { +- fscanf(fin, +- "No=%d #lines=%d qthr=%lf norm=%lf SNR=%lf bval=%lf\n", +- &j,&numlines[i],&qthr_s[i],&norm_s[i],&SNR[i],&bval_s[i]); +- if (j!=i) +- { printf("please check \"psy_data\""); +- exit(-1); +- } +- for(k=0;k<numlines[i];k++) +- partition_s[k2++] = i ; +- #ifdef DEBUG +- printf("No=%2d #lines=%2d qthr=%8.3f norm=%5.3f SNR=%6.3f bval=%8.3f\n", +- i,numlines[i],qthr_s[i],norm_s[i],SNR[i],bval_s[i] ); +- #endif +- } +- } +- else +- { +- for(j=0;j<cbmax_tp;j++) +- { +- fgets(tp,255,fin); +- sscanf(tp,"No=%d %s\n",&i,tp); +- if (j!=i) +- { printf("please check \"psy_data.\"\n"); +- exit(-1); +- } +- } +- } +- } +- +- /************************************************************************ +- * Now compute the spreading function, s[j][i], the value of the spread-* +- * ing function, centered at band j, for band i, store for later use * +- ************************************************************************/ +- #ifdef DEBUG_S3 +- fpp=fopen("s3_s","w"); +- #endif +- part_max = cbmax ; +- for(i=0;i<part_max;i++) +- { +- double tempx,x,tempy,temp; +- for(j=0;j<part_max;j++) +- { +- tempx = (bval_s[i] - bval_s[j])*1.05; +- if (j>=i) tempx = (bval_s[i] - bval_s[j])*3.0; +- else tempx = (bval_s[i] - bval_s[j])*1.5; +- if(tempx>=0.5 && tempx<=2.5) +- { +- temp = tempx - 0.5; +- x = 8.0 * (temp*temp - 2.0 * temp); +- } +- else x = 0.0; +- tempx += 0.474; +- tempy = 15.811389 + 7.5*tempx - 17.5*sqrt(1.0+tempx*tempx); +- if (tempy <= -60.0) s3_s[i][j] = 0.0; +- else s3_s[i][j] = exp( (x + tempy)*LN_TO_LOG10 ); +- #ifdef DEBUG_S3 +- fprintf(fpp,"s3_s[%d][%d]=%f\n",i,j,s3_s[i][j]); +- #endif +- #ifdef DEBUGP +- printf("j=%d i=%d tempy=%f s[i][j]=%f \n",i,j,tempy,s[i][j]); +- minval[i] = bmax[j-1]; +- printf("minval[%d] = %f, j-1=%d %f\n",i, minval[i] , j,fthr[i]) ; +- #endif +- } +- } +- #ifdef DEBUG_S3 +- fclose(fpp); +- #endif +- /* Read long block data for converting threshold calculation +- partitions to scale factor bands */ +- +- for(loop=0;loop<6;loop++) +- { +- fscanf(fin,"freq=%lf sb=%d\n",&freq_tp,&sbmax); +- sbmax++; +- #ifdef DEBUG +- printf("freq = %f sb = %d\n",freq_tp,sbmax); +- #endif +- +- if (sfreq == freq_tp) +- { +- for(i=0;i<sbmax;i++) +- { +- fscanf(fin, +- "sb=%d cbw=%d bu=%d bo=%d w1=%lf w2=%lf\n", +- &j,&cbw_l[i],&bu_l[i],&bo_l[i],&w1_l[i],&w2_l[i]); +- if (j!=i) +- { printf("30:please check \"psy_data\"\n"); +- exit(-1); +- } +- #ifdef DEBUG +- printf( +- "sb=%2d cbw=%1d bu=%2d bo=%2d w1=%5.3f w2=%5.3f\n", +- j,cbw_l[i],bu_l[i],bo_l[i],w1_l[i],w2_l[i]); +- #endif +- if (i!=0) +- if ( (bo_l[i] != (bu_l[i]+cbw_l[i])) || +- (fabs(1.0-w1_l[i]-w2_l[i-1]) > 0.01 ) ) +- { printf("31:please check \"psy_data.\"\n"); +- exit(-1); +- } +- } +- } +- else +- { +- for(j=0;j<sbmax;j++) +- { +- fgets(tp,255,fin); +- sscanf(tp,"sb=%d %s\n",&i,tp); +- if (j!=i) +- { printf("please check \"psy_data.\"\n"); +- exit(-1); +- } +- } +- } +- } +- +- /* Read short block data for converting threshold calculation +- partitions to scale factor bands */ +- +- for(loop=0;loop<6;loop++) +- { +- fscanf(fin,"freq=%lf sb=%d\n",&freq_tp,&sbmax); +- sbmax++; +- #ifdef DEBUG +- printf("freq = %f sb = %d\n",freq_tp,sbmax); +- #endif +- +- if (sfreq == freq_tp) +- { +- for(i=0;i<sbmax;i++) +- { +- fscanf(fin, +- "sb=%d cbw=%d bu=%d bo=%d w1=%lf w2=%lf\n", +- &j,&cbw_s[i],&bu_s[i],&bo_s[i],&w1_s[i],&w2_s[i]); +- if (j!=i) +- { printf("30:please check \"psy_data\"\n"); +- exit(-1); +- } +- #ifdef DEBUG +- printf( +- "sb=%2d cbw=%1d bu=%2d bo=%2d w1=%5.3f w2=%5.3f\n", +- j,cbw_s[i],bu_s[i],bo_s[i],w1_s[i],w2_s[i]); +- #endif +- if (i!=0) +- if ( (bo_s[i] != (bu_s[i]+cbw_s[i])) || +- (fabs(1.0-w1_s[i]-w2_s[i-1]) > 0.01 ) ) +- { printf("31:please check \"psy_data.\"\n"); +- exit(-1); +- } +- } +- } +- else +- { +- for(j=0;j<sbmax;j++) +- { +- fgets(tp,255,fin); +- sscanf(tp,"sb=%d %s\n",&i,tp); +- if (j!=i) +- { printf("please check \"psy_data.\"\n"); +- exit(-1); +- } +- } +- } +- } +- +- } +--- 0 ---- +diff -r -c -N encoder/l3psy.h lame3.50/l3psy.h +*** encoder/l3psy.h Wed Jan 22 02:43:16 1997 +--- lame3.50/l3psy.h Wed Dec 31 17:00:00 1969 +*************** +*** 1,37 **** +- /********************************************************************** +- * ISO MPEG Audio Subgroup Software Simulation Group (1996) +- * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- * +- * $Id: l3psy.h,v 1.1 1996/02/14 04:04:23 rowlands Exp $ +- * +- * $Log: l3psy.h,v $ +- * Revision 1.1 1996/02/14 04:04:23 rowlands +- * Initial revision +- * +- * Received from Mike Coleman +- **********************************************************************/ +- +- #ifndef L3PSY_DOT_H_ +- #define L3PSY_DOT_H_ +- /* #define CBANDS 63 */ +- #define CBANDS_s 42 +- #define BLKSIZE_s 256 +- #define HBLKSIZE_s 129 +- #define TCBMAX_l 63 +- #define TCBMAX_s 42 +- #define SBMAX_l 21 +- #define SBMAX_s 12 +- +- /* #define switch_pe 1800 */ +- #define NORM_TYPE 0 +- #define START_TYPE 1 +- #define SHORT_TYPE 2 +- #define STOP_TYPE 3 +- +- /* l3psy.c */ +- #include "l3side.h" +- void L3psycho_anal( short int *buffer, short int savebuf[1344], int chn, int lay, FLOAT snr32[32], +- double sfreq, double ratio_d[21], double ratio_ds[12][3], +- double *pe, gr_info *cod_info ); +- /* void read_absthr(float *absthr, int table); */ +- #endif +--- 0 ---- +diff -r -c -N encoder/l3side.h lame3.50/l3side.h +*** encoder/l3side.h Wed Jan 22 02:43:16 1997 +--- lame3.50/l3side.h Tue Sep 14 18:21:02 1999 +*************** +*** 25,32 **** + + /* #define CBLIMIT_SHORT 12 */ + +- void l3deco(); +- + typedef double D576[576]; + typedef int I576[576]; + typedef double D192_3[192][3]; +--- 25,30 ---- +*************** +*** 39,61 **** + + + typedef struct { +! double l[2][2][21]; +! double s[2][2][12][3]; + } III_psy_ratio; + + typedef struct { + double l[2][2][21]; + double s[2][2][12][3]; + } III_psy_xmin; +- +- typedef struct { +- double xr[576]; +- double xr_s[3][192]; +- double xmin[21]; +- double xmin_s[3][192]; +- int ix[576]; +- int ix_s[3][192]; +- } III_input3; /* ch */ + + typedef struct { + unsigned part2_3_length; +--- 37,50 ---- + + + typedef struct { +! double l[2][4][21]; +! double s[2][4][12][3]; + } III_psy_ratio; + + typedef struct { + double l[2][2][21]; + double s[2][2][12][3]; + } III_psy_xmin; + + typedef struct { + unsigned part2_3_length; +diff -r -c -N encoder/lame.c lame3.50/lame.c +*** encoder/lame.c Wed Dec 31 17:00:00 1969 +--- lame3.50/lame.c Wed Oct 27 16:14:58 1999 +*************** +*** 0 **** +--- 1,1431 ---- ++ /* ++ * LAME MP3 encoding engine ++ * ++ * Copyright (c) 1999 Mark Taylor ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License as published by ++ * the Free Software Foundation; either version 2, or (at your option) ++ * any later version. ++ * ++ * This program is distributed in the hope that it will be useful, ++ * but WITHOUT ANY WARRANTY; without even the implied warranty of ++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the ++ * GNU General Public License for more details. ++ * ++ * You should have received a copy of the GNU General Public License ++ * along with this program; see the file COPYING. If not, write to ++ * the Free Software Foundation, 675 Mass Ave, Cambridge, MA 02139, USA. ++ */ ++ ++ #include <assert.h> ++ ++ #ifdef HAVEGTK ++ #include "gtkanal.h" ++ #include <gtk/gtk.h> ++ #endif ++ #include "util.h" ++ #include "globalflags.h" ++ #include "psymodel.h" ++ #include "fastmdct.h" ++ #include "quantize.h" ++ #include "l3bitstream.h" ++ #include "reservoir.h" ++ #include "formatBitstream.h" ++ #include "version.h" ++ #include "VbrTag.h" ++ #include "id3tag.h" ++ #include "get_audio.h" ++ ++ ++ ++ /* Global flags. defined extern in globalflags.h */ ++ /* default values set in lame_init() */ ++ int allow_diff_short; ++ int ATHonly; ++ int autoconvert; ++ int experimentalX; ++ int experimentalY; ++ int experimentalZ; ++ int force_ms; ++ int fast_mode; ++ long int frameNum; ++ int gtkflag; ++ int g_bWriteVbrTag; ++ int gpsycho; ++ int highq; ++ sound_file_format input_format; ++ int lame_nowrite; ++ double resample_ratio; ++ int sfb21; ++ int silent; ++ int swapbytes; /* force byte swapping */ ++ long totalframes; /* frames: 0..totalframes-1 */ ++ int VBR; ++ int VBR_q; ++ int VBR_min_bitrate; /* 32kbs */ ++ int VBR_max_bitrate; /* 256kbs */ ++ int voice_mode; ++ ++ ++ ++ /* Global variable definitions for "musicin.c" */ ++ ID3TAGDATA id3tag; ++ struct stat sb; ++ Bit_stream_struc bs; ++ III_side_info_t l3_side; ++ frame_params fr_ps; ++ char inPath[MAX_NAME_SIZE]; ++ char outPath[MAX_NAME_SIZE]; ++ int target_bitrate; ++ char *programName; ++ unsigned long num_samples; ++ static layer info; ++ ++ #ifdef BRHIST ++ #include <string.h> ++ #include <termcap.h> ++ void brhist_init(int br_min, int br_max); ++ void brhist_add_count(void); ++ void brhist_disp(void); ++ void brhist_disp_total(void); ++ extern long brhist_temp[15]; ++ int disp_brhist = 1; ++ #endif ++ ++ ++ ++ /************************************************************************ ++ * ++ * usage ++ * ++ * PURPOSE: Writes command line syntax to the file specified by #stderr# ++ * ++ ************************************************************************/ ++ ++ void lame_usage(char *name) /* print syntax & exit */ ++ { ++ fprintf(stderr,"\n"); ++ fprintf(stderr,"USAGE : %s [options] <infile> [outfile]\n",name); ++ fprintf(stderr,"\n<infile> and/or <outfile> can be \"-\", which means stdin/stdout.\n"); ++ fprintf(stderr,"\n"); ++ fprintf(stderr,"OPTIONS :\n"); ++ fprintf(stderr," -m mode (s)tereo, (j)oint, (f)orce or (m)ono (default %c)\n",DFLT_MOD); ++ fprintf(stderr," force = force ms_stereo on all frames. Faster and\n"); ++ fprintf(stderr," uses special Mid & Side masking thresholds\n"); ++ fprintf(stderr," -b <bitrate> set the bitrate, default 128kbps\n"); ++ fprintf(stderr," (for VBR, this sets the allowed minimum bitrate)\n"); ++ fprintf(stderr," -s sfreq sampling frequency of input file(kHz) - default %4.1f\n",DFLT_SFQ); ++ fprintf(stderr," --resample sfreq sampling frequency of output file(kHz)- default=input sfreq\n"); ++ fprintf(stderr," --mp3input input file is a MP3 file\n"); ++ fprintf(stderr," --voice experimental voice mode\n"); ++ fprintf(stderr,"\n"); ++ fprintf(stderr," -v use variable bitrate (VBR)\n"); ++ fprintf(stderr," -V n quality setting for VBR. default n=%i\n",VBR_q); ++ fprintf(stderr," 0=high quality,bigger files. 9=smaller files\n"); ++ fprintf(stderr," -t disable Xing VBR informational tag\n"); ++ fprintf(stderr," --nohist disable VBR histogram display\n"); ++ fprintf(stderr,"\n"); ++ fprintf(stderr," -h use (maybe) quality improvements\n"); ++ fprintf(stderr," -f fast mode (low quality)\n"); ++ fprintf(stderr," -k disable sfb=21 cutoff\n"); ++ fprintf(stderr," -d allow channels to have different blocktypes\n"); ++ fprintf(stderr," --athonly only use the ATH for masking\n"); ++ fprintf(stderr,"\n"); ++ fprintf(stderr," -r input is raw pcm\n"); ++ fprintf(stderr," -x force byte-swapping of input\n"); ++ fprintf(stderr," -a downmix from stereo to mono file for mono encoding\n"); ++ fprintf(stderr," -e emp de-emphasis n/5/c (obsolete)\n"); ++ fprintf(stderr," -p error protection. adds 16bit checksum to every frame\n"); ++ fprintf(stderr," (the checksum is computed correctly)\n"); ++ fprintf(stderr," -c mark as copyright\n"); ++ fprintf(stderr," -o mark as non-original\n"); ++ fprintf(stderr," -S don't print progress report, VBR histograms\n"); ++ fprintf(stderr,"\n"); ++ fprintf(stderr," Specifying any of the following options will add an ID3 tag\n"); ++ fprintf(stderr," --tt <title> title of song (max 30 chars)\n"); ++ fprintf(stderr," --ta <artist> artist who did the song (max 30 chars)\n"); ++ fprintf(stderr," --tl <album> album where it came from (max 30 chars)\n"); ++ fprintf(stderr," --ty <year> year in which the song/album was made (max 4 chars)\n"); ++ fprintf(stderr," --tc <comment> additional info (max 30 chars)\n"); ++ fprintf(stderr,"\n"); ++ #ifdef HAVEGTK ++ fprintf(stderr," -g run graphical analysis on <infile>\n"); ++ #endif ++ display_bitrates(2); ++ exit(1); ++ } ++ ++ ++ ++ ++ ++ ++ /************************************************************************ ++ * ++ * parse_args ++ * ++ * PURPOSE: Sets encoding parameters to the specifications of the ++ * command line. Default settings are used for parameters ++ * not specified in the command line. ++ * ++ * If the input file is in WAVE or AIFF format, the sampling frequency is read ++ * from the AIFF header. ++ * ++ * The input and output filenames are read into #inpath# and #outpath#. ++ * ++ ************************************************************************/ ++ void lame_parse_args(int argc, char **argv) ++ { ++ FLOAT srate; ++ int brate; ++ layer *info = fr_ps.header; ++ int err = 0, i = 0; ++ int mode_given=0; ++ int num_channels=2; /* number of channels of INPUT file */ ++ int samplerate,resamplerate=0; ++ int framesize; ++ double compression_ratio; ++ ++ /* preset defaults */ ++ programName = argv[0]; ++ inPath[0] = '\0'; outPath[0] = '\0'; ++ info->lay = DFLT_LAY; ++ switch(DFLT_MOD) { ++ case 's': info->mode = MPG_MD_STEREO; info->mode_ext = MPG_MD_LR_LR; break; ++ case 'd': info->mode = MPG_MD_DUAL_CHANNEL; info->mode_ext=MPG_MD_LR_LR; break; ++ case 'j': info->mode = MPG_MD_JOINT_STEREO; info->mode_ext=MPG_MD_LR_LR; break; ++ case 'm': info->mode = MPG_MD_MONO; info->mode_ext = MPG_MD_LR_LR; break; ++ default: ++ fprintf(stderr, "%s: Bad mode dflt %c\n", programName, DFLT_MOD); ++ abort(); ++ } ++ samplerate = 1000*DFLT_SFQ; ++ if((info->sampling_frequency = SmpFrqIndex((long)samplerate, &info->version)) < 0) { ++ fprintf(stderr, "%s: bad sfrq default %.2f\n", programName, DFLT_SFQ); ++ abort(); ++ } ++ ++ info->bitrate_index = 9; ++ brate = 0; ++ switch(DFLT_EMP) { ++ case 'n': info->emphasis = 0; break; ++ case '5': info->emphasis = 1; break; ++ case 'c': info->emphasis = 3; break; ++ default: ++ fprintf(stderr, "%s: Bad emph dflt %c\n", programName, DFLT_EMP); ++ abort(); ++ } ++ info->copyright = 0; ++ info->original = 1; ++ info->error_protection = FALSE; ++ ++ #ifndef _BLADEDLL ++ id3_inittag(&id3tag); ++ id3tag.used = 0; ++ #endif ++ ++ /* process args */ ++ while(++i<argc && err == 0) { ++ char c, *token, *arg, *nextArg; ++ int argUsed; ++ ++ token = argv[i]; ++ if(*token++ == '-') { ++ if(i+1 < argc) nextArg = argv[i+1]; ++ else nextArg = ""; ++ argUsed = 0; ++ if (! *token) { ++ /* The user wants to use stdin and/or stdout. */ ++ if(inPath[0] == '\0') strncpy(inPath, argv[i],MAX_NAME_SIZE); ++ else if(outPath[0] == '\0') strncpy(outPath, argv[i],MAX_NAME_SIZE); ++ } ++ if (*token == '-') { ++ /* GNU style */ ++ token++; ++ ++ if (strcmp(token, "resample")==0) { ++ argUsed=1; ++ srate = atof( nextArg ); ++ /* samplerate = rint( 1000.0 * srate ); $A */ ++ resamplerate = (( 1000.0 * srate ) + 0.5); ++ if (srate < 1) { ++ fprintf(stderr,"Must specify samplerate with --resample\n"); ++ exit(1); ++ } ++ } ++ else if (strcmp(token, "mp3input")==0) { ++ input_format=sf_mp3; ++ } ++ else if (strcmp(token, "voice")==0) { ++ voice_mode=1; ++ } ++ else if (strcmp(token, "athonly")==0) { ++ ATHonly=1; ++ } ++ else if (strcmp(token, "nohist")==0) { ++ #ifdef BRHIST ++ disp_brhist = 0; ++ #endif ++ } ++ /* options for ID3 tag */ ++ else if (strcmp(token, "tt")==0) { ++ id3tag.used=1; argUsed = 1; ++ strncpy(id3tag.title, nextArg, 30); ++ } ++ else if (strcmp(token, "ta")==0) { ++ id3tag.used=1; argUsed = 1; ++ strncpy(id3tag.artist, nextArg, 30); ++ } ++ else if (strcmp(token, "tl")==0) { ++ id3tag.used=1; argUsed = 1; ++ strncpy(id3tag.album, nextArg, 30); ++ } ++ else if (strcmp(token, "ty")==0) { ++ id3tag.used=1; argUsed = 1; ++ strncpy(id3tag.year, nextArg, 4); ++ } ++ else if (strcmp(token, "tc")==0) { ++ id3tag.used=1; argUsed = 1; ++ strncpy(id3tag.comment, nextArg, 30); ++ } ++ else ++ { ++ fprintf(stderr,"%s: unrec option --%s\n", ++ programName, token); ++ } ++ i += argUsed; ++ ++ } else while( (c = *token++) ) { ++ if(*token ) arg = token; ++ else arg = nextArg; ++ switch(c) { ++ case 'm': argUsed = 1; ++ mode_given=1; ++ if (*arg == 's') ++ { info->mode = MPG_MD_STEREO; info->mode_ext = MPG_MD_LR_LR; } ++ else if (*arg == 'd') ++ { info->mode = MPG_MD_DUAL_CHANNEL; info->mode_ext=MPG_MD_LR_LR; } ++ else if (*arg == 'j') ++ { info->mode = MPG_MD_JOINT_STEREO; info->mode_ext=MPG_MD_LR_LR; } ++ else if (*arg == 'f') ++ { info->mode = MPG_MD_JOINT_STEREO; force_ms=1; info->mode_ext=MPG_MD_LR_LR; } ++ else if (*arg == 'm') ++ { info->mode = MPG_MD_MONO; info->mode_ext = MPG_MD_LR_LR; } ++ else { ++ fprintf(stderr,"%s: -m mode must be s/d/j/f/m not %s\n", ++ programName, arg); ++ err = 1; ++ } ++ break; ++ case 'V': argUsed = 1; VBR = 1; ++ if (*arg == '0') ++ { VBR_q=0; } ++ else if (*arg == '1') ++ { VBR_q=1; } ++ else if (*arg == '2') ++ { VBR_q=2; } ++ else if (*arg == '3') ++ { VBR_q=3; } ++ else if (*arg == '4') ++ { VBR_q=4; } ++ else if (*arg == '5') ++ { VBR_q=5; } ++ else if (*arg == '6') ++ { VBR_q=6; } ++ else if (*arg == '7') ++ { VBR_q=7; } ++ else if (*arg == '8') ++ { VBR_q=8; } ++ else if (*arg == '9') ++ { VBR_q=9; } ++ else { ++ fprintf(stderr,"%s: -V n must be 0-9 not %s\n", ++ programName, arg); ++ err = 1; ++ } ++ break; ++ ++ case 's': ++ argUsed = 1; ++ srate = atof( arg ); ++ /* samplerate = rint( 1000.0 * srate ); $A */ ++ samplerate = (( 1000.0 * srate ) + 0.5); ++ break; ++ case 'b': ++ argUsed = 1; ++ brate = atoi(arg); ++ break; ++ case 't': /* dont write VBR tag */ ++ g_bWriteVbrTag=0; ++ break; ++ case 'r': /* force raw pcm input file */ ++ #ifdef LIBSNDFILE ++ fprintf(stderr,"WARNING: libsndfile may ignore -r and perform fseek's on the input.\n"); ++ fprintf(stderr,"Compile without libsndfile if this is a problem.\n"); ++ #endif ++ input_format=sf_raw; ++ break; ++ case 'x': /* force byte swapping */ ++ swapbytes=TRUE; ++ break; ++ case 'p': /* (jo) error_protection: add crc16 information to stream */ ++ info->error_protection = 1; ++ break; ++ case 'a': /* autoconvert input file from stereo to mono - for mono mp3 encoding */ ++ autoconvert = TRUE; ++ break; ++ case 'h': ++ highq = TRUE; ++ break; ++ case 'k': ++ sfb21 = FALSE; ++ break; ++ case 'd': ++ allow_diff_short = 1; ++ break; ++ case 'v': ++ VBR = 1; ++ break; ++ case 'S': ++ silent = TRUE; ++ break; ++ case 'X': argUsed = 1; experimentalX = 0; ++ if (*arg == '0') ++ { experimentalX=0; } ++ else if (*arg == '1') ++ { experimentalX=1; } ++ else if (*arg == '2') ++ { experimentalX=2; } ++ else if (*arg == '3') ++ { experimentalX=3; } ++ else if (*arg == '4') ++ { experimentalX=4; } ++ else if (*arg == '5') ++ { experimentalX=5; } ++ else { ++ fprintf(stderr,"%s: -X n must be 0-5 not %s\n", ++ programName, arg); ++ err = 1; ++ } ++ break; ++ ++ ++ case 'Y': ++ experimentalY = TRUE; ++ break; ++ case 'Z': ++ experimentalZ = TRUE; ++ break; ++ case 'f': ++ fast_mode = 1; ++ break; ++ #ifdef HAVEGTK ++ case 'g': /* turn on gtk analysis */ ++ gtkflag = TRUE; ++ break; ++ #endif ++ case 'e': argUsed = 1; ++ if (*arg == 'n') info->emphasis = 0; ++ else if (*arg == '5') info->emphasis = 1; ++ else if (*arg == 'c') info->emphasis = 3; ++ else { ++ fprintf(stderr,"%s: -e emp must be n/5/c not %s\n", ++ programName, arg); ++ err = 1; ++ } ++ break; ++ case 'c': info->copyright = 1; break; ++ case 'o': info->original = 0; break; ++ default: fprintf(stderr,"%s: unrec option %c\n", ++ programName, c); ++ err = 1; break; ++ } ++ if(argUsed) { ++ if(arg == token) token = ""; /* no more from token */ ++ else ++i; /* skip arg we used */ ++ arg = ""; argUsed = 0; ++ } ++ } ++ } ++ else { ++ if(inPath[0] == '\0') strncpy(inPath, argv[i], MAX_NAME_SIZE); ++ else if(outPath[0] == '\0') strncpy(outPath, argv[i], MAX_NAME_SIZE); ++ else { ++ fprintf(stderr,"%s: excess arg %s\n", programName, argv[i]); ++ err = 1; ++ } ++ } ++ } ++ ++ // Do not write VBR tag if VBR flag is not specified ++ if (g_bWriteVbrTag==1 && VBR==0) g_bWriteVbrTag=0; ++ ++ if(err || inPath[0] == '\0') lame_usage(programName); /* never returns */ ++ if (inPath[0]=='-') silent=1; /* turn off status - it's broken for stdin */ ++ if(outPath[0] == '\0') { ++ if (inPath[0]=='-') { ++ /* if input is stdin, default output is stdout */ ++ strcpy(outPath,"-"); ++ }else { ++ strncpy(outPath, inPath, MAX_NAME_SIZE - strlen(DFLT_EXT)); ++ strncat(outPath, DFLT_EXT, strlen(DFLT_EXT) ); ++ } ++ } ++ /* some file options not allowed with stdout */ ++ if (outPath[0]=='-') { ++ g_bWriteVbrTag=0; /* turn off VBR tag */ ++ if (id3tag.used) { ++ id3tag.used=0; /* turn of id3 tagging */ ++ fprintf(stderr,"id3tag ignored: id3 tagging not supported for stdout.\n"); ++ } ++ } ++ ++ ++ /* if user did not explicitly specify input is mp3, check file name */ ++ if (input_format != sf_mp3) ++ if (!(strcmp((char *) &inPath[strlen(inPath)-4],".mp3"))) ++ input_format = sf_mp3; ++ ++ /* default guess for number of channels */ ++ if (autoconvert) num_channels=2; ++ else if (info->mode == MPG_MD_MONO) num_channels=1; ++ else num_channels=2; ++ ++ #ifdef _BLADEDLL ++ num_samples=0; ++ #else ++ /* open the input file */ ++ OpenSndFile(inPath,info,samplerate,num_channels); ++ /* if GetSndSampleRate is non zero, use it to overwrite the default */ ++ if (GetSndSampleRate()) samplerate=GetSndSampleRate(); ++ if (GetSndChannels()) num_channels=GetSndChannels(); ++ num_samples = GetSndSamples(); ++ #endif ++ ++ if (autoconvert==TRUE) { ++ info->mode = MPG_MD_MONO; ++ info->mode_ext = MPG_MD_LR_LR; ++ } ++ if (num_channels==1) { ++ autoconvert = FALSE; /* avoid 78rpm emulation mode from downmixing mono file! */ ++ info->mode = MPG_MD_MONO; ++ info->mode_ext = MPG_MD_LR_LR; ++ } ++ /* if user specified mono and there are 2 channels, autoconvert */ ++ if ((num_channels==2) && (info->mode == MPG_MD_MONO)) ++ autoconvert = TRUE; ++ ++ if ((num_channels==1) && (info->mode!=MPG_MD_MONO)) { ++ fprintf(stderr,"Error: mono input, stereo output not supported. \n"); ++ exit(1); ++ } ++ if (autoconvert==TRUE) { ++ fprintf(stderr, "Autoconverting from stereo to mono. Setting encoding to mono mode.\n"); ++ } ++ ++ ++ ++ /* set the output sampling rate, and resample options if necessary ++ samplerate = input sample rate ++ resamplerate = ouput sample rate ++ */ ++ if (resamplerate==0) { ++ /* user did not specify output sample rate */ ++ resamplerate=samplerate; /* default */ ++ } ++ ++ info->sampling_frequency = SmpFrqIndex((long)resamplerate, &info->version); ++ if( info->sampling_frequency < 0) { ++ display_bitrates(2); ++ exit(1); ++ } ++ ++ #ifndef _BLADEDLL ++ /* estimate total frames. must be done after setting sampling rate so ++ * we know the framesize. */ ++ totalframes=0; ++ framesize = (info->version==0) ? 576 : 1152; ++ if (num_samples == MAX_U_32_NUM) { ++ stat(inPath,&sb); /* try file size, assume 2 bytes per sample */ ++ if (input_format == sf_mp3) { ++ double totalseconds = (sb.st_size*8.0/(1000.0*GetSndBitrate())); ++ double framespersecond = (double)samplerate/(double)framesize; ++ totalframes = 1+(totalseconds*framespersecond); ++ }else{ ++ totalframes = 2+(sb.st_size/(2*framesize*num_channels)); ++ } ++ }else{ ++ totalframes = 2+(num_samples/framesize); ++ } ++ #endif /* _BLADEDLL */ ++ ++ ++ ++ if ( brate == 0 ) { ++ info->bitrate_index=9; ++ brate = bitrate[info->version][info->lay-1][info->bitrate_index]; ++ } else { ++ if( (info->bitrate_index = BitrateIndex(info->lay, brate, info->version,resamplerate)) < 0) { ++ display_bitrates(2); ++ exit(1); ++ } ++ /* a bit rate was specified. for VBR, take this to be the minimum */ ++ VBR_min_bitrate=info->bitrate_index; ++ } ++ ++ resample_ratio=0; ++ if (resamplerate != samplerate) { ++ fprintf(stderr,"Resampling: input=%iHz output=%iHz\n", ++ (int)samplerate,(int)resamplerate); ++ resample_ratio = (double)samplerate/(double)resamplerate; ++ } ++ compression_ratio = resamplerate*16*num_channels/(1000.0*brate); ++ if (compression_ratio > 12 ) { ++ /* print message suggesting downsampling or low-pass filter */ ++ } ++ ++ ++ /* default is STEREO for higher bitrates, not JSTEREO, unless of course ++ * the user specified a mode */ ++ if ((!mode_given) && (compression_ratio <= 8 )) { ++ if (info->mode != MPG_MD_MONO){ ++ info->mode = MPG_MD_STEREO; info->mode_ext = MPG_MD_LR_LR; ++ } ++ } ++ if ((!mode_given) && (VBR_q <= 3 )) { ++ if (info->mode != MPG_MD_MONO){ ++ info->mode = MPG_MD_STEREO; info->mode_ext = MPG_MD_LR_LR; ++ } ++ } ++ ++ ++ ++ if (info->bitrate_index==14) VBR=0; /* dont bother with VBR at 320kbs */ ++ if (VBR) { ++ /* default max bitrate is 256kbs */ ++ /* we do not normally allow 320bps frams with VBR, unless: */ ++ if (info->bitrate_index==13) VBR_max_bitrate=14; ++ if (VBR_q == 0) VBR_max_bitrate=14; /* allow 320kbs */ ++ if (VBR_q >= 5) VBR_max_bitrate=12; /* max = 224kbs */ ++ if (VBR_q >= 8) VBR_max_bitrate=9; /* low quality, max = 128kbs */ ++ if (voice_mode) VBR_max_bitrate=10; ++ highq = TRUE; ++ } ++ ++ ++ /* Should we disable the scalefactor band 21 cutoff? */ ++ /* NOTE: dont *enable* it here, since that could override the users' ++ * -k option, which disables sfb21 cufoff */ ++ /* remove coefficients in scalefactor band 21 (12 for short blocks) ++ * FhG does this for bps <= 128kbs, so we will too. ++ * This amounds to a 16kHz low-pass filter. If that offends you, you ++ * probably should not be encoding at 128kbs! ++ * There is no ratio[21] or xfsf[21], so when these coefficients are ++ * included they are just quantized as is. mt 5/99 ++ */ ++ /* disable sfb21 cutoff for high quality VBR */ ++ if (VBR && (VBR_q <4) ) sfb21 = 0; ++ /* 44.1kHz at 128kbs: compression factor of 11 */ ++ /* 44.1kHz at 160kbs: compression factor of 8.82 */ ++ if (compression_ratio<9.0) sfb21=0; ++ ++ /* dont allow forced mid/side stereo for mono output */ ++ if (info->mode == MPG_MD_MONO) force_ms=0; ++ ++ if (gtkflag) { ++ lame_nowrite=1; /* disable all file output */ ++ g_bWriteVbrTag=0; /* disable Xing VBR tag */ ++ } ++ ++ if (info->version==0) { ++ g_bWriteVbrTag=0; /* no MPEG2 Xing VBR tags yet */ ++ } ++ ++ ++ /* open the output file */ ++ /* if gtkflag, no output. but set outfile = stdout */ ++ open_bit_stream_w(&bs, outPath, BUFFER_SIZE,lame_nowrite); ++ ++ /* copy some header information */ ++ fr_ps.actual_mode = info->mode; ++ fr_ps.stereo = (info->mode == MPG_MD_MONO) ? 1 : 2; ++ ++ ++ #ifdef BRHIST ++ if (VBR) { ++ if (disp_brhist) ++ brhist_init(VBR_min_bitrate, VBR_max_bitrate); ++ } else ++ disp_brhist = 0; ++ #endif ++ ++ if (g_bWriteVbrTag) ++ { ++ // Write initial VBR Header to bitstream ++ InitVbrTag(&bs,info->version-1,info->mode,info->sampling_frequency); ++ /* flush VBR header frame to mp3 file */ ++ if (!lame_nowrite) ++ { ++ write_buffer(&bs); ++ empty_buffer(&bs); ++ } ++ /* note: if lame_nowrite==0, we do not empty the buffer, the VBR header will ++ * remain in the bit buffer and the first mp3 frame will be ++ * appeneded. this way, lame_encode() will return to the calling ++ * program the VBR header along with the first mp3 frame */ ++ } ++ ++ return; ++ } ++ ++ ++ ++ ++ /************************************************************************ ++ * ++ * print_config ++ * ++ * PURPOSE: Prints the encoding parameters used ++ * ++ ************************************************************************/ ++ void lame_print_config(void) ++ { ++ layer *info = fr_ps.header; ++ char *mode_names[4] = { "stereo", "j-stereo", "dual-ch", "single-ch" }; ++ ++ if (gtkflag) { ++ fprintf(stderr, "Analyzing %s \n",inPath); ++ } ++ else { ++ fprintf(stderr, "Encoding %s to %s\n", ++ (strcmp(inPath, "-")? inPath : "stdin"), ++ (strcmp(outPath, "-")? outPath : "stdout")); ++ if (VBR) ++ fprintf(stderr, "Encoding as %.1fkHz VBR(q=%i) %s MPEG%i LayerIII file\n", ++ s_freq[info->version][info->sampling_frequency], ++ VBR_q,mode_names[info->mode],2-info->version); ++ else ++ fprintf(stderr, "Encoding as %.1f kHz %d kbps %s MPEG%i LayerIII file\n", ++ s_freq[info->version][info->sampling_frequency], ++ bitrate[info->version][info->lay-1][info->bitrate_index], ++ mode_names[info->mode],2-info->version); ++ } ++ fflush(stderr); ++ } ++ ++ ++ ++ ++ #ifdef BRHIST ++ ++ #define BRHIST_BARMAX 50 ++ ++ long brhist_count[15]; ++ long brhist_temp[15]; ++ int brhist_vbrmin; ++ int brhist_vbrmax; ++ long brhist_max; ++ char brhist_bps[15][5]; ++ char brhist_backcur[200]; ++ char brhist_bar[BRHIST_BARMAX+10]; ++ char brhist_spc[BRHIST_BARMAX+1]; ++ ++ char stderr_buff[BUFSIZ]; ++ ++ ++ void brhist_init(int br_min, int br_max) ++ { ++ int i; ++ char term_buff[1024]; ++ layer *info = fr_ps.header; ++ char *termname; ++ char *tp; ++ char tc[10]; ++ ++ for(i = 0; i < 15; i++) ++ { ++ sprintf(brhist_bps[i], "%3d:", bitrate[info->version][info->lay-1][i]); ++ brhist_count[i] = 0; ++ brhist_temp[i] = 0; ++ } ++ ++ brhist_vbrmin = br_min; ++ brhist_vbrmax = br_max; ++ ++ brhist_max = 0; ++ ++ memset(&brhist_bar[0], '*', BRHIST_BARMAX); ++ brhist_bar[BRHIST_BARMAX] = '\0'; ++ memset(&brhist_spc[0], ' ', BRHIST_BARMAX); ++ brhist_spc[BRHIST_BARMAX] = '\0'; ++ brhist_backcur[0] = '\0'; ++ ++ if ((termname = getenv("TERM")) == NULL) ++ { ++ fprintf(stderr, "can't get TERM environment string.\n"); ++ disp_brhist = 0; ++ return; ++ } ++ ++ if (tgetent(term_buff, termname) != 1) ++ { ++ fprintf(stderr, "can't find termcap entry: %s\n", termname); ++ disp_brhist = 0; ++ return; ++ } ++ ++ tc[0] = '\0'; ++ tp = &tc[0]; ++ tgetstr("up", &tp); ++ brhist_backcur[0] = '\0'; ++ for(i = br_min-1; i <= br_max; i++) ++ strcat(brhist_backcur, tc); ++ ++ /* ++ tc[0] = '\0'; ++ tp = &tc[0]; ++ tgetstr("ce", &tp); ++ strcat(brhist_bar, tc); ++ */ ++ ++ setbuf(stderr, stderr_buff); ++ } ++ ++ void brhist_add_count(void) ++ { ++ int i; ++ ++ for(i = brhist_vbrmin; i <= brhist_vbrmax; i++) ++ { ++ brhist_count[i] += brhist_temp[i]; ++ if (brhist_count[i] > brhist_max) ++ brhist_max = brhist_count[i]; ++ brhist_temp[i] = 0; ++ } ++ } ++ ++ void brhist_disp(void) ++ { ++ int i; ++ long full; ++ int barlen; ++ ++ full = (brhist_max < BRHIST_BARMAX) ? BRHIST_BARMAX : brhist_max; ++ fputc('\n', stderr); ++ for(i = brhist_vbrmin; i <= brhist_vbrmax; i++) ++ { ++ barlen = (brhist_count[i]*BRHIST_BARMAX+full-1) / full; ++ fputs(brhist_bps[i], stderr); ++ fputs(&brhist_bar[BRHIST_BARMAX - barlen], stderr); ++ fputs(&brhist_spc[barlen], stderr); ++ fputc('\n', stderr); ++ } ++ fputs(brhist_backcur, stderr); ++ fflush(stderr); ++ } ++ ++ void brhist_disp_total(void) ++ { ++ int i; ++ double ave; ++ layer *info = fr_ps.header; ++ ++ for(i = brhist_vbrmin; i <= brhist_vbrmax; i++) ++ fputc('\n', stderr); ++ ++ ave=0; ++ for(i = brhist_vbrmin; i <= brhist_vbrmax; i++) ++ ave += bitrate[info->version][info->lay-1][i]* ++ (double)brhist_count[i] / totalframes; ++ fprintf(stderr, "\naverage: %2.0f kbs\n",ave); ++ ++ #if 0 ++ fprintf(stderr, "----- bitrate statistics -----\n"); ++ fprintf(stderr, " [kbps] frames\n"); ++ for(i = brhist_vbrmin; i <= brhist_vbrmax; i++) ++ { ++ fprintf(stderr, " %3d %8ld (%.1f%%)\n", ++ bitrate[info->version][info->lay-1][i], ++ brhist_count[i], ++ (double)brhist_count[i] / totalframes * 100.0); ++ } ++ #endif ++ fflush(stderr); ++ } ++ ++ #endif ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ /************************************************************************ ++ * ++ * encodeframe() ++ * ++ * SEMANTICS: One overlapping frame of audio of up to 2 channels are ++ * processed at a time in the following order: ++ * (associated routines are in parentheses) ++ * ++ * 1. Filter sliding window of data to get 32 subband ++ * samples per channel. ++ * (window_subband,filter_subband) ++ * ++ * 2. Calculate scalefactors for the frame, and if layer 2, ++ * also calculate scalefactor select information. ++ * (*_scale_factor_calc) ++ * ++ * 3. Calculate psychoacoustic masking levels using selected ++ * psychoacoustic model. ++ * (*_Psycho_One, psycho_anal) ++ * ++ * 4. Perform iterative bit allocation for subbands with low ++ * mask_to_noise ratios using masking levels from step 4. ++ * (*_main_bit_allocation) ++ * ++ * 5. Pack bit allocation and scalefactors onto bitstream. ++ * (*_encode_bit_alloc,*_encode_scale) ++ * ++ * 6. Quantize subbands and pack them into bitstream ++ * (*_subband_quantization, *_sample_encoding) ++ * ++ ************************************************************************/ ++ int lame_encode(short int Buffer[2][1152],char *mpg123bs) ++ { ++ static unsigned long frameBits; ++ static unsigned long bitsPerSlot; ++ static double frac_SpF; ++ static double slot_lag; ++ static int mode_gr; ++ static int old_bitrate; ++ static unsigned long sentBits = 0; ++ static L3SBS l3_sb_sample; ++ /* set to 56 to sync with FhG */ ++ #define EXTRADELAY 56 ++ static int samplesPerFrame; ++ static short int mfbuf[2][1152+576+EXTRADELAY]; /* 3 granule internal buffer */ ++ double xr[2][2][576]; ++ double win_que[64]; ++ int l3_enc[2][2][576]; ++ int mpg123count; ++ III_psy_ratio ratio; ++ III_scalefac_t scalefac; ++ ++ int j,ch,gr,mean_bits; ++ int bitsPerFrame; ++ double pe[2][4]; ++ double pe_use[2][2]; ++ double pe_sum_old=0; ++ static double pe_sum=0; ++ short *win_buf[2]; ++ int stereo; ++ layer *info; ++ int i; ++ int check_ms_stereo; ++ int bit_rate; ++ double samp; ++ double ms_ener_ratio[2]; ++ ++ stereo = fr_ps.stereo; ++ info = fr_ps.header; ++ ++ info->mode_ext = MPG_MD_LR_LR; ++ ++ /* use m/s stereo? */ ++ check_ms_stereo = ((info->mode == MPG_MD_JOINT_STEREO) && ++ (!force_ms) && (info->version == 1) && ++ (stereo==2)); ++ ++ ++ if (frameNum==0) old_bitrate=info->bitrate_index; ++ info->bitrate_index = old_bitrate; ++ ++ ++ bit_rate = bitrate[info->version][info->lay-1][info->bitrate_index]; ++ samp = s_freq[info->version][info->sampling_frequency]; ++ ++ if (frameNum==0) { ++ double avg_slots_per_frame; ++ int whole_SpF; ++ sentBits = 0; ++ memset((char *) mfbuf, 0, sizeof(mfbuf)); ++ ++ /* Figure average number of 'slots' per frame. */ ++ /* Bitrate means TOTAL for both channels, not per side. */ ++ bitsPerSlot = 8; ++ samplesPerFrame = info->version == 1 ? 1152 : 576; ++ avg_slots_per_frame = ((double)samplesPerFrame /samp) * ++ ((double)bit_rate / (double)bitsPerSlot); ++ whole_SpF = (int) avg_slots_per_frame; ++ frac_SpF = avg_slots_per_frame - (double)whole_SpF; ++ slot_lag = -frac_SpF; ++ info->padding = 1; ++ if (fabs(frac_SpF) < 1e-9) info->padding = 0; ++ mode_gr = (info->version == 1) ? 2 : 1; /* mode_gr = 2 */ ++ } ++ ++ ++ /* shift in new samples, delayed by 576+EXTRADELAY */ ++ for (ch=0; ch<stereo; ch++) ++ for (i=0; i<576+EXTRADELAY; i++) ++ mfbuf[ch][i]=mfbuf[ch][i+samplesPerFrame]; ++ for (ch=0; ch<stereo; ch++) ++ for (i=0; i<samplesPerFrame; i++) ++ mfbuf[ch][i+576+EXTRADELAY]=Buffer[ch][i]; ++ ++ ++ ++ ++ /********************** padding *****************************/ ++ if (VBR) { ++ /* leave info_padding as it was set above */ ++ } else { ++ if (frac_SpF != 0) { ++ if (slot_lag > (frac_SpF-1.0) ) { ++ slot_lag -= frac_SpF; ++ info->padding = 0; ++ } ++ else { ++ info->padding = 1; ++ slot_lag += (1-frac_SpF); ++ } ++ } ++ } ++ ++ #ifndef _BLADEDLL ++ /********************** status display *****************************/ ++ if (!gtkflag && !silent) { ++ int mod = info->version == 0 ? 100 : 20; ++ if (frameNum%mod==0) { ++ timestatus(info,frameNum,totalframes); ++ #ifdef BRHIST ++ if (disp_brhist) ++ { ++ brhist_add_count(); ++ brhist_disp(); ++ } ++ #endif ++ } ++ } ++ ++ #endif /* _BLADEDLL */ ++ ++ /********************** process InputBuffer *****************************/ ++ ++ ++ if (force_ms) ++ for (i=0; i< samplesPerFrame; i++) { ++ /* dont overflow the short int */ ++ double mid = ((int)mfbuf[0][576+i]+(int)mfbuf[1][576+i])/(2.0); ++ double side= ((int)mfbuf[0][576+i]-(int)mfbuf[1][576+i])/(2.0); ++ mfbuf[0][576+i]=mid; ++ mfbuf[1][576+i]=side; ++ } ++ ++ ++ ++ /***************************** Layer 3 ********************************** ++ * mfbuf contains 3 granules: [0 1 2 ] ++ * encoder will encode granules 0 1 ++ * psy-model will be fed granules 1 2, and because of its 1 granule delay ++ * it will return thresholds for granules 0 1 */ ++ ++ win_buf[0] = &mfbuf[0][0]; ++ win_buf[1] = &mfbuf[1][0]; ++ ++ ++ if (!fast_mode) { ++ /* psychoacoustic model ++ * psy model adds a 544 delay to sync with the filterbanks ++ * in addition to this delay, it also adds a 1 granule (576) delay ++ * that we must compensate for (mt 6/99). ++ */ ++ short int *bufp[2]; /* address of beginning of left & right granule */ ++ int blocktype[2]; ++ ++ for (gr=0; gr < mode_gr ; gr++) { ++ for ( ch = 0; ch < stereo; ch++ ) bufp[ch] = &mfbuf[ch][576 + gr*576]; ++ L3psycho_anal( bufp, stereo, gr, info, ++ s_freq[info->version][info->sampling_frequency] * 1000.0, ++ check_ms_stereo,&ms_ener_ratio[gr], ++ ratio.l[gr], ratio.s[gr], pe[gr], blocktype); ++ for ( ch = 0; ch < stereo; ch++ ) ++ l3_side.gr[gr].ch[ch].tt.block_type=blocktype[ch]; ++ } ++ }else{ ++ for (gr=0; gr < mode_gr ; gr++) ++ for ( ch = 0; ch < stereo; ch++ ) { ++ l3_side.gr[gr].ch[ch].tt.block_type=NORM_TYPE; ++ pe[gr][ch]=700; ++ } ++ } ++ ++ pe_sum_old=pe_sum; ++ pe_sum=0; ++ for( gr = 0; gr < mode_gr; gr++ ) ++ for ( ch = 0; ch < stereo; ch++ ) ++ pe_sum+=pe[gr][ch]; ++ ++ ++ ++ /* for VBR, what bitrate should we try first? */ ++ if (VBR){ ++ info->bitrate_index=old_bitrate-1; ++ if (abs(pe_sum_old - pe_sum)>200){ ++ if (pe_sum_old>pe_sum) ++ info->bitrate_index--; ++ } ++ ++ if (info->bitrate_index < VBR_min_bitrate) ++ info->bitrate_index=VBR_min_bitrate; ++ if (info->bitrate_index > VBR_max_bitrate) ++ info->bitrate_index=VBR_max_bitrate; ++ } ++ ++ ++ /* block type flags */ ++ for( gr = 0; gr < mode_gr; gr++ ) { ++ for ( ch = 0; ch < stereo; ch++ ) { ++ gr_info *cod_info = &l3_side.gr[gr].ch[ch].tt; ++ cod_info->mixed_block_flag = 0; /* never used by this model */ ++ if (cod_info->block_type == NORM_TYPE ) ++ cod_info->window_switching_flag = 0; ++ else ++ cod_info->window_switching_flag = 1; ++ } ++ } ++ ++ ++ /* polyphase filtering */ ++ for( gr = 0; gr < mode_gr; gr++ ) ++ for ( ch = 0; ch < stereo; ch++ ) ++ for ( j = 0; j < 18; j++ ) { ++ window_subband( &win_buf[ch], win_que, ch ); ++ filter_subband( win_que, &(l3_sb_sample)[ch][gr+1][j][0] ); } ++ ++ /* apply mdct to the polyphase outputs */ ++ mdct_sub( l3_sb_sample, xr, stereo, &l3_side, mode_gr ); ++ ++ /* data was scaled by 1/2. fix so effectively it was scaled by 1/sqrt(2) */ ++ if (force_ms) { ++ for ( gr = 0; gr < mode_gr; gr++ ) ++ for ( ch = 0; ch < stereo; ch++ ) ++ for (i =0 ; i< 576; i++) ++ xr[gr][ch][i] *= SQRT2; ++ } ++ ++ /* set a default for ms_ener_ratio since it was not computed in these cases */ ++ if (force_ms || fast_mode) { ++ ms_ener_ratio[0]=.25; ++ ms_ener_ratio[1]=.25; ++ } ++ ++ ++ if (check_ms_stereo) { ++ /* make sure block type is the same in each channel */ ++ check_ms_stereo = ++ (l3_side.gr[0].ch[0].tt.block_type==l3_side.gr[0].ch[1].tt.block_type) && ++ (l3_side.gr[1].ch[0].tt.block_type==l3_side.gr[1].ch[1].tt.block_type); ++ } ++ if (check_ms_stereo) { ++ if ( (ms_ener_ratio[0] + ms_ener_ratio[1])<.70) ++ info->mode_ext = MPG_MD_MS_LR; ++ ms_ener_ratio[0]=Min(ms_ener_ratio[0],.5); ++ ms_ener_ratio[1]=Min(ms_ener_ratio[1],.5); ++ } ++ ++ #ifdef HAVEGTK ++ if (gtkflag) { ++ for ( gr = 0; gr < mode_gr; gr++ ) ++ for ( ch = 0; ch < stereo; ch++ ) { ++ pinfo->ms_ratio[gr]=ms_ener_ratio[gr]; ++ pinfo->blocktype[gr][ch]= ++ l3_side.gr[gr].ch[ch].tt.block_type; ++ for ( j = 0; j < 576; j++ ) pinfo->xr[gr][ch][j]=xr[gr][ch][j]; ++ } ++ } ++ #endif ++ ++ /* if MS stereo, switch to MS psy data */ ++ for ( gr = 0; gr < mode_gr; gr++ ) { ++ for ( ch = 0; ch < stereo; ch++ ) { ++ if (highq && (info->mode_ext==MPG_MD_MS_LR)) { ++ pe_use[gr][ch]=pe[gr][ch+2]; ++ memcpy(ratio.l[gr][ch],ratio.l[gr][ch+2],sizeof(ratio.l[gr][ch])); ++ memcpy(ratio.s[gr][ch],ratio.s[gr][ch+2],sizeof(ratio.s[gr][ch])); ++ #ifdef HAVEGTK ++ if (gtkflag) { ++ pinfo->ers[gr][ch]=pinfo->ers[gr][ch+2]; ++ pinfo->pe[gr][ch]=pinfo->pe[gr][ch+2]; ++ memcpy(pinfo->energy[gr][ch],pinfo->energy[gr][ch+2], ++ sizeof(pinfo->energy[gr][ch])); ++ memcpy(pinfo->thr[gr][ch],pinfo->thr[gr][ch+2], ++ sizeof(pinfo->thr[gr][ch])); ++ memcpy(pinfo->en[gr][ch],pinfo->en[gr][ch+2], ++ sizeof(pinfo->en[gr][ch])); ++ memcpy(pinfo->thr_s[gr][ch],pinfo->thr_s[gr][ch+2], ++ sizeof(pinfo->thr_s[gr][ch])); ++ memcpy(pinfo->en_s[gr][ch],pinfo->en_s[gr][ch+2], ++ sizeof(pinfo->en_s[gr][ch])); ++ } ++ #endif ++ }else{ ++ pe_use[gr][ch]=pe[gr][ch]; ++ } ++ } ++ } ++ ++ /* bit and noise allocation */ ++ if (VBR) ++ VBR_iteration_loop( pe_use, ms_ener_ratio, xr, &ratio, &l3_side, l3_enc, ++ &scalefac, &fr_ps); ++ else ++ iteration_loop( pe_use, ms_ener_ratio, xr, &ratio, &l3_side, l3_enc, ++ &scalefac, &fr_ps); ++ ++ #ifdef BRHIST ++ brhist_temp[info->bitrate_index]++; ++ #endif ++ ++ /* flag for our ms_stereo with psy-model on mid & side channels */ ++ if (force_ms) info->mode_ext = MPG_MD_MS_LR; ++ ++ /* write the frame to the bitstream */ ++ old_bitrate = info->bitrate_index; ++ getframebits(info,stereo,&bitsPerFrame,&mean_bits); ++ III_format_bitstream( bitsPerFrame, &fr_ps, l3_enc, &l3_side, ++ &scalefac, &bs, xr, NULL, 0 ); ++ ++ ++ frameBits = bs.totbit - sentBits; ++ ++ ++ if ( frameBits % bitsPerSlot ) /* a program failure */ ++ fprintf( stderr, "Sent %ld bits = %ld slots plus %ld\n", ++ frameBits, frameBits/bitsPerSlot, ++ frameBits%bitsPerSlot ); ++ sentBits += frameBits; ++ ++ /* copy mp3 bit buffer into array */ ++ mpg123count = copy_buffer(mpg123bs,&bs); ++ if (!lame_nowrite) write_buffer(&bs); /* ouput to mp3 file */ ++ empty_buffer(&bs); /* empty buffer */ ++ ++ if (g_bWriteVbrTag) AddVbrFrame((int)(sentBits/8)); ++ ++ #ifdef HAVEGTK ++ if (gtkflag) { ++ int adj=0; ++ if (info->version == 0) adj=576; ++ for ( ch = 0; ch < stereo; ch++ ) { ++ for ( j = 0; j < WINDELAY; j++ ) ++ pinfo->pcmdata[ch][j] = pinfo->pcmdata[ch][j+1152-adj]; ++ for ( j = 0; j < 1152-adj; j++ ) { ++ pinfo->pcmdata[ch][j+WINDELAY] = mfbuf[ch][j]; ++ } ++ } ++ } ++ #endif ++ // fprintf(stderr,"finished frameNum=%i \n",frameNum); ++ frameNum++; ++ return mpg123count; ++ } ++ ++ ++ #ifndef _BLADEDLL ++ int lame_readframe(short int Buffer[2][1152]) ++ { ++ int iread,ifreq,ofreq; ++ static int eof; ++ ++ if (frameNum==0) eof=0; ++ iread = get_audio(Buffer,fr_ps.stereo,fr_ps.header); ++ /* note: if input is stereo and output is mono, get_audio() ++ * will return stereo=1 and .5*(L+R) in channel 0, ++ * and nothing in channel 1. */ ++ /* DOWNSAMPLE, ETC */ ++ ofreq=1000*s_freq[info.version][info.sampling_frequency]; /* output */ ++ ifreq=ofreq*resample_ratio; /* input */ ++ ++ #ifdef HAVEGTK ++ if (gtkflag) { ++ pinfo->frameNum = frameNum; ++ pinfo->sampfreq=ofreq; ++ pinfo->framesize=(fr_ps.header->version==0) ? 576 : 1152; ++ pinfo->stereo = fr_ps.stereo; ++ } ++ #endif ++ ++ if (resample_ratio>0) { ++ fprintf(stderr,"Error: resample code not yet written! \n"); ++ exit(1); ++ } ++ ++ /* check to see if we overestimated/underestemated totalframes */ ++ if (!eof) { ++ if (iread==0) { ++ /* this is the last frame */ ++ totalframes=frameNum+1; ++ eof=1; ++ }else{ ++ if (frameNum > (totalframes-1)) totalframes = frameNum; ++ } ++ } ++ ++ ++ ++ return iread; ++ } ++ #endif /* _BLADEDLL */ ++ ++ /* initialize mp3 encoder */ ++ void lame_init(int nowrite) ++ { ++ ++ #ifdef __FreeBSD__ ++ #include <floatingpoint.h> ++ { ++ /* seet floating point mask to the Linux default */ ++ fp_except_t mask; ++ mask=fpgetmask(); ++ /* if bit is set, we get SIGFPE on that error! */ ++ fpsetmask(mask & ~(FP_X_INV|FP_X_DZ)); ++ /* fprintf(stderr,"FreeBSD mask is 0x%x\n",mask); */ ++ } ++ #endif ++ #ifdef ABORTFP ++ { ++ #include <fpu_control.h> ++ unsigned int mask; ++ _FPU_GETCW(mask); ++ /* Set the Linux mask to abort on most FPE's */ ++ /* if bit is set, we _mask_ SIGFPE on that error! */ ++ // mask &= ~( _FPU_MASK_IM | _FPU_MASK_ZM | _FPU_MASK_OM | _FPU_MASK_UM ); ++ mask &= ~( _FPU_MASK_IM | _FPU_MASK_ZM | _FPU_MASK_OM ); ++ _FPU_SETCW(mask); ++ } ++ #endif ++ ++ ++ #ifndef _BLADEDLL ++ fprintf(stderr,"LAME version %s (www.sulaco.org/mp3) \n",get_lame_version()); ++ fprintf(stderr,"GPSYCHO: GPL psycho-acoustic model version %s. \n",get_psy_version()); ++ #ifdef LIBSNDFILE ++ fprintf(stderr,"Input handled by libsndfile (www.zip.com.au/~erikd/libsndfile)\n"); ++ #endif ++ #endif /* _BLADEDLL */ ++ ++ /* Global flags. set defaults here */ ++ allow_diff_short=0; ++ ATHonly=0; ++ autoconvert=FALSE; ++ experimentalX = 0; ++ experimentalY = 0; ++ experimentalZ = 0; ++ force_ms=FALSE; ++ fast_mode=0; ++ frameNum=0; ++ g_bWriteVbrTag=1; ++ gtkflag=0; ++ gpsycho=1; ++ highq=0; ++ input_format=sf_unknown; ++ lame_nowrite=nowrite; ++ resample_ratio=0; ++ sfb21=1; ++ silent=0; ++ swapbytes=0; ++ totalframes=0; ++ VBR=0; ++ VBR_q=4; ++ VBR_min_bitrate=1; /* 32kbs */ ++ VBR_max_bitrate=13; /* 256kbs */ ++ voice_mode=0; ++ ++ ++ // Clear info structure ++ memset(&info,0,sizeof(info)); ++ memset(&bs, 0, sizeof(Bit_stream_struc)); ++ memset(&fr_ps, 0, sizeof(frame_params)); ++ memset(&l3_side,0x00,sizeof(III_side_info_t)); ++ ++ fr_ps.header = &info; ++ fr_ps.tab_num = -1; /* no table loaded */ ++ fr_ps.alloc = NULL; ++ info.version = MPEG_AUDIO_ID; /* =1 Default: MPEG-1 */ ++ info.extension = 0; ++ ++ InitFormatBitStream(); ++ InitReservoir(); ++ ++ } ++ ++ ++ #ifndef _BLADEDLL ++ int lame_cleanup(char *mpg123bs) ++ { ++ int mpg123count; ++ frameNum--; ++ if (!gtkflag && !silent) { ++ timestatus(&info,frameNum,totalframes); ++ fprintf(stderr,"\n"); ++ #ifdef BRHIST ++ if (disp_brhist) ++ { ++ brhist_add_count(); ++ brhist_disp(); ++ } ++ #endif ++ #ifdef BRHIST ++ if (disp_brhist) ++ brhist_disp_total(); ++ #endif ++ fflush(stderr); ++ } ++ ++ ++ CloseSndFile(); ++ ++ ++ III_FlushBitstream(); ++ mpg123count = copy_buffer(mpg123bs,&bs); ++ ++ if (!lame_nowrite) ++ { ++ write_buffer(&bs); /* ouput to mp3 file */ ++ empty_buffer(&bs); /* empty buffer */ ++ ++ // Close the file ++ fclose(bs.pt); ++ ++ // Deallocate all buffers ++ desalloc_buffer(&bs); ++ ++ ++ // AF ADDED ++ if (g_bWriteVbrTag) ++ { ++ // Calculate relative quality of VBR stream ++ // 0=best, 100=worst ++ int nQuality=VBR_q*100/9; ++ ++ // Write Xing header again ++ PutVbrTag(outPath,nQuality); ++ } ++ // AF END ++ ++ // write an ID3 tag ++ if(id3tag.used) { ++ id3_buildtag(&id3tag); ++ id3_writetag(outPath, &id3tag); ++ } ++ } ++ return mpg123count; ++ } ++ #endif /* _BLADEDLL */ +diff -r -c -N encoder/lame.dsp lame3.50/lame.dsp +*** encoder/lame.dsp Wed Dec 31 17:00:00 1969 +--- lame3.50/lame.dsp Sun Oct 17 12:24:26 1999 +*************** +*** 0 **** +--- 1,352 ---- ++ # Microsoft Developer Studio Project File - Name="lame" - Package Owner=<4> ++ # Microsoft Developer Studio Generated Build File, Format Version 5.00 ++ # ** DO NOT EDIT ** ++ ++ # TARGTYPE "Win32 (x86) Console Application" 0x0103 ++ ++ CFG=lame - Win32 Debug GTK ++ !MESSAGE This is not a valid makefile. To build this project using NMAKE, ++ !MESSAGE use the Export Makefile command and run ++ !MESSAGE ++ !MESSAGE NMAKE /f "lame.mak". ++ !MESSAGE ++ !MESSAGE You can specify a configuration when running NMAKE ++ !MESSAGE by defining the macro CFG on the command line. For example: ++ !MESSAGE ++ !MESSAGE NMAKE /f "lame.mak" CFG="lame - Win32 Debug GTK" ++ !MESSAGE ++ !MESSAGE Possible choices for configuration are: ++ !MESSAGE ++ !MESSAGE "lame - Win32 Release" (based on "Win32 (x86) Console Application") ++ !MESSAGE "lame - Win32 Debug" (based on "Win32 (x86) Console Application") ++ !MESSAGE "lame - Win32 Debug GTK" (based on "Win32 (x86) Console Application") ++ !MESSAGE "lame - Win32 Release GTK" (based on\ ++ "Win32 (x86) Console Application") ++ !MESSAGE ++ ++ # Begin Project ++ # PROP Scc_ProjName "" ++ # PROP Scc_LocalPath "" ++ CPP=xicl.exe ++ RSC=rc.exe ++ ++ !IF "$(CFG)" == "lame - Win32 Release" ++ ++ # PROP BASE Use_MFC 0 ++ # PROP BASE Use_Debug_Libraries 0 ++ # PROP BASE Output_Dir "Release" ++ # PROP BASE Intermediate_Dir "Release" ++ # PROP BASE Target_Dir "" ++ # PROP Use_MFC 0 ++ # PROP Use_Debug_Libraries 0 ++ # PROP Output_Dir "Release" ++ # PROP Intermediate_Dir "Release" ++ # PROP Ignore_Export_Lib 0 ++ # PROP Target_Dir "" ++ # ADD BASE CPP /nologo /W3 /GX /O2 /D "WIN32" /D "NDEBUG" /D "_CONSOLE" /D "_MBCS" /YX /FD /c ++ # ADD CPP /nologo /W3 /GX /O2 /I "./WinGTK/gtk-plus" /I "./WinGTK/glib-1.2" /D "NDEBUG" /D "WIN32" /D "_CONSOLE" /D "_MBCS" /D BS_FORMAT=BINARY /YX /FD /c ++ # ADD BASE RSC /l 0x409 /d "NDEBUG" ++ # ADD RSC /l 0x409 /d "NDEBUG" ++ BSC32=bscmake.exe ++ # ADD BASE BSC32 /nologo ++ # ADD BSC32 /nologo ++ LINK32=xilink.exe ++ # ADD BASE LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:console /machine:I386 ++ # ADD LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:console /profile /map /machine:I386 ++ ++ !ELSEIF "$(CFG)" == "lame - Win32 Debug" ++ ++ # PROP BASE Use_MFC 0 ++ # PROP BASE Use_Debug_Libraries 1 ++ # PROP BASE Output_Dir "Debug" ++ # PROP BASE Intermediate_Dir "Debug" ++ # PROP BASE Target_Dir "" ++ # PROP Use_MFC 0 ++ # PROP Use_Debug_Libraries 1 ++ # PROP Output_Dir "Debug" ++ # PROP Intermediate_Dir "Debug" ++ # PROP Ignore_Export_Lib 0 ++ # PROP Target_Dir "" ++ # ADD BASE CPP /nologo /W3 /Gm /GX /Zi /Od /D "WIN32" /D "_DEBUG" /D "_CONSOLE" /D "_MBCS" /YX /FD /c ++ # ADD CPP /nologo /Zp4 /W3 /Gm /GX /Zi /Od /D "_DEBUG" /D "WIN32" /D "_CONSOLE" /D "_MBCS" /D BS_FORMAT=BINARY /YX /FD /c ++ # ADD BASE RSC /l 0x409 /d "_DEBUG" ++ # ADD RSC /l 0x409 /d "_DEBUG" ++ BSC32=bscmake.exe ++ # ADD BASE BSC32 /nologo ++ # ADD BSC32 /nologo ++ LINK32=xilink.exe ++ # ADD BASE LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:console /debug /machine:I386 /pdbtype:sept ++ # ADD LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:console /debug /machine:I386 /pdbtype:sept ++ ++ !ELSEIF "$(CFG)" == "lame - Win32 Debug GTK" ++ ++ # PROP BASE Use_MFC 0 ++ # PROP BASE Use_Debug_Libraries 1 ++ # PROP BASE Output_Dir "lame___W" ++ # PROP BASE Intermediate_Dir "lame___W" ++ # PROP BASE Ignore_Export_Lib 0 ++ # PROP BASE Target_Dir "" ++ # PROP Use_MFC 0 ++ # PROP Use_Debug_Libraries 1 ++ # PROP Output_Dir "DebugGTK" ++ # PROP Intermediate_Dir "DebugGTK" ++ # PROP Ignore_Export_Lib 0 ++ # PROP Target_Dir "" ++ # ADD BASE CPP /nologo /Zp4 /W3 /Gm /GX /Zi /Od /I "../WinGTK/gtk-plus" /I "../WinGTK/glib-1.2" /D "_DEBUG" /D "WIN32" /D "_CONSOLE" /D "_MBCS" /D BS_FORMAT=BINARY /D "HAVEGTK" /YX /FD /c ++ # ADD CPP /nologo /Zp4 /W3 /Gm /GX /Zi /Od /I "./WinGTK/gtk-plus" /I "./WinGTK/glib-1.2" /D "_DEBUG" /D "WIN32" /D "_CONSOLE" /D "_MBCS" /D BS_FORMAT=BINARY /D "HAVEGTK" /YX /FD /c ++ # ADD BASE RSC /l 0x409 /d "_DEBUG" ++ # ADD RSC /l 0x409 /d "_DEBUG" ++ BSC32=bscmake.exe ++ # ADD BASE BSC32 /nologo ++ # ADD BSC32 /nologo ++ LINK32=xilink.exe ++ # ADD BASE LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:console /debug /machine:I386 /pdbtype:sept ++ # ADD LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib glib-1.2.lib gdk-1.3.lib gtk-1.3.lib /nologo /subsystem:console /debug /machine:I386 /pdbtype:sept /libpath:"./WinGTK/glib-1.2" /libpath:"./WinGTK/gtk-plus/gdk" /libpath:"./WinGTK/gtk-plus/gtk" ++ ++ !ELSEIF "$(CFG)" == "lame - Win32 Release GTK" ++ ++ # PROP BASE Use_MFC 0 ++ # PROP BASE Use_Debug_Libraries 0 ++ # PROP BASE Output_Dir "lame___0" ++ # PROP BASE Intermediate_Dir "lame___0" ++ # PROP BASE Ignore_Export_Lib 0 ++ # PROP BASE Target_Dir "" ++ # PROP Use_MFC 0 ++ # PROP Use_Debug_Libraries 0 ++ # PROP Output_Dir "ReleaseGTK" ++ # PROP Intermediate_Dir "ReleaseGTK" ++ # PROP Ignore_Export_Lib 0 ++ # PROP Target_Dir "" ++ # ADD BASE CPP /nologo /W3 /GX /O2 /I "../WinGTK/gtk-plus" /I "../WinGTK/glib-1.2" /D "NDEBUG" /D "WIN32" /D "_CONSOLE" /D "_MBCS" /D BS_FORMAT=BINARY /D "HAVEGTK" /YX /FD /c ++ # ADD CPP /nologo /W3 /GX /O2 /I "./WinGTK/gtk-plus" /I "./WinGTK/glib-1.2" /D "NDEBUG" /D "WIN32" /D "_CONSOLE" /D "_MBCS" /D BS_FORMAT=BINARY /D "HAVEGTK" /YX /FD /c ++ # ADD BASE RSC /l 0x409 /d "NDEBUG" ++ # ADD RSC /l 0x409 /d "NDEBUG" ++ BSC32=bscmake.exe ++ # ADD BASE BSC32 /nologo ++ # ADD BSC32 /nologo ++ LINK32=xilink.exe ++ # ADD BASE LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:console /profile /map /machine:I386 ++ # ADD LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib glib-1.2.lib gdk-1.3.lib gtk-1.3.lib /nologo /subsystem:console /profile /map /machine:I386 /libpath:"./WinGTK/glib-1.2" /libpath:"./WinGTK/gtk-plus/gdk" /libpath:"./WinGTK/gtk-plus/gtk" ++ ++ !ENDIF ++ ++ # Begin Target ++ ++ # Name "lame - Win32 Release" ++ # Name "lame - Win32 Debug" ++ # Name "lame - Win32 Debug GTK" ++ # Name "lame - Win32 Release GTK" ++ # Begin Source File ++ ++ SOURCE=.\encoder.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\fastmdct.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\fastmdct.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\fft.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\fft.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\filterbank.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\formatBitstream.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\formatBitstream.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\get_audio.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\get_audio.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\globalflags.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\gpkplotting.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\gpkplotting.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\gtkanal.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\gtkanal.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\huffman.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\id3tag.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\id3tag.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\ieeefloat.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\ieeefloat.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=".\l3bitstream-pvt.h" ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\l3bitstream.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\l3bitstream.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\l3side.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\lame.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\lame.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\machine.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\main.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\musicin.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\portableio.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\portableio.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\profile.txt ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\psymodel.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\psymodel.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=".\quantize-pvt.h" ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\quantize.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\quantize.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\reservoir.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\reservoir.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\sqrttab.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\tables.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\tables.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\takehiro.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\timestatus.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\timestatus.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\util.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\util.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\VbrTag.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\VbrTag.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\version.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\version.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\wavheader.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\WinGTK.txt ++ # End Source File ++ # End Target ++ # End Project +diff -r -c -N encoder/lame.dsw lame3.50/lame.dsw +*** encoder/lame.dsw Wed Dec 31 17:00:00 1969 +--- lame3.50/lame.dsw Fri Sep 10 09:52:04 1999 +*************** +*** 0 **** +--- 1,44 ---- ++ Microsoft Developer Studio Workspace File, Format Version 5.00 ++ # WARNING: DO NOT EDIT OR DELETE THIS WORKSPACE FILE! ++ ++ ############################################################################### ++ ++ Project: "lame"=.\lame.dsp - Package Owner=<4> ++ ++ Package=<5> ++ {{{ ++ }}} ++ ++ Package=<4> ++ {{{ ++ Begin Project Dependency ++ Project_Dep_Name mpglib ++ End Project Dependency ++ }}} ++ ++ ############################################################################### ++ ++ Project: "mpglib"=.\mpglib\mpglib.dsp - Package Owner=<4> ++ ++ Package=<5> ++ {{{ ++ }}} ++ ++ Package=<4> ++ {{{ ++ }}} ++ ++ ############################################################################### ++ ++ Global: ++ ++ Package=<5> ++ {{{ ++ }}} ++ ++ Package=<3> ++ {{{ ++ }}} ++ ++ ############################################################################### ++ +diff -r -c -N encoder/lame.h lame3.50/lame.h +*** encoder/lame.h Wed Dec 31 17:00:00 1969 +--- lame3.50/lame.h Tue Sep 28 22:00:44 1999 +*************** +*** 0 **** +--- 1,62 ---- ++ /* ++ * Interface to MP3 LAME encoding engine ++ * ++ * Copyright (c) 1999 Mark Taylor ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License as published by ++ * the Free Software Foundation; either version 2, or (at your option) ++ * any later version. ++ * ++ * This program is distributed in the hope that it will be useful, ++ * but WITHOUT ANY WARRANTY; without even the implied warranty of ++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the ++ * GNU General Public License for more details. ++ * ++ * You should have received a copy of the GNU General Public License ++ * along with this program; see the file COPYING. If not, write to ++ * the Free Software Foundation, 675 Mass Ave, Cambridge, MA 02139, USA. ++ */ ++ ++ #include <stdio.h> ++ #define LAME_MAXMP3BUFFER 16384 ++ ++ /* if no_output=1, all LAME output is disabled, and it is up to the ++ * calling program to write the mp3 data (returned by lame_encode() and ++ * lame_cleanup()). In this case, the calling program is responsible ++ * for filling in the Xing VBR header data and adding the id3 tag. ++ * see lame_cleanup() for details. */ ++ void lame_init(int no_output); ++ ++ /* lame_cleanup will flush the buffers and may return a final mp3 frame */ ++ int lame_cleanup(char *mp3buf); ++ ++ void lame_usage(char *); ++ void lame_parse_args(int, char **); ++ void lame_print_config(void); ++ ++ /* read one frame of PCM data from audio input file opened by lame_parse_args*/ ++ /* input file can be either mp3 or uncompressed audio file */ ++ int lame_readframe(short int Buffer[2][1152]); ++ ++ /* ++ NOTE: for lame_encode and lame_decode, because of the bit reservoir ++ capability of mp3 frames, there can be a delay between the input ++ and output. lame_decode/lame_encode should always return exactly ++ one frame of output, but it may not return any output ++ until a second call with a second frame. (or even 3 or 4 more calls) ++ */ ++ ++ /* input 1 pcm frame, output (maybe) 1 mp3 frame. */ ++ int lame_encode(short int Buffer[2][1152],char *mp3buffer); ++ ++ /* input 1 mp3 frame, output (maybe) 1 pcm frame. */ ++ int lame_decode_init(void); ++ int lame_decode(char *mp3buf,int len,short pcm[][1152]); ++ ++ /* read mp3 file until mpglib returns one frame of PCM data */ ++ int lame_decode_initfile(FILE *fd,int *stereo,int *samp,int *bitrate); ++ int lame_decode_fromfile(FILE *fd,short int mpg123pcm[2][1152]); ++ ++ ++ +diff -r -c -N encoder/loop-pvt.h lame3.50/loop-pvt.h +*** encoder/loop-pvt.h Wed Jan 22 02:43:16 1997 +--- lame3.50/loop-pvt.h Wed Dec 31 17:00:00 1969 +*************** +*** 1,119 **** +- /********************************************************************** +- * ISO MPEG Audio Subgroup Software Simulation Group (1996) +- * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- * +- * $Id: loop-pvt.h,v 1.1 1996/02/14 04:04:23 rowlands Exp $ +- * +- * Private interface declarations for loop.c +- * +- * $Log: loop-pvt.h,v $ +- * Revision 1.1 1996/02/14 04:04:23 rowlands +- * Initial revision +- * +- * Received from Mike Coleman +- **********************************************************************/ +- +- #ifndef LOOP_PVT_H +- #define LOOP_PVT_H +- +- /* +- Revision History: +- +- Date Programmer Comment +- ========== ========================= =============================== +- 1995/10/01 mc@fivebats.com created +- +- */ +- +- int outer_loop( double xr[2][2][576], /*vector of the magnitudees of the spectral values */ +- int max_bits, +- III_psy_xmin *l3_xmin, /* the allowed distortion of the scalefactor */ +- int l3_enc[2][2][576], /* vector of quantized values ix(0..575) */ +- frame_params *fr_ps, +- III_scalefac_t *scalefac, /* scalefactors */ +- int gr, +- int ch, +- III_side_info_t *l3_side ); +- +- int part2_length( III_scalefac_t *scalefac, +- frame_params *fr_ps, +- int gr, +- int ch, +- III_side_info_t *si ); +- +- int quantanf_init( double xr[576] ); +- +- int inner_loop( double xr[2][2][576], +- int l3_enc[2][2][576], +- int max_bits, +- gr_info *cod_info, +- int gr, +- int ch ); +- void calc_xmin( double xr[2][2][576], +- III_psy_ratio *ratio, +- gr_info *cod_info, +- III_psy_xmin *l3_xmin, +- int gr, +- int ch ); +- double xr_max( double xr[576], +- unsigned int begin, +- unsigned int end ); +- +- void calc_scfsi( double xr[576], +- III_side_info_t *l3_side, +- III_psy_xmin *l3_xmin, +- int ch, +- int gr ); +- +- void gr_deco( gr_info *cod_info ); +- +- +- int count_bit( int ix[576], unsigned int start, unsigned int end, unsigned int table); +- int bigv_bitcount( int ix[576], gr_info *cod_info ); +- int choose_table( int max); +- void bigv_tab_select( int ix[576], gr_info *cod_info ); +- void subdivide( gr_info *cod_info ); +- int count1_bitcount( int ix[576], gr_info *cod_info ); +- void calc_runlen( int ix[576], +- gr_info *cod_info ); +- int scale_bitcount( III_scalefac_t *scalefac, +- gr_info *cod_info, +- int gr, +- int ch ); +- void calc_noise( double xr[576], +- int ix[576], +- gr_info *cod_info, +- double xfsf[4][CBLIMIT] ); +- +- +- int loop_break( III_scalefac_t *scalefac, +- gr_info *cod_info, +- int gr, +- int ch ); +- void preemphasis( double xr[576], +- double xfsf[4][CBLIMIT], +- III_psy_xmin *l3_xmin, +- int gr, +- int ch, +- III_side_info_t *l3_side ); +- int amp_scalefac_bands( double xr[576], +- double xfsf[4][CBLIMIT], +- III_psy_xmin *l3_xmin, +- III_side_info_t *l3_side, +- III_scalefac_t *scalefac, +- int gr, +- int ch, +- int iteration ); +- void quantize( double xr[576], +- int ix[576], +- gr_info *cod_info ); +- int ix_max( int ix[576], +- unsigned int begin, +- unsigned int end ); +- +- +- int +- new_choose_table( int ix[576], +- unsigned int begin, +- unsigned int end ); +- #endif +--- 0 ---- +diff -r -c -N encoder/loop.c lame3.50/loop.c +*** encoder/loop.c Wed Jan 22 02:43:16 1997 +--- lame3.50/loop.c Wed Dec 31 17:00:00 1969 +*************** +*** 1,2220 **** +- /********************************************************************** +- * ISO MPEG Audio Subgroup Software Simulation Group (1996) +- * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- * +- * $Id: loop.c,v 1.2 1997/01/19 22:28:29 rowlands Exp $ +- * +- * $Log: loop.c,v $ +- * Revision 1.2 1997/01/19 22:28:29 rowlands +- * Layer 3 bug fixes from Seymour Shlien +- * +- * Revision 1.1 1996/02/14 04:04:23 rowlands +- * Initial revision +- * +- * Received from Mike Coleman +- **********************************************************************/ +- /********************************************************************** +- * date programmers comment * +- *25. 6.92 Toshiyuki Ishino Ver 1.0 * +- *29.10.92 Masahiro Iwadare Ver 2.0 * +- *17. 4.93 Masahiro Iwadare Updated for IS Modification * +- *04.11.93 Seymour Shlien Speed up inner loop * +- *09.09.95 mc@fivebats.com Several changes for updated IS,* +- * and some MPEG2-LSF support * +- *20.12.96 seymour.shlien@crc.doc.ca Fixed some bugs and improved * +- * the appearance * +- *********************************************************************/ +- +- #include <stdio.h> +- #include <stdlib.h> +- #include <math.h> +- #include <assert.h> +- #include "l3side.h" +- #include "loop.h" +- #include "huffman.h" +- #include "l3bitstream.h" +- #include "reservoir.h" +- #include "loop-pvt.h" +- +- /* #define DEBUG */ +- /* #define DEBUGSC */ +- /* #define PERFORM 3 */ +- /* If PERFORM is defined to some number, then a file encode.log +- is preduced showing the intermediate results produced by the +- outer_loop code for the frame number = PERFORM. +- */ +- #define BIN_SEARCH +- /* for speeding up the iteration_loop algorithm */ +- +- #ifdef PERFORM +- FILE *log_output; +- extern int frameNum; +- float worst_xfsf_to_xmin_ratio(III_psy_xmin *l3_xmin, double xfsf[4][CBLIMIT] +- ,int block_type,int gr,int ch); +- #endif +- +- /* New SS 20-12-96 */ +- #ifdef BIN_SEARCH +- int bin_search_StepSize(int desired_rate, double start, int ix[576], +- double xrs[576], gr_info * cod_info); +- int count_bits(); +- float worst_xfsf_to_xmin_ratio(); +- #endif +- +- +- /* +- Here are MPEG1 Table B.8 and MPEG2 Table B.1 +- -- Layer III scalefactor bands. +- Index into this using a method such as: +- idx = fr_ps->header->sampling_frequency +- + (fr_ps->header->version * 3) +- */ +- +- struct scalefac_struct sfBandIndex[6] = +- { +- +- { /* Table B.2.b: 22.05 kHz */ +- {0,6,12,18,24,30,36,44,54,66,80,96,116,140,168,200,238,284,336,396,464,522,576}, +- {0,4,8,12,18,24,32,42,56,74,100,132,174,192} +- }, +- { /* Table B.2.c: 24 kHz */ +- {0,6,12,18,24,30,36,44,54,66,80,96,114,136,162,194,232,278,330,394,464,540,576}, +- {0,4,8,12,18,26,36,48,62,80,104,136,180,192} +- }, +- { /* Table B.2.a: 16 kHz */ +- {0,6,12,18,24,30,36,44,45,66,80,96,116,140,168,200,238,248,336,396,464,522,576}, +- {0,4,8,12,18,26,36,48,62,80,104,134,174,192} +- }, +- { /* Table B.8.b: 44.1 kHz */ +- {0,4,8,12,16,20,24,30,36,44,52,62,74,90,110,134,162,196,238,288,342,418,576}, +- {0,4,8,12,16,22,30,40,52,66,84,106,136,192} +- }, +- { /* Table B.8.c: 48 kHz */ +- {0,4,8,12,16,20,24,30,36,42,50,60,72,88,106,128,156,190,230,276,330,384,576}, +- {0,4,8,12,16,22,28,38,50,64,80,100,126,192} +- }, +- { /* Table B.8.a: 32 kHz */ +- {0,4,8,12,16,20,24,30,36,44,54,66,82,102,126,156,194,240,296,364,448,550,576}, +- {0,4,8,12,16,22,30,42,58,78,104,138,180,192} +- } +- }; +- +- /* +- The following table is used to implement the scalefactor +- partitioning for MPEG2 as described in section +- 2.4.3.2 of the IS. The indexing corresponds to the +- way the tables are presented in the IS: +- +- [table_number][row_in_table][column of nr_of_sfb] +- */ +- static unsigned nr_of_sfb_block[6][3][4] = +- { +- { +- {6, 5, 5, 5}, +- {9, 9, 9, 9}, +- {6, 9, 9, 9} +- }, +- { +- {6, 5, 7, 3}, +- {9, 9, 12, 6}, +- {6, 9, 12, 6} +- }, +- { +- {11, 10, 0, 0}, +- {18, 18, 0, 0}, +- {15,18,0,0} +- }, +- { +- {7, 7, 7, 0}, +- {12, 12, 12, 0}, +- {6, 15, 12, 0} +- }, +- { +- {6, 6, 6, 3}, +- {12, 9, 9, 6}, +- {6, 12, 9, 6} +- }, +- { +- {8, 8, 5, 0}, +- {15,12,9,0}, +- {6,18,9,0} +- } +- }; +- +- /* +- table of largest scalefactors for MPEG2 +- */ +- static unsigned max_sfac_tab[6][4] = +- { +- {4, 4, 3, 3}, +- {4, 4, 3, 0}, +- {3, 2, 0, 0}, +- {4, 5, 5, 0}, +- {3, 3, 3, 0}, +- {2, 2, 0, 0} +- }; +- +- /* Table B.6: layer3 preemphasis */ +- int pretab[21] = +- { +- 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, +- 1, 1, 1, 1, 2, 2, 3, 3, 3, 2 +- }; +- +- /* This is the scfsi_band table from 2.4.2.7 of the IS */ +- int scfsi_band_long[5] = { 0, 6, 11, 16, 21 }; +- +- int *scalefac_band_long = &sfBandIndex[3].l[0]; +- int *scalefac_band_short = &sfBandIndex[3].s[0]; +- +- +- +- /************************************************************************/ +- /* iteration_loop() */ +- /************************************************************************/ +- void +- iteration_loop( double pe[][2], double xr_org[2][2][576], III_psy_ratio *ratio, +- III_side_info_t *l3_side, int l3_enc[2][2][576], +- int mean_bits, int stereo, double xr_dec[2][2][576], +- III_scalefac_t *scalefac, frame_params *fr_ps, +- int ancillary_pad, int bitsPerFrame ) +- { +- static int firstcall = 1; +- III_psy_xmin l3_xmin; +- gr_info *cod_info; +- layer *info; +- int *main_data_begin; +- +- int max_bits; +- int ch, gr, sfb, i, mode_gr; +- static int huffman_read_flag = 0; +- +- double xr[2][2][576]; +- I576 *ix; +- +- main_data_begin = &l3_side->main_data_begin; +- l3_side->resvDrain = 0; +- +- if ( firstcall ) +- { +- *main_data_begin = 0; +- firstcall = 0; +- #ifdef PERFORM +- log_output = fopen("encode.log","w"); +- #endif +- } +- +- info = fr_ps->header; +- mode_gr = (info->version == 1) ? 2 : 1; +- +- scalefac_band_long = &sfBandIndex[info->sampling_frequency + (info->version * 3)].l[0]; +- scalefac_band_short = &sfBandIndex[info->sampling_frequency + (info->version * 3)].s[0]; +- +- /* reading huffman code table */ +- if (huffman_read_flag == 0) { +- FILE *fi = OpenTableFile( "huffcode" ); +- if ( fi == NULL ) +- exit( EXIT_FAILURE ); +- read_huffcodetab( fi ); +- huffman_read_flag++; +- fclose( fi ); +- } +- +- +- for ( gr = 0; gr < mode_gr; gr++ ) +- { +- for ( ch = 0; ch < stereo; ch++ ) +- { +- for ( i = 0; i < 576; i++ ) +- xr[gr][ch][i] = xr_org[gr][ch][i]; +- } +- } +- +- ResvFrameBegin( fr_ps, l3_side, mean_bits, bitsPerFrame ); +- +- for ( gr = 0; gr < mode_gr; gr++ ) +- { +- for ( ch = 0; ch < stereo; ch++ ) +- { +- ix = (I576 *) l3_enc[gr][ch]; +- cod_info = (gr_info *) &(l3_side->gr[gr].ch[ch]); +- gr_deco(cod_info); +- calc_xmin( xr, ratio, cod_info, &l3_xmin, gr, ch ); +- +- #ifdef DEBUG +- printf( "----- start gr[%d] ch[%1d] : block_type=%1d, window_switching_flag=%1d (loop)\n", +- gr,ch, cod_info->block_type, cod_info->window_switching_flag ); +- #endif +- +- if ( info->version == 1 ) +- calc_scfsi( xr[gr][ch], l3_side, &l3_xmin, ch, gr ); +- +- /* calculation of number of available bit( per granule ) */ +- max_bits = ResvMaxBits( fr_ps, l3_side, &pe[gr][ch], mean_bits ); +- #ifdef DEBUG +- printf( " max_bits = %d, mean_bits = %d (iteration_loop)\n", max_bits, mean_bits ); +- #endif +- +- /* reset of iteration variables */ +- +- for ( sfb = 0; sfb < 21; sfb++ ) +- scalefac->l[gr][ch][sfb] = 0; +- for ( sfb = 0; sfb < 13; sfb++ ) +- for ( i = 0; i < 3; i++ ) +- scalefac->s[gr][ch][sfb][i] = 0; +- +- for ( i = 0; i < 4; i++ ) +- cod_info->slen[i] = 0; +- cod_info->sfb_partition_table = &nr_of_sfb_block[0][0][0]; +- +- cod_info->part2_3_length = 0; +- cod_info->big_values = 0; +- cod_info->count1 = 0; +- cod_info->scalefac_compress = 0; +- cod_info->table_select[0] = 0; +- cod_info->table_select[1] = 0; +- cod_info->table_select[2] = 0; +- cod_info->subblock_gain[0] = 0; +- cod_info->subblock_gain[1] = 0; +- cod_info->subblock_gain[2] = 0; +- cod_info->region0_count = 0; +- cod_info->region1_count = 0; +- cod_info->part2_length = 0; +- cod_info->preflag = 0; +- cod_info->scalefac_scale = 0; +- cod_info->quantizerStepSize = 0.0; +- cod_info->count1table_select= 0; +- +- /* all spectral values zero ? */ +- if ( fabs(xr_max(xr[gr][ch], 0, 576)) != 0.0 ) +- { +- cod_info->quantizerStepSize = +- (double) quantanf_init( xr[gr][ch] ); +- cod_info->part2_3_length = outer_loop( xr, max_bits, &l3_xmin, +- l3_enc, fr_ps, scalefac, +- gr, ch, l3_side ); +- } +- ResvAdjust( fr_ps, cod_info, l3_side, mean_bits ); +- +- cod_info->global_gain = nint( cod_info->quantizerStepSize + 210.0 ); +- assert( cod_info->global_gain < 256 ); +- } /* for ch */ +- } /* for gr */ +- ResvFrameEnd( fr_ps, l3_side, mean_bits ); +- } +- +- +- +- /************************************************************************/ +- /* quantanf_init */ +- /************************************************************************/ +- int quantanf_init( double xr[576] ) +- /* Function: Calculate the first quantization step quantanf. */ +- { +- int i, tp = 0; +- double system_const, minlimit; +- double sfm = 0.0, sum1 = 0.0, sum2 = 0.0; +- +- system_const = 8.0; +- minlimit = -100.0; +- +- for ( i = 0; i < 576; i++ ) +- { +- if ( xr[i] != 0 ) +- { +- double tpd = xr[i] * xr[i]; +- sum1 += log( tpd ); +- sum2 += tpd; +- } +- } +- if ( sum2 != 0.0 ) +- { +- sfm = exp( sum1 / 576.0 ) / (sum2 / 576.0); +- tp = nint( system_const * log(sfm) ); +- if ( tp < minlimit ) +- tp = minlimit; +- #ifdef DEBUG +- printf(" quantanf = %d (quantanf_init)\n",tp ); +- #endif +- } +- return(tp-70.0); /* SS 19-12-96. Starting value of +- global_gain or quantizerStepSize +- has to be reduced for iteration_loop +- */ +- } +- +- +- +- +- +- /************************************************************************/ +- /* outer_loop */ +- /************************************************************************/ +- /* Function: The outer iteration loop controls the masking conditions */ +- /* of all scalefactorbands. It computes the best scalefac and */ +- /* global gain. This module calls the inner iteration loop */ +- /************************************************************************/ +- int outer_loop( +- double xr[2][2][576], /* magnitudes of the spectral values */ +- int max_bits, +- III_psy_xmin *l3_xmin, /* the allowed distortion of the scalefactor */ +- int l3_enc[2][2][576], /* vector of quantized values ix(0..575) */ +- frame_params *fr_ps, +- III_scalefac_t *scalefac, /* scalefactors */ +- int gr, int ch, III_side_info_t *l3_side ) +- { +- int status ; +- int scalesave_l[CBLIMIT], scalesave_s[CBLIMIT][3]; +- int sfb, bits, huff_bits, save_preflag, save_compress; +- double xfsf[4][CBLIMIT]; +- int i, over, iteration; +- float max_ratio; +- +- +- /* D576 *xrs; */ /* to eliminate warning messages from gcc compiler */ +- /* I576 *ix; */ /* replace this code with below. S. Shlien 15-1-97 */ +- +- double *xrs; +- int *ix; +- gr_info *cod_info = &l3_side->gr[gr].ch[ch].tt; +- +- /* xrs = (D576 *) &xr[gr][ch][0]; */ +- /* ix = (I576 *) l3_enc[gr][ch]; */ +- +- +- xrs = (double *) &(xr[gr][ch][0]); +- ix = (int *) &(l3_enc[gr][ch][0]); +- +- +- iteration = 0; +- #ifdef PERFORM +- if(frameNum == PERFORM) +- fprintf(log_output,"\n\nframe = %d ch = %d gr= %d\n",frameNum,ch,gr); +- #endif +- do +- { +- iteration += 1; +- cod_info->part2_length = part2_length( scalefac, fr_ps, gr, ch, l3_side ); +- huff_bits = max_bits - cod_info->part2_length; +- +- #ifdef BIN_SEARCH +- if(iteration == 1) +- { +- bin_search_StepSize(max_bits,cod_info->quantizerStepSize, +- ix,xrs,cod_info); /* speeds things up a bit */ +- } +- #endif +- #ifdef PERFORM +- if(frameNum==PERFORM) +- fprintf(log_output,"\n Interim Results %d\n\n",iteration); +- bits = test_inner_loop( xr, l3_enc, huff_bits, cod_info, gr, ch, +- xfsf,l3_xmin); +- #else +- bits = inner_loop( xr, l3_enc, huff_bits, cod_info, gr, ch ); +- #endif +- +- calc_noise( &xr[gr][ch][0], &l3_enc[gr][ch][0], cod_info, xfsf ); /* distortion calculation */ +- +- for ( sfb = 0; sfb < CBLIMIT; sfb++ ) /* save scaling factors */ +- scalesave_l[sfb] = scalefac->l[gr][ch][sfb]; +- +- for ( sfb = 0; sfb < SFB_SMAX; sfb++ ) +- for ( i = 0; i < 3; i++ ) +- scalesave_s[sfb][i] = scalefac->s[gr][ch][sfb][i]; +- +- save_preflag = cod_info->preflag; +- save_compress = cod_info->scalefac_compress; +- +- preemphasis( &xr[gr][ch][0], xfsf, l3_xmin, gr, ch, l3_side ); +- +- +- +- #ifdef PERFORM +- if(frameNum == PERFORM) +- { +- fprintf(log_output,"\nbits = %d huff_bits= %d ", bits,huff_bits); +- fprintf(log_output," max_bits = %d\n",max_bits); +- fprintf(log_output,"Stepsize = %f ",cod_info->quantizerStepSize); +- fprintf(log_output," scale_bits = %d \n", cod_info->part2_length ); +- print_scalefacs(log_output,scalefac,cod_info->block_type,gr,ch); +- /*if (gr==0 && ch==0) +- print_quantized_values(log_output,l3_enc[gr][ch] ,cod_info);*/ +- max_ratio = worst_xfsf_to_xmin_ratio(l3_xmin,xfsf,cod_info->block_type,gr,ch); +- fprintf(log_output,"max_ratio = %6.2f\n",max_ratio ); +- print_ratios(log_output,l3_xmin,xfsf,cod_info->block_type,gr,ch); +- fprintf(log_output,"\n\n"); +- fflush(log_output); +- } +- #endif +- over = amp_scalefac_bands( &xr[gr][ch][0], xfsf, l3_xmin, +- l3_side, scalefac, gr, ch, iteration ); +- +- #if 1 +- if ( (status = loop_break(scalefac, cod_info, gr, ch)) == 0 ) +- { +- if ( fr_ps->header->version == 1 ) +- status = scale_bitcount( scalefac, cod_info, gr, ch ); +- else +- status = scale_bitcount_lsf( scalefac, cod_info, gr, ch ); +- } +- #else +- status = loop_break( scalefac, cod_info, gr, ch ); +- if ( fr_ps->header->version == 1 ) +- status += scale_bitcount( scalefac, cod_info, gr, ch ); +- else +- status += scale_bitcount_lsf( scalefac, cod_info, gr, ch ); +- #endif +- +- } +- while ( (status == 0) && (over > 0) ); +- +- cod_info->preflag = save_preflag; +- cod_info->scalefac_compress = save_compress; +- +- for ( sfb = 0; sfb < 21; sfb++ ) +- scalefac->l[gr][ch][sfb] = scalesave_l[sfb]; +- +- for ( i = 0; i < 3; i++ ) +- for ( sfb = 0; sfb < 12; sfb++ ) +- scalefac->s[gr][ch][sfb][i] = scalesave_s[sfb][i]; +- +- cod_info->part2_length = part2_length( scalefac, fr_ps, gr, ch, l3_side ); +- cod_info->part2_3_length = cod_info->part2_length + bits; +- +- #ifdef PERFORM +- if(frameNum == PERFORM) +- { +- fprintf(log_output,"\n Final Results\n"); +- fprintf(log_output,"bits = %d huff_bits= %d", bits,huff_bits); +- fprintf(log_output," max_bits = %d\n",max_bits); +- fprintf(log_output," Stepsize = %f ",cod_info->quantizerStepSize); +- fprintf(log_output, " scale_bits = %d \n", cod_info->part2_length ); +- max_ratio = worst_xfsf_to_xmin_ratio(l3_xmin,xfsf,cod_info->block_type,gr,ch); +- print_scalefacs(log_output,scalefac,cod_info->block_type,gr,ch); +- fprintf(log_output,"max_ratio = %6.2f\n",max_ratio ); +- print_ratios(log_output,l3_xmin,xfsf,cod_info->block_type,gr,ch); +- fflush(log_output); +- } +- #endif +- return cod_info->part2_3_length; +- } +- +- +- +- +- /***************************************************************************/ +- /* inner_loop */ +- /***************************************************************************/ +- /* The code selects the best quantizerStepSize for a particular set +- /* of scalefacs */ +- +- int +- inner_loop( double xr[2][2][576], int l3_enc[2][2][576], int max_bits, +- gr_info *cod_info, int gr, int ch ) +- { +- int bits, c1bits, bvbits; +- double *xrs; /* D576 *xr; */ +- int *ix; /* I576 *ix; */ +- xrs = &xr[gr][ch][0]; +- ix = l3_enc[gr][ch]; +- +- assert( max_bits >= 0 ); +- cod_info->quantizerStepSize -= 1.0;; +- do +- { +- do +- { +- cod_info->quantizerStepSize += 1.0; +- quantize( xrs, ix, cod_info ); +- } +- while ( ix_max(ix, 0, 576) > 8191 + 14 ); /* within table range? */ +- +- calc_runlen( ix, cod_info ); /*rzero,count1,big_values*/ +- bits = c1bits = count1_bitcount( ix, cod_info ); /*count1_table selection*/ +- subdivide( cod_info ); /* bigvalues sfb division */ +- bigv_tab_select( ix, cod_info ); /* codebook selection*/ +- bits += bvbits = bigv_bitcount( ix, cod_info ); /* bit count */ +- +- #ifdef PERFORM +- if(frameNum == PERFORM) +- fprintf(log_output,"StepSize=%f bits = %d huff_bits = %d\n", +- cod_info->quantizerStepSize,bits,max_bits); +- #endif +- +- } +- while ( bits > max_bits ); +- +- return bits; +- } +- +- +- +- /***************************************************************************/ +- /* calc_scfsi */ +- /***************************************************************************/ +- /* calculation of the scalefactor select information ( scfsi ) */ +- +- void calc_scfsi( double xr[576], III_side_info_t *l3_side, +- III_psy_xmin *l3_xmin, int ch, int gr ) +- { +- static int en_tot[2][2]; /* ch,gr */ +- static int en[2][2][21]; +- static int xm[2][2][21]; +- static int xrmax[2][2]; +- +- int en_tot_krit = 10; +- int en_dif_krit = 100; +- int en_scfsi_band_krit = 10; +- int xm_scfsi_band_krit = 10; +- +- int scfsi_band; +- unsigned scfsi_set; +- +- int sfb, start, end, i; +- int condition = 0; +- double temp, log2 = log( 2.0 ); +- gr_info *cod_info = &l3_side->gr[gr].ch[ch].tt; +- +- xrmax[gr][ch] = xr_max( xr, 0, 576 ); +- scfsi_set = 0; +- +- /* the total energy of the granule */ +- for ( temp = 0.0, i = 0; i < 576; i++ ) +- temp += xr[i] * xr[i]; +- if ( temp == 0.0 ) +- en_tot[gr][ch] = 0.0; +- else +- en_tot[gr][ch] = log( temp ) / log2 ; +- +- /* the energy of each scalefactor band, en */ +- /* the allowed distortion of each scalefactor band, xm */ +- +- if ( cod_info->window_switching_flag == 0 || +- cod_info->block_type != 2 ) +- for ( sfb = 0; sfb < 21; sfb++ ) +- { +- start = scalefac_band_long[ sfb ]; +- end = scalefac_band_long[ sfb+1 ]; +- +- for ( temp = 0.0, i = start; i < end; i++ ) +- temp += xr[i] * xr[i]; +- if ( temp == 0.0 ) +- en[gr][ch][sfb] = 0.0; +- else +- en[gr][ch][sfb] = log( temp )/ log2; +- +- if ( l3_xmin->l[gr][ch][sfb] == 0.0 ) +- xm[gr][ch][sfb] = 0.0; +- else +- xm[gr][ch][sfb] = log( l3_xmin->l[gr][ch][sfb] ) / log2; +- } +- if ( gr == 1 ) +- { +- int gr2, tp; +- +- for ( gr2 = 0; gr2 < 2; gr2++ ) +- { +- /* The spectral values are not all zero */ +- if ( xrmax[ch][gr2] != 0.0 ) +- condition++; +- /* None of the granules contains short blocks */ +- if ( (cod_info->window_switching_flag == 0) || +- (cod_info->block_type != 2) ) +- condition++; +- } +- if ( abs(en_tot[0] - en_tot[1]) < en_tot_krit ) +- condition++; +- for ( tp = 0, sfb = 0; sfb < 21; sfb++ ) +- tp += abs( en[ch][0][sfb] - en[ch][1][sfb] ); +- if ( tp < en_dif_krit ) +- condition++; +- +- if ( condition == 6 ) +- { +- for ( scfsi_band = 0; scfsi_band < 4; scfsi_band++ ) +- { +- int sum0 = 0, sum1 = 0; +- l3_side->scfsi[ch][scfsi_band] = 0; +- start = scfsi_band_long[scfsi_band]; +- end = scfsi_band_long[scfsi_band+1]; +- for ( sfb = start; sfb < end; sfb++ ) +- { +- sum0 += abs( en[ch][0][sfb] - en[ch][1][sfb] ); +- sum1 += abs( xm[ch][0][sfb] - xm[ch][1][sfb] ); +- } +- +- if ( sum0 < en_scfsi_band_krit && sum1 < xm_scfsi_band_krit ) +- { +- l3_side->scfsi[ch][scfsi_band] = 1; +- scfsi_set |= (1 << scfsi_band); +- } +- else +- l3_side->scfsi[ch][scfsi_band] = 0; +- } /* for scfsi_band */ +- #ifdef DEBUG +- fprintf( stderr, "calc_scfsi: scfsi_set = 0x%02x\n", scfsi_set ); +- #endif +- } /* if condition == 6 */ +- else +- for ( scfsi_band = 0; scfsi_band < 4; scfsi_band++ ) +- l3_side->scfsi[ch][scfsi_band] = 0; +- } /* if gr == 1 */ +- } +- +- +- +- /***************************************************************************/ +- /* part2_length */ +- /***************************************************************************/ +- +- /* calculates the number of bits needed to encode the scalefacs in the */ +- /* main data block */ +- +- int part2_length( III_scalefac_t *scalefac, frame_params *fr_ps, +- int gr, int ch, III_side_info_t *si ) +- { +- int slen1, slen2, slen3, slen4, bits, sfb, window, partition; +- gr_info *gi = &si->gr[gr].ch[ch].tt; +- +- bits = 0; +- if ( fr_ps->header->version == 1 ) +- { +- static int slen1_tab[16] = { 0, 0, 0, 0, 3, 1, 1, 1, 2, 2, 2, 3, 3, 3, 4, 4 }; +- static int slen2_tab[16] = { 0, 1, 2, 3, 0, 1, 2, 3, 1, 2, 3, 1, 2, 3, 2, 3 }; +- +- slen1 = slen1_tab[ gi->scalefac_compress ]; +- slen2 = slen2_tab[ gi->scalefac_compress ]; +- +- if ( (gi->window_switching_flag == 1) && (gi->block_type == 2) ) +- { +- if ( gi->mixed_block_flag ) +- { +- bits += (8 * slen1) + (9 * slen1) + (18 * slen2); +- } +- else +- { +- bits += (18 * slen1) + (18 * slen2); +- } +- } +- else +- { +- if ( (gr == 0) || (si->scfsi[ch][0] == 0) ) +- bits += (6 * slen1); +- +- if ( (gr == 0) || (si->scfsi[ch][1] == 0) ) +- /* bits += (6 * slen1); This is wrong SS 19-12-96 */ +- bits += (5 * slen1); +- +- if ( (gr == 0) || (si->scfsi[ch][2] == 0) ) +- /* bits += (6 * slen1); This is wrong SS 19-12-96 */ +- bits += (5 * slen2); +- +- if ( (gr == 0) || (si->scfsi[ch][3] == 0) ) +- /* bits += (6 * slen1); This is wrong SS 19-12-96 */ +- bits += (5 * slen2); +- } +- } +- else +- { /* MPEG 2 */ +- assert( gi->sfb_partition_table ); +- for ( partition = 0; partition < 4; partition++ ) +- bits += gi->slen[partition] * gi->sfb_partition_table[partition]; +- } +- return bits; +- } +- +- +- +- /*************************************************************************/ +- /* scale_bitcount */ +- /*************************************************************************/ +- +- /* Also calculates the number of bits necessary to code the scalefactors. */ +- +- int scale_bitcount( III_scalefac_t *scalefac, gr_info *cod_info, +- int gr, int ch ) +- { +- int i, k, sfb, max_slen1 = 0, max_slen2 = 0, /*a, b, */ ep = 2; +- +- static int slen1[16] = { 0, 0, 0, 0, 3, 1, 1, 1, 2, 2, 2, 3, 3, 3, 4, 4 }; +- static int slen2[16] = { 0, 1, 2, 3, 0, 1, 2, 3, 1, 2, 3, 1, 2, 3, 2, 3 }; +- static int pow2[5] = { 1, 2, 4, 8, 16 }; +- +- if ( cod_info->window_switching_flag != 0 && cod_info->block_type == 2 ) +- { +- if ( cod_info->mixed_block_flag == 0 ) +- { +- /* a = 18; b = 18; */ +- for ( i = 0; i < 3; i++ ) +- { +- for ( sfb = 0; sfb < 6; sfb++ ) +- if ( scalefac->s[gr][ch][sfb][i] > max_slen1 ) +- max_slen1 = scalefac->s[gr][ch][sfb][i]; +- for (sfb = 6; sfb < 12; sfb++ ) +- if ( scalefac->s[gr][ch][sfb][i] > max_slen2 ) +- max_slen2 = scalefac->s[gr][ch][sfb][i]; +- } +- } +- else +- {/* mixed_block_flag = 1 */ +- /* a = 17; b = 18; */ +- for ( sfb = 0; sfb < 8; sfb++ ) +- if ( scalefac->l[gr][ch][sfb] > max_slen1 ) +- max_slen1 = scalefac->l[gr][ch][sfb]; +- for ( i = 0; i < 3; i++ ) +- { +- for ( sfb = 3; sfb < 6; sfb++ ) +- if ( scalefac->s[gr][ch][sfb][i] > max_slen1 ) +- max_slen1 = scalefac->s[gr][ch][sfb][i]; +- for ( sfb = 6; sfb < 12; sfb++ ) +- if ( scalefac->s[gr][ch][sfb][i] > max_slen2 ) +- max_slen2 = scalefac->s[gr][ch][sfb][i]; +- } +- } +- } +- else +- { /* block_type == 1,2,or 3 */ +- /* a = 11; b = 10; */ +- for ( sfb = 0; sfb < 11; sfb++ ) +- if ( scalefac->l[gr][ch][sfb] > max_slen1 ) +- max_slen1 = scalefac->l[gr][ch][sfb]; +- for ( sfb = 11; sfb < 21; sfb++ ) +- if ( scalefac->l[gr][ch][sfb] > max_slen2 ) +- max_slen2 = scalefac->l[gr][ch][sfb]; +- } +- +- for ( k = 0; k < 16; k++ ) +- { +- if ( (max_slen1 < pow2[slen1[k]]) && (max_slen2 < pow2[slen2[k]]) ) +- { +- ep = 0; +- break; +- } +- } +- +- if ( ep == 0 ) +- cod_info->scalefac_compress = k; +- #ifdef DEBUG +- if ( ep != 0 ) +- printf( "---WARNING !! Amplification of some bands over limits\n" ); +- #endif +- return ep; +- } +- +- +- +- +- /*************************************************************************/ +- /* scale_bitcount_lsf */ +- /*************************************************************************/ +- +- /* Also counts the number of bits to encode the scalefacs but for MPEG 2 */ +- /* Lower sampling frequencies (24, 22.05 and 16 kHz.) */ +- +- /* This is reverse-engineered from section 2.4.3.2 of the MPEG2 IS, */ +- /* "Audio Decoding Layer III" */ +- +- int scale_bitcount_lsf( III_scalefac_t *scalefac, gr_info *cod_info, +- int gr, int ch ) +- { +- int table_number, row_in_table, partition, nr_sfb, window, over; +- int i, k, sfb, max_sfac[ 4 ]; +- unsigned *partition_table; +- +- /* +- Set partition table. Note that should try to use table one, +- but do not yet... +- */ +- if ( cod_info->preflag ) +- table_number = 2; +- else +- table_number = 0; +- +- for ( i = 0; i < 4; i++ ) +- max_sfac[i] = 0; +- +- if ( cod_info->window_switching_flag != 0 && cod_info->block_type == 2 ) +- { +- if ( cod_info->mixed_block_flag == 0 ) +- { +- row_in_table = 1; +- partition_table = &nr_of_sfb_block[table_number][row_in_table][0]; +- for ( sfb = 0, partition = 0; partition < 4; partition++ ) +- { +- nr_sfb = partition_table[ partition ] / 3; +- for ( i = 0; i < nr_sfb; i++, sfb++ ) +- for ( window = 0; window < 3; window++ ) +- if ( scalefac->s[gr][ch][sfb][window] > max_sfac[partition] ) +- max_sfac[partition] = scalefac->s[gr][ch][sfb][window]; +- } +- } +- else +- {/* mixed_block_flag = 1 */ +- row_in_table = 2; +- partition_table = &nr_of_sfb_block[table_number][row_in_table][0]; +- partition = 0; +- nr_sfb = partition_table[ partition ]; +- for ( i = 0; i < nr_sfb; i++, sfb++ ) +- if ( scalefac->l[gr][ch][sfb] > max_sfac[partition] ) +- max_sfac[partition] = scalefac->l[gr][ch][sfb]; +- +- for ( sfb = 0, partition = 1; partition < 4; partition++ ) +- { +- nr_sfb = partition_table[ partition ] / 3; +- for ( i = 0; i < nr_sfb; i++, sfb++ ) +- for ( window = 0; window < 3; window++ ) +- if ( scalefac->s[gr][ch][sfb][window] > max_sfac[partition] ) +- max_sfac[partition] = scalefac->s[gr][ch][sfb][window]; +- } +- } +- } +- else +- { +- row_in_table = 0; +- partition_table = &nr_of_sfb_block[table_number][row_in_table][0]; +- partition = 0; +- for ( sfb = 0, partition = 0; partition < 4; partition++ ) +- { +- nr_sfb = partition_table[ partition ]; +- for ( i = 0; i < nr_sfb; i++, sfb++ ) +- if ( scalefac->l[gr][ch][sfb] > max_sfac[partition] ) +- max_sfac[partition] = scalefac->l[gr][ch][sfb]; +- } +- } +- +- for ( over = 0, partition = 0; partition < 4; partition++ ) +- { +- if ( max_sfac[partition] > max_sfac_tab[table_number][partition] ) +- over++; +- } +- if ( !over ) +- { +- /* +- Since no bands have been over-amplified, we can set scalefac_compress +- and slen[] for the formatter +- */ +- static int log2tab[] = { 0, 1, 2, 2, 3, 3, 3, 3, 4, 4, 4, 4, 4, 4, 4, 4 }; +- +- unsigned slen1, slen2, slen3, slen4; +- +- cod_info->sfb_partition_table = &nr_of_sfb_block[table_number][row_in_table][0]; +- for ( partition = 0; partition < 4; partition++ ) +- cod_info->slen[partition] = log2tab[max_sfac[partition]]; +- +- /* set scalefac_compress */ +- slen1 = cod_info->slen[ 0 ]; +- slen2 = cod_info->slen[ 1 ]; +- slen3 = cod_info->slen[ 2 ]; +- slen4 = cod_info->slen[ 3 ]; +- +- switch ( table_number ) +- { +- case 0: +- cod_info->scalefac_compress = (((slen1 * 5) + slen2) << 4) +- + (slen3 << 2) +- + slen4; +- break; +- +- case 1: +- cod_info->scalefac_compress = 400 +- + (((slen1 * 5) + slen2) << 2) +- + slen3; +- break; +- +- case 2: +- cod_info->scalefac_compress = 500 + (slen1 * 3) + slen2; +- break; +- +- default: +- fprintf( stderr, "intensity stereo not implemented yet\n" ); +- exit( EXIT_FAILURE ); +- break; +- } +- } +- #ifdef DEBUG +- if ( over ) +- printf( "---WARNING !! Amplification of some bands over limits\n" ); +- #endif +- return over; +- } +- +- +- /*************************************************************************/ +- /* calc_noise */ +- /*************************************************************************/ +- +- /* Function: Calculate the distortion introduced by the quantization */ +- /* in each scale factor band. */ +- +- void calc_noise( double xr[576], int ix[576], gr_info *cod_info, +- double xfsf[4][CBLIMIT] ) +- { +- int start, end, sfb, l, i; +- double sum,step,bw; +- +- D192_3 *xr_s; +- I192_3 *ix_s; +- +- xr_s = (D192_3 *) xr; +- ix_s = (I192_3 *) ix; +- +- step = pow( 2.0, (cod_info->quantizerStepSize) * 0.25 ); +- for ( sfb = 0; sfb < cod_info->sfb_lmax; sfb++ ) +- { +- start = scalefac_band_long[ sfb ]; +- end = scalefac_band_long[ sfb+1 ]; +- bw = end - start; +- +- for ( sum = 0.0, l = start; l < end; l++ ) +- { +- double temp; +- temp = fabs( xr[l] ) - pow( (double) ix[l], 4.0 / 3.0 ) * step; +- sum += temp * temp; +- } +- xfsf[0][sfb] = sum / bw; +- } +- +- for ( i = 0; i < 3; i++ ) +- { +- step = pow( 2.0, (cod_info->quantizerStepSize) * 0.25 ); /* subblock_gain ? */ +- for ( sfb = cod_info->sfb_smax; sfb < 12; sfb++ ) +- { +- start = scalefac_band_short[ sfb ]; +- end = scalefac_band_short[ sfb+1 ]; +- bw = end - start; +- +- for ( sum = 0.0, l = start; l < end; l++ ) +- { +- double temp; +- temp = fabs( (*xr_s)[l][i] ) - pow( (double) (*ix_s)[l][i], 4.0 / 3.0 ) * step; +- sum += temp * temp; +- } +- xfsf[i+1][sfb] = sum / bw; +- } +- } +- } +- +- +- +- +- /*************************************************************************/ +- /* calc_xmin */ +- /*************************************************************************/ +- +- /* +- Calculate the allowed distortion for each scalefactor band, +- as determined by the psychoacoustic model. +- xmin(sb) = ratio(sb) * en(sb) / bw(sb) +- */ +- +- void calc_xmin( double xr[2][2][576], III_psy_ratio *ratio, +- gr_info *cod_info, III_psy_xmin *l3_xmin, +- int gr, int ch ) +- { +- int start, end, sfb, l, b; +- double en, bw; +- +- D192_3 *xr_s; +- +- xr_s = (D192_3 *) xr[gr][ch] ; +- +- for ( sfb = cod_info->sfb_smax; sfb < SFB_SMAX - 1; sfb++ ) +- { +- start = scalefac_band_short[ sfb ]; +- end = scalefac_band_short[ sfb + 1 ]; +- bw = end - start; +- for ( b = 0; b < 3; b++ ) +- { +- for ( en = 0.0, l = start; l < end; l++ ) +- en += (*xr_s)[l][b] * (*xr_s)[l][b]; +- l3_xmin->s[gr][ch][sfb][b] = ratio->s[gr][ch][sfb][b] * en / bw; +- } +- } +- +- for ( sfb = 0; sfb < cod_info->sfb_lmax; sfb++ ) +- { +- start = scalefac_band_long[ sfb ]; +- end = scalefac_band_long[ sfb+1 ]; +- bw = end - start; +- +- for ( en = 0.0, l = start; l < end; l++ ) +- en += xr[gr][ch][l] * xr[gr][ch][l]; +- l3_xmin->l[gr][ch][sfb] = ratio->l[gr][ch][sfb] * en / bw; +- } +- } +- +- +- +- /*************************************************************************/ +- /* loop_break */ +- /*************************************************************************/ +- +- /* Function: Returns zero if there is a scalefac which has not been +- amplified. Otherwise it returns one. +- */ +- +- int loop_break( III_scalefac_t *scalefac, gr_info *cod_info, +- int gr, int ch ) +- { +- int i, sfb, temp = 1; +- +- for ( sfb = 0; sfb < cod_info->sfb_lmax; sfb++ ) +- if ( scalefac->l[gr][ch][sfb] == 0 ) +- temp = 0; +- +- for ( sfb = cod_info->sfb_smax; sfb < 12; sfb++ ) +- for ( i = 0; i < 3; i++ ) +- if ( scalefac->s[gr][ch][sfb][i] == 0 ) +- temp = 0; +- #ifdef DEBUG +- if ( temp != 0 ) +- printf( "---WARNING !! All scalefactor bands amplified\n" ); +- #endif +- return temp; +- } +- +- +- +- /*************************************************************************/ +- /* preemphasis */ +- /*************************************************************************/ +- +- /* +- See ISO 11172-3 section C.1.5.4.3.4 +- */ +- +- void preemphasis( double xr[576], double xfsf[4][CBLIMIT], +- III_psy_xmin *l3_xmin, +- int gr, int ch, III_side_info_t *l3_side ) +- { +- int i, sfb, start, end, scfsi_band, over; +- double ifqstep; +- gr_info *cod_info = &l3_side->gr[gr].ch[ch].tt; +- +- if ( gr == 1 ) +- { +- /* +- If the second granule is being coded and scfsi is active in +- at least one scfsi_band, the preemphasis in the second granule +- is set equal to the setting in the first granule +- */ +- for ( scfsi_band = 0; scfsi_band < 4; scfsi_band++ ) +- if ( l3_side->scfsi[ch][scfsi_band] ) +- { +- cod_info->preflag = l3_side->gr[0].ch[ch].tt.preflag; +- return; +- } +- +- } +- +- /* +- Preemphasis is switched on if in all the upper four scalefactor +- bands the actual distortion exceeds the threshold after the +- first call of the inner loop +- */ +- if ( cod_info->block_type != 2 && cod_info->preflag == 0 ) +- { +- over = 0; +- for ( sfb = 17; sfb < 21; sfb++ ) +- if ( xfsf[0][sfb] > l3_xmin->l[gr][ch][sfb] ) +- over++; +- +- if (over == 4 ) +- { +- cod_info->preflag = 1; +- ifqstep = ( cod_info->scalefac_scale == 0 ) ? sqrt(2.) +- : pow( 2.0, (0.5 * (1.0 + (double) cod_info->scalefac_scale)) ); +- +- for ( sfb = 0; sfb < cod_info->sfb_lmax; sfb++ ) +- { +- l3_xmin->l[gr][ch][sfb] *= pow( ifqstep, 2.0 * (double) pretab[sfb] ); +- start = scalefac_band_long[ sfb ]; +- end = scalefac_band_long[ sfb+1 ]; +- for( i = start; i < end; i++ ) +- xr[i] *= pow( ifqstep, (double) pretab[sfb] ); +- } +- } +- } +- } +- +- +- /*************************************************************************/ +- /* amp_scalefac_bands */ +- /*************************************************************************/ +- +- /* +- Amplify the scalefactor bands that violate the masking threshold. +- See ISO 11172-3 Section C.1.5.4.3.5 +- */ +- +- int amp_scalefac_bands( double xr[576], double xfsf[4][CBLIMIT], +- III_psy_xmin *l3_xmin, III_side_info_t *l3_side, +- III_scalefac_t *scalefac, +- int gr, int ch, int iteration ) +- { +- int start, end, l, sfb, i, scfsi_band, over = 0; +- double ifqstep, ifqstep2; +- D192_3 *xr_s; +- gr_info *cod_info, *gr0; +- int copySF, preventSF; +- cod_info = &l3_side->gr[gr].ch[ch].tt; +- gr0 = &l3_side->gr[0].ch[ch].tt; +- +- xr_s = (D192_3 *) xr; +- copySF = 0; +- preventSF = 0; +- +- if ( cod_info->scalefac_scale == 0 ) +- ifqstep = sqrt( 2.0 ); +- else +- ifqstep = pow( 2.0, 0.5 * (1.0 + (double) cod_info->scalefac_scale) ); +- +- if ( gr == 1 ) +- { +- /* +- If the second granule is being coded and scfsi is active in at +- least one scfsi_band... +- */ +- for ( scfsi_band = 0; scfsi_band < 4; scfsi_band++ ) +- if ( l3_side->scfsi[ch][scfsi_band] ) +- { +- /* +- a) ifqstep has to be set similar to the +- first granule... +- */ +- if ( gr0->scalefac_scale == 0 ) +- ifqstep = sqrt( 2.0 ); +- else +- ifqstep = pow( 2.0, 0.5 * (1.0 + (double) gr0->scalefac_scale) ); +- +- if ( iteration == 1 ) +- { +- /* +- b) If it is the first iteration, the scalefactors +- of scalefactor bands in which scfsi is enabled +- must be taken from the first granule +- */ +- copySF = 1; +- } +- else +- { +- /* +- c) If it is not the first iteration, the amplification +- must be prevented for scalefactor bands in which +- scfsi is enabled +- */ +- preventSF = 1; +- } +- break; +- } +- +- } +- +- ifqstep2 = ifqstep * ifqstep; +- scfsi_band = 0; +- +- for ( sfb = 0; sfb < cod_info->sfb_lmax; sfb++ ) +- { +- if ( copySF || preventSF ) +- { +- if ( sfb == scfsi_band_long[scfsi_band + 1] ) +- scfsi_band += 1; +- if ( l3_side->scfsi[ch][scfsi_band] ) +- { +- if ( copySF ) +- scalefac->l[gr][ch][sfb] = scalefac->l[0][ch][sfb]; +- continue; +- } +- } +- if ( xfsf[0][sfb] > l3_xmin->l[gr][ch][sfb] ) +- { +- over++; +- l3_xmin->l[gr][ch][sfb] *= ifqstep2; +- scalefac->l[gr][ch][sfb]++; +- start = scalefac_band_long[sfb]; +- end = scalefac_band_long[sfb+1]; +- for ( l = start; l < end; l++ ) +- xr[l] *= ifqstep; +- } +- } +- +- /* +- Note that scfsi is not enabled for frames containing +- short blocks +- */ +- for ( i = 0; i < 3; i++ ) +- for ( sfb = cod_info->sfb_smax; sfb < 12; sfb++ ) +- if ( xfsf[i+1][sfb] > l3_xmin->s[gr][ch][sfb][i] ) +- { +- over++; +- l3_xmin->s[gr][ch][sfb][i] *= ifqstep2; +- scalefac->s[gr][ch][sfb][i]++; +- #ifdef DEBUGSC +- printf( "cod_info->scalefac[%d][%d] = %d (amp_scale)\n", +- i,sfb,scalefac->s[gr][ch][sfb][i] ); +- #endif +- start = scalefac_band_short[sfb]; +- end = scalefac_band_short[sfb+1]; +- for ( l = start; l < end; l++ ) +- (*xr_s)[l][i] *= ifqstep; +- } +- return over; +- } +- +- +- +- +- /*************************************************************************/ +- /* quantize */ +- /*************************************************************************/ +- +- /* +- Function: Quantization of the vector xr ( -> ix) +- */ +- +- void quantize( double xr[576], int ix[576], gr_info *cod_info ) +- { +- int i, b, l_end, s_start; +- double step, quantizerStepSize; +- +- D192_3 *xr_s; +- I192_3 *ix_s; +- +- xr_s = (D192_3 *) xr; +- ix_s = (I192_3 *) ix; +- +- quantizerStepSize = (double) cod_info->quantizerStepSize; +- +- for ( i = 0; i < 576; i++ ) +- ix[i] = 0; +- +- if ( cod_info->quantizerStepSize == 0.0 ) +- step = 1.0; +- else +- step = pow ( 2.0, quantizerStepSize * 0.25 ); +- +- if ( cod_info->window_switching_flag != 0 && cod_info->block_type == 2 ) +- if ( cod_info->mixed_block_flag == 0 ) +- { +- l_end = 0; +- s_start = 0; +- } +- else +- { +- l_end = 18 * 2; +- s_start = 6 * 2; +- } +- else +- { +- l_end = 576; +- s_start = 192; +- } +- +- for ( i = 0; i < l_end; i++ ) +- ix[i] = nint( pow(fabs(xr[i]) / step, 0.75) - 0.0946 ); +- +- if ( s_start < 192 ) +- for ( b = 0; b < 3; b++ ) +- { +- step = pow( 2.0, (quantizerStepSize + 8.0 * (double) cod_info->subblock_gain[b]) * 0.25 ); +- for ( i = s_start; i < 192; i++ ) +- (*ix_s)[i][b] = nint( pow(fabs((*xr_s)[i][b]) / step, 0.75) - 0.0946 ); +- } +- } +- +- +- +- +- /*************************************************************************/ +- /* ix_max */ +- /*************************************************************************/ +- +- /* +- Function: Calculate the maximum of ix from 0 to 575 +- */ +- +- int ix_max( int ix[576], unsigned int begin, unsigned int end ) +- { +- int i, max = 0; +- +- for ( i = begin; i < end; i++ ) +- { +- int x = abs( ix[i] ); +- if ( x > max ) +- max = x; +- } +- return max; +- } +- +- +- /*************************************************************************/ +- /* xr_max */ +- /*************************************************************************/ +- +- /* +- Function: Calculate the maximum of xr[576] from 0 to 575 +- */ +- +- double xr_max( double xr[576], unsigned int begin, unsigned int end ) +- { +- int i; +- double max = 0.0, temp; +- +- for ( i = begin; i < end; i++ ) +- if( (temp = fabs(xr[i])) > max ) +- max = temp; +- return max; +- } +- +- +- +- /* Noiseless coding -- Huffman coding */ +- +- +- /*************************************************************************/ +- /* calc_runlen */ +- /*************************************************************************/ +- +- /* +- Function: Calculation of rzero, count1, big_values +- (Partitions ix into big values, quadruples and zeros). +- */ +- +- void calc_runlen( int ix[576], gr_info *cod_info ) +- { +- int i; +- int rzero = 0; +- +- if ( cod_info->window_switching_flag && (cod_info->block_type == 2) ) +- { /* short blocks */ +- cod_info->count1 = 0; +- cod_info->big_values = 288; +- } +- else +- { +- for ( i = 576; i > 1; i -= 2 ) +- if ( ix[i-1] == 0 && ix[i-2] == 0 ) +- rzero++; +- else +- break; +- +- cod_info->count1 = 0 ; +- for ( ; i > 3; i -= 4 ) +- if ( abs(ix[i-1]) <= 1 +- && abs(ix[i-2]) <= 1 +- && abs(ix[i-3]) <= 1 +- && abs(ix[i-4]) <= 1 ) +- cod_info->count1++; +- else +- break; +- +- cod_info->big_values = i/2; +- } +- assert( (2 * rzero + 4 * cod_info->count1 + 2 * cod_info->big_values) == 576 ); +- } +- +- +- +- /*************************************************************************/ +- /* count1_bitcount */ +- /*************************************************************************/ +- +- /* +- Determines the number of bits to encode the quadruples. +- */ +- +- int count1_bitcount( int ix[ 576 ], gr_info *cod_info ) +- { +- int abs_and_sign( int *x ); +- +- int p, i, k, bitsum_count1; +- int v, w, x, y, signbits; +- int sum0 = 0, sum1 = 0; +- +- for ( i = cod_info->big_values * 2, k = 0; k < cod_info->count1; i += 4, k++ ) +- { +- v = ix[ i ]; +- w = ix[ i + 1 ]; +- x = ix[ i + 2 ]; +- y = ix[ i + 3 ]; +- +- abs_and_sign( &v ); +- abs_and_sign( &w ); +- abs_and_sign( &x ); +- abs_and_sign( &y ); +- +- p = v + (w << 1) + (x << 2) + (y << 3); +- +- signbits = 0; +- +- if ( v != 0 ) +- signbits += 1; +- if ( w != 0 ) +- signbits += 1; +- if ( x != 0 ) +- signbits += 1; +- if ( y != 0 ) +- signbits += 1; +- +- sum0 += signbits; +- sum1 += signbits; +- +- sum0 += ht[ 32 ].hlen[ p ]; +- sum1 += ht[ 33 ].hlen[ p ]; +- } +- +- if ( sum0 < sum1 ) +- { +- bitsum_count1 = sum0; +- cod_info->count1table_select = 0; +- } +- else +- { +- bitsum_count1 = sum1; +- cod_info->count1table_select = 1; +- } +- return( bitsum_count1 ); +- } +- +- +- +- +- +- struct +- { +- unsigned region0_count; +- unsigned region1_count; +- } subdv_table[ 23 ] = +- { +- {0, 0}, /* 0 bands */ +- {0, 0}, /* 1 bands */ +- {0, 0}, /* 2 bands */ +- {0, 0}, /* 3 bands */ +- {0, 0}, /* 4 bands */ +- {0, 1}, /* 5 bands */ +- {1, 1}, /* 6 bands */ +- {1, 1}, /* 7 bands */ +- {1, 2}, /* 8 bands */ +- {2, 2}, /* 9 bands */ +- {2, 3}, /* 10 bands */ +- {2, 3}, /* 11 bands */ +- {3, 4}, /* 12 bands */ +- {3, 4}, /* 13 bands */ +- {3, 4}, /* 14 bands */ +- {4, 5}, /* 15 bands */ +- {4, 5}, /* 16 bands */ +- {4, 6}, /* 17 bands */ +- {5, 6}, /* 18 bands */ +- {5, 6}, /* 19 bands */ +- {5, 7}, /* 20 bands */ +- {6, 7}, /* 21 bands */ +- {6, 7}, /* 22 bands */ +- }; +- +- +- +- +- /*************************************************************************/ +- /* subdivide */ +- /*************************************************************************/ +- +- /* presumable subdivides the bigvalue region which will +- use separate Huffman tables. +- */ +- +- void subdivide( gr_info *cod_info ) +- { +- int scfb_anz = 0; +- int bigvalues_region; +- +- if ( cod_info->big_values == 0 ) +- { /* no big_values region */ +- cod_info->region0_count = 0; +- cod_info->region1_count = 0; +- } +- else +- { +- bigvalues_region = 2 * cod_info->big_values; +- +- if ( (cod_info->window_switching_flag == 0) ) +- { /* long blocks */ +- int thiscount, index; +- /* Calculate scfb_anz */ +- while ( scalefac_band_long[scfb_anz] < bigvalues_region ) +- scfb_anz++; +- assert( scfb_anz < 23 ); +- +- cod_info->region0_count = subdv_table[scfb_anz].region0_count; +- thiscount = cod_info->region0_count; +- index = thiscount + 1; +- while ( thiscount && (scalefac_band_long[index] > bigvalues_region) ) +- { +- thiscount -= 1; +- index -= 1; +- } +- cod_info->region0_count = thiscount; +- +- cod_info->region1_count = subdv_table[scfb_anz].region1_count; +- index = cod_info->region0_count + cod_info->region1_count + 2; +- thiscount = cod_info->region1_count; +- while ( thiscount && (scalefac_band_long[index] > bigvalues_region) ) +- { +- thiscount -= 1; +- index -= 1; +- } +- cod_info->region1_count = thiscount; +- cod_info->address1 = scalefac_band_long[cod_info->region0_count+1]; +- cod_info->address2 = scalefac_band_long[cod_info->region0_count +- + cod_info->region1_count + 2 ]; +- cod_info->address3 = bigvalues_region; +- } +- else +- { +- if ( (cod_info->block_type == 2) && (cod_info->mixed_block_flag == 0) ) +- { +- cod_info->region0_count = 8; +- cod_info->region1_count = 36; +- cod_info->address1 = 36; +- cod_info->address2 = bigvalues_region; +- cod_info->address3 = 0; +- } +- else +- { +- cod_info->region0_count = 7; +- cod_info->region1_count = 13; +- cod_info->address1 = scalefac_band_long[cod_info->region0_count+1]; +- cod_info->address2 = bigvalues_region; +- cod_info->address3 = 0; +- } +- } +- } +- } +- +- +- +- +- /*************************************************************************/ +- /* bigv_tab_select */ +- /*************************************************************************/ +- +- /* +- /* Function: Select huffman code tables for bigvalues regions +- */ +- +- void bigv_tab_select( int ix[576], gr_info *cod_info ) +- { +- /* int max; */ +- +- cod_info->table_select[0] = 0; +- cod_info->table_select[1] = 0; +- cod_info->table_select[2] = 0; +- +- if ( cod_info->window_switching_flag && (cod_info->block_type == 2) ) +- { +- /* +- Within each scalefactor band, data is given for successive +- time windows, beginning with window 0 and ending with window 2. +- Within each window, the quantized values are then arranged in +- order of increasing frequency... +- */ +- int sfb, window, line, start, end, max1, max2, x, y; +- int region1Start; +- int *pmax; +- +- region1Start = 12; +- max1 = max2 = 0; +- +- for ( sfb = 0; sfb < 13; sfb++ ) +- { +- start = scalefac_band_short[ sfb ]; +- end = scalefac_band_short[ sfb+1 ]; +- +- if ( start < region1Start ) +- pmax = &max1; +- else +- pmax = &max2; +- +- for ( window = 0; window < 3; window++ ) +- for ( line = start; line < end; line += 2 ) +- { +- assert( line >= 0 ); +- assert( line < 576 ); +- x = abs( ix[ (line * 3) + window ] ); +- y = abs( ix[ ((line + 1) * 3) + window ]); +- *pmax = *pmax > x ? *pmax : x; +- *pmax = *pmax > y ? *pmax : y; +- } +- } +- cod_info->table_select[0] = choose_table( max1 ); +- cod_info->table_select[1] = choose_table( max2 ); +- } +- else +- { +- if ( cod_info->address1 > 0 ) +- cod_info->table_select[0] = new_choose_table( ix, 0, cod_info->address1 ); +- +- if ( cod_info->address2 > cod_info->address1 ) +- cod_info->table_select[1] = new_choose_table( ix, cod_info->address1, cod_info->address2 ); +- +- if ( cod_info->big_values * 2 > cod_info->address2 ) +- cod_info->table_select[2] = new_choose_table( ix, cod_info->address2, cod_info->big_values * 2 ); +- } +- } +- +- +- +- +- /*************************************************************************/ +- /* new_choose table */ +- /*************************************************************************/ +- +- /* +- Choose the Huffman table that will encode ix[begin..end] with +- the fewest bits. +- +- Note: This code contains knowledge about the sizes and characteristics +- of the Huffman tables as defined in the IS (Table B.7), and will not work +- with any arbitrary tables. +- */ +- +- int new_choose_table( int ix[576], unsigned int begin, unsigned int end ) +- { +- int i, max; +- int choice[ 2 ]; +- int sum[ 2 ]; +- +- max = ix_max( ix, begin, end ); +- +- if ( max == 0 ) +- return 0; +- +- max = abs( max ); +- +- choice[ 0 ] = 0; +- choice[ 1 ] = 0; +- +- if ( max < 15 ) +- { +- /* try tables with no linbits */ +- for ( i = 0; i < 14; i++ ) +- if ( ht[i].xlen > max ) +- { +- choice[ 0 ] = i; +- break; +- } +- assert( choice[0] ); +- +- sum[ 0 ] = count_bit( ix, begin, end, choice[0] ); +- +- switch ( choice[0] ) +- { +- case 2: +- sum[ 1 ] = count_bit( ix, begin, end, 3 ); +- if ( sum[1] <= sum[0] ) +- choice[ 0 ] = 3; +- break; +- +- case 5: +- sum[ 1 ] = count_bit( ix, begin, end, 6 ); +- if ( sum[1] <= sum[0] ) +- choice[ 0 ] = 6; +- break; +- +- case 7: +- sum[ 1 ] = count_bit( ix, begin, end, 8 ); +- if ( sum[1] <= sum[0] ) +- { +- choice[ 0 ] = 8; +- sum[ 0 ] = sum[ 1 ]; +- } +- sum[ 1 ] = count_bit( ix, begin, end, 9 ); +- if ( sum[1] <= sum[0] ) +- choice[ 0 ] = 9; +- break; +- +- case 10: +- sum[ 1 ] = count_bit( ix, begin, end, 11 ); +- if ( sum[1] <= sum[0] ) +- { +- choice[ 0 ] = 11; +- sum[ 0 ] = sum[ 1 ]; +- } +- sum[ 1 ] = count_bit( ix, begin, end, 12 ); +- if ( sum[1] <= sum[0] ) +- choice[ 0 ] = 12; +- break; +- +- case 13: +- sum[ 1 ] = count_bit( ix, begin, end, 15 ); +- if ( sum[1] <= sum[0] ) +- choice[ 0 ] = 15; +- break; +- +- default: +- break; +- } +- } +- else +- { +- /* try tables with linbits */ +- max -= 15; +- +- for ( i = 15; i < 24; i++ ) +- { +- if ( ht[i].linmax >= max ) +- { +- choice[ 0 ] = i; +- break; +- } +- } +- for ( i = 24; i < 32; i++ ) +- { +- if ( ht[i].linmax >= max ) +- { +- choice[ 1 ] = i; +- break; +- } +- } +- assert( choice[0] ); +- assert( choice[1] ); +- +- sum[ 0 ] = count_bit( ix, begin, end, choice[0] ); +- sum[ 1 ] = count_bit( ix, begin, end, choice[1] ); +- if ( sum[1] < sum[0] ) +- choice[ 0 ] = choice[ 1 ]; +- } +- return choice[ 0 ]; +- } +- +- +- +- /*************************************************************************/ +- /* choose table */ +- /*************************************************************************/ +- +- int choose_table( int max ) +- { +- int i, choice; +- +- if ( max == 0 ) +- return 0; +- +- max = abs( max ); +- choice = 0; +- +- if ( max < 15 ) +- { +- for ( i = 0; i < 15; i++ ) +- { +- if ( ht[i].xlen > max ) +- { +- choice = i; +- break; +- } +- } +- } +- else +- { +- max -= 15; +- for (i = 15; i < 32; i++ ) +- { +- if ( ht[i].linmax >= max ) +- { +- choice = i; +- break; +- } +- } +- } +- assert( choice ); +- return choice; +- } +- +- +- /*************************************************************************/ +- /* bigv_bitcount */ +- /*************************************************************************/ +- +- /* +- Function: Count the number of bits necessary to code the bigvalues region. +- */ +- +- int bigv_bitcount( int ix[576], gr_info *gi ) +- { +- int bits = 0; +- +- if ( gi->window_switching_flag && gi->block_type == 2 ) +- { +- /* +- Within each scalefactor band, data is given for successive +- time windows, beginning with window 0 and ending with window 2. +- Within each window, the quantized values are then arranged in +- order of increasing frequency... +- */ +- int sfb, window, line, start, end; +- I192_3 *ix_s; +- +- if ( gi->mixed_block_flag ) +- { +- unsigned int table; +- +- if ( (table = gi->table_select[0]) != 0 ) +- bits += count_bit( ix, 0, gi->address1, table ); +- sfb = 2; +- } +- else +- sfb = 0; +- +- ix_s = (I192_3 *) &ix[0]; +- +- for ( ; sfb < 13; sfb++ ) +- { +- unsigned tableindex = 100; +- +- start = scalefac_band_short[ sfb ]; +- end = scalefac_band_short[ sfb+1 ]; +- +- if ( start < 12 ) +- tableindex = gi->table_select[ 0 ]; +- else +- tableindex = gi->table_select[ 1 ]; +- assert( tableindex < 32 ); +- +- for ( window = 0; window < 3; window++ ) +- for ( line = start; line < end; line += 2 ) +- { +- unsigned int code, ext; +- int cbits, xbits; +- int x = (*ix_s)[line][window]; +- int y = (*ix_s)[line + 1][window]; +- bits += HuffmanCode( tableindex, x, y, &code, &ext, &cbits, &xbits ); +- } +- } +- } +- else +- { +- unsigned int table; +- +- if( (table=gi->table_select[0]) != 0 ) /* region0 */ +- bits += count_bit(ix, 0, gi->address1, table ); +- if( (table=gi->table_select[1]) != 0 ) /* region1 */ +- bits += count_bit(ix, gi->address1, gi->address2, table ); +- if( (table=gi->table_select[2]) != 0 ) /* region2 */ +- bits += count_bit(ix, gi->address2, gi->address3, table ); +- } +- return bits; +- } +- +- +- +- /*************************************************************************/ +- /* count_bit */ +- /*************************************************************************/ +- +- /* +- Function: Count the number of bits necessary to code the subregion. +- */ +- +- int count_bit( int ix[576], unsigned int start, unsigned int end, unsigned int table ) +- { +- int i, sum; +- +- sum = 0; +- for ( i = start; i < end; i += 2 ) +- { +- unsigned int code, ext; +- int cbits, xbits; +- sum += HuffmanCode( table, ix[i], ix[i+1], &code, &ext, &cbits, &xbits ); +- } +- return sum; +- } +- +- +- +- #ifndef HAVE_NINT +- int +- nint( double in ) +- { +- int temp; +- +- if( in < 0 ) temp = (int)(in - 0.5); +- else temp = (int)(in + 0.5); +- +- return(temp); +- } +- +- double +- aint(double in) { +- return((long) in); +- } +- #endif +- +- +- +- /* +- Seymour's comment: Jan 8 1995 +- When mixed_block_flag is set, the low subbands 0-1 undergo the long +- window transform and are each split into 18 frequency lines, while +- the remaining 30 subbands undergo the short window transform and are +- each split into 6 frequency lines. A problem now arises, as neither +- the short or long scale factor bands apply to this mixed spectrum. +- The standard resolves this situation by using the first 8 long scale +- factor bands for the low spectrum and the short scale factor bands +- in the range of 3 to 11 (inclusive) for the remaining frequency lines. +- These scale factor bands do not match exactly to the 0-1 subbands +- for all sampling frequencies (32,44.1 and 48 kHz); however they +- were designed so that there would not be a frequency gap or overlap +- at the switch over point. (Note multiply short frequency lines by 3 +- to account for wider frequency line.) +- */ +- +- +- +- /*************************************************************************/ +- /* gr_deco */ +- /*************************************************************************/ +- +- void gr_deco( gr_info *cod_info ) +- { +- if ( cod_info->window_switching_flag != 0 && cod_info->block_type == 2 ) +- if ( cod_info->mixed_block_flag == 0 ) +- { +- cod_info->sfb_lmax = 0; /* No sb*/ +- cod_info->sfb_smax = 0; +- } +- else +- { +- cod_info->sfb_lmax = 8; +- cod_info->sfb_smax = 3; +- } +- else +- { +- cod_info->sfb_lmax = SFB_LMAX - 1; +- cod_info->sfb_smax = SFB_SMAX - 1; /* No sb */ +- } +- } +- +- +- +- +- +- +- +- +- /* The following optional code written by Seymour Shlien +- will speed up the outer_loop code which is called +- by iteration_loop. When BIN_SEARCH is defined, the +- outer_loop function precedes the call to the function inner_loop +- with a call to bin_search gain defined below, which +- returns a good starting quantizerStepSize. +- */ +- +- #if defined(BIN_SEARCH) || defined(PERFORM) +- int count_bits(ix,cod_info) +- int *ix; /* I576 *ix; */ +- gr_info *cod_info; +- { +- int bits,max; +- calc_runlen(ix,cod_info); /*rzero,count1,big_values*/ +- max = ix_max( ix, 0,576); +- if(max > 8192) return 100000; /* report unsuitable quantizer */ +- bits = count1_bitcount(ix, cod_info); /*count1_table selection*/ +- subdivide(cod_info); /* bigvalues sfb division */ +- bigv_tab_select(ix,cod_info); /* codebook selection*/ +- bits += bigv_bitcount(ix,cod_info); /* bit count */ +- /* printf("\nglobal_gain = %f bits= %d ",cod_info->quantizerStepSize,bits);*/ +- return bits; +- } +- #endif +- +- +- #ifdef BIN_SEARCH +- +- int bin_search_StepSize(int desired_rate, double start, int *ix, +- double xrs[576], gr_info * cod_info) +- { +- double top,bot,next,last; +- int bit; +- top = start; +- bot = 200; +- next = start; +- do +- { +- last = next; +- next = aint((top+bot)/2.0); +- cod_info->quantizerStepSize = next; +- quantize(xrs,ix,cod_info); +- bit = count_bits(ix,cod_info); +- if (bit>desired_rate) top = next; +- else bot = next; +- /* printf("\n%f %f %f %d %d",next, top,bot,bit,desired_rate);*/ +- } +- while ((bit != desired_rate) && fabs(last - next) > 1.0); +- return next; +- } +- +- #endif +- +- +- +- +- #ifdef PERFORM +- +- /* The following code is used for exposing some problems with +- the outer_loop code. PERFORM should be defined to the +- frame number you wish to have additional output recorded +- in the file encode.log - Seymour Shlien 14-Jan-97 +- */ +- +- /* +- float worst_xfsf_to_xmin_ratio(l3_xmin,xfsf,block_type,gr,ch) +- double xfsf[4][CBLIMIT]; +- III_psy_xmin *l3_xmin; +- int block_type,gr,ch; +- */ +- float worst_xfsf_to_xmin_ratio(III_psy_xmin *l3_xmin, double xfsf[4][CBLIMIT] +- ,int block_type,int gr,int ch) +- +- { +- float ratio,maxratio; +- int i,j; +- maxratio =-100.0; +- if (block_type != 2) +- for(i=0;i<21;i++) +- { +- ratio = 10.*log10(xfsf[0][i] /l3_xmin->l[gr][ch][i]); +- if (ratio > maxratio) maxratio = ratio; +- } +- else +- { +- for(j=0;j<3;j++) +- /* for(i = cod_info->sfb_smax; i <SFB_SMAX; i++) */ +- for(i = 0; i <11; i++) +- { +- ratio = 10.*log10(xfsf[j+1][i] /l3_xmin->s[gr][ch][i][j]); +- if (ratio > maxratio) maxratio = ratio; +- } +- } +- return maxratio; +- } +- +- +- print_ratios(handle_out,l3_xmin,xfsf,block_type,gr,ch) +- FILE *handle_out; +- double xfsf[4][CBLIMIT]; +- III_psy_xmin *l3_xmin; +- int gr,ch; +- int block_type; +- { +- float ratio; +- int i,j; +- if(block_type !=2) +- for (i=0;i<21;i++) +- { +- ratio = 100.0; /* signals undefined value in output */ +- if(l3_xmin->l[gr][ch][i] >1.0e-20) +- ratio = 10.*log10(xfsf[0][i] /l3_xmin->l[gr][ch][i]); +- fprintf(handle_out,"%6.2f ",ratio); +- if(i%5==4) fprintf(handle_out,"\n"); +- } +- else +- +- for(j=0;j<3;j++) +- { fprintf(handle_out,"\n block %d\n",j); +- for(i = 0; i <11; i++) +- { +- ratio = 10.*log10(xfsf[j+1][i] /l3_xmin->s[gr][ch][i][j]); +- fprintf(handle_out,"%6.2f ",ratio); +- if(i%5==4) fprintf(handle_out,"\n"); +- } +- } +- fprintf(handle_out,"\n"); +- } +- +- +- print_scalefacs(handle_out,scalefac,block_type,gr,ch) +- FILE *handle_out; +- III_scalefac_t *scalefac; +- int gr,ch; +- int block_type; +- { +- int sfb,j; +- +- if(block_type !=2) +- for ( sfb = 0; sfb < 21; sfb++ ) +- { +- fprintf(handle_out,"%6d ", scalefac->l[gr][ch][sfb]); +- if(sfb%5==4) fprintf(handle_out,"\n"); +- } +- else +- for (j=0;j<3;j++) +- { +- fprintf(handle_out,"\n block %d\n",j); +- for (sfb=0;sfb<11;sfb++) +- { +- fprintf(handle_out,"%6d ",scalefac->s[gr][ch][sfb][j]); +- if(sfb%5==4) fprintf(handle_out,"\n"); +- } +- } +- fprintf(handle_out,"\n"); +- } +- +- +- print_quantized_values(FILE *handle, int ix[576], gr_info *cod_info) +- { +- int sfb,start,end,i,bw; +- for (sfb=0;sfb<cod_info->sfb_lmax;sfb++) +- { +- start = scalefac_band_long[sfb]; +- end = scalefac_band_long[sfb+1]; +- bw = end - start; +- fprintf(handle,"scalefac band %d from %d to %d\n",sfb,start,end); +- for (i=0;i<bw;i++) +- { +- fprintf(handle,"%8d",ix[start+i]); +- if(i%5==4) fprintf(handle,"\n"); +- } +- fprintf(handle,"\n"); +- } +- } +- +- +- test_inner_loop(double xr[2][2][576], int l3_enc[2][2][576], int max_bits, +- gr_info *cod_info, int gr, int ch, double xfsf[4][CBLIMIT], +- III_psy_xmin *l3_xmin) +- { +- int bits, c1bits, bvbits; +- double *xrs; /* D576 *xr; */ +- int *ix; /* I576 *ix; */ +- xrs = &xr[gr][ch][0]; +- ix = l3_enc[gr][ch]; +- +- assert( max_bits >= 0 ); +- cod_info->quantizerStepSize -= 1.0;; +- do +- { +- do +- { +- cod_info->quantizerStepSize += 1.0; +- quantize( xrs, ix, cod_info ); +- } +- while ( ix_max(ix, 0, 576) > 8191 + 14 ); /* within table range? */ +- +- bits = count_bits(ix,cod_info); +- +- if(frameNum == PERFORM) +- { +- fprintf(log_output,"StepSize=%f bits = %d huff_bits = %d\n", +- cod_info->quantizerStepSize,bits,max_bits); +- calc_noise( &xr[gr][ch][0], &l3_enc[gr][ch][0], cod_info, xfsf ); +- /* distortion calculation */ +- print_ratios(log_output,l3_xmin,xfsf,cod_info->block_type,gr,ch); +- fprintf(log_output,"\n\n"); +- } +- +- } +- while ( bits > max_bits ); +- +- return bits; +- } +- +- #endif PERFORM +--- 0 ---- +diff -r -c -N encoder/loop.h lame3.50/loop.h +*** encoder/loop.h Wed Jan 22 02:43:16 1997 +--- lame3.50/loop.h Wed Dec 31 17:00:00 1969 +*************** +*** 1,90 **** +- /********************************************************************** +- * ISO MPEG Audio Subgroup Software Simulation Group (1996) +- * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- * +- * $Id: loop.h,v 1.1 1996/02/14 04:04:23 rowlands Exp $ +- * +- * $Log: loop.h,v $ +- * Revision 1.1 1996/02/14 04:04:23 rowlands +- * Initial revision +- * +- * Received from Mike Coleman +- **********************************************************************/ +- +- #ifndef LOOP_DOT_H +- #define LOOP_DOT_H +- #include "common.h" +- #include "l3side.h" +- +- /********************************************************************** +- * date programmers comment * +- * 25. 6.92 Toshiyuki Ishino Ver 1.0 * +- * 29.10.92 Masahiro Iwadare Ver 2.0 * +- * 17. 4.93 Masahiro Iwadare Updated for IS Modification * +- * * +- *********************************************************************/ +- +- extern int cont_flag; +- +- #define e 2.71828182845 +- +- /*#define SBLIMIT 32*/ +- #define CBLIMIT 21 +- +- #define SFB_LMAX 22 +- #define SFB_SMAX 13 +- +- extern int pretab[]; +- +- struct scalefac_struct +- { +- int l[23]; +- int s[14]; +- }; +- +- extern struct scalefac_struct sfBandIndex[]; /* Table B.8 -- in loop.c */ +- +- +- void iteration_loop( double pe[][2], double xr_org[2][2][576], III_psy_ratio *ratio, +- III_side_info_t *l3_side, int l3_enc[2][2][576], int mean_bits, +- int stereo, double xr_dec[2][2][576], +- III_scalefac_t *scalefac, frame_params *fr_ps, +- int ancillary_pad, int bitsPerFrame ); +- +- +- int nint( double in ); +- +- /* #define PI 3.1415926535 */ +- #define maximum(A,B) ( (A) > (B) ? (A) : (B) ) +- #define minimum(A,B) ( (A) < (B) ? (A) : (B) ) +- #define signum( A ) ( (A) > 0 ? 1 : -1 ) +- +- /* GLOBALE VARIABLE */ +- +- /*extern FILE *debp; +- extern FILE *huffcp,*huffdp; +- extern FILE *musicin,*cod_music;*/ +- +- /* Beachte:Partitonen nur fuer 48 kHz */ +- /* andere ev. fehlerhaft */ +- +- /* static int scalefac_band_long[22]; +- static int scalefac_band_short[13]; +- */ +- /*static int huffman_tab_quad[2][16][2]; +- extern int bigv_cod_tab[17][2][16][16]; +- extern int bigv_dec_tab[17][512][5]; +- extern int bigv_root_node[17]; +- extern int count1_root_node[2];*/ +- /* extern int cod_tab_info[2][31]; */ +- +- +- +- +- /* in max spalte war um 1 zuviel */ +- /* [0][i] : maximalwert der Tabelle mit index i +- [1][i] : anzahl der linbits der TAb. i */ +- +- extern int bit_buffer[50000]; +- +- #endif +--- 0 ---- +diff -r -c -N encoder/loopold.c lame3.50/loopold.c +*** encoder/loopold.c Wed Dec 31 17:00:00 1969 +--- lame3.50/loopold.c Mon Oct 11 12:17:30 1999 +*************** +*** 0 **** +--- 1,662 ---- ++ /************************************************************************/ ++ /* init_outer_loop mt 6/99 */ ++ /************************************************************************/ ++ void init_outer_loop_dual( ++ double xr[2][2][576], /* could be L/R OR MID/SIDE */ ++ double xr_org[2][2][576], ++ III_psy_xmin *l3_xmin, /* the allowed distortion of the scalefactor */ ++ III_scalefac_t *scalefac, /* scalefactors */ ++ int gr, int stereo, III_side_info_t *l3_side, ++ III_psy_ratio *ratio, int ch) ++ { ++ int sfb,i; ++ gr_info *cod_info; ++ cod_info = &l3_side->gr[gr].ch[ch].tt; ++ ++ /* compute max allowed distortion */ ++ if (convert_psy) /* l/r thresholds, use unconverted data */ ++ calc_xmin( xr_org, ratio, cod_info, l3_xmin, gr, ch ); ++ else ++ calc_xmin( xr, ratio, cod_info, l3_xmin, gr, ch ); ++ ++ /* if ( info->version == 1 ) ++ calc_scfsi( xr[gr][ch], l3_side, &l3_xmin, ch, gr ); ++ */ ++ ++ ++ /* reset of iteration variables */ ++ ++ for ( sfb = 0; sfb < SFB_LMAX; sfb++ ) ++ scalefac->l[gr][ch][sfb] = 0; ++ for ( sfb = 0; sfb < SFB_SMAX; sfb++ ) ++ for ( i = 0; i < 3; i++ ) ++ scalefac->s[gr][ch][sfb][i] = 0; ++ ++ for ( i = 0; i < 4; i++ ) ++ cod_info->slen[i] = 0; ++ cod_info->sfb_partition_table = &nr_of_sfb_block[0][0][0]; ++ ++ cod_info->part2_3_length = 0; ++ cod_info->big_values = ((cod_info->block_type==2)?288:0); ++ cod_info->count1 = 0; ++ cod_info->scalefac_compress = 0; ++ cod_info->table_select[0] = 0; ++ cod_info->table_select[1] = 0; ++ cod_info->table_select[2] = 0; ++ cod_info->subblock_gain[0] = 0; ++ cod_info->subblock_gain[1] = 0; ++ cod_info->subblock_gain[2] = 0; ++ cod_info->region0_count = 0; ++ cod_info->region1_count = 0; ++ cod_info->part2_length = 0; ++ cod_info->preflag = 0; ++ cod_info->scalefac_scale = 0; ++ cod_info->quantizerStepSize = 0.0; ++ cod_info->count1table_select= 0; ++ cod_info->address1 = 0; ++ cod_info->address2 = 0; ++ cod_info->address3 = 0; ++ ++ ++ if (experimentalZ) { ++ /* compute subblock gains */ ++ int j,b; double en[3],mx; ++ if ((cod_info->block_type ==2) ) { ++ /* estimate energy within each subblock */ ++ for (b=0; b<3; b++) en[b]=0; ++ for ( i=0,j = 0; j < 192; j++ ) { ++ for (b=0; b<3; b++) { ++ en[b]+=xr[gr][ch][i]*xr[gr][ch][i]; ++ i++; ++ } ++ } ++ mx = 1e-12; ++ for (b=0; b<3; b++) mx=Max(mx,en[b]); ++ for (b=0; b<3; b++) en[b] = Max(en[b],1e-12)/mx; ++ /* pick gain so that 2^(2gain)*en[0] = 1 */ ++ /* gain = .5* log( 1/en[0] )/log(2) = -.5*log(en[])/log(2) */ ++ for (b=0; b<3; b++) { ++ cod_info->subblock_gain[b]=nint2(-.5*log(en[b])/log(2.0)); ++ if (cod_info->subblock_gain[b] > 2) ++ cod_info->subblock_gain[b]=2; ++ if (cod_info->subblock_gain[b] < 0) ++ cod_info->subblock_gain[b]=0; ++ } ++ } ++ } ++ } ++ ++ ++ ++ ++ void quant_compare_dual(int better[2], int notdone[2], int stereo, ++ double ms_ener_ratio, ++ int best_over[2],double best_tot_noise[2],double best_over_noise[2],double best_max_noise[2], ++ int over[2],double tot_noise[2], double over_noise[2], double max_noise[2]) ++ { ++ int ch; ++ /* ++ noise is given in decibals (db) relative to masking thesholds. ++ ++ over_noise: sum of quantization noise > masking ++ tot_noise: sum of all quantization noise ++ max_noise: max quantization noise ++ ++ */ ++ ++ ++ for (ch=0 ; ch < stereo ; ch ++ ) { ++ better[ch]=0; ++ if (notdone[ch]) { ++ if (convert_psy) { ++ better[ch] = (over[0]+over[1]) < (best_over[0]+best_over[1]); ++ if ((over[0]+over[1])==(best_over[0]+best_over[1])) { ++ better[ch] = (over_noise[0]+over_noise[1]) < (best_over_noise[0]+best_over_noise[1]); ++ } ++ }else{ ++ better[ch] = ((over[ch] < best_over[ch]) || ++ ((over[ch]==best_over[ch]) && (over_noise[ch]<best_over_noise[ch])) ) ; ++ } ++ } ++ } ++ } ++ ++ ++ ++ ++ /************************************************************************/ ++ /* outer_loop */ ++ /************************************************************************/ ++ /* Function: The outer iteration loop controls the masking conditions */ ++ /* of all scalefactorbands. It computes the best scalefac and */ ++ /* global gain. This module calls the inner iteration loop ++ * ++ * mt 5/99 completely rewritten to allow for bit reservoir control, ++ * mid/side channels with L/R or mid/side masking thresholds, ++ * and chooses best quantization instead of last quantization when ++ * no distortion free quantization can be found. ++ * ++ * added VBR support mt 5/99 ++ ************************************************************************/ ++ void outer_loop_dual( ++ double xr[2][2][576], /* could be L/R OR MID/SIDE */ ++ double xr_org[2][2][576], ++ int mean_bits, ++ int VBRbits[2][2], ++ int bit_rate, ++ int best_over[2], ++ III_psy_xmin *l3_xmin, /* the allowed distortion of the scalefactor */ ++ int l3_enc[2][2][576], /* vector of quantized values ix(0..575) */ ++ frame_params *fr_ps, ++ III_scalefac_t *scalefac, /* scalefactors */ ++ int gr, int stereo, III_side_info_t *l3_side, ++ III_psy_ratio *ratio, double pe[2][2], double ms_ener_ratio[2]) ++ { ++ int status[2],notdone[2]={0,0},count[2]={0,0},bits_found[2]; ++ int targ_bits[2],real_bits[2],tbits,extra_bits; ++ int scalesave_l[2][CBLIMIT], scalesave_s[2][CBLIMIT][3]; ++ int sfb, bits, huff_bits, save_preflag[2], save_compress[2]; ++ double xfsf[2][4][CBLIMIT]; ++ double xrpow[2][2][576],temp; ++ double distort[2][4][CBLIMIT]; ++ int save_l3_enc[2][576]; ++ int save_real_bits[2]; ++ int i,over[2], iteration, ch, compute_stepsize; ++ int better[2]; ++ double max_noise[2]; ++ double best_max_noise[2]; ++ double over_noise[2]; ++ double best_over_noise[2]; ++ double tot_noise[2]; ++ double best_tot_noise[2]; ++ gr_info save_cod_info[2]; ++ gr_info *cod_info[2]; ++ ++ cod_info[0] = &l3_side->gr[gr].ch[0].tt; ++ cod_info[1] = &l3_side->gr[gr].ch[1].tt; ++ for (ch=0 ; ch < stereo ; ch ++) ++ best_over[ch] = 100; ++ ++ ++ if (convert_mdct) ms_convert(xr[gr],xr_org[gr]); ++ else memcpy(xr[gr],xr_org[gr],sizeof(double)*2*576); ++ for (ch=0; ch<stereo; ch++) ++ init_outer_loop_dual(xr,xr_org,l3_xmin,scalefac,gr,stereo,l3_side,ratio,ch); ++ ++ for (ch=0 ; ch < stereo ; ch ++) { ++ count[ch]=0; ++ for (i=0; i<576; i++) { ++ if ( fabs(xr[gr][ch][i]) > 0 ) count[ch]++; ++ } ++ notdone[ch]=count[ch]; ++ if (count[ch]==0) best_over[ch]=0; ++ } ++ ++ /****************************************************************** ++ * allocate bits for each channel ++ ******************************************************************/ ++ if (VBR) { ++ for (ch=0 ; ch < stereo ; ch ++ ) ++ targ_bits[ch]=VBRbits[gr][ch]; ++ ++ }else { ++ int add_bits[2]; ++ ++ /* allocate targ_bits for granule */ ++ ResvMaxBits2( mean_bits, &tbits, &extra_bits, gr); ++ ++ for (ch=0 ; ch < stereo ; ch ++ ) ++ targ_bits[ch]=tbits/stereo; ++ ++ ++ // allocate extra bits from reservoir based on PE ++ bits=0; ++ for (ch=0; ch<stereo; ch++) { ++ double pe_temp=pe[gr][ch]; ++ if (convert_psy) pe_temp=Max(pe[gr][0],pe[gr][1]); ++ ++ /* extra bits based on PE > 700 */ ++ add_bits[ch]=(pe_temp-750)/1.4; /* 2.0; */ ++ ++ ++ /* short blocks need extra, no matter what the pe */ ++ if (cod_info[ch]->block_type==2) ++ if (add_bits[ch]<500) add_bits[ch]=500; ++ ++ if (add_bits[ch] < 0) add_bits[ch]=0; ++ bits += add_bits[ch]; ++ } ++ for (ch=0; ch<stereo; ch++) { ++ if (bits > extra_bits) add_bits[ch] = (extra_bits*add_bits[ch])/bits; ++ targ_bits[ch] = targ_bits[ch] + add_bits[ch]; ++ } ++ for (ch=0; ch<stereo; ch++) ++ extra_bits -= add_bits[ch]; ++ } ++ ++ if (reduce_sidechannel) { ++ /* ms_ener_ratio = 0: allocate 66/33 mid/side fac=.33 ++ * ms_ener_ratio =.5: allocate 50/50 mid/side fac= 0 */ ++ /* 75/25 split is fac=.5 */ ++ /* float fac = .50*(.5-ms_ener_ratio[gr])/.5;*/ ++ float fac = .33*(.5-ms_ener_ratio[gr])/.5; ++ if (fac<0) fac=0; ++ ++ /* dont reduce side channel below 125 bits */ ++ if (targ_bits[1]-targ_bits[1]*fac > 125) { ++ targ_bits[0] += targ_bits[1]*fac; ++ targ_bits[1] -= targ_bits[1]*fac; ++ } ++ ++ ++ /* dont allow to many bits per channel */ ++ for (ch=0; ch<stereo; ch++) { ++ int max_bits = Min(4095,mean_bits/2 + 1200); ++ if (targ_bits[ch] > max_bits) { ++ extra_bits += (targ_bits[ch] - max_bits); ++ targ_bits[ch] = max_bits; ++ } ++ } ++ } ++ ++ ++ /* BEGIN MAIN LOOP */ ++ iteration = 0; ++ compute_stepsize=1; ++ ++ while ( (notdone[0] || notdone[1]) ) { ++ int pre_just_turned_on[2]; ++ iteration ++; ++ ++ if (compute_stepsize) { ++ /* compute initial quantization step */ ++ compute_stepsize=0; ++ for (ch=0 ; ch < stereo ; ch ++ ) ++ if (notdone[ch]) { ++ for(i=0;i<576;i++) { ++ temp=fabs(xr[gr][ch][i]); ++ xrpow[gr][ch][i]=sqrt(sqrt(temp)*temp); ++ } ++ bits_found[ch]=bin_search_StepSize2(targ_bits[ch],-211.0,46, ++ l3_enc[gr][ch],xr[gr][ch],xrpow[gr][ch],cod_info[ch]); ++ } ++ } ++ ++ ++ ++ /* inner_loop starts with the initial quantization step computed above ++ * and slowly increases until the bits < huff_bits. ++ * Thus is it important not to start with too large of an inital ++ * quantization step. Too small is ok, but inner_loop will take longer ++ */ ++ for (ch=0 ; ch < stereo ; ch ++ ) { ++ if (notdone[ch]) { ++ // cod_info[ch]->part2_length = part2_length( scalefac, fr_ps->header->version, cod_info[ch]); ++ huff_bits = targ_bits[ch] - cod_info[ch]->part2_length; ++ if (huff_bits < 0) { ++ if (iteration==1) { ++ fprintf(stderr,"ERROR: outer_loop(): huff_bits < 0. \n"); ++ exit(-5); ++ }else{ ++ /* scale factors too large, not enough bits. use previous quantizaton */ ++ notdone[ch]=0; ++ over[ch]=999; ++ } ++ }else{ ++ /* if this is the first iteration, see if we can reuse the quantization ++ * computed in bin_search_StepSize above */ ++ if (iteration==1) { ++ if(bits_found[ch]>huff_bits) { ++ cod_info[ch]->quantizerStepSize+=1.0; ++ real_bits[ch] = inner_loop( xr, xrpow[gr][ch], l3_enc, huff_bits, cod_info[ch], gr, ch ); ++ } else real_bits[ch]=bits_found[ch]; ++ } ++ else ++ real_bits[ch]=inner_loop( xr, xrpow[gr][ch], l3_enc, huff_bits, cod_info[ch], gr, ch ); ++ } ++ } ++ } ++ ++ ++ ++ /* compute the distortion in this quantization */ ++ if (fast_mode) { ++ for (ch=0; ch<stereo; ch++) ++ over[ch]=0; ++ }else{ ++ if (convert_psy) { ++ /* mid/side coefficiets, l/r thresholds */ ++ calc_noise2( xr[gr], l3_enc[gr], cod_info, xfsf, ++ distort, l3_xmin,gr,stereo,over,over_noise,tot_noise,max_noise); ++ } else { ++ /* coefficients and thresholds both l/r (or both mid/side) */ ++ for (ch=0; ch<stereo; ch++) ++ if (notdone[ch]) ++ over[ch]=calc_noise1( xr[gr][ch], l3_enc[gr][ch], cod_info[ch], ++ xfsf[ch],distort[ch], l3_xmin,gr,ch, &over_noise[ch], ++ &tot_noise[ch], &max_noise[ch]); ++ } ++ } ++ ++ /* check if this quantization is better the our saved quantization */ ++ if (iteration == 1) ++ for (ch=0; ch<stereo; ch++) better[ch]=1; ++ else quant_compare_dual(better,notdone,stereo,ms_ener_ratio[gr], ++ best_over,best_tot_noise,best_over_noise,best_max_noise, ++ over,tot_noise,over_noise,max_noise); ++ ++ ++ ++ /* save data so we can restore this quantization later */ ++ for (ch=0 ; ch < stereo ; ch ++ ) { ++ if (better[ch]) { ++ best_over[ch]=over[ch]; ++ best_over_noise[ch]=over_noise[ch]; ++ best_tot_noise[ch]=tot_noise[ch]; ++ best_max_noise[ch]=max_noise[ch]; ++ if (notdone[ch]) { ++ for ( sfb = 0; sfb < CBLIMIT; sfb++ ) /* save scaling factors */ ++ scalesave_l[ch][sfb] = scalefac->l[gr][ch][sfb]; ++ ++ for ( sfb = 0; sfb < SFB_SMAX; sfb++ ) ++ for ( i = 0; i < 3; i++ ) ++ scalesave_s[ch][sfb][i] = scalefac->s[gr][ch][sfb][i]; ++ ++ save_preflag[ch] = cod_info[ch]->preflag; ++ save_compress[ch] = cod_info[ch]->scalefac_compress; ++ ++ memcpy(save_l3_enc[ch],l3_enc[gr][ch],sizeof(l3_enc[gr][ch])); ++ memcpy(&save_cod_info[ch],cod_info[ch],sizeof(save_cod_info[ch])); ++ save_real_bits[ch]=real_bits[ch]; ++ ++ #ifdef HAVEGTK ++ if (gtkflag) { ++ for ( i = 0; i < 3; i++ ) { ++ for ( sfb = cod_info[ch]->sfb_smax; sfb < 12; sfb++ ) { ++ pinfo->xfsf_s[gr][ch][3*sfb+i] = ++ pinfo->thr_s[gr][ch][3*sfb+i]*xfsf[ch][i+1][sfb]/ ++ (1e-20+l3_xmin->s[gr][ch][sfb][i]); ++ } ++ } ++ for ( sfb = 0; sfb < cod_info[ch]->sfb_lmax; sfb++ ) { ++ pinfo->xfsf[gr][ch][sfb] = ++ pinfo->thr[gr][ch][sfb]*xfsf[ch][0][sfb]/ ++ (1e-20 + l3_xmin->l[gr][ch][sfb]); ++ } ++ pinfo->over[gr][ch]=over[ch]; ++ pinfo->over_noise[gr][ch]=over_noise[ch]; ++ } ++ #endif ++ ++ ++ } ++ } ++ } ++ ++ /* if no bands with distortion, we are done */ ++ for (ch=0 ; ch < stereo ; ch ++ ) ++ if (notdone[ch]) { ++ if (convert_psy) ++ notdone[ch] = (over[0] || over[1]); ++ else ++ notdone[ch] = over[ch]; ++ } ++ ++ ++ ++ ++ /* see if we should apply preemphasis */ ++ for (ch=0 ; ch < stereo ; ch ++ ) { ++ pre_just_turned_on[ch]=0; ++ if (notdone[ch] ) pre_just_turned_on[ch]= ++ preemphasis( xr[gr][ch], xrpow[gr][ch], l3_xmin, ++ gr, ch, l3_side, distort[ch]); ++ } ++ ++ ++ /* if we didn't just apply pre-emph, let us see if we should ++ * amplify some scale factor bands */ ++ for (ch=0 ; ch < stereo ; ch ++ ) ++ if (notdone[ch] && (!pre_just_turned_on[ch]) ) { ++ amp_scalefac_bands( xr[gr][ch], xrpow[gr][ch], l3_xmin, ++ l3_side, scalefac, gr, ch, iteration,distort[ch]); ++ } ++ ++ ++ /* check to make sure we have not amplified too much */ ++ for (ch=0 ; ch < stereo ; ch ++ ) { ++ if (notdone[ch]) { ++ if ( (status[ch] = loop_break(scalefac, cod_info[ch], gr, ch)) == 0 ) { ++ if ( fr_ps->header->version == 1 ) { ++ status[ch] = scale_bitcount( scalefac, cod_info[ch], gr, ch ); ++ }else{ ++ status[ch] = scale_bitcount_lsf( scalefac, cod_info[ch], gr, ch ); ++ } ++ } ++ notdone[ch] = !status[ch]; ++ } ++ } ++ } /* done with main iteration */ ++ ++ ++ ++ /* restore some data */ ++ for (ch=0 ; ch < stereo ; ch ++ ) { ++ if (count[ch] ) { ++ cod_info[ch]->preflag = save_preflag[ch]; ++ cod_info[ch]->scalefac_compress = save_compress[ch]; ++ ++ for ( sfb = 0; sfb < CBLIMIT; sfb++ ) { ++ scalefac->l[gr][ch][sfb] = scalesave_l[ch][sfb]; ++ } ++ ++ for ( i = 0; i < 3; i++ ) ++ for ( sfb = 0; sfb < SFB_SMAX; sfb++ ) { ++ scalefac->s[gr][ch][sfb][i] = scalesave_s[ch][sfb][i]; ++ } ++ ++ { ++ real_bits[ch]=save_real_bits[ch]; ++ memcpy(l3_enc[gr][ch],save_l3_enc[ch],sizeof(l3_enc[gr][ch])); ++ memcpy(cod_info[ch],&save_cod_info[ch],sizeof(save_cod_info[ch])); ++ ++ if ( fr_ps->header->version == 1 ) ++ status[ch] = scale_bitcount( scalefac, cod_info[ch], gr, ch ); ++ else ++ status[ch] = scale_bitcount_lsf( scalefac, cod_info[ch], gr, ch ); ++ if (status[ch]) { ++ fprintf(stderr,"Error recomputing scalefac_compress...this should not happen"); ++ exit(-10); ++ } ++ } ++ // cod_info[ch]->part2_length = part2_length( scalefac, fr_ps->header->version, cod_info[ch]); ++ cod_info[ch]->part2_3_length = cod_info[ch]->part2_length + real_bits[ch]; ++ ++ #ifdef HAVEGTK ++ if (gtkflag) ++ pinfo->LAMEmainbits[gr][ch]=cod_info[ch]->part2_3_length; ++ #endif ++ } ++ } ++ ++ /* finish up */ ++ for (ch=0 ; ch < stereo ; ch ++ ) { ++ if (!VBR) ResvAdjust( fr_ps, cod_info[ch], l3_side, mean_bits ); ++ cod_info[ch]->global_gain = nint2( cod_info[ch]->quantizerStepSize + 210.0 ); ++ assert( cod_info[ch]->global_gain < 256 ); ++ } ++ } ++ ++ ++ /*************************************************************************/ ++ /* calc_noise2 */ ++ /*************************************************************************/ ++ /* Improved version of calc_noise for dual channel. This routine is */ ++ /* used when you are quantizaing mid and side channels using masking */ ++ /* thresholds from L and R channels. mt 5/99 */ ++ ++ void calc_noise2( double xr[2][576], int ix[2][576], gr_info *cod_info[2], ++ double xfsf[2][4][CBLIMIT], double distort[2][4][CBLIMIT], ++ III_psy_xmin *l3_xmin,int gr,int stereo, int over[2], ++ double over_noise[2], double tot_noise[2], double max_noise[2]) ++ { ++ int start, end, sfb, l, i; ++ double sum[2],step_s[3][2],step[2],bw; ++ ++ D192_3 *xr_s[2]; ++ I192_3 *ix_s[2]; ++ ++ static double pow43[PRECALC_SIZE]; ++ int ch; ++ static int init=0; ++ double diff[2]; ++ double noise; ++ int index; ++ ++ if (init==0) { ++ init++; ++ for(i=0;i<PRECALC_SIZE;i++) ++ pow43[i] = pow((double)i, 4.0/3.0); ++ } ++ ++ ++ ++ /* calc_noise2: we can assume block types of both channels must be the same ++ */ ++ if (cod_info[0]->block_type != 2) { ++ for (ch=0 ; ch < stereo ; ch ++ ) { ++ over[ch]=0; ++ over_noise[ch]=0; ++ tot_noise[ch]=0; ++ max_noise[ch]=-999; ++ step[ch] = pow( 2.0, (cod_info[ch]->quantizerStepSize) * 0.25 ); ++ } ++ for ( sfb = 0; sfb < SFB_LMAX-1; sfb++ ) { ++ start = scalefac_band_long[ sfb ]; ++ end = scalefac_band_long[ sfb+1 ]; ++ bw = end - start; ++ ++ // for (ch=0 ; ch < stereo ; ch ++ ) sum[ch]=0; ++ for ( sum[0]=0, sum[1]=0, l = start; l < end; l++ ) { ++ index=ix[0][l]; ++ if (index==0) { ++ diff[0]=xr[0][l]; ++ } else { ++ if (xr[0][l]<0) { ++ diff[0]=xr[0][l] + pow43[index] * step[0]; ++ } else ++ diff[0]=xr[0][l] - pow43[index] * step[0]; ++ } ++ index=ix[1][l]; ++ if (index==0) { ++ diff[1]=xr[1][l]; ++ } else { ++ if (xr[1][l]<0) { ++ diff[1]=xr[1][l] + pow43[index] * step[1]; ++ } else ++ diff[1]=xr[1][l] - pow43[index] * step[1]; ++ } ++ sum[0] += (diff[0]+diff[1])*(diff[0]+diff[1]); ++ sum[1] += (diff[0]-diff[1])*(diff[0]-diff[1]); ++ } ++ sum[0] *= 0.5; ++ sum[1] *= 0.5; ++ ++ for (ch=0 ; ch < stereo ; ch ++ ) { ++ xfsf[ch][0][sfb] = sum[ch] / bw; ++ noise = 10*log10(Max(.001,xfsf[ch][0][sfb]/l3_xmin->l[gr][ch][sfb])); ++ distort[ch][0][sfb] = noise; ++ if (noise>0) { ++ over[ch]++; ++ //over_noise[ch] += xfsf[ch][0][sfb] - l3_xmin->l[gr][ch][sfb]; ++ over_noise[ch] += noise; ++ } ++ tot_noise[ch] += noise; ++ max_noise[ch] = Max(max_noise[ch],noise); ++ } ++ ++ /* if there is audible distortion in left or right channel, set flags ++ * to denote distortion in both mid and side channels */ ++ for (ch=0 ; ch < stereo ; ch ++ ) { ++ distort[ch][0][sfb] = Max(distort[0][0][sfb],distort[1][0][sfb]); ++ } ++ } ++ } ++ ++ /* calc_noise2: we can assume block types of both channels must be the same ++ */ ++ if (cod_info[0]->block_type == 2) { ++ for (ch=0 ; ch < stereo ; ch ++ ) { ++ ++ for (i=0;i<3;i++){ ++ step_s[i][ch] = pow( 2.0, (cod_info[ch]->quantizerStepSize) * 0.25 ); ++ /* subblock_gain ? */ ++ if (cod_info[ch]->subblock_gain[i] ) ++ step_s[i][ch] *= pow(2.0,-2.0*cod_info[ch]->subblock_gain[i]); ++ } ++ ++ over[ch] = 0; ++ xr_s[ch] = (D192_3 *) xr[ch]; ++ ix_s[ch] = (I192_3 *) ix[ch]; ++ } ++ ++ for ( sfb = 0 ; sfb < SFB_SMAX-1; sfb++ ) { ++ start = scalefac_band_short[ sfb ]; ++ end = scalefac_band_short[ sfb+1 ]; ++ bw = end - start; ++ for ( i = 0; i < 3; i++ ) { ++ for (ch=0 ; ch < stereo ; ch ++ ) sum[ch] = 0.0; ++ for ( l = start; l < end; l++ ) { ++ index=(*ix_s[0])[l][i]; ++ if (index==0) ++ diff[0] = (*xr_s[0])[l][i]; ++ else { ++ if ((*xr_s[0])[l][i] < 0) { ++ diff[0] = (*xr_s[0])[l][i] + pow43[index] * step_s[i][0]; ++ } else { ++ diff[0] = (*xr_s[0])[l][i] - pow43[index] * step_s[i][0]; ++ } ++ } ++ index=(*ix_s[1])[l][i]; ++ if (index==0) ++ diff[1] = (*xr_s[1])[l][i]; ++ else { ++ if ((*xr_s[1])[l][i] < 0) { ++ diff[1] = (*xr_s[1])[l][i] + pow43[index] * step_s[i][1]; ++ } else { ++ diff[1] = (*xr_s[1])[l][i] - pow43[index] * step_s[i][1]; ++ } ++ } ++ sum[0] += (diff[0]+diff[1])*(diff[0]+diff[1])/(2.0); ++ sum[1] += (diff[0]-diff[1])*(diff[0]-diff[1])/(2.0); ++ } ++ ++ for (ch=0 ; ch < stereo ; ch ++ ) { ++ xfsf[ch][i+1][sfb] = sum[ch] / bw; ++ noise = ++ 10*log10(Max(.001,xfsf[ch][i+1][sfb]/l3_xmin->s[gr][ch][sfb][i])); ++ distort[ch][i+1][sfb] = noise>0; ++ if (noise>0) { ++ over[ch]++; ++ //over_noise[ch] += xfsf[ch][i+1][sfb]-l3_xmin->s[gr][ch][sfb][i]; ++ over_noise[ch] += noise; ++ } ++ tot_noise[ch] += noise; ++ max_noise[ch]=Max(max_noise[ch],noise); ++ } ++ /* if there is audible distortion in left or right channel, set flags ++ * to denote distortion in both mid and side channels */ ++ for (ch=0 ; ch < stereo ; ch ++ ) ++ distort[ch][i+1][sfb] = Max ++ (distort[0][i+1][sfb],distort[1][i+1][sfb] ); ++ } ++ } ++ } ++ } ++ ++ ++ ++ ++ ++ ++ +diff -r -c -N encoder/machine.h lame3.50/machine.h +*** encoder/machine.h Wed Dec 31 17:00:00 1969 +--- lame3.50/machine.h Tue Oct 26 12:15:05 1999 +*************** +*** 0 **** +--- 1,66 ---- ++ /* ++ * Machine dependent defines/includes for LAME. ++ * ++ * Copyright (c) 1999 A.L. Faber ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License as published by ++ * the Free Software Foundation; either version 2, or (at your option) ++ * any later version. ++ * ++ * This program is distributed in the hope that it will be useful, ++ * but WITHOUT ANY WARRANTY; without even the implied warranty of ++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the ++ * GNU General Public License for more details. ++ * ++ * You should have received a copy of the GNU General Public License ++ * along with this program; see the file COPYING. If not, write to ++ * the Free Software Foundation, 675 Mass Ave, Cambridge, MA 02139, USA. ++ */ ++ ++ ++ #ifndef MACHINE_H_INCLUDED ++ #define MACHINE_H_INCLUDED ++ ++ ++ #include <stdio.h> ++ #include <string.h> ++ #include <math.h> ++ #include <stdlib.h> ++ #include <ctype.h> ++ #include <signal.h> ++ #include <sys/types.h> ++ #include <sys/stat.h> ++ #include <fcntl.h> ++ #include <errno.h> ++ ++ /* ++ #ifndef OS_AMIGAOS ++ #include <memory.h> ++ #endif ++ #include <float.h> ++ */ ++ ++ #if ( defined(_MSC_VER) || defined(__BORLANDC__) ) ++ #define WIN32_LEAN_AND_MEAN ++ #include <windows.h> ++ #else ++ typedef float FLOAT; ++ #endif ++ ++ ++ ++ #ifdef _WIN32 ++ # define M_PI 3.14159265358979323846 ++ # define M_SQRT2 1.41421356237309504880 ++ typedef unsigned long u_long; ++ typedef unsigned int u_int; ++ typedef unsigned short u_short; ++ typedef unsigned char u_char; ++ ++ #include <fcntl.h> ++ #include <io.h> ++ #endif ++ ++ ++ #endif +diff -r -c -N encoder/main.c lame3.50/main.c +*** encoder/main.c Wed Dec 31 17:00:00 1969 +--- lame3.50/main.c Tue Sep 28 21:57:13 1999 +*************** +*** 0 **** +--- 1,56 ---- ++ #include "lame.h" ++ ++ #ifdef HAVEGTK ++ #include "gtkanal.h" ++ #include <gtk/gtk.h> ++ #endif ++ ++ char mp3buffer[LAME_MAXMP3BUFFER]; ++ ++ ++ /********************************************************************** ++ * read one frame and encode it ++ **********************************************************************/ ++ int makeframe(void) ++ { ++ int iread; ++ static short int Buffer[2][1152]; ++ iread=lame_readframe(Buffer); ++ /* even if iread=0, get_audio hit EOF and returned Buffer=all 0's. call ++ * makeframe() one last time to flush all buffers */ ++ lame_encode(Buffer,mp3buffer); ++ return iread; ++ } ++ ++ ++ ++ ++ /************************************************************************ ++ * ++ * main ++ * ++ * PURPOSE: MPEG-1,2 Layer III encoder with GPSYCHO ++ * psychoacoustic model. ++ * ++ ************************************************************************/ ++ ++ ++ int main(int argc, char **argv) ++ { ++ lame_init(0); ++ if(argc==1) lame_usage(argv[0]); /* no command-line args */ ++ lame_parse_args(argc, argv); ++ lame_print_config(); ++ ++ #ifdef HAVEGTK ++ if (gtkflag) gtk_init (&argc, &argv); ++ if (gtkflag) gtkcontrol(); ++ else ++ #endif ++ while (makeframe()); ++ ++ ++ lame_cleanup(mp3buffer); ++ return 0; ++ } ++ +diff -r -c -N encoder/makefile.unix lame3.50/makefile.unix +*** encoder/makefile.unix Wed Jan 22 02:43:23 1997 +--- lame3.50/makefile.unix Wed Dec 31 17:00:00 1969 +*************** +*** 1,77 **** +- ############################################################################ +- ## ISO MPEG Audio Subgroup Software Simulation Group (1996) +- ## ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- ## +- ## $Id: makefile.unix,v 1.1 1996/02/14 05:28:25 rowlands Exp $ +- ## +- ## Makefile for encoder for vanilla UNIX platform using standard make. +- ## +- ## $Log: makefile.unix,v $ +- ## Revision 1.1 1996/02/14 05:28:25 rowlands +- ## Initial revision +- ## +- ############################################################################ +- +- # MODE= -O -DUNIX +- MODE= -g -DUNIX +- #MODE= -DMSC60 -AH -Zi -Gt +- OBJ_SUFFIX=.o +- EXE_SUFFIX= +- +- PROGRAM = encode$(EXE_SUFFIX) +- +- all: $(PROGRAM) +- +- $(PROGRAM) : \ +- common.c common.h encode.c encoder.h formatBitstream.c \ +- formatBitstream.h huffman.c huffman.h ieeefloat.c ieeefloat.h \ +- l3bitstream-pvt.h l3bitstream.c l3bitstream.h l3psy.c l3psy.h \ +- l3side.h loop-pvt.h loop.c loop.h mdct.c mdct.h musicin.c \ +- portableio.c portableio.h psy.c reservoir.c reservoir.h subs.c tonal.c +- +- HEDS = \ +- common.h encoder.h formatBitstream.h huffman.h ieeefloat.h \ +- l3bitstream-pvt.h l3bitstream.h l3psy.h l3side.h loop-pvt.h loop.h \ +- mdct.h portableio.h reservoir.h +- +- OBJS = \ +- common$(OBJ_SUFFIX) \ +- encode$(OBJ_SUFFIX) \ +- formatBitstream$(OBJ_SUFFIX) \ +- huffman$(OBJ_SUFFIX) \ +- ieeefloat$(OBJ_SUFFIX) \ +- l3bitstream$(OBJ_SUFFIX) \ +- l3psy$(OBJ_SUFFIX) \ +- loop$(OBJ_SUFFIX) \ +- mdct$(OBJ_SUFFIX) \ +- musicin$(OBJ_SUFFIX) \ +- portableio$(OBJ_SUFFIX) \ +- psy$(OBJ_SUFFIX) \ +- reservoir$(OBJ_SUFFIX) \ +- subs$(OBJ_SUFFIX) \ +- tonal$(OBJ_SUFFIX) +- +- LFLAG = +- +- CC = gcc +- LN = gcc +- LNEXTRA=-lm +- #CC = cl +- #LN = cl +- +- .c$(OBJ_SUFFIX): $(HEDS) +- $(CC) -c $(MODE) $*.c +- +- all: $(PROGRAM) +- +- default: all +- +- +- $(PROGRAM) : $(OBJS) $(HEDS) +- $(LN) -o $(PROGRAM) $(MODE) $(LFLAG) $(OBJS) $(LNEXTRA) +- +- clean: +- rm -f $(OBJS) +- +- veryclean: +- rm -f $(PROGRAM) $(OBJS) +--- 0 ---- +diff -r -c -N encoder/mdct.c lame3.50/mdct.c +*** encoder/mdct.c Wed Jan 22 02:43:17 1997 +--- lame3.50/mdct.c Wed Dec 31 17:00:00 1969 +*************** +*** 1,212 **** +- /********************************************************************** +- * ISO MPEG Audio Subgroup Software Simulation Group (1996) +- * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- * +- * $Id: mdct.c,v 1.1 1996/02/14 04:04:23 rowlands Exp $ +- * +- * $Log: mdct.c,v $ +- * Revision 1.1 1996/02/14 04:04:23 rowlands +- * Initial revision +- * +- * Received from Mike Coleman +- **********************************************************************/ +- +- #include "common.h" +- #include "l3side.h" +- #include "mdct.h" +- +- double ca[8], cs[8]; +- +- /* +- This is table B.9: coefficients for aliasing reduction +- */ +- static double c[8] = { -0.6,-0.535,-0.33,-0.185,-0.095,-0.041,-0.0142, -0.0037 }; +- +- void mdct_sub( L3SBS (*sb_sample), double (*mdct_freq)[2][576], int stereo, III_side_info_t *l3_side, int mode_gr ) +- { +- gr_info *cod_info; +- double mdct_in[36]; +- int ch,gr,band,k,j; +- double bu,bd; +- static int init = 0; +- int block_type; +- double (*mdct_enc)[2][32][18] = (double (*)[2][32][18]) mdct_freq; +- +- if ( init == 0 ) +- { +- /* prepare the aliasing reduction butterflies */ +- for ( k = 0; k < 8; k++ ) +- { +- double sq; +- sq = sqrt( 1.0 + c[k] * c[k] ); +- ca[k] = c[k] / sq; +- cs[k] = 1.0 / sq; +- } +- init++; +- } +- +- for ( gr = 0; gr < mode_gr; gr++ ) +- for ( ch = 0; ch < stereo; ch++ ) +- { +- cod_info = (gr_info *) &(l3_side->gr[gr].ch[ch]) ; +- block_type = cod_info->block_type; +- +- /* +- Compensate for inversion in the analysis filter +- */ +- for ( band = 0; band < 32; band++ ) +- for ( k = 0; k < 18; k++ ) +- if ( (band & 1) && (k & 1) ) +- (*sb_sample)[ch][gr+1][k][band] *= -1.0; +- +- /* +- Perform imdct of 18 previous subband samples +- + 18 current subband samples +- */ +- for ( band = 0; band < 32; band++ ) +- { +- for ( k = 0; k < 18; k++ ) +- { +- mdct_in[k] = (*sb_sample)[ch][ gr ][k][band]; +- mdct_in[k+18] = (*sb_sample)[ch][gr+1][k][band]; +- } +- if ( cod_info->mixed_block_flag && (band < 2) ) +- block_type = 0; +- +- mdct( mdct_in, &mdct_enc[gr][ch][band][0], block_type ); +- } +- +- /* +- Perform aliasing reduction butterfly +- on long blocks +- */ +- if ( block_type != 2 ) +- for ( band = 0; band < 31; band++ ) +- for ( k = 0; k < 8; k++ ) +- { +- bu = mdct_enc[gr][ch][band][17-k] * cs[k] + mdct_enc[gr][ch][band+1][k] * ca[k]; +- bd = mdct_enc[gr][ch][band+1][k] * cs[k] - mdct_enc[gr][ch][band][17-k] * ca[k]; +- mdct_enc[gr][ch][band][17-k] = bu; +- mdct_enc[gr][ch][band+1][k] = bd; +- } +- +- } +- +- /* +- Save latest granule's subband samples to be used in +- the next mdct call +- */ +- for ( ch = 0; ch < stereo; ch++ ) +- for ( j = 0; j < 18; j++ ) +- for ( band = 0; band < 32; band++ ) +- (*sb_sample)[ch][0][j][band] = (*sb_sample)[ch][mode_gr][j][band]; +- } +- +- void mdct( double *in, double *out, int block_type ) +- { +- /*-------------------------------------------------------------------*/ +- /* */ +- /* Function: Calculation of the MDCT */ +- /* In the case of long blocks ( block_type 0,1,3 ) there are */ +- /* 36 coefficents in the time domain and 18 in the frequency */ +- /* domain. */ +- /* In the case of short blocks (block_type 2 ) there are 3 */ +- /* transformations with short length. This leads to 12 coefficents */ +- /* in the time and 6 in the frequency domain. In this case the */ +- /* results are stored side by side in the vector out[]. */ +- /* */ +- /* New layer3 */ +- /* */ +- /*-------------------------------------------------------------------*/ +- +- int l,k,i,m,N; +- double sum; +- static double win[4][36]; +- static int init = 0; +- static double cos_s[6][12], cos_l[18][36]; +- +- +- if ( init == 0 ) +- { +- /* type 0 */ +- for ( i = 0; i < 36; i++ ) +- win[0][i] = sin( PI/36 * (i + 0.5) ); +- /* type 1*/ +- for ( i = 0; i < 18; i++ ) +- win[1][i] = sin( PI/36 * (i + 0.5) ); +- for ( i = 18; i < 24; i++ ) +- win[1][i] = 1.0; +- for ( i = 24; i < 30; i++ ) +- win[1][i] = sin( PI/12 * ( i + 0.5 - 18) ); +- for ( i = 30; i < 36; i++ ) +- win[1][i] = 0.0; +- /* type 3*/ +- for ( i = 0; i < 6; i++ ) +- win[3][i] = 0.0; +- for ( i = 6; i < 12; i++ ) +- win[3][i] = sin( PI/12 * (i + 0.5 - 6) ); +- for ( i = 12; i < 18; i++ ) +- win[3][i] = 1.0; +- for ( i = 18; i < 36; i++ ) +- win[3][i] = sin( PI/36 * (i + 0.5) ); +- /* type 2*/ +- for ( i = 0; i < 12; i++ ) +- win[2][i] = sin( PI/12 * (i + 0.5) ); +- for ( i = 12; i < 36; i++ ) +- win[2][i] = 0.0; +- +- N = 12; +- for ( m = 0; m < N / 2; m++ ) +- for ( k = 0; k < N; k++ ) +- cos_s[m][k] = cos( (PI /(2 * N)) * (2 * k + 1 + N / 2) * +- (2 * m + 1) ) / (N / 4); +- +- N = 36; +- for ( m = 0; m < N / 2; m++ ) +- for ( k = 0; k < N; k++ ) +- cos_l[m][k] = cos( (PI / (2 * N)) * (2 * k + 1 + N / 2) * +- (2 * m + 1) ) / (N / 4); +- +- init++; +- } +- +- if ( block_type == 2 ) +- { +- N = 12; +- for ( l = 0; l < 3; l++ ) +- { +- for ( m = 0; m < N / 2; m++ ) +- { +- for ( sum = 0.0, k = 0; k < N; k++ ) +- sum += win[block_type][k] * in[k + 6 * l + 6] * cos_s[m][k]; +- out[ 3 * m + l] = sum; +- } +- } +- } +- else +- { +- N = 36; +- for ( m = 0; m < N / 2; m++ ) +- { +- for ( sum = 0.0, k = 0; k < N; k++ ) +- sum += win[block_type][k] * in[k] * cos_l[m][k]; +- out[m] = sum; +- } +- } +- } +- +- void +- delay( double (*xr)[2][576], int stereo ) +- { +- static double xr_buff[2][576]; +- double xr_buff2[2][576]; +- unsigned int i,j; +- +- for (i=0;i<stereo;i++) +- { +- for (j=0;j<576;j++) xr_buff2[i][j] = xr_buff[i][j]; +- for (j=0;j<576;j++) xr_buff[i][j] = xr[1][i][j]; +- for (j=0;j<576;j++) xr[1][i][j] = xr[0][i][j]; +- for (j=0;j<576;j++) xr[0][i][j] = xr_buff2[i][j]; +- } +- } +--- 0 ---- +diff -r -c -N encoder/mdct.h lame3.50/mdct.h +*** encoder/mdct.h Wed Jan 22 02:43:17 1997 +--- lame3.50/mdct.h Wed Dec 31 17:00:00 1969 +*************** +*** 1,25 **** +- /********************************************************************** +- * ISO MPEG Audio Subgroup Software Simulation Group (1996) +- * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- * +- * $Id: mdct.h,v 1.1 1996/02/14 04:04:23 rowlands Exp $ +- * +- * $Log: mdct.h,v $ +- * Revision 1.1 1996/02/14 04:04:23 rowlands +- * Initial revision +- * +- * Received from Mike Coleman +- **********************************************************************/ +- +- #ifndef MDCT_DOT_H +- #define MDCT_DOT_H +- void mdct(double *in, double *out, int block_type); +- void inv_mdct(double *in, double *out, int block_type); +- +- typedef double D32_18[SBLIMIT][18]; +- typedef double L3SBS[2][3][18][SBLIMIT]; /* [gr][ch] */ +- +- void mdct_sub(L3SBS (*sb_sample), double (*mdct_freq)[2][576], int stereo, III_side_info_t *l3_side, int mode_gr ); +- void mdct_sub_dec(double (*mdct_freq)[2][576], double inv_mdct_dec[3][2][18][32], int stereo, III_side_info_t *l3_side); +- void delay(double (*xr)[2][576], int stereo); +- #endif +--- 0 ---- +diff -r -c -N encoder/mlame lame3.50/mlame +*** encoder/mlame Wed Dec 31 17:00:00 1969 +--- lame3.50/mlame Fri Sep 10 09:52:07 1999 +*************** +*** 0 **** +--- 1,81 ---- ++ #!/bin/bash ++ #!/usr/local/bin/bash ++ ############################################################################ ++ # ++ # Run the LAME encoder on multible files, with option to delete .wav files ++ # after encoding. "mlame -h" will give instructions. ++ # ++ # Robert Hegemann <Robert.Hegemann@gmx.de> ++ # ++ ############################################################################ ++ ++ mp3coder="lame" ++ options="-h -m f -b 128" ++ rmsrc=false ++ ++ helptext="\ ++ \nThis script runs the LAME mp3 encoder on multiple files: \n\n\ ++ $0 [options] <file 1> ... <file n>\n\ ++ \n\ ++ options:\n\ ++ -h this help text\n\ ++ -r remove files after encoding\n\ ++ -o \"<lame options>\" overrides script default options \"${options}\"\n\ ++ \n\ ++ example:\n\ ++ $0 -r -o \"-v -V 0 -b 112\" a*.wav z*.aif\n\ ++ \n\ ++ " ++ ++ # process command-line options ++ # this could be extended to fake the ++ # commandline interface of the mp3encoder ++ ++ while getopts ":o:r" optn; do ++ case $optn in ++ o ) options=$OPTARG # replace default options ++ ;; ++ r ) rmsrc=true ++ ;; ++ \? ) printf "$helptext" ++ exit 1 ++ ;; ++ esac ++ done ++ shift $(($OPTIND - 1)) ++ ++ # process input-files ++ ++ for filename in "$@"; do ++ case $filename in ++ *[*?]* ) # means shell couldn�t extend *.wav, etc. ++ echo "warning: no $filename file(s) found" ++ ;; ++ *[.][wW][aA][vV] ) ++ name=${filename%[.][wW][aA][vV]} ++ if $mp3coder $options $filename ${name}.mp3 ++ then ++ if [ $rmsrc = true ]; then ++ rm -f $filename ++ fi ++ fi ++ ;; ++ *[.][aA][iI][fF] ) ++ name=${filename%[.][aA][iI][fF]} ++ if $mp3coder $options $filename ${name}.mp3 ++ then ++ if [ $rmsrc = true ]; then ++ rm -f $filename ++ fi ++ fi ++ ;; ++ * ) ++ if $mp3coder $options $filename ${filename}.mp3 ++ then ++ if [ $rmsrc = true ]; then ++ rm -f $filename ++ fi ++ fi ++ ;; ++ esac ++ done +diff -r -c -N encoder/mp3x.c lame3.50/mp3x.c +*** encoder/mp3x.c Wed Dec 31 17:00:00 1969 +--- lame3.50/mp3x.c Tue Sep 28 21:58:18 1999 +*************** +*** 0 **** +--- 1,33 ---- ++ #include "lame.h" ++ ++ #include "gtkanal.h" ++ #include <gtk/gtk.h> ++ ++ ++ ++ ++ /************************************************************************ ++ * ++ * main ++ * ++ * PURPOSE: MPEG-1,2 Layer III encoder with GPSYCHO ++ * psychoacoustic model. ++ * ++ ************************************************************************/ ++ int main(int argc, char **argv) ++ { ++ char mp3buffer[LAME_MAXMP3BUFFER]; ++ lame_init(1); ++ ++ if(argc==1) lame_usage(argv[0]); /* no command-line args */ ++ else lame_parse_args(argc, argv); ++ lame_print_config(); ++ ++ gtkflag=1; ++ gtk_init (&argc, &argv); ++ gtkcontrol(); ++ ++ lame_cleanup(mp3buffer); ++ return 0; ++ } ++ +diff -r -c -N encoder/mpglib/Makefile lame3.50/mpglib/Makefile +*** encoder/mpglib/Makefile Wed Dec 31 17:00:00 1969 +--- lame3.50/mpglib/Makefile Fri Sep 10 09:51:56 1999 +*************** +*** 0 **** +--- 1,22 ---- ++ ++ CC=gcc ++ CFLAGS=-Wall -g ++ ++ all: mpglib ++ ++ ++ *.o: mpg123.h mpglib.h ++ ++ mpglib: common.o dct64_i386.o decode_i386.o layer3.o tabinit.o interface.o main.o ++ $(CC) -o mpglib common.o dct64_i386.o decode_i386.o layer3.o \ ++ tabinit.o interface.o main.o -lm ++ ++ lib: common.o dct64_i386.o decode_i386.o layer3.o tabinit.o interface.o main.o ++ ar -svru libmpg123.a main.o common.o dct64_i386.o decode_i386.o layer3.o \ ++ tabinit.o interface.o ++ ++ ++ clean: ++ rm *.o mpglib ++ ++ +diff -r -c -N encoder/mpglib/README lame3.50/mpglib/README +*** encoder/mpglib/README Wed Dec 31 17:00:00 1969 +--- lame3.50/mpglib/README Fri Sep 10 09:51:56 1999 +*************** +*** 0 **** +--- 1,33 ---- ++ MP3 library ++ ----------- ++ Version 0.2 ++ ++ This decoder is a 'light' version (thrown out all unnecessay parts) ++ from the mpg123 package. I made this for a company. ++ ++ Currently only Layer3 is enabled to save some space. Layer1,2 isn't ++ tested at all. The interface will not change significantly. ++ A backport to the mpg123 package is planed. ++ ++ comiled and tested only on Solaris 2.6 ++ main.c contains a simple demo application for library. ++ ++ COPYING: you may use this source under GPL terms! ++ ++ PLEASE NOTE: This software may contain patented alogrithm (at least ++ patented in some countries). It may be not allowed to sell/use products ++ based on this source code in these countries. Check this out first! ++ ++ COPYRIGHT of MP3 music: ++ Please note, that the duplicating of copyrighted music without explicit ++ permission violates the rights of the owner. ++ ++ SENDING PATCHES: ++ Maybe I change the copyright policy (ie some kind of more free BSD licencse). ++ Please consider this when sending patches/changes. ++ ++ FEEDBACK: ++ I'm interessted to here from you, when you use this package as part ++ of another project. ++ ++ +diff -r -c -N encoder/mpglib/TODO lame3.50/mpglib/TODO +*** encoder/mpglib/TODO Wed Dec 31 17:00:00 1969 +--- lame3.50/mpglib/TODO Fri Sep 10 09:51:56 1999 +*************** +*** 0 **** +--- 1,2 ---- ++ ++ apply 'VBR' bug +diff -r -c -N encoder/mpglib/common.c lame3.50/mpglib/common.c +*** encoder/mpglib/common.c Wed Dec 31 17:00:00 1969 +--- lame3.50/mpglib/common.c Thu Sep 16 00:12:43 1999 +*************** +*** 0 **** +--- 1,262 ---- ++ #include <ctype.h> ++ #include <stdlib.h> ++ #include <signal.h> ++ ++ #include <sys/types.h> ++ #include <sys/stat.h> ++ #include <fcntl.h> ++ ++ #include "mpg123.h" ++ ++ struct parameter param = { 1 , 1 , 0 , 0 }; ++ ++ int tabsel_123[2][3][16] = { ++ { {0,32,64,96,128,160,192,224,256,288,320,352,384,416,448,}, ++ {0,32,48,56, 64, 80, 96,112,128,160,192,224,256,320,384,}, ++ {0,32,40,48, 56, 64, 80, 96,112,128,160,192,224,256,320,} }, ++ ++ { {0,32,48,56,64,80,96,112,128,144,160,176,192,224,256,}, ++ {0,8,16,24,32,40,48,56,64,80,96,112,128,144,160,}, ++ {0,8,16,24,32,40,48,56,64,80,96,112,128,144,160,} } ++ }; ++ ++ long freqs[9] = { 44100, 48000, 32000, ++ 22050, 24000, 16000 , ++ 11025 , 12000 , 8000 }; ++ ++ int bitindex; ++ unsigned char *wordpointer; ++ unsigned char *pcm_sample; ++ int pcm_point = 0; ++ ++ ++ #if 0 ++ static void get_II_stuff(struct frame *fr) ++ { ++ static int translate[3][2][16] = ++ { { { 0,2,2,2,2,2,2,0,0,0,1,1,1,1,1,0 } , ++ { 0,2,2,0,0,0,1,1,1,1,1,1,1,1,1,0 } } , ++ { { 0,2,2,2,2,2,2,0,0,0,0,0,0,0,0,0 } , ++ { 0,2,2,0,0,0,0,0,0,0,0,0,0,0,0,0 } } , ++ { { 0,3,3,3,3,3,3,0,0,0,1,1,1,1,1,0 } , ++ { 0,3,3,0,0,0,1,1,1,1,1,1,1,1,1,0 } } }; ++ ++ int table,sblim; ++ static struct al_table *tables[5] = ++ { alloc_0, alloc_1, alloc_2, alloc_3 , alloc_4 }; ++ static int sblims[5] = { 27 , 30 , 8, 12 , 30 }; ++ ++ if(fr->lsf) ++ table = 4; ++ else ++ table = translate[fr->sampling_frequency][2-fr->stereo][fr->bitrate_index]; ++ sblim = sblims[table]; ++ ++ fr->alloc = tables[table]; ++ fr->II_sblimit = sblim; ++ } ++ #endif ++ ++ #define HDRCMPMASK 0xfffffd00 ++ ++ #if 0 ++ int head_check(unsigned long head) ++ { ++ if( (head & 0xffe00000) != 0xffe00000) ++ return FALSE; ++ if(!((head>>17)&3)) ++ return FALSE; ++ if( ((head>>12)&0xf) == 0xf) ++ return FALSE; ++ if( ((head>>10)&0x3) == 0x3 ) ++ return FALSE; ++ return TRUE; ++ } ++ #endif ++ ++ /* ++ * the code a header and write the information ++ * into the frame structure ++ */ ++ int decode_header(struct frame *fr,unsigned long newhead) ++ { ++ ++ ++ if( newhead & (1<<20) ) { ++ fr->lsf = (newhead & (1<<19)) ? 0x0 : 0x1; ++ fr->mpeg25 = 0; ++ } ++ else { ++ fr->lsf = 1; ++ fr->mpeg25 = 1; ++ } ++ ++ fr->lay = 4-((newhead>>17)&3); ++ if( ((newhead>>10)&0x3) == 0x3) { ++ fprintf(stderr,"Stream error\n"); ++ exit(1); ++ } ++ if(fr->mpeg25) { ++ fr->sampling_frequency = 6 + ((newhead>>10)&0x3); ++ } ++ else ++ fr->sampling_frequency = ((newhead>>10)&0x3) + (fr->lsf*3); ++ fr->error_protection = ((newhead>>16)&0x1)^0x1; ++ ++ if(fr->mpeg25) /* allow Bitrate change for 2.5 ... */ ++ fr->bitrate_index = ((newhead>>12)&0xf); ++ ++ fr->bitrate_index = ((newhead>>12)&0xf); ++ fr->padding = ((newhead>>9)&0x1); ++ fr->extension = ((newhead>>8)&0x1); ++ fr->mode = ((newhead>>6)&0x3); ++ fr->mode_ext = ((newhead>>4)&0x3); ++ fr->copyright = ((newhead>>3)&0x1); ++ fr->original = ((newhead>>2)&0x1); ++ fr->emphasis = newhead & 0x3; ++ ++ fr->stereo = (fr->mode == MPG_MD_MONO) ? 1 : 2; ++ ++ if(!fr->bitrate_index) ++ { ++ fprintf(stderr,"Free format not supported.\n"); ++ return (0); ++ } ++ ++ switch(fr->lay) ++ { ++ case 1: ++ #if 0 ++ fr->do_layer = do_layer1; ++ fr->jsbound = (fr->mode == MPG_MD_JOINT_STEREO) ? ++ (fr->mode_ext<<2)+4 : 32; ++ fr->framesize = (long) tabsel_123[fr->lsf][0][fr->bitrate_index] * 12000; ++ fr->framesize /= freqs[fr->sampling_frequency]; ++ fr->framesize = ((fr->framesize+fr->padding)<<2)-4; ++ #else ++ fprintf(stderr,"Not supported!\n"); ++ #endif ++ break; ++ case 2: ++ #if 0 ++ fr->do_layer = do_layer2; ++ get_II_stuff(fr); ++ fr->jsbound = (fr->mode == MPG_MD_JOINT_STEREO) ? ++ (fr->mode_ext<<2)+4 : fr->II_sblimit; ++ fr->framesize = (long) tabsel_123[fr->lsf][1][fr->bitrate_index] * 144000; ++ fr->framesize /= freqs[fr->sampling_frequency]; ++ fr->framesize += fr->padding - 4; ++ #else ++ fprintf(stderr,"Not supported!\n"); ++ #endif ++ break; ++ case 3: ++ #if 0 ++ fr->do_layer = do_layer3; ++ if(fr->lsf) ++ ssize = (fr->stereo == 1) ? 9 : 17; ++ else ++ ssize = (fr->stereo == 1) ? 17 : 32; ++ #endif ++ ++ #if 0 ++ if(fr->error_protection) ++ ssize += 2; ++ #endif ++ fr->framesize = (long) tabsel_123[fr->lsf][2][fr->bitrate_index] * 144000; ++ fr->framesize /= freqs[fr->sampling_frequency]<<(fr->lsf); ++ fr->framesize = fr->framesize + fr->padding - 4; ++ break; ++ default: ++ fprintf(stderr,"Sorry, unknown layer type.\n"); ++ return (0); ++ } ++ ++ /* print_header(fr); */ ++ ++ return 1; ++ } ++ ++ ++ #if 1 ++ void print_header(struct frame *fr) ++ { ++ static char *modes[4] = { "Stereo", "Joint-Stereo", "Dual-Channel", "Single-Channel" }; ++ static char *layers[4] = { "Unknown" , "I", "II", "III" }; ++ ++ fprintf(stderr,"MPEG %s, Layer: %s, Freq: %ld, mode: %s, modext: %d, BPF : %d\n", ++ fr->mpeg25 ? "2.5" : (fr->lsf ? "2.0" : "1.0"), ++ layers[fr->lay],freqs[fr->sampling_frequency], ++ modes[fr->mode],fr->mode_ext,fr->framesize+4); ++ fprintf(stderr,"Channels: %d, copyright: %s, original: %s, CRC: %s, emphasis: %d.\n", ++ fr->stereo,fr->copyright?"Yes":"No", ++ fr->original?"Yes":"No",fr->error_protection?"Yes":"No", ++ fr->emphasis); ++ fprintf(stderr,"Bitrate: %d Kbits/s, Extension value: %d\n", ++ tabsel_123[fr->lsf][fr->lay-1][fr->bitrate_index],fr->extension); ++ } ++ ++ void print_header_compact(struct frame *fr) ++ { ++ static char *modes[4] = { "stereo", "joint-stereo", "dual-channel", "mono" }; ++ static char *layers[4] = { "Unknown" , "I", "II", "III" }; ++ ++ fprintf(stderr,"MPEG %s layer %s, %d kbit/s, %ld Hz %s\n", ++ fr->mpeg25 ? "2.5" : (fr->lsf ? "2.0" : "1.0"), ++ layers[fr->lay], ++ tabsel_123[fr->lsf][fr->lay-1][fr->bitrate_index], ++ freqs[fr->sampling_frequency], modes[fr->mode]); ++ } ++ ++ #endif ++ ++ unsigned int getbits(int number_of_bits) ++ { ++ unsigned long rval; ++ ++ if(!number_of_bits) ++ return 0; ++ ++ { ++ rval = wordpointer[0]; ++ rval <<= 8; ++ rval |= wordpointer[1]; ++ rval <<= 8; ++ rval |= wordpointer[2]; ++ rval <<= bitindex; ++ rval &= 0xffffff; ++ ++ bitindex += number_of_bits; ++ ++ rval >>= (24-number_of_bits); ++ ++ wordpointer += (bitindex>>3); ++ bitindex &= 7; ++ } ++ return rval; ++ } ++ ++ unsigned int getbits_fast(int number_of_bits) ++ { ++ unsigned long rval; ++ ++ { ++ rval = wordpointer[0]; ++ rval <<= 8; ++ rval |= wordpointer[1]; ++ rval <<= bitindex; ++ rval &= 0xffff; ++ bitindex += number_of_bits; ++ ++ rval >>= (16-number_of_bits); ++ ++ wordpointer += (bitindex>>3); ++ bitindex &= 7; ++ } ++ return rval; ++ } ++ ++ ++ ++ ++ +diff -r -c -N encoder/mpglib/dct64_i386.c lame3.50/mpglib/dct64_i386.c +*** encoder/mpglib/dct64_i386.c Wed Dec 31 17:00:00 1969 +--- lame3.50/mpglib/dct64_i386.c Thu Sep 16 00:12:03 1999 +*************** +*** 0 **** +--- 1,315 ---- ++ /* ++ * Discrete Cosine Tansform (DCT) for subband synthesis ++ * optimized for machines with no auto-increment. ++ * The performance is highly compiler dependend. Maybe ++ * the dct64.c version for 'normal' processor may be faster ++ * even for Intel processors. ++ */ ++ ++ #include "mpg123.h" ++ ++ static void dct64_1(real *out0,real *out1,real *b1,real *b2,real *samples) ++ { ++ ++ { ++ register real *costab = pnts[0]; ++ ++ b1[0x00] = samples[0x00] + samples[0x1F]; ++ b1[0x1F] = (samples[0x00] - samples[0x1F]) * costab[0x0]; ++ ++ b1[0x01] = samples[0x01] + samples[0x1E]; ++ b1[0x1E] = (samples[0x01] - samples[0x1E]) * costab[0x1]; ++ ++ b1[0x02] = samples[0x02] + samples[0x1D]; ++ b1[0x1D] = (samples[0x02] - samples[0x1D]) * costab[0x2]; ++ ++ b1[0x03] = samples[0x03] + samples[0x1C]; ++ b1[0x1C] = (samples[0x03] - samples[0x1C]) * costab[0x3]; ++ ++ b1[0x04] = samples[0x04] + samples[0x1B]; ++ b1[0x1B] = (samples[0x04] - samples[0x1B]) * costab[0x4]; ++ ++ b1[0x05] = samples[0x05] + samples[0x1A]; ++ b1[0x1A] = (samples[0x05] - samples[0x1A]) * costab[0x5]; ++ ++ b1[0x06] = samples[0x06] + samples[0x19]; ++ b1[0x19] = (samples[0x06] - samples[0x19]) * costab[0x6]; ++ ++ b1[0x07] = samples[0x07] + samples[0x18]; ++ b1[0x18] = (samples[0x07] - samples[0x18]) * costab[0x7]; ++ ++ b1[0x08] = samples[0x08] + samples[0x17]; ++ b1[0x17] = (samples[0x08] - samples[0x17]) * costab[0x8]; ++ ++ b1[0x09] = samples[0x09] + samples[0x16]; ++ b1[0x16] = (samples[0x09] - samples[0x16]) * costab[0x9]; ++ ++ b1[0x0A] = samples[0x0A] + samples[0x15]; ++ b1[0x15] = (samples[0x0A] - samples[0x15]) * costab[0xA]; ++ ++ b1[0x0B] = samples[0x0B] + samples[0x14]; ++ b1[0x14] = (samples[0x0B] - samples[0x14]) * costab[0xB]; ++ ++ b1[0x0C] = samples[0x0C] + samples[0x13]; ++ b1[0x13] = (samples[0x0C] - samples[0x13]) * costab[0xC]; ++ ++ b1[0x0D] = samples[0x0D] + samples[0x12]; ++ b1[0x12] = (samples[0x0D] - samples[0x12]) * costab[0xD]; ++ ++ b1[0x0E] = samples[0x0E] + samples[0x11]; ++ b1[0x11] = (samples[0x0E] - samples[0x11]) * costab[0xE]; ++ ++ b1[0x0F] = samples[0x0F] + samples[0x10]; ++ b1[0x10] = (samples[0x0F] - samples[0x10]) * costab[0xF]; ++ } ++ ++ ++ { ++ register real *costab = pnts[1]; ++ ++ b2[0x00] = b1[0x00] + b1[0x0F]; ++ b2[0x0F] = (b1[0x00] - b1[0x0F]) * costab[0]; ++ b2[0x01] = b1[0x01] + b1[0x0E]; ++ b2[0x0E] = (b1[0x01] - b1[0x0E]) * costab[1]; ++ b2[0x02] = b1[0x02] + b1[0x0D]; ++ b2[0x0D] = (b1[0x02] - b1[0x0D]) * costab[2]; ++ b2[0x03] = b1[0x03] + b1[0x0C]; ++ b2[0x0C] = (b1[0x03] - b1[0x0C]) * costab[3]; ++ b2[0x04] = b1[0x04] + b1[0x0B]; ++ b2[0x0B] = (b1[0x04] - b1[0x0B]) * costab[4]; ++ b2[0x05] = b1[0x05] + b1[0x0A]; ++ b2[0x0A] = (b1[0x05] - b1[0x0A]) * costab[5]; ++ b2[0x06] = b1[0x06] + b1[0x09]; ++ b2[0x09] = (b1[0x06] - b1[0x09]) * costab[6]; ++ b2[0x07] = b1[0x07] + b1[0x08]; ++ b2[0x08] = (b1[0x07] - b1[0x08]) * costab[7]; ++ ++ b2[0x10] = b1[0x10] + b1[0x1F]; ++ b2[0x1F] = (b1[0x1F] - b1[0x10]) * costab[0]; ++ b2[0x11] = b1[0x11] + b1[0x1E]; ++ b2[0x1E] = (b1[0x1E] - b1[0x11]) * costab[1]; ++ b2[0x12] = b1[0x12] + b1[0x1D]; ++ b2[0x1D] = (b1[0x1D] - b1[0x12]) * costab[2]; ++ b2[0x13] = b1[0x13] + b1[0x1C]; ++ b2[0x1C] = (b1[0x1C] - b1[0x13]) * costab[3]; ++ b2[0x14] = b1[0x14] + b1[0x1B]; ++ b2[0x1B] = (b1[0x1B] - b1[0x14]) * costab[4]; ++ b2[0x15] = b1[0x15] + b1[0x1A]; ++ b2[0x1A] = (b1[0x1A] - b1[0x15]) * costab[5]; ++ b2[0x16] = b1[0x16] + b1[0x19]; ++ b2[0x19] = (b1[0x19] - b1[0x16]) * costab[6]; ++ b2[0x17] = b1[0x17] + b1[0x18]; ++ b2[0x18] = (b1[0x18] - b1[0x17]) * costab[7]; ++ } ++ ++ { ++ register real *costab = pnts[2]; ++ ++ b1[0x00] = b2[0x00] + b2[0x07]; ++ b1[0x07] = (b2[0x00] - b2[0x07]) * costab[0]; ++ b1[0x01] = b2[0x01] + b2[0x06]; ++ b1[0x06] = (b2[0x01] - b2[0x06]) * costab[1]; ++ b1[0x02] = b2[0x02] + b2[0x05]; ++ b1[0x05] = (b2[0x02] - b2[0x05]) * costab[2]; ++ b1[0x03] = b2[0x03] + b2[0x04]; ++ b1[0x04] = (b2[0x03] - b2[0x04]) * costab[3]; ++ ++ b1[0x08] = b2[0x08] + b2[0x0F]; ++ b1[0x0F] = (b2[0x0F] - b2[0x08]) * costab[0]; ++ b1[0x09] = b2[0x09] + b2[0x0E]; ++ b1[0x0E] = (b2[0x0E] - b2[0x09]) * costab[1]; ++ b1[0x0A] = b2[0x0A] + b2[0x0D]; ++ b1[0x0D] = (b2[0x0D] - b2[0x0A]) * costab[2]; ++ b1[0x0B] = b2[0x0B] + b2[0x0C]; ++ b1[0x0C] = (b2[0x0C] - b2[0x0B]) * costab[3]; ++ ++ b1[0x10] = b2[0x10] + b2[0x17]; ++ b1[0x17] = (b2[0x10] - b2[0x17]) * costab[0]; ++ b1[0x11] = b2[0x11] + b2[0x16]; ++ b1[0x16] = (b2[0x11] - b2[0x16]) * costab[1]; ++ b1[0x12] = b2[0x12] + b2[0x15]; ++ b1[0x15] = (b2[0x12] - b2[0x15]) * costab[2]; ++ b1[0x13] = b2[0x13] + b2[0x14]; ++ b1[0x14] = (b2[0x13] - b2[0x14]) * costab[3]; ++ ++ b1[0x18] = b2[0x18] + b2[0x1F]; ++ b1[0x1F] = (b2[0x1F] - b2[0x18]) * costab[0]; ++ b1[0x19] = b2[0x19] + b2[0x1E]; ++ b1[0x1E] = (b2[0x1E] - b2[0x19]) * costab[1]; ++ b1[0x1A] = b2[0x1A] + b2[0x1D]; ++ b1[0x1D] = (b2[0x1D] - b2[0x1A]) * costab[2]; ++ b1[0x1B] = b2[0x1B] + b2[0x1C]; ++ b1[0x1C] = (b2[0x1C] - b2[0x1B]) * costab[3]; ++ } ++ ++ { ++ register real const cos0 = pnts[3][0]; ++ register real const cos1 = pnts[3][1]; ++ ++ b2[0x00] = b1[0x00] + b1[0x03]; ++ b2[0x03] = (b1[0x00] - b1[0x03]) * cos0; ++ b2[0x01] = b1[0x01] + b1[0x02]; ++ b2[0x02] = (b1[0x01] - b1[0x02]) * cos1; ++ ++ b2[0x04] = b1[0x04] + b1[0x07]; ++ b2[0x07] = (b1[0x07] - b1[0x04]) * cos0; ++ b2[0x05] = b1[0x05] + b1[0x06]; ++ b2[0x06] = (b1[0x06] - b1[0x05]) * cos1; ++ ++ b2[0x08] = b1[0x08] + b1[0x0B]; ++ b2[0x0B] = (b1[0x08] - b1[0x0B]) * cos0; ++ b2[0x09] = b1[0x09] + b1[0x0A]; ++ b2[0x0A] = (b1[0x09] - b1[0x0A]) * cos1; ++ ++ b2[0x0C] = b1[0x0C] + b1[0x0F]; ++ b2[0x0F] = (b1[0x0F] - b1[0x0C]) * cos0; ++ b2[0x0D] = b1[0x0D] + b1[0x0E]; ++ b2[0x0E] = (b1[0x0E] - b1[0x0D]) * cos1; ++ ++ b2[0x10] = b1[0x10] + b1[0x13]; ++ b2[0x13] = (b1[0x10] - b1[0x13]) * cos0; ++ b2[0x11] = b1[0x11] + b1[0x12]; ++ b2[0x12] = (b1[0x11] - b1[0x12]) * cos1; ++ ++ b2[0x14] = b1[0x14] + b1[0x17]; ++ b2[0x17] = (b1[0x17] - b1[0x14]) * cos0; ++ b2[0x15] = b1[0x15] + b1[0x16]; ++ b2[0x16] = (b1[0x16] - b1[0x15]) * cos1; ++ ++ b2[0x18] = b1[0x18] + b1[0x1B]; ++ b2[0x1B] = (b1[0x18] - b1[0x1B]) * cos0; ++ b2[0x19] = b1[0x19] + b1[0x1A]; ++ b2[0x1A] = (b1[0x19] - b1[0x1A]) * cos1; ++ ++ b2[0x1C] = b1[0x1C] + b1[0x1F]; ++ b2[0x1F] = (b1[0x1F] - b1[0x1C]) * cos0; ++ b2[0x1D] = b1[0x1D] + b1[0x1E]; ++ b2[0x1E] = (b1[0x1E] - b1[0x1D]) * cos1; ++ } ++ ++ { ++ register real const cos0 = pnts[4][0]; ++ ++ b1[0x00] = b2[0x00] + b2[0x01]; ++ b1[0x01] = (b2[0x00] - b2[0x01]) * cos0; ++ b1[0x02] = b2[0x02] + b2[0x03]; ++ b1[0x03] = (b2[0x03] - b2[0x02]) * cos0; ++ b1[0x02] += b1[0x03]; ++ ++ b1[0x04] = b2[0x04] + b2[0x05]; ++ b1[0x05] = (b2[0x04] - b2[0x05]) * cos0; ++ b1[0x06] = b2[0x06] + b2[0x07]; ++ b1[0x07] = (b2[0x07] - b2[0x06]) * cos0; ++ b1[0x06] += b1[0x07]; ++ b1[0x04] += b1[0x06]; ++ b1[0x06] += b1[0x05]; ++ b1[0x05] += b1[0x07]; ++ ++ b1[0x08] = b2[0x08] + b2[0x09]; ++ b1[0x09] = (b2[0x08] - b2[0x09]) * cos0; ++ b1[0x0A] = b2[0x0A] + b2[0x0B]; ++ b1[0x0B] = (b2[0x0B] - b2[0x0A]) * cos0; ++ b1[0x0A] += b1[0x0B]; ++ ++ b1[0x0C] = b2[0x0C] + b2[0x0D]; ++ b1[0x0D] = (b2[0x0C] - b2[0x0D]) * cos0; ++ b1[0x0E] = b2[0x0E] + b2[0x0F]; ++ b1[0x0F] = (b2[0x0F] - b2[0x0E]) * cos0; ++ b1[0x0E] += b1[0x0F]; ++ b1[0x0C] += b1[0x0E]; ++ b1[0x0E] += b1[0x0D]; ++ b1[0x0D] += b1[0x0F]; ++ ++ b1[0x10] = b2[0x10] + b2[0x11]; ++ b1[0x11] = (b2[0x10] - b2[0x11]) * cos0; ++ b1[0x12] = b2[0x12] + b2[0x13]; ++ b1[0x13] = (b2[0x13] - b2[0x12]) * cos0; ++ b1[0x12] += b1[0x13]; ++ ++ b1[0x14] = b2[0x14] + b2[0x15]; ++ b1[0x15] = (b2[0x14] - b2[0x15]) * cos0; ++ b1[0x16] = b2[0x16] + b2[0x17]; ++ b1[0x17] = (b2[0x17] - b2[0x16]) * cos0; ++ b1[0x16] += b1[0x17]; ++ b1[0x14] += b1[0x16]; ++ b1[0x16] += b1[0x15]; ++ b1[0x15] += b1[0x17]; ++ ++ b1[0x18] = b2[0x18] + b2[0x19]; ++ b1[0x19] = (b2[0x18] - b2[0x19]) * cos0; ++ b1[0x1A] = b2[0x1A] + b2[0x1B]; ++ b1[0x1B] = (b2[0x1B] - b2[0x1A]) * cos0; ++ b1[0x1A] += b1[0x1B]; ++ ++ b1[0x1C] = b2[0x1C] + b2[0x1D]; ++ b1[0x1D] = (b2[0x1C] - b2[0x1D]) * cos0; ++ b1[0x1E] = b2[0x1E] + b2[0x1F]; ++ b1[0x1F] = (b2[0x1F] - b2[0x1E]) * cos0; ++ b1[0x1E] += b1[0x1F]; ++ b1[0x1C] += b1[0x1E]; ++ b1[0x1E] += b1[0x1D]; ++ b1[0x1D] += b1[0x1F]; ++ } ++ ++ out0[0x10*16] = b1[0x00]; ++ out0[0x10*12] = b1[0x04]; ++ out0[0x10* 8] = b1[0x02]; ++ out0[0x10* 4] = b1[0x06]; ++ out0[0x10* 0] = b1[0x01]; ++ out1[0x10* 0] = b1[0x01]; ++ out1[0x10* 4] = b1[0x05]; ++ out1[0x10* 8] = b1[0x03]; ++ out1[0x10*12] = b1[0x07]; ++ ++ b1[0x08] += b1[0x0C]; ++ out0[0x10*14] = b1[0x08]; ++ b1[0x0C] += b1[0x0a]; ++ out0[0x10*10] = b1[0x0C]; ++ b1[0x0A] += b1[0x0E]; ++ out0[0x10* 6] = b1[0x0A]; ++ b1[0x0E] += b1[0x09]; ++ out0[0x10* 2] = b1[0x0E]; ++ b1[0x09] += b1[0x0D]; ++ out1[0x10* 2] = b1[0x09]; ++ b1[0x0D] += b1[0x0B]; ++ out1[0x10* 6] = b1[0x0D]; ++ b1[0x0B] += b1[0x0F]; ++ out1[0x10*10] = b1[0x0B]; ++ out1[0x10*14] = b1[0x0F]; ++ ++ b1[0x18] += b1[0x1C]; ++ out0[0x10*15] = b1[0x10] + b1[0x18]; ++ out0[0x10*13] = b1[0x18] + b1[0x14]; ++ b1[0x1C] += b1[0x1a]; ++ out0[0x10*11] = b1[0x14] + b1[0x1C]; ++ out0[0x10* 9] = b1[0x1C] + b1[0x12]; ++ b1[0x1A] += b1[0x1E]; ++ out0[0x10* 7] = b1[0x12] + b1[0x1A]; ++ out0[0x10* 5] = b1[0x1A] + b1[0x16]; ++ b1[0x1E] += b1[0x19]; ++ out0[0x10* 3] = b1[0x16] + b1[0x1E]; ++ out0[0x10* 1] = b1[0x1E] + b1[0x11]; ++ b1[0x19] += b1[0x1D]; ++ out1[0x10* 1] = b1[0x11] + b1[0x19]; ++ out1[0x10* 3] = b1[0x19] + b1[0x15]; ++ b1[0x1D] += b1[0x1B]; ++ out1[0x10* 5] = b1[0x15] + b1[0x1D]; ++ out1[0x10* 7] = b1[0x1D] + b1[0x13]; ++ b1[0x1B] += b1[0x1F]; ++ out1[0x10* 9] = b1[0x13] + b1[0x1B]; ++ out1[0x10*11] = b1[0x1B] + b1[0x17]; ++ out1[0x10*13] = b1[0x17] + b1[0x1F]; ++ out1[0x10*15] = b1[0x1F]; ++ } ++ ++ /* ++ * the call via dct64 is a trick to force GCC to use ++ * (new) registers for the b1,b2 pointer to the bufs[xx] field ++ */ ++ void dct64(real *a,real *b,real *c) ++ { ++ real bufs[0x40]; ++ dct64_1(a,b,bufs,bufs+0x20,c); ++ } ++ ++ +diff -r -c -N encoder/mpglib/decode_i386.c lame3.50/mpglib/decode_i386.c +*** encoder/mpglib/decode_i386.c Wed Dec 31 17:00:00 1969 +--- lame3.50/mpglib/decode_i386.c Thu Sep 16 00:12:15 1999 +*************** +*** 0 **** +--- 1,154 ---- ++ /* ++ * Mpeg Layer-1,2,3 audio decoder ++ * ------------------------------ ++ * copyright (c) 1995,1996,1997 by Michael Hipp, All rights reserved. ++ * See also 'README' ++ * ++ * slighlty optimized for machines without autoincrement/decrement. ++ * The performance is highly compiler dependend. Maybe ++ * the decode.c version for 'normal' processor may be faster ++ * even for Intel processors. ++ */ ++ ++ #include <stdlib.h> ++ #include <math.h> ++ #include <string.h> ++ ++ #include "mpg123.h" ++ #include "mpglib.h" ++ ++ extern struct mpstr *gmp; ++ ++ /* old WRITE_SAMPLE */ ++ #define WRITE_SAMPLE(samples,sum,clip) \ ++ if( (sum) > 32767.0) { *(samples) = 0x7fff; (clip)++; } \ ++ else if( (sum) < -32768.0) { *(samples) = -0x8000; (clip)++; } \ ++ else { *(samples) = sum; } ++ ++ int synth_1to1_mono(real *bandPtr,unsigned char *samples,int *pnt) ++ { ++ short samples_tmp[64]; ++ short *tmp1 = samples_tmp; ++ int i,ret; ++ int pnt1 = 0; ++ ++ ret = synth_1to1(bandPtr,0,(unsigned char *) samples_tmp,&pnt1); ++ samples += *pnt; ++ ++ for(i=0;i<32;i++) { ++ *( (short *) samples) = *tmp1; ++ samples += 2; ++ tmp1 += 2; ++ } ++ *pnt += 64; ++ ++ return ret; ++ } ++ ++ ++ int synth_1to1(real *bandPtr,int channel,unsigned char *out,int *pnt) ++ { ++ static const int step = 2; ++ int bo; ++ short *samples = (short *) (out + *pnt); ++ ++ real *b0,(*buf)[0x110]; ++ int clip = 0; ++ int bo1; ++ ++ bo = gmp->synth_bo; ++ ++ if(!channel) { ++ bo--; ++ bo &= 0xf; ++ buf = gmp->synth_buffs[0]; ++ } ++ else { ++ samples++; ++ buf = gmp->synth_buffs[1]; ++ } ++ ++ if(bo & 0x1) { ++ b0 = buf[0]; ++ bo1 = bo; ++ dct64(buf[1]+((bo+1)&0xf),buf[0]+bo,bandPtr); ++ } ++ else { ++ b0 = buf[1]; ++ bo1 = bo+1; ++ dct64(buf[0]+bo,buf[1]+bo+1,bandPtr); ++ } ++ ++ gmp->synth_bo = bo; ++ ++ { ++ register int j; ++ real *window = decwin + 16 - bo1; ++ ++ for (j=16;j;j--,b0+=0x10,window+=0x20,samples+=step) ++ { ++ real sum; ++ sum = window[0x0] * b0[0x0]; ++ sum -= window[0x1] * b0[0x1]; ++ sum += window[0x2] * b0[0x2]; ++ sum -= window[0x3] * b0[0x3]; ++ sum += window[0x4] * b0[0x4]; ++ sum -= window[0x5] * b0[0x5]; ++ sum += window[0x6] * b0[0x6]; ++ sum -= window[0x7] * b0[0x7]; ++ sum += window[0x8] * b0[0x8]; ++ sum -= window[0x9] * b0[0x9]; ++ sum += window[0xA] * b0[0xA]; ++ sum -= window[0xB] * b0[0xB]; ++ sum += window[0xC] * b0[0xC]; ++ sum -= window[0xD] * b0[0xD]; ++ sum += window[0xE] * b0[0xE]; ++ sum -= window[0xF] * b0[0xF]; ++ ++ WRITE_SAMPLE(samples,sum,clip); ++ } ++ ++ { ++ real sum; ++ sum = window[0x0] * b0[0x0]; ++ sum += window[0x2] * b0[0x2]; ++ sum += window[0x4] * b0[0x4]; ++ sum += window[0x6] * b0[0x6]; ++ sum += window[0x8] * b0[0x8]; ++ sum += window[0xA] * b0[0xA]; ++ sum += window[0xC] * b0[0xC]; ++ sum += window[0xE] * b0[0xE]; ++ WRITE_SAMPLE(samples,sum,clip); ++ b0-=0x10,window-=0x20,samples+=step; ++ } ++ window += bo1<<1; ++ ++ for (j=15;j;j--,b0-=0x10,window-=0x20,samples+=step) ++ { ++ real sum; ++ sum = -window[-0x1] * b0[0x0]; ++ sum -= window[-0x2] * b0[0x1]; ++ sum -= window[-0x3] * b0[0x2]; ++ sum -= window[-0x4] * b0[0x3]; ++ sum -= window[-0x5] * b0[0x4]; ++ sum -= window[-0x6] * b0[0x5]; ++ sum -= window[-0x7] * b0[0x6]; ++ sum -= window[-0x8] * b0[0x7]; ++ sum -= window[-0x9] * b0[0x8]; ++ sum -= window[-0xA] * b0[0x9]; ++ sum -= window[-0xB] * b0[0xA]; ++ sum -= window[-0xC] * b0[0xB]; ++ sum -= window[-0xD] * b0[0xC]; ++ sum -= window[-0xE] * b0[0xD]; ++ sum -= window[-0xF] * b0[0xE]; ++ sum -= window[-0x0] * b0[0xF]; ++ ++ WRITE_SAMPLE(samples,sum,clip); ++ } ++ } ++ *pnt += 128; ++ ++ return clip; ++ } ++ ++ +diff -r -c -N encoder/mpglib/huffman.h lame3.50/mpglib/huffman.h +*** encoder/mpglib/huffman.h Wed Dec 31 17:00:00 1969 +--- lame3.50/mpglib/huffman.h Fri Sep 10 09:51:57 1999 +*************** +*** 0 **** +--- 1,332 ---- ++ /* ++ * huffman tables ... recalcualted to work with my optimzed ++ * decoder scheme (MH) ++ * ++ * probably we could save a few bytes of memory, because the ++ * smaller tables are often the part of a bigger table ++ */ ++ ++ struct newhuff ++ { ++ unsigned int linbits; ++ short *table; ++ }; ++ ++ static short tab0[] = ++ { ++ 0 ++ }; ++ ++ static short tab1[] = ++ { ++ -5, -3, -1, 17, 1, 16, 0 ++ }; ++ ++ static short tab2[] = ++ { ++ -15, -11, -9, -5, -3, -1, 34, 2, 18, -1, 33, 32, 17, -1, 1, ++ 16, 0 ++ }; ++ ++ static short tab3[] = ++ { ++ -13, -11, -9, -5, -3, -1, 34, 2, 18, -1, 33, 32, 16, 17, -1, ++ 1, 0 ++ }; ++ ++ static short tab5[] = ++ { ++ -29, -25, -23, -15, -7, -5, -3, -1, 51, 35, 50, 49, -3, -1, 19, ++ 3, -1, 48, 34, -3, -1, 18, 33, -1, 2, 32, 17, -1, 1, 16, ++ 0 ++ }; ++ ++ static short tab6[] = ++ { ++ -25, -19, -13, -9, -5, -3, -1, 51, 3, 35, -1, 50, 48, -1, 19, ++ 49, -3, -1, 34, 2, 18, -3, -1, 33, 32, 1, -1, 17, -1, 16, ++ 0 ++ }; ++ ++ static short tab7[] = ++ { ++ -69, -65, -57, -39, -29, -17, -11, -7, -3, -1, 85, 69, -1, 84, 83, ++ -1, 53, 68, -3, -1, 37, 82, 21, -5, -1, 81, -1, 5, 52, -1, ++ 80, -1, 67, 51, -5, -3, -1, 36, 66, 20, -1, 65, 64, -11, -7, ++ -3, -1, 4, 35, -1, 50, 3, -1, 19, 49, -3, -1, 48, 34, 18, ++ -5, -1, 33, -1, 2, 32, 17, -1, 1, 16, 0 ++ }; ++ ++ static short tab8[] = ++ { ++ -65, -63, -59, -45, -31, -19, -13, -7, -5, -3, -1, 85, 84, 69, 83, ++ -3, -1, 53, 68, 37, -3, -1, 82, 5, 21, -5, -1, 81, -1, 52, ++ 67, -3, -1, 80, 51, 36, -5, -3, -1, 66, 20, 65, -3, -1, 4, ++ 64, -1, 35, 50, -9, -7, -3, -1, 19, 49, -1, 3, 48, 34, -1, ++ 2, 32, -1, 18, 33, 17, -3, -1, 1, 16, 0 ++ }; ++ ++ static short tab9[] = ++ { ++ -63, -53, -41, -29, -19, -11, -5, -3, -1, 85, 69, 53, -1, 83, -1, ++ 84, 5, -3, -1, 68, 37, -1, 82, 21, -3, -1, 81, 52, -1, 67, ++ -1, 80, 4, -7, -3, -1, 36, 66, -1, 51, 64, -1, 20, 65, -5, ++ -3, -1, 35, 50, 19, -1, 49, -1, 3, 48, -5, -3, -1, 34, 2, ++ 18, -1, 33, 32, -3, -1, 17, 1, -1, 16, 0 ++ }; ++ ++ static short tab10[] = ++ { ++ -125,-121,-111, -83, -55, -35, -21, -13, -7, -3, -1, 119, 103, -1, 118, ++ 87, -3, -1, 117, 102, 71, -3, -1, 116, 86, -1, 101, 55, -9, -3, ++ -1, 115, 70, -3, -1, 85, 84, 99, -1, 39, 114, -11, -5, -3, -1, ++ 100, 7, 112, -1, 98, -1, 69, 53, -5, -1, 6, -1, 83, 68, 23, ++ -17, -5, -1, 113, -1, 54, 38, -5, -3, -1, 37, 82, 21, -1, 81, ++ -1, 52, 67, -3, -1, 22, 97, -1, 96, -1, 5, 80, -19, -11, -7, ++ -3, -1, 36, 66, -1, 51, 4, -1, 20, 65, -3, -1, 64, 35, -1, ++ 50, 3, -3, -1, 19, 49, -1, 48, 34, -7, -3, -1, 18, 33, -1, ++ 2, 32, 17, -1, 1, 16, 0 ++ }; ++ ++ static short tab11[] = ++ { ++ -121,-113, -89, -59, -43, -27, -17, -7, -3, -1, 119, 103, -1, 118, 117, ++ -3, -1, 102, 71, -1, 116, -1, 87, 85, -5, -3, -1, 86, 101, 55, ++ -1, 115, 70, -9, -7, -3, -1, 69, 84, -1, 53, 83, 39, -1, 114, ++ -1, 100, 7, -5, -1, 113, -1, 23, 112, -3, -1, 54, 99, -1, 96, ++ -1, 68, 37, -13, -7, -5, -3, -1, 82, 5, 21, 98, -3, -1, 38, ++ 6, 22, -5, -1, 97, -1, 81, 52, -5, -1, 80, -1, 67, 51, -1, ++ 36, 66, -15, -11, -7, -3, -1, 20, 65, -1, 4, 64, -1, 35, 50, ++ -1, 19, 49, -5, -3, -1, 3, 48, 34, 33, -5, -1, 18, -1, 2, ++ 32, 17, -3, -1, 1, 16, 0 ++ }; ++ ++ static short tab12[] = ++ { ++ -115, -99, -73, -45, -27, -17, -9, -5, -3, -1, 119, 103, 118, -1, 87, ++ 117, -3, -1, 102, 71, -1, 116, 101, -3, -1, 86, 55, -3, -1, 115, ++ 85, 39, -7, -3, -1, 114, 70, -1, 100, 23, -5, -1, 113, -1, 7, ++ 112, -1, 54, 99, -13, -9, -3, -1, 69, 84, -1, 68, -1, 6, 5, ++ -1, 38, 98, -5, -1, 97, -1, 22, 96, -3, -1, 53, 83, -1, 37, ++ 82, -17, -7, -3, -1, 21, 81, -1, 52, 67, -5, -3, -1, 80, 4, ++ 36, -1, 66, 20, -3, -1, 51, 65, -1, 35, 50, -11, -7, -5, -3, ++ -1, 64, 3, 48, 19, -1, 49, 34, -1, 18, 33, -7, -5, -3, -1, ++ 2, 32, 0, 17, -1, 1, 16 ++ }; ++ ++ static short tab13[] = ++ { ++ -509,-503,-475,-405,-333,-265,-205,-153,-115, -83, -53, -35, -21, -13, -9, ++ -7, -5, -3, -1, 254, 252, 253, 237, 255, -1, 239, 223, -3, -1, 238, ++ 207, -1, 222, 191, -9, -3, -1, 251, 206, -1, 220, -1, 175, 233, -1, ++ 236, 221, -9, -5, -3, -1, 250, 205, 190, -1, 235, 159, -3, -1, 249, ++ 234, -1, 189, 219, -17, -9, -3, -1, 143, 248, -1, 204, -1, 174, 158, ++ -5, -1, 142, -1, 127, 126, 247, -5, -1, 218, -1, 173, 188, -3, -1, ++ 203, 246, 111, -15, -7, -3, -1, 232, 95, -1, 157, 217, -3, -1, 245, ++ 231, -1, 172, 187, -9, -3, -1, 79, 244, -3, -1, 202, 230, 243, -1, ++ 63, -1, 141, 216, -21, -9, -3, -1, 47, 242, -3, -1, 110, 156, 15, ++ -5, -3, -1, 201, 94, 171, -3, -1, 125, 215, 78, -11, -5, -3, -1, ++ 200, 214, 62, -1, 185, -1, 155, 170, -1, 31, 241, -23, -13, -5, -1, ++ 240, -1, 186, 229, -3, -1, 228, 140, -1, 109, 227, -5, -1, 226, -1, ++ 46, 14, -1, 30, 225, -15, -7, -3, -1, 224, 93, -1, 213, 124, -3, ++ -1, 199, 77, -1, 139, 184, -7, -3, -1, 212, 154, -1, 169, 108, -1, ++ 198, 61, -37, -21, -9, -5, -3, -1, 211, 123, 45, -1, 210, 29, -5, ++ -1, 183, -1, 92, 197, -3, -1, 153, 122, 195, -7, -5, -3, -1, 167, ++ 151, 75, 209, -3, -1, 13, 208, -1, 138, 168, -11, -7, -3, -1, 76, ++ 196, -1, 107, 182, -1, 60, 44, -3, -1, 194, 91, -3, -1, 181, 137, ++ 28, -43, -23, -11, -5, -1, 193, -1, 152, 12, -1, 192, -1, 180, 106, ++ -5, -3, -1, 166, 121, 59, -1, 179, -1, 136, 90, -11, -5, -1, 43, ++ -1, 165, 105, -1, 164, -1, 120, 135, -5, -1, 148, -1, 119, 118, 178, ++ -11, -3, -1, 27, 177, -3, -1, 11, 176, -1, 150, 74, -7, -3, -1, ++ 58, 163, -1, 89, 149, -1, 42, 162, -47, -23, -9, -3, -1, 26, 161, ++ -3, -1, 10, 104, 160, -5, -3, -1, 134, 73, 147, -3, -1, 57, 88, ++ -1, 133, 103, -9, -3, -1, 41, 146, -3, -1, 87, 117, 56, -5, -1, ++ 131, -1, 102, 71, -3, -1, 116, 86, -1, 101, 115, -11, -3, -1, 25, ++ 145, -3, -1, 9, 144, -1, 72, 132, -7, -5, -1, 114, -1, 70, 100, ++ 40, -1, 130, 24, -41, -27, -11, -5, -3, -1, 55, 39, 23, -1, 113, ++ -1, 85, 7, -7, -3, -1, 112, 54, -1, 99, 69, -3, -1, 84, 38, ++ -1, 98, 53, -5, -1, 129, -1, 8, 128, -3, -1, 22, 97, -1, 6, ++ 96, -13, -9, -5, -3, -1, 83, 68, 37, -1, 82, 5, -1, 21, 81, ++ -7, -3, -1, 52, 67, -1, 80, 36, -3, -1, 66, 51, 20, -19, -11, ++ -5, -1, 65, -1, 4, 64, -3, -1, 35, 50, 19, -3, -1, 49, 3, ++ -1, 48, 34, -3, -1, 18, 33, -1, 2, 32, -3, -1, 17, 1, 16, ++ 0 ++ }; ++ ++ static short tab15[] = ++ { ++ -495,-445,-355,-263,-183,-115, -77, -43, -27, -13, -7, -3, -1, 255, 239, ++ -1, 254, 223, -1, 238, -1, 253, 207, -7, -3, -1, 252, 222, -1, 237, ++ 191, -1, 251, -1, 206, 236, -7, -3, -1, 221, 175, -1, 250, 190, -3, ++ -1, 235, 205, -1, 220, 159, -15, -7, -3, -1, 249, 234, -1, 189, 219, ++ -3, -1, 143, 248, -1, 204, 158, -7, -3, -1, 233, 127, -1, 247, 173, ++ -3, -1, 218, 188, -1, 111, -1, 174, 15, -19, -11, -3, -1, 203, 246, ++ -3, -1, 142, 232, -1, 95, 157, -3, -1, 245, 126, -1, 231, 172, -9, ++ -3, -1, 202, 187, -3, -1, 217, 141, 79, -3, -1, 244, 63, -1, 243, ++ 216, -33, -17, -9, -3, -1, 230, 47, -1, 242, -1, 110, 240, -3, -1, ++ 31, 241, -1, 156, 201, -7, -3, -1, 94, 171, -1, 186, 229, -3, -1, ++ 125, 215, -1, 78, 228, -15, -7, -3, -1, 140, 200, -1, 62, 109, -3, ++ -1, 214, 227, -1, 155, 185, -7, -3, -1, 46, 170, -1, 226, 30, -5, ++ -1, 225, -1, 14, 224, -1, 93, 213, -45, -25, -13, -7, -3, -1, 124, ++ 199, -1, 77, 139, -1, 212, -1, 184, 154, -7, -3, -1, 169, 108, -1, ++ 198, 61, -1, 211, 210, -9, -5, -3, -1, 45, 13, 29, -1, 123, 183, ++ -5, -1, 209, -1, 92, 208, -1, 197, 138, -17, -7, -3, -1, 168, 76, ++ -1, 196, 107, -5, -1, 182, -1, 153, 12, -1, 60, 195, -9, -3, -1, ++ 122, 167, -1, 166, -1, 192, 11, -1, 194, -1, 44, 91, -55, -29, -15, ++ -7, -3, -1, 181, 28, -1, 137, 152, -3, -1, 193, 75, -1, 180, 106, ++ -5, -3, -1, 59, 121, 179, -3, -1, 151, 136, -1, 43, 90, -11, -5, ++ -1, 178, -1, 165, 27, -1, 177, -1, 176, 105, -7, -3, -1, 150, 74, ++ -1, 164, 120, -3, -1, 135, 58, 163, -17, -7, -3, -1, 89, 149, -1, ++ 42, 162, -3, -1, 26, 161, -3, -1, 10, 160, 104, -7, -3, -1, 134, ++ 73, -1, 148, 57, -5, -1, 147, -1, 119, 9, -1, 88, 133, -53, -29, ++ -13, -7, -3, -1, 41, 103, -1, 118, 146, -1, 145, -1, 25, 144, -7, ++ -3, -1, 72, 132, -1, 87, 117, -3, -1, 56, 131, -1, 102, 71, -7, ++ -3, -1, 40, 130, -1, 24, 129, -7, -3, -1, 116, 8, -1, 128, 86, ++ -3, -1, 101, 55, -1, 115, 70, -17, -7, -3, -1, 39, 114, -1, 100, ++ 23, -3, -1, 85, 113, -3, -1, 7, 112, 54, -7, -3, -1, 99, 69, ++ -1, 84, 38, -3, -1, 98, 22, -3, -1, 6, 96, 53, -33, -19, -9, ++ -5, -1, 97, -1, 83, 68, -1, 37, 82, -3, -1, 21, 81, -3, -1, ++ 5, 80, 52, -7, -3, -1, 67, 36, -1, 66, 51, -1, 65, -1, 20, ++ 4, -9, -3, -1, 35, 50, -3, -1, 64, 3, 19, -3, -1, 49, 48, ++ 34, -9, -7, -3, -1, 18, 33, -1, 2, 32, 17, -3, -1, 1, 16, ++ 0 ++ }; ++ ++ static short tab16[] = ++ { ++ -509,-503,-461,-323,-103, -37, -27, -15, -7, -3, -1, 239, 254, -1, 223, ++ 253, -3, -1, 207, 252, -1, 191, 251, -5, -1, 175, -1, 250, 159, -3, ++ -1, 249, 248, 143, -7, -3, -1, 127, 247, -1, 111, 246, 255, -9, -5, ++ -3, -1, 95, 245, 79, -1, 244, 243, -53, -1, 240, -1, 63, -29, -19, ++ -13, -7, -5, -1, 206, -1, 236, 221, 222, -1, 233, -1, 234, 217, -1, ++ 238, -1, 237, 235, -3, -1, 190, 205, -3, -1, 220, 219, 174, -11, -5, ++ -1, 204, -1, 173, 218, -3, -1, 126, 172, 202, -5, -3, -1, 201, 125, ++ 94, 189, 242, -93, -5, -3, -1, 47, 15, 31, -1, 241, -49, -25, -13, ++ -5, -1, 158, -1, 188, 203, -3, -1, 142, 232, -1, 157, 231, -7, -3, ++ -1, 187, 141, -1, 216, 110, -1, 230, 156, -13, -7, -3, -1, 171, 186, ++ -1, 229, 215, -1, 78, -1, 228, 140, -3, -1, 200, 62, -1, 109, -1, ++ 214, 155, -19, -11, -5, -3, -1, 185, 170, 225, -1, 212, -1, 184, 169, ++ -5, -1, 123, -1, 183, 208, 227, -7, -3, -1, 14, 224, -1, 93, 213, ++ -3, -1, 124, 199, -1, 77, 139, -75, -45, -27, -13, -7, -3, -1, 154, ++ 108, -1, 198, 61, -3, -1, 92, 197, 13, -7, -3, -1, 138, 168, -1, ++ 153, 76, -3, -1, 182, 122, 60, -11, -5, -3, -1, 91, 137, 28, -1, ++ 192, -1, 152, 121, -1, 226, -1, 46, 30, -15, -7, -3, -1, 211, 45, ++ -1, 210, 209, -5, -1, 59, -1, 151, 136, 29, -7, -3, -1, 196, 107, ++ -1, 195, 167, -1, 44, -1, 194, 181, -23, -13, -7, -3, -1, 193, 12, ++ -1, 75, 180, -3, -1, 106, 166, 179, -5, -3, -1, 90, 165, 43, -1, ++ 178, 27, -13, -5, -1, 177, -1, 11, 176, -3, -1, 105, 150, -1, 74, ++ 164, -5, -3, -1, 120, 135, 163, -3, -1, 58, 89, 42, -97, -57, -33, ++ -19, -11, -5, -3, -1, 149, 104, 161, -3, -1, 134, 119, 148, -5, -3, ++ -1, 73, 87, 103, 162, -5, -1, 26, -1, 10, 160, -3, -1, 57, 147, ++ -1, 88, 133, -9, -3, -1, 41, 146, -3, -1, 118, 9, 25, -5, -1, ++ 145, -1, 144, 72, -3, -1, 132, 117, -1, 56, 131, -21, -11, -5, -3, ++ -1, 102, 40, 130, -3, -1, 71, 116, 24, -3, -1, 129, 128, -3, -1, ++ 8, 86, 55, -9, -5, -1, 115, -1, 101, 70, -1, 39, 114, -5, -3, ++ -1, 100, 85, 7, 23, -23, -13, -5, -1, 113, -1, 112, 54, -3, -1, ++ 99, 69, -1, 84, 38, -3, -1, 98, 22, -1, 97, -1, 6, 96, -9, ++ -5, -1, 83, -1, 53, 68, -1, 37, 82, -1, 81, -1, 21, 5, -33, ++ -23, -13, -7, -3, -1, 52, 67, -1, 80, 36, -3, -1, 66, 51, 20, ++ -5, -1, 65, -1, 4, 64, -1, 35, 50, -3, -1, 19, 49, -3, -1, ++ 3, 48, 34, -3, -1, 18, 33, -1, 2, 32, -3, -1, 17, 1, 16, ++ 0 ++ }; ++ ++ static short tab24[] = ++ { ++ -451,-117, -43, -25, -15, -7, -3, -1, 239, 254, -1, 223, 253, -3, -1, ++ 207, 252, -1, 191, 251, -5, -1, 250, -1, 175, 159, -1, 249, 248, -9, ++ -5, -3, -1, 143, 127, 247, -1, 111, 246, -3, -1, 95, 245, -1, 79, ++ 244, -71, -7, -3, -1, 63, 243, -1, 47, 242, -5, -1, 241, -1, 31, ++ 240, -25, -9, -1, 15, -3, -1, 238, 222, -1, 237, 206, -7, -3, -1, ++ 236, 221, -1, 190, 235, -3, -1, 205, 220, -1, 174, 234, -15, -7, -3, ++ -1, 189, 219, -1, 204, 158, -3, -1, 233, 173, -1, 218, 188, -7, -3, ++ -1, 203, 142, -1, 232, 157, -3, -1, 217, 126, -1, 231, 172, 255,-235, ++ -143, -77, -45, -25, -15, -7, -3, -1, 202, 187, -1, 141, 216, -5, -3, ++ -1, 14, 224, 13, 230, -5, -3, -1, 110, 156, 201, -1, 94, 186, -9, ++ -5, -1, 229, -1, 171, 125, -1, 215, 228, -3, -1, 140, 200, -3, -1, ++ 78, 46, 62, -15, -7, -3, -1, 109, 214, -1, 227, 155, -3, -1, 185, ++ 170, -1, 226, 30, -7, -3, -1, 225, 93, -1, 213, 124, -3, -1, 199, ++ 77, -1, 139, 184, -31, -15, -7, -3, -1, 212, 154, -1, 169, 108, -3, ++ -1, 198, 61, -1, 211, 45, -7, -3, -1, 210, 29, -1, 123, 183, -3, ++ -1, 209, 92, -1, 197, 138, -17, -7, -3, -1, 168, 153, -1, 76, 196, ++ -3, -1, 107, 182, -3, -1, 208, 12, 60, -7, -3, -1, 195, 122, -1, ++ 167, 44, -3, -1, 194, 91, -1, 181, 28, -57, -35, -19, -7, -3, -1, ++ 137, 152, -1, 193, 75, -5, -3, -1, 192, 11, 59, -3, -1, 176, 10, ++ 26, -5, -1, 180, -1, 106, 166, -3, -1, 121, 151, -3, -1, 160, 9, ++ 144, -9, -3, -1, 179, 136, -3, -1, 43, 90, 178, -7, -3, -1, 165, ++ 27, -1, 177, 105, -1, 150, 164, -17, -9, -5, -3, -1, 74, 120, 135, ++ -1, 58, 163, -3, -1, 89, 149, -1, 42, 162, -7, -3, -1, 161, 104, ++ -1, 134, 119, -3, -1, 73, 148, -1, 57, 147, -63, -31, -15, -7, -3, ++ -1, 88, 133, -1, 41, 103, -3, -1, 118, 146, -1, 25, 145, -7, -3, ++ -1, 72, 132, -1, 87, 117, -3, -1, 56, 131, -1, 102, 40, -17, -7, ++ -3, -1, 130, 24, -1, 71, 116, -5, -1, 129, -1, 8, 128, -1, 86, ++ 101, -7, -5, -1, 23, -1, 7, 112, 115, -3, -1, 55, 39, 114, -15, ++ -7, -3, -1, 70, 100, -1, 85, 113, -3, -1, 54, 99, -1, 69, 84, ++ -7, -3, -1, 38, 98, -1, 22, 97, -5, -3, -1, 6, 96, 53, -1, ++ 83, 68, -51, -37, -23, -15, -9, -3, -1, 37, 82, -1, 21, -1, 5, ++ 80, -1, 81, -1, 52, 67, -3, -1, 36, 66, -1, 51, 20, -9, -5, ++ -1, 65, -1, 4, 64, -1, 35, 50, -1, 19, 49, -7, -5, -3, -1, ++ 3, 48, 34, 18, -1, 33, -1, 2, 32, -3, -1, 17, 1, -1, 16, ++ 0 ++ }; ++ ++ static short tab_c0[] = ++ { ++ -29, -21, -13, -7, -3, -1, 11, 15, -1, 13, 14, -3, -1, 7, 5, ++ 9, -3, -1, 6, 3, -1, 10, 12, -3, -1, 2, 1, -1, 4, 8, ++ 0 ++ }; ++ ++ static short tab_c1[] = ++ { ++ -15, -7, -3, -1, 15, 14, -1, 13, 12, -3, -1, 11, 10, -1, 9, ++ 8, -7, -3, -1, 7, 6, -1, 5, 4, -3, -1, 3, 2, -1, 1, ++ 0 ++ }; ++ ++ ++ ++ static struct newhuff ht[] = ++ { ++ { /* 0 */ 0 , tab0 } , ++ { /* 2 */ 0 , tab1 } , ++ { /* 3 */ 0 , tab2 } , ++ { /* 3 */ 0 , tab3 } , ++ { /* 0 */ 0 , tab0 } , ++ { /* 4 */ 0 , tab5 } , ++ { /* 4 */ 0 , tab6 } , ++ { /* 6 */ 0 , tab7 } , ++ { /* 6 */ 0 , tab8 } , ++ { /* 6 */ 0 , tab9 } , ++ { /* 8 */ 0 , tab10 } , ++ { /* 8 */ 0 , tab11 } , ++ { /* 8 */ 0 , tab12 } , ++ { /* 16 */ 0 , tab13 } , ++ { /* 0 */ 0 , tab0 } , ++ { /* 16 */ 0 , tab15 } , ++ ++ { /* 16 */ 1 , tab16 } , ++ { /* 16 */ 2 , tab16 } , ++ { /* 16 */ 3 , tab16 } , ++ { /* 16 */ 4 , tab16 } , ++ { /* 16 */ 6 , tab16 } , ++ { /* 16 */ 8 , tab16 } , ++ { /* 16 */ 10, tab16 } , ++ { /* 16 */ 13, tab16 } , ++ { /* 16 */ 4 , tab24 } , ++ { /* 16 */ 5 , tab24 } , ++ { /* 16 */ 6 , tab24 } , ++ { /* 16 */ 7 , tab24 } , ++ { /* 16 */ 8 , tab24 } , ++ { /* 16 */ 9 , tab24 } , ++ { /* 16 */ 11, tab24 } , ++ { /* 16 */ 13, tab24 } ++ }; ++ ++ static struct newhuff htc[] = ++ { ++ { /* 1 , 1 , */ 0 , tab_c0 } , ++ { /* 1 , 1 , */ 0 , tab_c1 } ++ }; ++ ++ +diff -r -c -N encoder/mpglib/interface.c lame3.50/mpglib/interface.c +*** encoder/mpglib/interface.c Wed Dec 31 17:00:00 1969 +--- lame3.50/mpglib/interface.c Tue Oct 12 10:46:30 1999 +*************** +*** 0 **** +--- 1,218 ---- ++ #include <stdlib.h> ++ #include <stdio.h> ++ ++ #include "mpg123.h" ++ #include "mpglib.h" ++ ++ ++ /* Global mp .. it's a hack */ ++ struct mpstr *gmp; ++ ++ ++ BOOL InitMP3(struct mpstr *mp) ++ { ++ memset(mp,0,sizeof(struct mpstr)); ++ ++ mp->framesize = 0; ++ mp->fsizeold = -1; ++ mp->bsize = 0; ++ mp->head = mp->tail = NULL; ++ mp->fr.single = -1; ++ mp->bsnum = 0; ++ mp->synth_bo = 1; ++ ++ make_decode_tables(32767); ++ init_layer3(SBLIMIT); ++ ++ return !0; ++ } ++ ++ void ExitMP3(struct mpstr *mp) ++ { ++ struct buf *b,*bn; ++ ++ b = mp->tail; ++ while(b) { ++ free(b->pnt); ++ bn = b->next; ++ free(b); ++ b = bn; ++ } ++ } ++ ++ static struct buf *addbuf(struct mpstr *mp,char *buf,int size) ++ { ++ struct buf *nbuf; ++ ++ nbuf = (struct buf*) malloc( sizeof(struct buf) ); ++ if(!nbuf) { ++ fprintf(stderr,"Out of memory!\n"); ++ return NULL; ++ } ++ nbuf->pnt = (unsigned char*) malloc(size); ++ if(!nbuf->pnt) { ++ free(nbuf); ++ return NULL; ++ } ++ nbuf->size = size; ++ memcpy(nbuf->pnt,buf,size); ++ nbuf->next = NULL; ++ nbuf->prev = mp->head; ++ nbuf->pos = 0; ++ ++ if(!mp->tail) { ++ mp->tail = nbuf; ++ } ++ else { ++ mp->head->next = nbuf; ++ } ++ ++ mp->head = nbuf; ++ mp->bsize += size; ++ ++ return nbuf; ++ } ++ ++ static void remove_buf(struct mpstr *mp) ++ { ++ struct buf *buf = mp->tail; ++ ++ mp->tail = buf->next; ++ if(mp->tail) ++ mp->tail->prev = NULL; ++ else { ++ mp->tail = mp->head = NULL; ++ } ++ ++ free(buf->pnt); ++ free(buf); ++ ++ } ++ ++ static int read_buf_byte(struct mpstr *mp) ++ { ++ unsigned int b; ++ ++ int pos; ++ ++ pos = mp->tail->pos; ++ while(pos >= mp->tail->size) { ++ remove_buf(mp); ++ pos = mp->tail->pos; ++ if(!mp->tail) { ++ fprintf(stderr,"Fatal error!\n"); ++ exit(1); ++ } ++ } ++ ++ b = mp->tail->pnt[pos]; ++ mp->bsize--; ++ mp->tail->pos++; ++ ++ ++ return b; ++ } ++ ++ static void read_head(struct mpstr *mp) ++ { ++ unsigned long head; ++ ++ head = read_buf_byte(mp); ++ head <<= 8; ++ head |= read_buf_byte(mp); ++ head <<= 8; ++ head |= read_buf_byte(mp); ++ head <<= 8; ++ head |= read_buf_byte(mp); ++ ++ mp->header = head; ++ } ++ ++ int decodeMP3(struct mpstr *mp,char *in,int isize,char *out, ++ int osize,int *done) ++ { ++ int len; ++ ++ gmp = mp; ++ ++ if(osize < 4608) { ++ fprintf(stderr,"To less out space\n"); ++ return MP3_ERR; ++ } ++ ++ if(in) { ++ if(addbuf(mp,in,isize) == NULL) { ++ return MP3_ERR; ++ } ++ } ++ ++ ++ /* First decode header */ ++ if(mp->framesize == 0) { ++ if(mp->bsize < 4) { ++ return MP3_NEED_MORE; ++ } ++ read_head(mp); ++ decode_header(&mp->fr,mp->header); ++ mp->framesize = mp->fr.framesize; ++ } ++ ++ /* printf(" fr.framesize = %i \n",mp->fr.framesize); ++ printf(" bsize = %i \n",mp->bsize); ++ */ ++ ++ if(mp->fr.framesize > mp->bsize) { ++ return MP3_NEED_MORE; ++ } ++ wordpointer = mp->bsspace[mp->bsnum] + 512; ++ mp->bsnum = (mp->bsnum + 1) & 0x1; ++ bitindex = 0; ++ ++ len = 0; ++ while(len < mp->framesize) { ++ int nlen; ++ int blen = mp->tail->size - mp->tail->pos; ++ if( (mp->framesize - len) <= blen) { ++ nlen = mp->framesize-len; ++ } ++ else { ++ nlen = blen; ++ } ++ memcpy(wordpointer+len,mp->tail->pnt+mp->tail->pos,nlen); ++ len += nlen; ++ mp->tail->pos += nlen; ++ mp->bsize -= nlen; ++ if(mp->tail->pos == mp->tail->size) { ++ remove_buf(mp); ++ } ++ } ++ ++ *done = 0; ++ if(mp->fr.error_protection) ++ getbits(16); ++ do_layer3(&mp->fr,(unsigned char *) out,done); ++ ++ mp->fsizeold = mp->framesize; ++ mp->framesize = 0; ++ return MP3_OK; ++ } ++ ++ int set_pointer(long backstep) ++ { ++ unsigned char *bsbufold; ++ if(gmp->fsizeold < 0 && backstep > 0) { ++ fprintf(stderr,"Can't step back %ld!\n",backstep); ++ return MP3_ERR; ++ } ++ bsbufold = gmp->bsspace[gmp->bsnum] + 512; ++ wordpointer -= backstep; ++ if (backstep) ++ memcpy(wordpointer,bsbufold+gmp->fsizeold-backstep,backstep); ++ bitindex = 0; ++ return MP3_OK; ++ } ++ ++ ++ ++ ++ +diff -r -c -N encoder/mpglib/layer3.c lame3.50/mpglib/layer3.c +*** encoder/mpglib/layer3.c Wed Dec 31 17:00:00 1969 +--- lame3.50/mpglib/layer3.c Mon Oct 25 12:17:41 1999 +*************** +*** 0 **** +--- 1,1681 ---- ++ /* ++ * Mpeg Layer-3 audio decoder ++ * -------------------------- ++ * copyright (c) 1995,1996,1997 by Michael Hipp. ++ * All rights reserved. See also 'README' ++ */ ++ ++ #include <stdlib.h> ++ #include "mpg123.h" ++ #include "mpglib.h" ++ #include "huffman.h" ++ #ifdef OS_AMIGAOS ++ #include "/gtkanal.h" ++ #else ++ #include "../gtkanal.h" ++ #endif ++ ++ ++ extern struct mpstr *gmp; ++ ++ #define MPEG1 ++ ++ ++ static real ispow[8207]; ++ static real aa_ca[8],aa_cs[8]; ++ static real COS1[12][6]; ++ static real win[4][36]; ++ static real win1[4][36]; ++ static real gainpow2[256+118+4]; ++ static real COS9[9]; ++ static real COS6_1,COS6_2; ++ static real tfcos36[9]; ++ static real tfcos12[3]; ++ ++ struct bandInfoStruct { ++ short longIdx[23]; ++ short longDiff[22]; ++ short shortIdx[14]; ++ short shortDiff[13]; ++ }; ++ ++ int longLimit[9][23]; ++ int shortLimit[9][14]; ++ ++ struct bandInfoStruct bandInfo[9] = { ++ ++ /* MPEG 1.0 */ ++ { {0,4,8,12,16,20,24,30,36,44,52,62,74, 90,110,134,162,196,238,288,342,418,576}, ++ {4,4,4,4,4,4,6,6,8, 8,10,12,16,20,24,28,34,42,50,54, 76,158}, ++ {0,4*3,8*3,12*3,16*3,22*3,30*3,40*3,52*3,66*3, 84*3,106*3,136*3,192*3}, ++ {4,4,4,4,6,8,10,12,14,18,22,30,56} } , ++ ++ { {0,4,8,12,16,20,24,30,36,42,50,60,72, 88,106,128,156,190,230,276,330,384,576}, ++ {4,4,4,4,4,4,6,6,6, 8,10,12,16,18,22,28,34,40,46,54, 54,192}, ++ {0,4*3,8*3,12*3,16*3,22*3,28*3,38*3,50*3,64*3, 80*3,100*3,126*3,192*3}, ++ {4,4,4,4,6,6,10,12,14,16,20,26,66} } , ++ ++ { {0,4,8,12,16,20,24,30,36,44,54,66,82,102,126,156,194,240,296,364,448,550,576} , ++ {4,4,4,4,4,4,6,6,8,10,12,16,20,24,30,38,46,56,68,84,102, 26} , ++ {0,4*3,8*3,12*3,16*3,22*3,30*3,42*3,58*3,78*3,104*3,138*3,180*3,192*3} , ++ {4,4,4,4,6,8,12,16,20,26,34,42,12} } , ++ ++ /* MPEG 2.0 */ ++ { {0,6,12,18,24,30,36,44,54,66,80,96,116,140,168,200,238,284,336,396,464,522,576}, ++ {6,6,6,6,6,6,8,10,12,14,16,20,24,28,32,38,46,52,60,68,58,54 } , ++ {0,4*3,8*3,12*3,18*3,24*3,32*3,42*3,56*3,74*3,100*3,132*3,174*3,192*3} , ++ {4,4,4,6,6,8,10,14,18,26,32,42,18 } } , ++ ++ { {0,6,12,18,24,30,36,44,54,66,80,96,114,136,162,194,232,278,330,394,464,540,576}, ++ {6,6,6,6,6,6,8,10,12,14,16,18,22,26,32,38,46,52,64,70,76,36 } , ++ {0,4*3,8*3,12*3,18*3,26*3,36*3,48*3,62*3,80*3,104*3,136*3,180*3,192*3} , ++ {4,4,4,6,8,10,12,14,18,24,32,44,12 } } , ++ ++ { {0,6,12,18,24,30,36,44,54,66,80,96,116,140,168,200,238,284,336,396,464,522,576}, ++ {6,6,6,6,6,6,8,10,12,14,16,20,24,28,32,38,46,52,60,68,58,54 }, ++ {0,4*3,8*3,12*3,18*3,26*3,36*3,48*3,62*3,80*3,104*3,134*3,174*3,192*3}, ++ {4,4,4,6,8,10,12,14,18,24,30,40,18 } } , ++ /* MPEG 2.5 */ ++ { {0,6,12,18,24,30,36,44,54,66,80,96,116,140,168,200,238,284,336,396,464,522,576} , ++ {6,6,6,6,6,6,8,10,12,14,16,20,24,28,32,38,46,52,60,68,58,54}, ++ {0,12,24,36,54,78,108,144,186,240,312,402,522,576}, ++ {4,4,4,6,8,10,12,14,18,24,30,40,18} }, ++ { {0,6,12,18,24,30,36,44,54,66,80,96,116,140,168,200,238,284,336,396,464,522,576} , ++ {6,6,6,6,6,6,8,10,12,14,16,20,24,28,32,38,46,52,60,68,58,54}, ++ {0,12,24,36,54,78,108,144,186,240,312,402,522,576}, ++ {4,4,4,6,8,10,12,14,18,24,30,40,18} }, ++ { {0,12,24,36,48,60,72,88,108,132,160,192,232,280,336,400,476,566,568,570,572,574,576}, ++ {12,12,12,12,12,12,16,20,24,28,32,40,48,56,64,76,90,2,2,2,2,2}, ++ {0, 24, 48, 72,108,156,216,288,372,480,486,492,498,576}, ++ {8,8,8,12,16,20,24,28,36,2,2,2,26} } , ++ }; ++ ++ static int mapbuf0[9][152]; ++ static int mapbuf1[9][156]; ++ static int mapbuf2[9][44]; ++ static int *map[9][3]; ++ static int *mapend[9][3]; ++ ++ static unsigned int n_slen2[512]; /* MPEG 2.0 slen for 'normal' mode */ ++ static unsigned int i_slen2[256]; /* MPEG 2.0 slen for intensity stereo */ ++ ++ static real tan1_1[16],tan2_1[16],tan1_2[16],tan2_2[16]; ++ static real pow1_1[2][16],pow2_1[2][16],pow1_2[2][16],pow2_2[2][16]; ++ ++ static unsigned int get1bit(void) ++ { ++ unsigned char rval; ++ rval = *wordpointer << bitindex; ++ ++ bitindex++; ++ wordpointer += (bitindex>>3); ++ bitindex &= 7; ++ ++ return rval>>7; ++ } ++ ++ ++ ++ ++ /* ++ * init tables for layer-3 ++ */ ++ void init_layer3(int down_sample_sblimit) ++ { ++ int i,j,k,l; ++ ++ for(i=-256;i<118+4;i++) ++ gainpow2[i+256] = pow((double)2.0,-0.25 * (double) (i+210) ); ++ ++ for(i=0;i<8207;i++) ++ ispow[i] = pow((double)i,(double)4.0/3.0); ++ ++ for (i=0;i<8;i++) ++ { ++ static double Ci[8]={-0.6,-0.535,-0.33,-0.185,-0.095,-0.041,-0.0142,-0.0037}; ++ double sq=sqrt(1.0+Ci[i]*Ci[i]); ++ aa_cs[i] = 1.0/sq; ++ aa_ca[i] = Ci[i]/sq; ++ } ++ ++ for(i=0;i<18;i++) ++ { ++ win[0][i] = win[1][i] = 0.5 * sin( M_PI / 72.0 * (double) (2*(i+0) +1) ) / cos ( M_PI * (double) (2*(i+0) +19) / 72.0 ); ++ win[0][i+18] = win[3][i+18] = 0.5 * sin( M_PI / 72.0 * (double) (2*(i+18)+1) ) / cos ( M_PI * (double) (2*(i+18)+19) / 72.0 ); ++ } ++ for(i=0;i<6;i++) ++ { ++ win[1][i+18] = 0.5 / cos ( M_PI * (double) (2*(i+18)+19) / 72.0 ); ++ win[3][i+12] = 0.5 / cos ( M_PI * (double) (2*(i+12)+19) / 72.0 ); ++ win[1][i+24] = 0.5 * sin( M_PI / 24.0 * (double) (2*i+13) ) / cos ( M_PI * (double) (2*(i+24)+19) / 72.0 ); ++ win[1][i+30] = win[3][i] = 0.0; ++ win[3][i+6 ] = 0.5 * sin( M_PI / 24.0 * (double) (2*i+1) ) / cos ( M_PI * (double) (2*(i+6 )+19) / 72.0 ); ++ } ++ ++ for(i=0;i<9;i++) ++ COS9[i] = cos( M_PI / 18.0 * (double) i); ++ ++ for(i=0;i<9;i++) ++ tfcos36[i] = 0.5 / cos ( M_PI * (double) (i*2+1) / 36.0 ); ++ for(i=0;i<3;i++) ++ tfcos12[i] = 0.5 / cos ( M_PI * (double) (i*2+1) / 12.0 ); ++ ++ COS6_1 = cos( M_PI / 6.0 * (double) 1); ++ COS6_2 = cos( M_PI / 6.0 * (double) 2); ++ ++ for(i=0;i<12;i++) ++ { ++ win[2][i] = 0.5 * sin( M_PI / 24.0 * (double) (2*i+1) ) / cos ( M_PI * (double) (2*i+7) / 24.0 ); ++ for(j=0;j<6;j++) ++ COS1[i][j] = cos( M_PI / 24.0 * (double) ((2*i+7)*(2*j+1)) ); ++ } ++ ++ for(j=0;j<4;j++) { ++ static int len[4] = { 36,36,12,36 }; ++ for(i=0;i<len[j];i+=2) ++ win1[j][i] = + win[j][i]; ++ for(i=1;i<len[j];i+=2) ++ win1[j][i] = - win[j][i]; ++ } ++ ++ for(i=0;i<16;i++) ++ { ++ double t = tan( (double) i * M_PI / 12.0 ); ++ tan1_1[i] = t / (1.0+t); ++ tan2_1[i] = 1.0 / (1.0 + t); ++ tan1_2[i] = M_SQRT2 * t / (1.0+t); ++ tan2_2[i] = M_SQRT2 / (1.0 + t); ++ ++ for(j=0;j<2;j++) { ++ double base = pow(2.0,-0.25*(j+1.0)); ++ double p1=1.0,p2=1.0; ++ if(i > 0) { ++ if( i & 1 ) ++ p1 = pow(base,(i+1.0)*0.5); ++ else ++ p2 = pow(base,i*0.5); ++ } ++ pow1_1[j][i] = p1; ++ pow2_1[j][i] = p2; ++ pow1_2[j][i] = M_SQRT2 * p1; ++ pow2_2[j][i] = M_SQRT2 * p2; ++ } ++ } ++ ++ for(j=0;j<9;j++) ++ { ++ struct bandInfoStruct *bi = &bandInfo[j]; ++ int *mp; ++ int cb,lwin; ++ short *bdf; ++ ++ mp = map[j][0] = mapbuf0[j]; ++ bdf = bi->longDiff; ++ for(i=0,cb = 0; cb < 8 ; cb++,i+=*bdf++) { ++ *mp++ = (*bdf) >> 1; ++ *mp++ = i; ++ *mp++ = 3; ++ *mp++ = cb; ++ } ++ bdf = bi->shortDiff+3; ++ for(cb=3;cb<13;cb++) { ++ int l = (*bdf++) >> 1; ++ for(lwin=0;lwin<3;lwin++) { ++ *mp++ = l; ++ *mp++ = i + lwin; ++ *mp++ = lwin; ++ *mp++ = cb; ++ } ++ i += 6*l; ++ } ++ mapend[j][0] = mp; ++ ++ mp = map[j][1] = mapbuf1[j]; ++ bdf = bi->shortDiff+0; ++ for(i=0,cb=0;cb<13;cb++) { ++ int l = (*bdf++) >> 1; ++ for(lwin=0;lwin<3;lwin++) { ++ *mp++ = l; ++ *mp++ = i + lwin; ++ *mp++ = lwin; ++ *mp++ = cb; ++ } ++ i += 6*l; ++ } ++ mapend[j][1] = mp; ++ ++ mp = map[j][2] = mapbuf2[j]; ++ bdf = bi->longDiff; ++ for(cb = 0; cb < 22 ; cb++) { ++ *mp++ = (*bdf++) >> 1; ++ *mp++ = cb; ++ } ++ mapend[j][2] = mp; ++ ++ } ++ ++ for(j=0;j<9;j++) { ++ for(i=0;i<23;i++) { ++ longLimit[j][i] = (bandInfo[j].longIdx[i] - 1 + 8) / 18 + 1; ++ if(longLimit[j][i] > (down_sample_sblimit) ) ++ longLimit[j][i] = down_sample_sblimit; ++ } ++ for(i=0;i<14;i++) { ++ shortLimit[j][i] = (bandInfo[j].shortIdx[i] - 1) / 18 + 1; ++ if(shortLimit[j][i] > (down_sample_sblimit) ) ++ shortLimit[j][i] = down_sample_sblimit; ++ } ++ } ++ ++ for(i=0;i<5;i++) { ++ for(j=0;j<6;j++) { ++ for(k=0;k<6;k++) { ++ int n = k + j * 6 + i * 36; ++ i_slen2[n] = i|(j<<3)|(k<<6)|(3<<12); ++ } ++ } ++ } ++ for(i=0;i<4;i++) { ++ for(j=0;j<4;j++) { ++ for(k=0;k<4;k++) { ++ int n = k + j * 4 + i * 16; ++ i_slen2[n+180] = i|(j<<3)|(k<<6)|(4<<12); ++ } ++ } ++ } ++ for(i=0;i<4;i++) { ++ for(j=0;j<3;j++) { ++ int n = j + i * 3; ++ i_slen2[n+244] = i|(j<<3) | (5<<12); ++ n_slen2[n+500] = i|(j<<3) | (2<<12) | (1<<15); ++ } ++ } ++ ++ for(i=0;i<5;i++) { ++ for(j=0;j<5;j++) { ++ for(k=0;k<4;k++) { ++ for(l=0;l<4;l++) { ++ int n = l + k * 4 + j * 16 + i * 80; ++ n_slen2[n] = i|(j<<3)|(k<<6)|(l<<9)|(0<<12); ++ } ++ } ++ } ++ } ++ for(i=0;i<5;i++) { ++ for(j=0;j<5;j++) { ++ for(k=0;k<4;k++) { ++ int n = k + j * 4 + i * 20; ++ n_slen2[n+400] = i|(j<<3)|(k<<6)|(1<<12); ++ } ++ } ++ } ++ } ++ ++ /* ++ * read additional side information ++ */ ++ #ifdef MPEG1 ++ static void III_get_side_info_1(struct III_sideinfo *si,int stereo, ++ int ms_stereo,long sfreq,int single) ++ { ++ int ch, gr; ++ int powdiff = (single == 3) ? 4 : 0; ++ ++ si->main_data_begin = getbits(9); ++ if (stereo == 1) ++ si->private_bits = getbits_fast(5); ++ else ++ si->private_bits = getbits_fast(3); ++ ++ for (ch=0; ch<stereo; ch++) { ++ si->ch[ch].gr[0].scfsi = -1; ++ si->ch[ch].gr[1].scfsi = getbits_fast(4); ++ } ++ ++ for (gr=0; gr<2; gr++) ++ { ++ for (ch=0; ch<stereo; ch++) ++ { ++ register struct gr_info_s *gr_info = &(si->ch[ch].gr[gr]); ++ ++ gr_info->part2_3_length = getbits(12); ++ gr_info->big_values = getbits_fast(9); ++ if(gr_info->big_values > 288) { ++ fprintf(stderr,"big_values too large!\n"); ++ gr_info->big_values = 288; ++ } ++ { ++ unsigned int qss = getbits_fast(8); ++ gr_info->pow2gain = gainpow2+256 - qss + powdiff; ++ #ifdef HAVEGTK ++ if (gtkflag) ++ pinfo->qss[gr][ch]=qss; ++ #endif ++ } ++ if(ms_stereo) ++ gr_info->pow2gain += 2; ++ gr_info->scalefac_compress = getbits_fast(4); ++ /* window-switching flag == 1 for block_Type != 0 .. and block-type == 0 -> win-sw-flag = 0 */ ++ if(get1bit()) ++ { ++ int i; ++ gr_info->block_type = getbits_fast(2); ++ gr_info->mixed_block_flag = get1bit(); ++ gr_info->table_select[0] = getbits_fast(5); ++ gr_info->table_select[1] = getbits_fast(5); ++ ++ ++ /* ++ * table_select[2] not needed, because there is no region2, ++ * but to satisfy some verifications tools we set it either. ++ */ ++ gr_info->table_select[2] = 0; ++ for(i=0;i<3;i++) { ++ unsigned int sbg = (getbits_fast(3)<<3); ++ gr_info->full_gain[i] = gr_info->pow2gain + sbg; ++ #ifdef HAVEGTK ++ if (gtkflag) ++ pinfo->sub_gain[gr][ch][i]=sbg/8; ++ #endif ++ } ++ ++ ++ if(gr_info->block_type == 0) { ++ fprintf(stderr,"Blocktype == 0 and window-switching == 1 not allowed.\n"); ++ exit(1); ++ } ++ /* region_count/start parameters are implicit in this case. */ ++ gr_info->region1start = 36>>1; ++ gr_info->region2start = 576>>1; ++ } ++ else ++ { ++ int i,r0c,r1c; ++ for (i=0; i<3; i++) ++ gr_info->table_select[i] = getbits_fast(5); ++ r0c = getbits_fast(4); ++ r1c = getbits_fast(3); ++ gr_info->region1start = bandInfo[sfreq].longIdx[r0c+1] >> 1 ; ++ gr_info->region2start = bandInfo[sfreq].longIdx[r0c+1+r1c+1] >> 1; ++ gr_info->block_type = 0; ++ gr_info->mixed_block_flag = 0; ++ } ++ gr_info->preflag = get1bit(); ++ gr_info->scalefac_scale = get1bit(); ++ gr_info->count1table_select = get1bit(); ++ #ifdef HAVEGTK ++ if (gtkflag) ++ pinfo->scalefac_scale[gr][ch]=gr_info->scalefac_scale; ++ #endif ++ } ++ } ++ } ++ #endif ++ ++ /* ++ * Side Info for MPEG 2.0 / LSF ++ */ ++ static void III_get_side_info_2(struct III_sideinfo *si,int stereo, ++ int ms_stereo,long sfreq,int single) ++ { ++ int ch; ++ int powdiff = (single == 3) ? 4 : 0; ++ ++ si->main_data_begin = getbits(8); ++ if (stereo == 1) ++ si->private_bits = get1bit(); ++ else ++ si->private_bits = getbits_fast(2); ++ ++ for (ch=0; ch<stereo; ch++) ++ { ++ register struct gr_info_s *gr_info = &(si->ch[ch].gr[0]); ++ ++ gr_info->part2_3_length = getbits(12); ++ gr_info->big_values = getbits_fast(9); ++ if(gr_info->big_values > 288) { ++ fprintf(stderr,"big_values too large!\n"); ++ gr_info->big_values = 288; ++ } ++ gr_info->pow2gain = gainpow2+256 - getbits_fast(8) + powdiff; ++ if(ms_stereo) ++ gr_info->pow2gain += 2; ++ gr_info->scalefac_compress = getbits(9); ++ /* window-switching flag == 1 for block_Type != 0 .. and block-type == 0 -> win-sw-flag = 0 */ ++ if(get1bit()) ++ { ++ int i; ++ gr_info->block_type = getbits_fast(2); ++ gr_info->mixed_block_flag = get1bit(); ++ gr_info->table_select[0] = getbits_fast(5); ++ gr_info->table_select[1] = getbits_fast(5); ++ /* ++ * table_select[2] not needed, because there is no region2, ++ * but to satisfy some verifications tools we set it either. ++ */ ++ gr_info->table_select[2] = 0; ++ for(i=0;i<3;i++) { ++ gr_info->full_gain[i] = gr_info->pow2gain + (getbits_fast(3)<<3); ++ } ++ ++ if(gr_info->block_type == 0) { ++ fprintf(stderr,"Blocktype == 0 and window-switching == 1 not allowed.\n"); ++ exit(1); ++ } ++ /* region_count/start parameters are implicit in this case. */ ++ /* check this again! */ ++ if(gr_info->block_type == 2) ++ gr_info->region1start = 36>>1; ++ else if(sfreq == 8) ++ /* check this for 2.5 and sfreq=8 */ ++ gr_info->region1start = 108>>1; ++ else ++ gr_info->region1start = 54>>1; ++ gr_info->region2start = 576>>1; ++ } ++ else ++ { ++ int i,r0c,r1c; ++ for (i=0; i<3; i++) ++ gr_info->table_select[i] = getbits_fast(5); ++ r0c = getbits_fast(4); ++ r1c = getbits_fast(3); ++ gr_info->region1start = bandInfo[sfreq].longIdx[r0c+1] >> 1 ; ++ gr_info->region2start = bandInfo[sfreq].longIdx[r0c+1+r1c+1] >> 1; ++ gr_info->block_type = 0; ++ gr_info->mixed_block_flag = 0; ++ } ++ gr_info->scalefac_scale = get1bit(); ++ gr_info->count1table_select = get1bit(); ++ #ifdef HAVEGTK ++ if (gtkflag) ++ pinfo->scalefac_scale[0][ch]=gr_info->scalefac_scale; ++ #endif ++ } ++ } ++ ++ /* ++ * read scalefactors ++ */ ++ #ifdef MPEG1 ++ static int III_get_scale_factors_1(int *scf,struct gr_info_s *gr_info) ++ { ++ static unsigned char slen[2][16] = { ++ {0, 0, 0, 0, 3, 1, 1, 1, 2, 2, 2, 3, 3, 3, 4, 4}, ++ {0, 1, 2, 3, 0, 1, 2, 3, 1, 2, 3, 1, 2, 3, 2, 3} ++ }; ++ int numbits; ++ int num0 = slen[0][gr_info->scalefac_compress]; ++ int num1 = slen[1][gr_info->scalefac_compress]; ++ ++ if (gr_info->block_type == 2) ++ { ++ int i=18; ++ numbits = (num0 + num1) * 18; ++ ++ if (gr_info->mixed_block_flag) { ++ for (i=8;i;i--) ++ *scf++ = getbits_fast(num0); ++ i = 9; ++ numbits -= num0; /* num0 * 17 + num1 * 18 */ ++ } ++ ++ for (;i;i--) ++ *scf++ = getbits_fast(num0); ++ for (i = 18; i; i--) ++ *scf++ = getbits_fast(num1); ++ *scf++ = 0; *scf++ = 0; *scf++ = 0; /* short[13][0..2] = 0 */ ++ } ++ else ++ { ++ int i; ++ int scfsi = gr_info->scfsi; ++ ++ if(scfsi < 0) { /* scfsi < 0 => granule == 0 */ ++ for(i=11;i;i--) ++ *scf++ = getbits_fast(num0); ++ for(i=10;i;i--) ++ *scf++ = getbits_fast(num1); ++ numbits = (num0 + num1) * 10 + num0; ++ } ++ else { ++ numbits = 0; ++ if(!(scfsi & 0x8)) { ++ for (i=6;i;i--) ++ *scf++ = getbits_fast(num0); ++ numbits += num0 * 6; ++ } ++ else { ++ *scf++ = 0; *scf++ = 0; *scf++ = 0; /* set to ZERO necessary? */ ++ *scf++ = 0; *scf++ = 0; *scf++ = 0; ++ } ++ ++ if(!(scfsi & 0x4)) { ++ for (i=5;i;i--) ++ *scf++ = getbits_fast(num0); ++ numbits += num0 * 5; ++ } ++ else { ++ *scf++ = 0; *scf++ = 0; *scf++ = 0; /* set to ZERO necessary? */ ++ *scf++ = 0; *scf++ = 0; ++ } ++ ++ if(!(scfsi & 0x2)) { ++ for(i=5;i;i--) ++ *scf++ = getbits_fast(num1); ++ numbits += num1 * 5; ++ } ++ else { ++ *scf++ = 0; *scf++ = 0; *scf++ = 0; /* set to ZERO necessary? */ ++ *scf++ = 0; *scf++ = 0; ++ } ++ ++ if(!(scfsi & 0x1)) { ++ for (i=5;i;i--) ++ *scf++ = getbits_fast(num1); ++ numbits += num1 * 5; ++ } ++ else { ++ *scf++ = 0; *scf++ = 0; *scf++ = 0; /* set to ZERO necessary? */ ++ *scf++ = 0; *scf++ = 0; ++ } ++ } ++ ++ *scf++ = 0; /* no l[21] in original sources */ ++ } ++ return numbits; ++ } ++ #endif ++ ++ static int III_get_scale_factors_2(int *scf,struct gr_info_s *gr_info,int i_stereo) ++ { ++ unsigned char *pnt; ++ int i,j; ++ unsigned int slen; ++ int n = 0; ++ int numbits = 0; ++ ++ static unsigned char stab[3][6][4] = { ++ { { 6, 5, 5,5 } , { 6, 5, 7,3 } , { 11,10,0,0} , ++ { 7, 7, 7,0 } , { 6, 6, 6,3 } , { 8, 8,5,0} } , ++ { { 9, 9, 9,9 } , { 9, 9,12,6 } , { 18,18,0,0} , ++ {12,12,12,0 } , {12, 9, 9,6 } , { 15,12,9,0} } , ++ { { 6, 9, 9,9 } , { 6, 9,12,6 } , { 15,18,0,0} , ++ { 6,15,12,0 } , { 6,12, 9,6 } , { 6,18,9,0} } }; ++ ++ if(i_stereo) /* i_stereo AND second channel -> do_layer3() checks this */ ++ slen = i_slen2[gr_info->scalefac_compress>>1]; ++ else ++ slen = n_slen2[gr_info->scalefac_compress]; ++ ++ gr_info->preflag = (slen>>15) & 0x1; ++ ++ n = 0; ++ if( gr_info->block_type == 2 ) { ++ n++; ++ if(gr_info->mixed_block_flag) ++ n++; ++ } ++ ++ pnt = stab[n][(slen>>12)&0x7]; ++ ++ for(i=0;i<4;i++) { ++ int num = slen & 0x7; ++ slen >>= 3; ++ if(num) { ++ for(j=0;j<(int)(pnt[i]);j++) ++ *scf++ = getbits_fast(num); ++ numbits += pnt[i] * num; ++ } ++ else { ++ for(j=0;j<(int)(pnt[i]);j++) ++ *scf++ = 0; ++ } ++ } ++ ++ n = (n << 1) + 1; ++ for(i=0;i<n;i++) ++ *scf++ = 0; ++ ++ return numbits; ++ } ++ ++ static int pretab1[22] = {0,0,0,0,0,0,0,0,0,0,0,1,1,1,1,2,2,3,3,3,2,0}; ++ static int pretab2[22] = {0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0}; ++ ++ /* ++ * don't forget to apply the same changes to III_dequantize_sample_ms() !!! ++ */ ++ static int III_dequantize_sample(real xr[SBLIMIT][SSLIMIT],int *scf, ++ struct gr_info_s *gr_info,int sfreq,int part2bits) ++ { ++ int shift = 1 + gr_info->scalefac_scale; ++ real *xrpnt = (real *) xr; ++ int l[3],l3; ++ int part2remain = gr_info->part2_3_length - part2bits; ++ int *me; ++ ++ { ++ int bv = gr_info->big_values; ++ int region1 = gr_info->region1start; ++ int region2 = gr_info->region2start; ++ ++ l3 = ((576>>1)-bv)>>1; ++ /* ++ * we may lose the 'odd' bit here !! ++ * check this later again ++ */ ++ if(bv <= region1) { ++ l[0] = bv; l[1] = 0; l[2] = 0; ++ } ++ else { ++ l[0] = region1; ++ if(bv <= region2) { ++ l[1] = bv - l[0]; l[2] = 0; ++ } ++ else { ++ l[1] = region2 - l[0]; l[2] = bv - region2; ++ } ++ } ++ } ++ ++ if(gr_info->block_type == 2) { ++ /* ++ * decoding with short or mixed mode BandIndex table ++ */ ++ int i,max[4]; ++ int step=0,lwin=0,cb=0; ++ register real v = 0.0; ++ register int *m,mc; ++ ++ if(gr_info->mixed_block_flag) { ++ max[3] = -1; ++ max[0] = max[1] = max[2] = 2; ++ m = map[sfreq][0]; ++ me = mapend[sfreq][0]; ++ } ++ else { ++ max[0] = max[1] = max[2] = max[3] = -1; ++ /* max[3] not really needed in this case */ ++ m = map[sfreq][1]; ++ me = mapend[sfreq][1]; ++ } ++ ++ mc = 0; ++ for(i=0;i<2;i++) { ++ int lp = l[i]; ++ struct newhuff *h = ht+gr_info->table_select[i]; ++ for(;lp;lp--,mc--) { ++ register int x,y; ++ if( (!mc) ) { ++ mc = *m++; ++ xrpnt = ((real *) xr) + (*m++); ++ lwin = *m++; ++ cb = *m++; ++ if(lwin == 3) { ++ v = gr_info->pow2gain[(*scf++) << shift]; ++ step = 1; ++ } ++ else { ++ v = gr_info->full_gain[lwin][(*scf++) << shift]; ++ step = 3; ++ } ++ } ++ { ++ register short *val = h->table; ++ while((y=*val++)<0) { ++ if (get1bit()) ++ val -= y; ++ part2remain--; ++ } ++ x = y >> 4; ++ y &= 0xf; ++ } ++ if(x == 15) { ++ max[lwin] = cb; ++ part2remain -= h->linbits+1; ++ x += getbits(h->linbits); ++ if(get1bit()) ++ *xrpnt = -ispow[x] * v; ++ else ++ *xrpnt = ispow[x] * v; ++ } ++ else if(x) { ++ max[lwin] = cb; ++ if(get1bit()) ++ *xrpnt = -ispow[x] * v; ++ else ++ *xrpnt = ispow[x] * v; ++ part2remain--; ++ } ++ else ++ *xrpnt = 0.0; ++ xrpnt += step; ++ if(y == 15) { ++ max[lwin] = cb; ++ part2remain -= h->linbits+1; ++ y += getbits(h->linbits); ++ if(get1bit()) ++ *xrpnt = -ispow[y] * v; ++ else ++ *xrpnt = ispow[y] * v; ++ } ++ else if(y) { ++ max[lwin] = cb; ++ if(get1bit()) ++ *xrpnt = -ispow[y] * v; ++ else ++ *xrpnt = ispow[y] * v; ++ part2remain--; ++ } ++ else ++ *xrpnt = 0.0; ++ xrpnt += step; ++ } ++ } ++ for(;l3 && (part2remain > 0);l3--) { ++ struct newhuff *h = htc+gr_info->count1table_select; ++ register short *val = h->table,a; ++ ++ while((a=*val++)<0) { ++ part2remain--; ++ if(part2remain < 0) { ++ part2remain++; ++ a = 0; ++ break; ++ } ++ if (get1bit()) ++ val -= a; ++ } ++ for(i=0;i<4;i++) { ++ if(!(i & 1)) { ++ if(!mc) { ++ mc = *m++; ++ xrpnt = ((real *) xr) + (*m++); ++ lwin = *m++; ++ cb = *m++; ++ if(lwin == 3) { ++ v = gr_info->pow2gain[(*scf++) << shift]; ++ step = 1; ++ } ++ else { ++ v = gr_info->full_gain[lwin][(*scf++) << shift]; ++ step = 3; ++ } ++ } ++ mc--; ++ } ++ if( (a & (0x8>>i)) ) { ++ max[lwin] = cb; ++ part2remain--; ++ if(part2remain < 0) { ++ part2remain++; ++ break; ++ } ++ if(get1bit()) ++ *xrpnt = -v; ++ else ++ *xrpnt = v; ++ } ++ else ++ *xrpnt = 0.0; ++ xrpnt += step; ++ } ++ } ++ ++ while( m < me ) { ++ if(!mc) { ++ mc = *m++; ++ xrpnt = ((real *) xr) + *m++; ++ if( (*m++) == 3) ++ step = 1; ++ else ++ step = 3; ++ m++; /* cb */ ++ } ++ mc--; ++ *xrpnt = 0.0; ++ xrpnt += step; ++ *xrpnt = 0.0; ++ xrpnt += step; ++ /* we could add a little opt. here: ++ * if we finished a band for window 3 or a long band ++ * further bands could copied in a simple loop without a ++ * special 'map' decoding ++ */ ++ } ++ ++ gr_info->maxband[0] = max[0]+1; ++ gr_info->maxband[1] = max[1]+1; ++ gr_info->maxband[2] = max[2]+1; ++ gr_info->maxbandl = max[3]+1; ++ ++ { ++ int rmax = max[0] > max[1] ? max[0] : max[1]; ++ rmax = (rmax > max[2] ? rmax : max[2]) + 1; ++ gr_info->maxb = rmax ? shortLimit[sfreq][rmax] : longLimit[sfreq][max[3]+1]; ++ } ++ ++ } ++ else { ++ /* ++ * decoding with 'long' BandIndex table (block_type != 2) ++ */ ++ int *pretab = gr_info->preflag ? pretab1 : pretab2; ++ int i,max = -1; ++ int cb = 0; ++ register int *m = map[sfreq][2]; ++ register real v = 0.0; ++ register int mc = 0; ++ #if 0 ++ me = mapend[sfreq][2]; ++ #endif ++ ++ /* ++ * long hash table values ++ */ ++ for(i=0;i<3;i++) { ++ int lp = l[i]; ++ struct newhuff *h = ht+gr_info->table_select[i]; ++ ++ for(;lp;lp--,mc--) { ++ int x,y; ++ ++ if(!mc) { ++ mc = *m++; ++ v = gr_info->pow2gain[((*scf++) + (*pretab++)) << shift]; ++ cb = *m++; ++ } ++ { ++ register short *val = h->table; ++ while((y=*val++)<0) { ++ if (get1bit()) ++ val -= y; ++ part2remain--; ++ } ++ x = y >> 4; ++ y &= 0xf; ++ } ++ if (x == 15) { ++ max = cb; ++ part2remain -= h->linbits+1; ++ x += getbits(h->linbits); ++ if(get1bit()) ++ *xrpnt++ = -ispow[x] * v; ++ else ++ *xrpnt++ = ispow[x] * v; ++ } ++ else if(x) { ++ max = cb; ++ if(get1bit()) ++ *xrpnt++ = -ispow[x] * v; ++ else ++ *xrpnt++ = ispow[x] * v; ++ part2remain--; ++ } ++ else ++ *xrpnt++ = 0.0; ++ ++ if (y == 15) { ++ max = cb; ++ part2remain -= h->linbits+1; ++ y += getbits(h->linbits); ++ if(get1bit()) ++ *xrpnt++ = -ispow[y] * v; ++ else ++ *xrpnt++ = ispow[y] * v; ++ } ++ else if(y) { ++ max = cb; ++ if(get1bit()) ++ *xrpnt++ = -ispow[y] * v; ++ else ++ *xrpnt++ = ispow[y] * v; ++ part2remain--; ++ } ++ else ++ *xrpnt++ = 0.0; ++ } ++ } ++ ++ /* ++ * short (count1table) values ++ */ ++ for(;l3 && (part2remain > 0);l3--) { ++ struct newhuff *h = htc+gr_info->count1table_select; ++ register short *val = h->table,a; ++ ++ while((a=*val++)<0) { ++ part2remain--; ++ if(part2remain < 0) { ++ part2remain++; ++ a = 0; ++ break; ++ } ++ if (get1bit()) ++ val -= a; ++ } ++ for(i=0;i<4;i++) { ++ if(!(i & 1)) { ++ if(!mc) { ++ mc = *m++; ++ cb = *m++; ++ v = gr_info->pow2gain[((*scf++) + (*pretab++)) << shift]; ++ } ++ mc--; ++ } ++ if ( (a & (0x8>>i)) ) { ++ max = cb; ++ part2remain--; ++ if(part2remain < 0) { ++ part2remain++; ++ break; ++ } ++ if(get1bit()) ++ *xrpnt++ = -v; ++ else ++ *xrpnt++ = v; ++ } ++ else ++ *xrpnt++ = 0.0; ++ } ++ } ++ ++ /* ++ * zero part ++ */ ++ for(i=(&xr[SBLIMIT][0]-xrpnt)>>1;i;i--) { ++ *xrpnt++ = 0.0; ++ *xrpnt++ = 0.0; ++ } ++ ++ gr_info->maxbandl = max+1; ++ gr_info->maxb = longLimit[sfreq][gr_info->maxbandl]; ++ } ++ ++ while( part2remain > 16 ) { ++ getbits(16); /* Dismiss stuffing Bits */ ++ part2remain -= 16; ++ } ++ if(part2remain > 0) ++ getbits(part2remain); ++ else if(part2remain < 0) { ++ fprintf(stderr,"mpg123: Can't rewind stream by %d bits!\n",-part2remain); ++ return 1; /* -> error */ ++ } ++ return 0; ++ } ++ ++ ++ /* ++ * III_stereo: calculate real channel values for Joint-I-Stereo-mode ++ */ ++ static void III_i_stereo(real xr_buf[2][SBLIMIT][SSLIMIT],int *scalefac, ++ struct gr_info_s *gr_info,int sfreq,int ms_stereo,int lsf) ++ { ++ real (*xr)[SBLIMIT*SSLIMIT] = (real (*)[SBLIMIT*SSLIMIT] ) xr_buf; ++ struct bandInfoStruct *bi = &bandInfo[sfreq]; ++ real *tab1,*tab2; ++ ++ if(lsf) { ++ int p = gr_info->scalefac_compress & 0x1; ++ if(ms_stereo) { ++ tab1 = pow1_2[p]; tab2 = pow2_2[p]; ++ } ++ else { ++ tab1 = pow1_1[p]; tab2 = pow2_1[p]; ++ } ++ } ++ else { ++ if(ms_stereo) { ++ tab1 = tan1_2; tab2 = tan2_2; ++ } ++ else { ++ tab1 = tan1_1; tab2 = tan2_1; ++ } ++ } ++ ++ if (gr_info->block_type == 2) ++ { ++ int lwin,do_l = 0; ++ if( gr_info->mixed_block_flag ) ++ do_l = 1; ++ ++ for (lwin=0;lwin<3;lwin++) /* process each window */ ++ { ++ /* get first band with zero values */ ++ int is_p,sb,idx,sfb = gr_info->maxband[lwin]; /* sfb is minimal 3 for mixed mode */ ++ if(sfb > 3) ++ do_l = 0; ++ ++ for(;sfb<12;sfb++) ++ { ++ is_p = scalefac[sfb*3+lwin-gr_info->mixed_block_flag]; /* scale: 0-15 */ ++ if(is_p != 7) { ++ real t1,t2; ++ sb = bi->shortDiff[sfb]; ++ idx = bi->shortIdx[sfb] + lwin; ++ t1 = tab1[is_p]; t2 = tab2[is_p]; ++ for (; sb > 0; sb--,idx+=3) ++ { ++ real v = xr[0][idx]; ++ xr[0][idx] = v * t1; ++ xr[1][idx] = v * t2; ++ } ++ } ++ } ++ ++ #if 1 ++ /* in the original: copy 10 to 11 , here: copy 11 to 12 ++ maybe still wrong??? (copy 12 to 13?) */ ++ is_p = scalefac[11*3+lwin-gr_info->mixed_block_flag]; /* scale: 0-15 */ ++ sb = bi->shortDiff[12]; ++ idx = bi->shortIdx[12] + lwin; ++ #else ++ is_p = scalefac[10*3+lwin-gr_info->mixed_block_flag]; /* scale: 0-15 */ ++ sb = bi->shortDiff[11]; ++ idx = bi->shortIdx[11] + lwin; ++ #endif ++ if(is_p != 7) ++ { ++ real t1,t2; ++ t1 = tab1[is_p]; t2 = tab2[is_p]; ++ for ( ; sb > 0; sb--,idx+=3 ) ++ { ++ real v = xr[0][idx]; ++ xr[0][idx] = v * t1; ++ xr[1][idx] = v * t2; ++ } ++ } ++ } /* end for(lwin; .. ; . ) */ ++ ++ if (do_l) ++ { ++ /* also check l-part, if ALL bands in the three windows are 'empty' ++ * and mode = mixed_mode ++ */ ++ int sfb = gr_info->maxbandl; ++ int idx = bi->longIdx[sfb]; ++ ++ for ( ; sfb<8; sfb++ ) ++ { ++ int sb = bi->longDiff[sfb]; ++ int is_p = scalefac[sfb]; /* scale: 0-15 */ ++ if(is_p != 7) { ++ real t1,t2; ++ t1 = tab1[is_p]; t2 = tab2[is_p]; ++ for ( ; sb > 0; sb--,idx++) ++ { ++ real v = xr[0][idx]; ++ xr[0][idx] = v * t1; ++ xr[1][idx] = v * t2; ++ } ++ } ++ else ++ idx += sb; ++ } ++ } ++ } ++ else /* ((gr_info->block_type != 2)) */ ++ { ++ int sfb = gr_info->maxbandl; ++ int is_p,idx = bi->longIdx[sfb]; ++ for ( ; sfb<21; sfb++) ++ { ++ int sb = bi->longDiff[sfb]; ++ is_p = scalefac[sfb]; /* scale: 0-15 */ ++ if(is_p != 7) { ++ real t1,t2; ++ t1 = tab1[is_p]; t2 = tab2[is_p]; ++ for ( ; sb > 0; sb--,idx++) ++ { ++ real v = xr[0][idx]; ++ xr[0][idx] = v * t1; ++ xr[1][idx] = v * t2; ++ } ++ } ++ else ++ idx += sb; ++ } ++ ++ is_p = scalefac[20]; /* copy l-band 20 to l-band 21 */ ++ if(is_p != 7) ++ { ++ int sb; ++ real t1 = tab1[is_p],t2 = tab2[is_p]; ++ ++ for ( sb = bi->longDiff[21]; sb > 0; sb--,idx++ ) ++ { ++ real v = xr[0][idx]; ++ xr[0][idx] = v * t1; ++ xr[1][idx] = v * t2; ++ } ++ } ++ } /* ... */ ++ } ++ ++ static void III_antialias(real xr[SBLIMIT][SSLIMIT],struct gr_info_s *gr_info) ++ { ++ int sblim; ++ ++ if(gr_info->block_type == 2) ++ { ++ if(!gr_info->mixed_block_flag) ++ return; ++ sblim = 1; ++ } ++ else { ++ sblim = gr_info->maxb-1; ++ } ++ ++ /* 31 alias-reduction operations between each pair of sub-bands */ ++ /* with 8 butterflies between each pair */ ++ ++ { ++ int sb; ++ real *xr1=(real *) xr[1]; ++ ++ for(sb=sblim;sb;sb--,xr1+=10) ++ { ++ int ss; ++ real *cs=aa_cs,*ca=aa_ca; ++ real *xr2 = xr1; ++ ++ for(ss=7;ss>=0;ss--) ++ { /* upper and lower butterfly inputs */ ++ register real bu = *--xr2,bd = *xr1; ++ *xr2 = (bu * (*cs) ) - (bd * (*ca) ); ++ *xr1++ = (bd * (*cs++) ) + (bu * (*ca++) ); ++ } ++ } ++ } ++ } ++ ++ /* ++ DCT insipired by Jeff Tsay's DCT from the maplay package ++ this is an optimized version with manual unroll. ++ ++ References: ++ [1] S. Winograd: "On Computing the Discrete Fourier Transform", ++ Mathematics of Computation, Volume 32, Number 141, January 1978, ++ Pages 175-199 ++ */ ++ ++ static void dct36(real *inbuf,real *o1,real *o2,real *wintab,real *tsbuf) ++ { ++ { ++ register real *in = inbuf; ++ ++ in[17]+=in[16]; in[16]+=in[15]; in[15]+=in[14]; ++ in[14]+=in[13]; in[13]+=in[12]; in[12]+=in[11]; ++ in[11]+=in[10]; in[10]+=in[9]; in[9] +=in[8]; ++ in[8] +=in[7]; in[7] +=in[6]; in[6] +=in[5]; ++ in[5] +=in[4]; in[4] +=in[3]; in[3] +=in[2]; ++ in[2] +=in[1]; in[1] +=in[0]; ++ ++ in[17]+=in[15]; in[15]+=in[13]; in[13]+=in[11]; in[11]+=in[9]; ++ in[9] +=in[7]; in[7] +=in[5]; in[5] +=in[3]; in[3] +=in[1]; ++ ++ ++ { ++ ++ #define MACRO0(v) { \ ++ real tmp; \ ++ out2[9+(v)] = (tmp = sum0 + sum1) * w[27+(v)]; \ ++ out2[8-(v)] = tmp * w[26-(v)]; } \ ++ sum0 -= sum1; \ ++ ts[SBLIMIT*(8-(v))] = out1[8-(v)] + sum0 * w[8-(v)]; \ ++ ts[SBLIMIT*(9+(v))] = out1[9+(v)] + sum0 * w[9+(v)]; ++ #define MACRO1(v) { \ ++ real sum0,sum1; \ ++ sum0 = tmp1a + tmp2a; \ ++ sum1 = (tmp1b + tmp2b) * tfcos36[(v)]; \ ++ MACRO0(v); } ++ #define MACRO2(v) { \ ++ real sum0,sum1; \ ++ sum0 = tmp2a - tmp1a; \ ++ sum1 = (tmp2b - tmp1b) * tfcos36[(v)]; \ ++ MACRO0(v); } ++ ++ register const real *c = COS9; ++ register real *out2 = o2; ++ register real *w = wintab; ++ register real *out1 = o1; ++ register real *ts = tsbuf; ++ ++ real ta33,ta66,tb33,tb66; ++ ++ ta33 = in[2*3+0] * c[3]; ++ ta66 = in[2*6+0] * c[6]; ++ tb33 = in[2*3+1] * c[3]; ++ tb66 = in[2*6+1] * c[6]; ++ ++ { ++ real tmp1a,tmp2a,tmp1b,tmp2b; ++ tmp1a = in[2*1+0] * c[1] + ta33 + in[2*5+0] * c[5] + in[2*7+0] * c[7]; ++ tmp1b = in[2*1+1] * c[1] + tb33 + in[2*5+1] * c[5] + in[2*7+1] * c[7]; ++ tmp2a = in[2*0+0] + in[2*2+0] * c[2] + in[2*4+0] * c[4] + ta66 + in[2*8+0] * c[8]; ++ tmp2b = in[2*0+1] + in[2*2+1] * c[2] + in[2*4+1] * c[4] + tb66 + in[2*8+1] * c[8]; ++ ++ MACRO1(0); ++ MACRO2(8); ++ } ++ ++ { ++ real tmp1a,tmp2a,tmp1b,tmp2b; ++ tmp1a = ( in[2*1+0] - in[2*5+0] - in[2*7+0] ) * c[3]; ++ tmp1b = ( in[2*1+1] - in[2*5+1] - in[2*7+1] ) * c[3]; ++ tmp2a = ( in[2*2+0] - in[2*4+0] - in[2*8+0] ) * c[6] - in[2*6+0] + in[2*0+0]; ++ tmp2b = ( in[2*2+1] - in[2*4+1] - in[2*8+1] ) * c[6] - in[2*6+1] + in[2*0+1]; ++ ++ MACRO1(1); ++ MACRO2(7); ++ } ++ ++ { ++ real tmp1a,tmp2a,tmp1b,tmp2b; ++ tmp1a = in[2*1+0] * c[5] - ta33 - in[2*5+0] * c[7] + in[2*7+0] * c[1]; ++ tmp1b = in[2*1+1] * c[5] - tb33 - in[2*5+1] * c[7] + in[2*7+1] * c[1]; ++ tmp2a = in[2*0+0] - in[2*2+0] * c[8] - in[2*4+0] * c[2] + ta66 + in[2*8+0] * c[4]; ++ tmp2b = in[2*0+1] - in[2*2+1] * c[8] - in[2*4+1] * c[2] + tb66 + in[2*8+1] * c[4]; ++ ++ MACRO1(2); ++ MACRO2(6); ++ } ++ ++ { ++ real tmp1a,tmp2a,tmp1b,tmp2b; ++ tmp1a = in[2*1+0] * c[7] - ta33 + in[2*5+0] * c[1] - in[2*7+0] * c[5]; ++ tmp1b = in[2*1+1] * c[7] - tb33 + in[2*5+1] * c[1] - in[2*7+1] * c[5]; ++ tmp2a = in[2*0+0] - in[2*2+0] * c[4] + in[2*4+0] * c[8] + ta66 - in[2*8+0] * c[2]; ++ tmp2b = in[2*0+1] - in[2*2+1] * c[4] + in[2*4+1] * c[8] + tb66 - in[2*8+1] * c[2]; ++ ++ MACRO1(3); ++ MACRO2(5); ++ } ++ ++ { ++ real sum0,sum1; ++ sum0 = in[2*0+0] - in[2*2+0] + in[2*4+0] - in[2*6+0] + in[2*8+0]; ++ sum1 = (in[2*0+1] - in[2*2+1] + in[2*4+1] - in[2*6+1] + in[2*8+1] ) * tfcos36[4]; ++ MACRO0(4); ++ } ++ } ++ ++ } ++ } ++ ++ /* ++ * new DCT12 ++ */ ++ static void dct12(real *in,real *rawout1,real *rawout2,register real *wi,register real *ts) ++ { ++ #define DCT12_PART1 \ ++ in5 = in[5*3]; \ ++ in5 += (in4 = in[4*3]); \ ++ in4 += (in3 = in[3*3]); \ ++ in3 += (in2 = in[2*3]); \ ++ in2 += (in1 = in[1*3]); \ ++ in1 += (in0 = in[0*3]); \ ++ \ ++ in5 += in3; in3 += in1; \ ++ \ ++ in2 *= COS6_1; \ ++ in3 *= COS6_1; \ ++ ++ #define DCT12_PART2 \ ++ in0 += in4 * COS6_2; \ ++ \ ++ in4 = in0 + in2; \ ++ in0 -= in2; \ ++ \ ++ in1 += in5 * COS6_2; \ ++ \ ++ in5 = (in1 + in3) * tfcos12[0]; \ ++ in1 = (in1 - in3) * tfcos12[2]; \ ++ \ ++ in3 = in4 + in5; \ ++ in4 -= in5; \ ++ \ ++ in2 = in0 + in1; \ ++ in0 -= in1; ++ ++ ++ { ++ real in0,in1,in2,in3,in4,in5; ++ register real *out1 = rawout1; ++ ts[SBLIMIT*0] = out1[0]; ts[SBLIMIT*1] = out1[1]; ts[SBLIMIT*2] = out1[2]; ++ ts[SBLIMIT*3] = out1[3]; ts[SBLIMIT*4] = out1[4]; ts[SBLIMIT*5] = out1[5]; ++ ++ DCT12_PART1 ++ ++ { ++ real tmp0,tmp1 = (in0 - in4); ++ { ++ real tmp2 = (in1 - in5) * tfcos12[1]; ++ tmp0 = tmp1 + tmp2; ++ tmp1 -= tmp2; ++ } ++ ts[(17-1)*SBLIMIT] = out1[17-1] + tmp0 * wi[11-1]; ++ ts[(12+1)*SBLIMIT] = out1[12+1] + tmp0 * wi[6+1]; ++ ts[(6 +1)*SBLIMIT] = out1[6 +1] + tmp1 * wi[1]; ++ ts[(11-1)*SBLIMIT] = out1[11-1] + tmp1 * wi[5-1]; ++ } ++ ++ DCT12_PART2 ++ ++ ts[(17-0)*SBLIMIT] = out1[17-0] + in2 * wi[11-0]; ++ ts[(12+0)*SBLIMIT] = out1[12+0] + in2 * wi[6+0]; ++ ts[(12+2)*SBLIMIT] = out1[12+2] + in3 * wi[6+2]; ++ ts[(17-2)*SBLIMIT] = out1[17-2] + in3 * wi[11-2]; ++ ++ ts[(6+0)*SBLIMIT] = out1[6+0] + in0 * wi[0]; ++ ts[(11-0)*SBLIMIT] = out1[11-0] + in0 * wi[5-0]; ++ ts[(6+2)*SBLIMIT] = out1[6+2] + in4 * wi[2]; ++ ts[(11-2)*SBLIMIT] = out1[11-2] + in4 * wi[5-2]; ++ } ++ ++ in++; ++ ++ { ++ real in0,in1,in2,in3,in4,in5; ++ register real *out2 = rawout2; ++ ++ DCT12_PART1 ++ ++ { ++ real tmp0,tmp1 = (in0 - in4); ++ { ++ real tmp2 = (in1 - in5) * tfcos12[1]; ++ tmp0 = tmp1 + tmp2; ++ tmp1 -= tmp2; ++ } ++ out2[5-1] = tmp0 * wi[11-1]; ++ out2[0+1] = tmp0 * wi[6+1]; ++ ts[(12+1)*SBLIMIT] += tmp1 * wi[1]; ++ ts[(17-1)*SBLIMIT] += tmp1 * wi[5-1]; ++ } ++ ++ DCT12_PART2 ++ ++ out2[5-0] = in2 * wi[11-0]; ++ out2[0+0] = in2 * wi[6+0]; ++ out2[0+2] = in3 * wi[6+2]; ++ out2[5-2] = in3 * wi[11-2]; ++ ++ ts[(12+0)*SBLIMIT] += in0 * wi[0]; ++ ts[(17-0)*SBLIMIT] += in0 * wi[5-0]; ++ ts[(12+2)*SBLIMIT] += in4 * wi[2]; ++ ts[(17-2)*SBLIMIT] += in4 * wi[5-2]; ++ } ++ ++ in++; ++ ++ { ++ real in0,in1,in2,in3,in4,in5; ++ register real *out2 = rawout2; ++ out2[12]=out2[13]=out2[14]=out2[15]=out2[16]=out2[17]=0.0; ++ ++ DCT12_PART1 ++ ++ { ++ real tmp0,tmp1 = (in0 - in4); ++ { ++ real tmp2 = (in1 - in5) * tfcos12[1]; ++ tmp0 = tmp1 + tmp2; ++ tmp1 -= tmp2; ++ } ++ out2[11-1] = tmp0 * wi[11-1]; ++ out2[6 +1] = tmp0 * wi[6+1]; ++ out2[0+1] += tmp1 * wi[1]; ++ out2[5-1] += tmp1 * wi[5-1]; ++ } ++ ++ DCT12_PART2 ++ ++ out2[11-0] = in2 * wi[11-0]; ++ out2[6 +0] = in2 * wi[6+0]; ++ out2[6 +2] = in3 * wi[6+2]; ++ out2[11-2] = in3 * wi[11-2]; ++ ++ out2[0+0] += in0 * wi[0]; ++ out2[5-0] += in0 * wi[5-0]; ++ out2[0+2] += in4 * wi[2]; ++ out2[5-2] += in4 * wi[5-2]; ++ } ++ } ++ ++ /* ++ * III_hybrid ++ */ ++ static void III_hybrid(real fsIn[SBLIMIT][SSLIMIT],real tsOut[SSLIMIT][SBLIMIT], ++ int ch,struct gr_info_s *gr_info) ++ { ++ real *tspnt = (real *) tsOut; ++ real (*block)[2][SBLIMIT*SSLIMIT] = gmp->hybrid_block; ++ int *blc = gmp->hybrid_blc; ++ real *rawout1,*rawout2; ++ int bt; ++ int sb = 0; ++ ++ { ++ int b = blc[ch]; ++ rawout1=block[b][ch]; ++ b=-b+1; ++ rawout2=block[b][ch]; ++ blc[ch] = b; ++ } ++ ++ ++ if(gr_info->mixed_block_flag) { ++ sb = 2; ++ dct36(fsIn[0],rawout1,rawout2,win[0],tspnt); ++ dct36(fsIn[1],rawout1+18,rawout2+18,win1[0],tspnt+1); ++ rawout1 += 36; rawout2 += 36; tspnt += 2; ++ } ++ ++ bt = gr_info->block_type; ++ if(bt == 2) { ++ for (; sb<gr_info->maxb; sb+=2,tspnt+=2,rawout1+=36,rawout2+=36) { ++ dct12(fsIn[sb],rawout1,rawout2,win[2],tspnt); ++ dct12(fsIn[sb+1],rawout1+18,rawout2+18,win1[2],tspnt+1); ++ } ++ } ++ else { ++ for (; sb<gr_info->maxb; sb+=2,tspnt+=2,rawout1+=36,rawout2+=36) { ++ dct36(fsIn[sb],rawout1,rawout2,win[bt],tspnt); ++ dct36(fsIn[sb+1],rawout1+18,rawout2+18,win1[bt],tspnt+1); ++ } ++ } ++ ++ for(;sb<SBLIMIT;sb++,tspnt++) { ++ int i; ++ for(i=0;i<SSLIMIT;i++) { ++ tspnt[i*SBLIMIT] = *rawout1++; ++ *rawout2++ = 0.0; ++ } ++ } ++ } ++ ++ /* ++ * main layer3 handler ++ */ ++ int do_layer3(struct frame *fr,unsigned char *pcm_sample,int *pcm_point) ++ { ++ int gr, ch, ss,clip=0; ++ int scalefacs[39]; /* max 39 for short[13][3] mode, mixed: 38, long: 22 */ ++ struct III_sideinfo sideinfo; ++ int stereo = fr->stereo; ++ int single = fr->single; ++ int ms_stereo,i_stereo; ++ int sfreq = fr->sampling_frequency; ++ int stereo1,granules; ++ ++ ++ ++ if(stereo == 1) { /* stream is mono */ ++ stereo1 = 1; ++ single = 0; ++ } ++ else if(single >= 0) /* stream is stereo, but force to mono */ ++ stereo1 = 1; ++ else ++ stereo1 = 2; ++ ++ if(fr->mode == MPG_MD_JOINT_STEREO) { ++ ms_stereo = fr->mode_ext & 0x2; ++ i_stereo = fr->mode_ext & 0x1; ++ } ++ else ++ ms_stereo = i_stereo = 0; ++ ++ ++ if(fr->lsf) { ++ granules = 1; ++ III_get_side_info_2(&sideinfo,stereo,ms_stereo,sfreq,single); ++ } ++ else { ++ granules = 2; ++ #ifdef MPEG1 ++ III_get_side_info_1(&sideinfo,stereo,ms_stereo,sfreq,single); ++ #else ++ fprintf(stderr,"Not supported\n"); ++ #endif ++ } ++ ++ if(set_pointer(sideinfo.main_data_begin) == MP3_ERR) ++ return 0; ++ ++ ++ for (gr=0;gr<granules;gr++) ++ { ++ static real hybridIn[2][SBLIMIT][SSLIMIT]; ++ static real hybridOut[2][SSLIMIT][SBLIMIT]; ++ ++ { ++ struct gr_info_s *gr_info = &(sideinfo.ch[0].gr[gr]); ++ long part2bits; ++ if(fr->lsf) ++ part2bits = III_get_scale_factors_2(scalefacs,gr_info,0); ++ else { ++ #ifdef MPEG1 ++ part2bits = III_get_scale_factors_1(scalefacs,gr_info); ++ #else ++ fprintf(stderr,"Not supported\n"); ++ #endif ++ } ++ if(III_dequantize_sample(hybridIn[0], scalefacs,gr_info,sfreq,part2bits)) ++ return clip; ++ } ++ if(stereo == 2) { ++ struct gr_info_s *gr_info = &(sideinfo.ch[1].gr[gr]); ++ long part2bits; ++ if(fr->lsf) ++ part2bits = III_get_scale_factors_2(scalefacs,gr_info,i_stereo); ++ else { ++ #ifdef MPEG1 ++ part2bits = III_get_scale_factors_1(scalefacs,gr_info); ++ #else ++ fprintf(stderr,"Not supported\n"); ++ #endif ++ } ++ ++ if(III_dequantize_sample(hybridIn[1],scalefacs,gr_info,sfreq,part2bits)) ++ return clip; ++ ++ if(ms_stereo) { ++ int i; ++ for(i=0;i<SBLIMIT*SSLIMIT;i++) { ++ real tmp0,tmp1; ++ tmp0 = ((real *) hybridIn[0])[i]; ++ tmp1 = ((real *) hybridIn[1])[i]; ++ ((real *) hybridIn[1])[i] = tmp0 - tmp1; ++ ((real *) hybridIn[0])[i] = tmp0 + tmp1; ++ } ++ } ++ ++ if(i_stereo) ++ III_i_stereo(hybridIn,scalefacs,gr_info,sfreq,ms_stereo,fr->lsf); ++ ++ if(ms_stereo || i_stereo || (single == 3) ) { ++ if(gr_info->maxb > sideinfo.ch[0].gr[gr].maxb) ++ sideinfo.ch[0].gr[gr].maxb = gr_info->maxb; ++ else ++ gr_info->maxb = sideinfo.ch[0].gr[gr].maxb; ++ } ++ ++ switch(single) { ++ case 3: ++ { ++ register int i; ++ register real *in0 = (real *) hybridIn[0],*in1 = (real *) hybridIn[1]; ++ for(i=0;i<SSLIMIT*gr_info->maxb;i++,in0++) ++ *in0 = (*in0 + *in1++); /* *0.5 done by pow-scale */ ++ } ++ break; ++ case 1: ++ { ++ register int i; ++ register real *in0 = (real *) hybridIn[0],*in1 = (real *) hybridIn[1]; ++ for(i=0;i<SSLIMIT*gr_info->maxb;i++) ++ *in0++ = *in1++; ++ } ++ break; ++ } ++ } ++ ++ #ifdef HAVEGTK ++ if (gtkflag) { ++ extern int tabsel_123[2][3][16]; ++ int sb; ++ ++ pinfo->bitrate = ++ tabsel_123[fr->lsf][fr->lay-1][fr->bitrate_index]; ++ pinfo->sampfreq = freqs[sfreq]; ++ pinfo->emph = fr->emphasis; ++ pinfo->stereo = fr->stereo; ++ pinfo->js = (fr->mode == MPG_MD_JOINT_STEREO); ++ pinfo->ms_stereo = ms_stereo; ++ pinfo->i_stereo = i_stereo; ++ pinfo->maindata = sideinfo.main_data_begin; ++ ++ for(ch=0;ch<stereo1;ch++) { ++ struct gr_info_s *gr_info = &(sideinfo.ch[ch].gr[gr]); ++ pinfo->mixed[gr][ch] = gr_info->mixed_block_flag; ++ pinfo->mpg123blocktype[gr][ch]=gr_info->block_type; ++ pinfo->mainbits[gr][ch] = gr_info->part2_3_length; ++ pinfo->scfsi = gr_info->scfsi; ++ } ++ for(ch=0;ch<stereo1;ch++) { ++ int j=0; ++ for (sb=0;sb<SBLIMIT;sb++) ++ for(ss=0;ss<SSLIMIT;ss++,j++) ++ pinfo->mpg123xr[gr][ch][j]=hybridIn[ch][sb][ss]; ++ } ++ } ++ ++ #endif ++ ++ for(ch=0;ch<stereo1;ch++) { ++ struct gr_info_s *gr_info = &(sideinfo.ch[ch].gr[gr]); ++ III_antialias(hybridIn[ch],gr_info); ++ III_hybrid(hybridIn[ch], hybridOut[ch], ch,gr_info); ++ } ++ ++ for(ss=0;ss<SSLIMIT;ss++) { ++ if(single >= 0) { ++ clip += synth_1to1_mono(hybridOut[0][ss],pcm_sample,pcm_point); ++ } ++ else { ++ int p1 = *pcm_point; ++ clip += synth_1to1(hybridOut[0][ss],0,pcm_sample,&p1); ++ clip += synth_1to1(hybridOut[1][ss],1,pcm_sample,pcm_point); ++ } ++ } ++ } ++ ++ return clip; ++ } ++ ++ ++ +diff -r -c -N encoder/mpglib/main.c lame3.50/mpglib/main.c +*** encoder/mpglib/main.c Wed Dec 31 17:00:00 1969 +--- lame3.50/mpglib/main.c Mon Oct 25 12:24:07 1999 +*************** +*** 0 **** +--- 1,178 ---- ++ #include "mpg123.h" ++ #include "mpglib.h" ++ ++ #ifdef OS_AMIGAOS ++ #include "/lame.h" ++ #else ++ #include "../lame.h" ++ #endif /* OS_AMIGAOS */ ++ ++ #include <stdlib.h> ++ ++ static char buf[16384]; ++ #define FSIZE 8192 ++ static char out[FSIZE]; ++ struct mpstr mp; ++ ++ #if 0 ++ void main(void) ++ { ++ int size; ++ char out[8192]; ++ int len,ret; ++ ++ ++ InitMP3(&mp); ++ ++ while(1) { ++ len = read(0,buf,16384); ++ if(len <= 0) ++ break; ++ ret = decodeMP3(&mp,buf,len,out,8192,&size); ++ while(ret == MP3_OK) { ++ write(1,out,size); ++ ret = decodeMP3(&mp,NULL,0,out,8192,&size); ++ } ++ } ++ ++ } ++ #endif ++ ++ int is_syncword(char *header) ++ { ++ ++ /* ++ unsigned int s0,s1; ++ s0 = (unsigned char) header[0]; ++ s1 = (unsigned char) header[1] ; ++ printf(" syncword: %2X %2X \n ",s0, s1); ++ */ ++ ++ /* ++ printf(" integer %i \n",(int) ( header[0] == (char) 0xFF)); ++ printf(" integer %i \n",(int) ( (header[1] & (char) 0xF0) == (char) 0xF0)); ++ */ ++ ++ return ++ ((int) ( header[0] == (char) 0xFF)) && ++ ((int) ( (header[1] & (char) 0xF0) == (char) 0xF0)); ++ ++ ++ } ++ ++ ++ int lame_decode_initfile(FILE *fd, int *stereo, int *samp, int *bitrate) ++ { ++ extern int tabsel_123[2][3][16]; ++ int ret,size; ++ size_t len; ++ InitMP3(&mp); ++ memset(buf, 0, sizeof(buf)); ++ ++ /* skip RIFF type proprietary headers */ ++ /* look for sync word FFF */ ++ while (!is_syncword(buf)) { ++ buf[0]=buf[1]; ++ // if (read(fd,&buf[1],1) <= 0) return -1; /* failed */ ++ if (fread(&buf[1],1,1,fd) <= 0) return -1; /* failed */ ++ } ++ ret = decodeMP3(&mp,buf,2,out,FSIZE,&size); ++ ++ /* read the header */ ++ //len = read(fd,buf,8); ++ len = fread(buf,1,8,fd); ++ if (len <=0 ) return -1; ++ ret = decodeMP3(&mp,buf,len,out,FSIZE,&size); ++ ++ *stereo = mp.fr.stereo; ++ *samp = freqs[mp.fr.sampling_frequency]; ++ *bitrate = tabsel_123[mp.fr.lsf][mp.fr.lay-1][mp.fr.bitrate_index]; ++ ++ ++ /* ++ printf("ret = %i NEED_MORE=%i \n",ret,MP3_NEED_MORE); ++ printf("stereo = %i \n",mp.fr.stereo); ++ printf("samp = %i \n",(int)freqs[mp.fr.sampling_frequency]); ++ */ ++ ++ return 0; ++ } ++ ++ ++ ++ int lame_decode_fromfile(FILE *fd, short pcm[][1152]) ++ { ++ int size,stereo; ++ int outsize=0,j,i,ret; ++ size_t len; ++ ++ size=0; ++ //len = read(fd,buf,64); ++ len = fread(buf,1,64,fd); ++ if (len <=0 ) return 0; ++ ret = decodeMP3(&mp,buf,len,out,FSIZE,&size); ++ ++ /* read more until we get a valid output frame */ ++ while((ret == MP3_NEED_MORE) || !size) { ++ // len = read(fd,buf,100); ++ len = fread(buf,1,100,fd); ++ if (len <=0 ) return 0; ++ ret = decodeMP3(&mp,buf,len,out,FSIZE,&size); ++ /* if (ret ==MP3_ERR) return 0; lets ignore errors and keep reading... */ ++ /* ++ printf("ret = %i size= %i %i %i %i \n",ret,size, ++ MP3_NEED_MORE,MP3_ERR,MP3_OK); ++ */ ++ } ++ ++ stereo=mp.fr.stereo; ++ ++ if (ret == MP3_OK) ++ { ++ /* write(1,out,size); */ ++ outsize = size/(2*(stereo)); ++ if ((outsize!=576) && (outsize!=1152)) { ++ fprintf(stderr,"Opps: mpg123 returned more than one frame! Cant handle this... \n"); ++ exit(-50); ++ } ++ ++ for (j=0; j<stereo; j++) ++ for (i=0; i<outsize; i++) ++ pcm[j][i] = ((short *) out)[stereo*i+j]; ++ } ++ return outsize; ++ } ++ ++ ++ int lame_decode_init(void) ++ { ++ InitMP3(&mp); ++ memset(buf, 0, sizeof(buf)); ++ return 0; ++ } ++ ++ ++ int lame_decode(char *buf,int len,short pcm[][1152]) ++ { ++ int size; ++ int outsize=0,j,i,ret; ++ ++ ret = decodeMP3(&mp,buf,len,out,FSIZE,&size); ++ if (ret==MP3_OK) { ++ /* printf("mpg123 output one frame out=%i \n",size/4); */ ++ outsize = size/(2*mp.fr.stereo); ++ if (outsize > 1152) { ++ fprintf(stderr,"Opps: mpg123 returned more than one frame! shouldn't happen... \n"); ++ exit(-50); ++ } ++ ++ for (j=0; j<mp.fr.stereo; j++) ++ for (i=0; i<outsize; i++) ++ pcm[j][i] = ((short *) out)[mp.fr.stereo*i+j]; ++ } ++ return outsize; ++ } ++ ++ ++ ++ +diff -r -c -N encoder/mpglib/mpg123.h lame3.50/mpglib/mpg123.h +*** encoder/mpglib/mpg123.h Wed Dec 31 17:00:00 1969 +--- lame3.50/mpglib/mpg123.h Tue Oct 12 10:45:52 1999 +*************** +*** 0 **** +--- 1,179 ---- ++ #include <stdio.h> ++ #include <string.h> ++ #include <signal.h> ++ ++ ++ #include <math.h> ++ ++ #ifdef _MSC_VER ++ ++ # define M_PI 3.14159265358979323846 ++ # define M_SQRT2 1.41421356237309504880 ++ # define REAL_IS_FLOAT ++ # define NEW_DCT9 ++ ++ # define random rand ++ # define srandom srand ++ ++ #endif ++ #ifdef __EMX__ ++ #define M_PI 3.14159265358979323846 ++ #define M_SQRT2 1.41421356237309504880 ++ #endif ++ ++ ++ #ifdef REAL_IS_FLOAT ++ # define real float ++ #elif defined(REAL_IS_LONG_DOUBLE) ++ # define real long double ++ #else ++ # define real double ++ #endif ++ ++ #ifdef __GNUC__ ++ #define INLINE inline ++ #else ++ #define INLINE ++ #endif ++ ++ /* AUDIOBUFSIZE = n*64 with n=1,2,3 ... */ ++ #define AUDIOBUFSIZE 16384 ++ ++ #define FALSE 0 ++ #define TRUE 1 ++ ++ #define SBLIMIT 32 ++ #define SSLIMIT 18 ++ ++ #define MPG_MD_STEREO 0 ++ #define MPG_MD_JOINT_STEREO 1 ++ #define MPG_MD_DUAL_CHANNEL 2 ++ #define MPG_MD_MONO 3 ++ ++ #define MAXFRAMESIZE 1792 ++ ++ ++ /* Pre Shift fo 16 to 8 bit converter table */ ++ #define AUSHIFT (3) ++ ++ struct frame { ++ int stereo; ++ int jsbound; ++ int single; ++ int lsf; ++ int mpeg25; ++ int header_change; ++ int lay; ++ int error_protection; ++ int bitrate_index; ++ int sampling_frequency; ++ int padding; ++ int extension; ++ int mode; ++ int mode_ext; ++ int copyright; ++ int original; ++ int emphasis; ++ int framesize; /* computed framesize */ ++ }; ++ ++ struct parameter { ++ int quiet; /* shut up! */ ++ int tryresync; /* resync stream after error */ ++ int verbose; /* verbose level */ ++ int checkrange; ++ }; ++ ++ /* extern unsigned int get1bit(void); */ ++ extern unsigned int getbits(int); ++ extern unsigned int getbits_fast(int); ++ extern int set_pointer(long); ++ ++ extern unsigned char *wordpointer; ++ extern int bitindex; ++ ++ extern void make_decode_tables(long scaleval); ++ extern int do_layer3(struct frame *fr,unsigned char *,int *); ++ extern int decode_header(struct frame *fr,unsigned long newhead); ++ ++ ++ ++ struct gr_info_s { ++ int scfsi; ++ unsigned part2_3_length; ++ unsigned big_values; ++ unsigned scalefac_compress; ++ unsigned block_type; ++ unsigned mixed_block_flag; ++ unsigned table_select[3]; ++ unsigned subblock_gain[3]; ++ unsigned maxband[3]; ++ unsigned maxbandl; ++ unsigned maxb; ++ unsigned region1start; ++ unsigned region2start; ++ unsigned preflag; ++ unsigned scalefac_scale; ++ unsigned count1table_select; ++ real *full_gain[3]; ++ real *pow2gain; ++ }; ++ ++ struct III_sideinfo ++ { ++ unsigned main_data_begin; ++ unsigned private_bits; ++ struct { ++ struct gr_info_s gr[2]; ++ } ch[2]; ++ }; ++ ++ extern int synth_1to1 (real *,int,unsigned char *,int *); ++ extern int synth_1to1_8bit (real *,int,unsigned char *,int *); ++ extern int synth_1to1_mono (real *,unsigned char *,int *); ++ extern int synth_1to1_mono2stereo (real *,unsigned char *,int *); ++ extern int synth_1to1_8bit_mono (real *,unsigned char *,int *); ++ extern int synth_1to1_8bit_mono2stereo (real *,unsigned char *,int *); ++ ++ extern int synth_2to1 (real *,int,unsigned char *,int *); ++ extern int synth_2to1_8bit (real *,int,unsigned char *,int *); ++ extern int synth_2to1_mono (real *,unsigned char *,int *); ++ extern int synth_2to1_mono2stereo (real *,unsigned char *,int *); ++ extern int synth_2to1_8bit_mono (real *,unsigned char *,int *); ++ extern int synth_2to1_8bit_mono2stereo (real *,unsigned char *,int *); ++ ++ extern int synth_4to1 (real *,int,unsigned char *,int *); ++ extern int synth_4to1_8bit (real *,int,unsigned char *,int *); ++ extern int synth_4to1_mono (real *,unsigned char *,int *); ++ extern int synth_4to1_mono2stereo (real *,unsigned char *,int *); ++ extern int synth_4to1_8bit_mono (real *,unsigned char *,int *); ++ extern int synth_4to1_8bit_mono2stereo (real *,unsigned char *,int *); ++ ++ extern int synth_ntom (real *,int,unsigned char *,int *); ++ extern int synth_ntom_8bit (real *,int,unsigned char *,int *); ++ extern int synth_ntom_mono (real *,unsigned char *,int *); ++ extern int synth_ntom_mono2stereo (real *,unsigned char *,int *); ++ extern int synth_ntom_8bit_mono (real *,unsigned char *,int *); ++ extern int synth_ntom_8bit_mono2stereo (real *,unsigned char *,int *); ++ ++ extern void rewindNbits(int bits); ++ extern int hsstell(void); ++ extern int get_songlen(struct frame *fr,int no); ++ ++ extern void init_layer3(int); ++ extern void init_layer2(void); ++ extern void make_decode_tables(long scale); ++ extern void make_conv16to8_table(int); ++ extern void dct64(real *,real *,real *); ++ ++ extern void synth_ntom_set_step(long,long); ++ ++ extern unsigned char *conv16to8; ++ extern long freqs[9]; ++ extern real muls[27][64]; ++ extern real decwin[512+32]; ++ extern real *pnts[5]; ++ ++ extern struct parameter param; ++ ++ +diff -r -c -N encoder/mpglib/mpglib.dsp lame3.50/mpglib/mpglib.dsp +*** encoder/mpglib/mpglib.dsp Wed Dec 31 17:00:00 1969 +--- lame3.50/mpglib/mpglib.dsp Sun Oct 24 21:49:54 1999 +*************** +*** 0 **** +--- 1,164 ---- ++ # Microsoft Developer Studio Project File - Name="mpglib" - Package Owner=<4> ++ # Microsoft Developer Studio Generated Build File, Format Version 5.00 ++ # ** DO NOT EDIT ** ++ ++ # TARGTYPE "Win32 (x86) Static Library" 0x0104 ++ ++ CFG=mpglib - Win32 Debug GTK ++ !MESSAGE This is not a valid makefile. To build this project using NMAKE, ++ !MESSAGE use the Export Makefile command and run ++ !MESSAGE ++ !MESSAGE NMAKE /f "mpglib.mak". ++ !MESSAGE ++ !MESSAGE You can specify a configuration when running NMAKE ++ !MESSAGE by defining the macro CFG on the command line. For example: ++ !MESSAGE ++ !MESSAGE NMAKE /f "mpglib.mak" CFG="mpglib - Win32 Debug GTK" ++ !MESSAGE ++ !MESSAGE Possible choices for configuration are: ++ !MESSAGE ++ !MESSAGE "mpglib - Win32 Release" (based on "Win32 (x86) Static Library") ++ !MESSAGE "mpglib - Win32 Debug" (based on "Win32 (x86) Static Library") ++ !MESSAGE "mpglib - Win32 Debug GTK" (based on "Win32 (x86) Static Library") ++ !MESSAGE "mpglib - Win32 Release GTK" (based on "Win32 (x86) Static Library") ++ !MESSAGE ++ ++ # Begin Project ++ # PROP Scc_ProjName "" ++ # PROP Scc_LocalPath "" ++ CPP=xicl.exe ++ ++ !IF "$(CFG)" == "mpglib - Win32 Release" ++ ++ # PROP BASE Use_MFC 0 ++ # PROP BASE Use_Debug_Libraries 0 ++ # PROP BASE Output_Dir "Release" ++ # PROP BASE Intermediate_Dir "Release" ++ # PROP BASE Target_Dir "" ++ # PROP Use_MFC 0 ++ # PROP Use_Debug_Libraries 0 ++ # PROP Output_Dir "ReleaseGTK" ++ # PROP Intermediate_Dir "ReleaseGTK" ++ # PROP Target_Dir "" ++ # ADD BASE CPP /nologo /W3 /GX /O2 /D "WIN32" /D "NDEBUG" /D "_WINDOWS" /YX /FD /c ++ # ADD CPP /nologo /W3 /GX /O2 /D "WIN32" /D "NDEBUG" /D "_WINDOWS" /YX /FD /c ++ BSC32=bscmake.exe ++ # ADD BASE BSC32 /nologo ++ # ADD BSC32 /nologo ++ LIB32=link.exe -lib ++ # ADD BASE LIB32 /nologo ++ # ADD LIB32 /nologo ++ ++ !ELSEIF "$(CFG)" == "mpglib - Win32 Debug" ++ ++ # PROP BASE Use_MFC 0 ++ # PROP BASE Use_Debug_Libraries 1 ++ # PROP BASE Output_Dir "Debug" ++ # PROP BASE Intermediate_Dir "Debug" ++ # PROP BASE Target_Dir "" ++ # PROP Use_MFC 0 ++ # PROP Use_Debug_Libraries 1 ++ # PROP Output_Dir "DebugGTK" ++ # PROP Intermediate_Dir "DebugGTK" ++ # PROP Target_Dir "" ++ # ADD BASE CPP /nologo /W3 /GX /Z7 /Od /D "WIN32" /D "_DEBUG" /D "_WINDOWS" /YX /FD /c ++ # ADD CPP /nologo /W3 /GX /Z7 /Od /D "WIN32" /D "_DEBUG" /D "_WINDOWS" /YX /FD /c ++ BSC32=bscmake.exe ++ # ADD BASE BSC32 /nologo ++ # ADD BSC32 /nologo ++ LIB32=link.exe -lib ++ # ADD BASE LIB32 /nologo ++ # ADD LIB32 /nologo ++ ++ !ELSEIF "$(CFG)" == "mpglib - Win32 Debug GTK" ++ ++ # PROP BASE Use_MFC 0 ++ # PROP BASE Use_Debug_Libraries 1 ++ # PROP BASE Output_Dir "mpglib__" ++ # PROP BASE Intermediate_Dir "mpglib__" ++ # PROP BASE Target_Dir "" ++ # PROP Use_MFC 0 ++ # PROP Use_Debug_Libraries 1 ++ # PROP Output_Dir "mpglib__" ++ # PROP Intermediate_Dir "mpglib__" ++ # PROP Target_Dir "" ++ # ADD BASE CPP /nologo /W3 /GX /Z7 /Od /D "WIN32" /D "_DEBUG" /D "_WINDOWS" /YX /FD /c ++ # ADD CPP /nologo /W3 /GX /Z7 /Od /D "WIN32" /D "_DEBUG" /D "_WINDOWS" /D "HAVEGTK" /YX /FD /c ++ BSC32=bscmake.exe ++ # ADD BASE BSC32 /nologo ++ # ADD BSC32 /nologo ++ LIB32=link.exe -lib ++ # ADD BASE LIB32 /nologo ++ # ADD LIB32 /nologo ++ ++ !ELSEIF "$(CFG)" == "mpglib - Win32 Release GTK" ++ ++ # PROP BASE Use_MFC 0 ++ # PROP BASE Use_Debug_Libraries 0 ++ # PROP BASE Output_Dir "mpglib_0" ++ # PROP BASE Intermediate_Dir "mpglib_0" ++ # PROP BASE Target_Dir "" ++ # PROP Use_MFC 0 ++ # PROP Use_Debug_Libraries 0 ++ # PROP Output_Dir "mpglib_0" ++ # PROP Intermediate_Dir "mpglib_0" ++ # PROP Target_Dir "" ++ # ADD BASE CPP /nologo /W3 /GX /O2 /D "WIN32" /D "NDEBUG" /D "_WINDOWS" /YX /FD /c ++ # ADD CPP /nologo /W3 /GX /O2 /D "WIN32" /D "NDEBUG" /D "_WINDOWS" /D "HAVEGTK" /YX /FD /c ++ BSC32=bscmake.exe ++ # ADD BASE BSC32 /nologo ++ # ADD BSC32 /nologo ++ LIB32=link.exe -lib ++ # ADD BASE LIB32 /nologo ++ # ADD LIB32 /nologo ++ ++ !ENDIF ++ ++ # Begin Target ++ ++ # Name "mpglib - Win32 Release" ++ # Name "mpglib - Win32 Debug" ++ # Name "mpglib - Win32 Debug GTK" ++ # Name "mpglib - Win32 Release GTK" ++ # Begin Source File ++ ++ SOURCE=.\common.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\dct64_i386.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\decode_i386.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\huffman.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\interface.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\layer3.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\main.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\mpg123.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\mpglib.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\tabinit.c ++ # End Source File ++ # End Target ++ # End Project +diff -r -c -N encoder/mpglib/mpglib.h lame3.50/mpglib/mpglib.h +*** encoder/mpglib/mpglib.h Wed Dec 31 17:00:00 1969 +--- lame3.50/mpglib/mpglib.h Fri Sep 10 09:51:57 1999 +*************** +*** 0 **** +--- 1,44 ---- ++ ++ struct buf { ++ unsigned char *pnt; ++ long size; ++ long pos; ++ struct buf *next; ++ struct buf *prev; ++ }; ++ ++ struct framebuf { ++ struct buf *buf; ++ long pos; ++ struct frame *next; ++ struct frame *prev; ++ }; ++ ++ struct mpstr { ++ struct buf *head,*tail; ++ int bsize; ++ int framesize; ++ int fsizeold; ++ struct frame fr; ++ unsigned char bsspace[2][MAXFRAMESIZE+512]; /* MAXFRAMESIZE */ ++ real hybrid_block[2][2][SBLIMIT*SSLIMIT]; ++ int hybrid_blc[2]; ++ unsigned long header; ++ int bsnum; ++ real synth_buffs[2][2][0x110]; ++ int synth_bo; ++ ++ }; ++ ++ #define BOOL int ++ ++ #define MP3_ERR -1 ++ #define MP3_OK 0 ++ #define MP3_NEED_MORE 1 ++ ++ ++ BOOL InitMP3(struct mpstr *mp); ++ int decodeMP3(struct mpstr *mp,char *inmemory,int inmemsize, ++ char *outmemory,int outmemsize,int *done); ++ void ExitMP3(struct mpstr *mp); ++ +diff -r -c -N encoder/mpglib/tabinit.c lame3.50/mpglib/tabinit.c +*** encoder/mpglib/tabinit.c Wed Dec 31 17:00:00 1969 +--- lame3.50/mpglib/tabinit.c Thu Sep 16 00:12:46 1999 +*************** +*** 0 **** +--- 1,80 ---- ++ #include <stdlib.h> ++ ++ #include "mpg123.h" ++ ++ real decwin[512+32]; ++ static real cos64[16],cos32[8],cos16[4],cos8[2],cos4[1]; ++ real *pnts[] = { cos64,cos32,cos16,cos8,cos4 }; ++ ++ #if 0 ++ static unsigned char *conv16to8_buf = NULL; ++ unsigned char *conv16to8; ++ #endif ++ ++ static long intwinbase[] = { ++ 0, -1, -1, -1, -1, -1, -1, -2, -2, -2, ++ -2, -3, -3, -4, -4, -5, -5, -6, -7, -7, ++ -8, -9, -10, -11, -13, -14, -16, -17, -19, -21, ++ -24, -26, -29, -31, -35, -38, -41, -45, -49, -53, ++ -58, -63, -68, -73, -79, -85, -91, -97, -104, -111, ++ -117, -125, -132, -139, -147, -154, -161, -169, -176, -183, ++ -190, -196, -202, -208, -213, -218, -222, -225, -227, -228, ++ -228, -227, -224, -221, -215, -208, -200, -189, -177, -163, ++ -146, -127, -106, -83, -57, -29, 2, 36, 72, 111, ++ 153, 197, 244, 294, 347, 401, 459, 519, 581, 645, ++ 711, 779, 848, 919, 991, 1064, 1137, 1210, 1283, 1356, ++ 1428, 1498, 1567, 1634, 1698, 1759, 1817, 1870, 1919, 1962, ++ 2001, 2032, 2057, 2075, 2085, 2087, 2080, 2063, 2037, 2000, ++ 1952, 1893, 1822, 1739, 1644, 1535, 1414, 1280, 1131, 970, ++ 794, 605, 402, 185, -45, -288, -545, -814, -1095, -1388, ++ -1692, -2006, -2330, -2663, -3004, -3351, -3705, -4063, -4425, -4788, ++ -5153, -5517, -5879, -6237, -6589, -6935, -7271, -7597, -7910, -8209, ++ -8491, -8755, -8998, -9219, -9416, -9585, -9727, -9838, -9916, -9959, ++ -9966, -9935, -9863, -9750, -9592, -9389, -9139, -8840, -8492, -8092, ++ -7640, -7134, -6574, -5959, -5288, -4561, -3776, -2935, -2037, -1082, ++ -70, 998, 2122, 3300, 4533, 5818, 7154, 8540, 9975, 11455, ++ 12980, 14548, 16155, 17799, 19478, 21189, 22929, 24694, 26482, 28289, ++ 30112, 31947, 33791, 35640, 37489, 39336, 41176, 43006, 44821, 46617, ++ 48390, 50137, 51853, 53534, 55178, 56778, 58333, 59838, 61289, 62684, ++ 64019, 65290, 66494, 67629, 68692, 69679, 70590, 71420, 72169, 72835, ++ 73415, 73908, 74313, 74630, 74856, 74992, 75038 }; ++ ++ void make_decode_tables(long scaleval) ++ { ++ int i,j,k,kr,divv; ++ real *table,*costab; ++ ++ ++ for(i=0;i<5;i++) ++ { ++ kr=0x10>>i; divv=0x40>>i; ++ costab = pnts[i]; ++ for(k=0;k<kr;k++) ++ costab[k] = 1.0 / (2.0 * cos(M_PI * ((double) k * 2.0 + 1.0) / (double) divv)); ++ } ++ ++ table = decwin; ++ scaleval = -scaleval; ++ for(i=0,j=0;i<256;i++,j++,table+=32) ++ { ++ if(table < decwin+512+16) ++ table[16] = table[0] = (double) intwinbase[j] / 65536.0 * (double) scaleval; ++ if(i % 32 == 31) ++ table -= 1023; ++ if(i % 64 == 63) ++ scaleval = - scaleval; ++ } ++ ++ for( /* i=256 */ ;i<512;i++,j--,table+=32) ++ { ++ if(table < decwin+512+16) ++ table[16] = table[0] = (double) intwinbase[j] / 65536.0 * (double) scaleval; ++ if(i % 32 == 31) ++ table -= 1023; ++ if(i % 64 == 63) ++ scaleval = - scaleval; ++ } ++ } ++ ++ ++ +diff -r -c -N encoder/musicin.c lame3.50/musicin.c +*** encoder/musicin.c Wed Jan 22 02:43:17 1997 +--- lame3.50/musicin.c Wed Dec 31 17:00:00 1969 +*************** +*** 1,1151 **** +- /********************************************************************** +- * ISO MPEG Audio Subgroup Software Simulation Group (1996) +- * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- * +- * $Id: musicin.c,v 1.2 1997/01/19 22:28:29 rowlands Exp $ +- * +- * $Log: musicin.c,v $ +- * Revision 1.2 1997/01/19 22:28:29 rowlands +- * Layer 3 bug fixes from Seymour Shlien +- * +- * Revision 1.1 1996/02/14 04:04:23 rowlands +- * Initial revision +- * +- * Received from Mike Coleman +- **********************************************************************/ +- /********************************************************************** +- * date programmers comment * +- * 3/01/91 Douglas Wong, start of version 1.1 records * +- * Davis Pan * +- * 3/06/91 Douglas Wong, rename: setup.h to endef.h * +- * removed extraneous variables * +- * 3/21/91 J.Georges Fritsch introduction of the bit-stream * +- * package. This package allows you * +- * to generate the bit-stream in a * +- * binary or ascii format * +- * 3/31/91 Bill Aspromonte replaced the read of the SB matrix * +- * by an "code generated" one * +- * 5/10/91 W. Joseph Carter Ported to Macintosh and Unix. * +- * Incorporated Jean-Georges Fritsch's * +- * "bitstream.c" package. * +- * Modified to strictly adhere to * +- * encoded bitstream specs, including * +- * "Berlin changes". * +- * Modified user interface dialog & code * +- * to accept any input & output * +- * filenames desired. Also added * +- * de-emphasis prompt and final bail-out * +- * opportunity before encoding. * +- * Added AIFF PCM sound file reading * +- * capability. * +- * Modified PCM sound file handling to * +- * process all incoming samples and fill * +- * out last encoded frame with zeros * +- * (silence) if needed. * +- * Located and fixed numerous software * +- * bugs and table data errors. * +- * 27jun91 dpwe (Aware Inc) Used new frame_params struct. * +- * Clear all automatic arrays. * +- * Changed some variable names, * +- * simplified some code. * +- * Track number of bits actually sent. * +- * Fixed padding slot, stereo bitrate * +- * Added joint-stereo : scales L+R. * +- * 6/12/91 Earle Jennings added fix for MS_DOS in obtain_param * +- * 6/13/91 Earle Jennings added stack length adjustment before * +- * main for MS_DOS * +- * 7/10/91 Earle Jennings conversion of all float to FLOAT * +- * port to MsDos from MacIntosh completed* +- * 8/ 8/91 Jens Spille Change for MS-C6.00 * +- * 8/22/91 Jens Spille new obtain_parameters() * +- *10/ 1/91 S.I. Sudharsanan, Ported to IBM AIX platform. * +- * Don H. Lee, * +- * Peter W. Farrett * +- *10/ 3/91 Don H. Lee implemented CRC-16 error protection * +- * newly introduced functions are * +- * I_CRC_calc, II_CRC_calc and encode_CRC* +- * Additions and revisions are marked * +- * with "dhl" for clarity * +- *11/11/91 Katherine Wang Documentation of code. * +- * (variables in documentation are * +- * surround by the # symbol, and an '*'* +- * denotes layer I or II versions) * +- * 2/11/92 W. Joseph Carter Ported new code to Macintosh. Most * +- * important fixes involved changing * +- * 16-bit ints to long or unsigned in * +- * bit alloc routines for quant of 65535 * +- * and passing proper function args. * +- * Removed "Other Joint Stereo" option * +- * and made bitrate be total channel * +- * bitrate, irrespective of the mode. * +- * Fixed many small bugs & reorganized. * +- * 2/25/92 Masahiro Iwadare made code cleaner and more consistent * +- * 8/07/92 Mike Coleman make exit() codes return error status * +- * made slight changes for portability * +- *19 aug 92 Soren H. Nielsen Changed MS-DOS file name extensions. * +- * 8/25/92 Shaun Astarabadi Replaced rint() function with explicit* +- * rounding for portability with MSDOS. * +- * 9/22/92 jddevine@aware.com Fixed _scale_factor_calc() calls. * +- *10/19/92 Masahiro Iwadare added info->mode and info->mode_ext * +- * updates for AIFF format files * +- * 3/10/93 Kevin Peterson In parse_args, only set non default * +- * bit rate if specified in arg list. * +- * Use return value from aiff_read_hdrs * +- * to fseek to start of sound data * +- * 7/26/93 Davis Pan fixed bug in printing info->mode_ext * +- * value for joint stereo condition * +- * 8/27/93 Seymour Shlien, Fixes in Unix and MSDOS ports, * +- * Daniel Lauzon, and * +- * Bill Truerniet * +- * 11/7/95 Soeren H. Nielsen LSF added. Bug fix in MSDOS ext. * +- * 8/02/95 mc@fivebats.com Changed default bitrate selection so * +- * it works with the new LSF stuff * +- *10/01/95 mc@fivebats.com Added layer3 * +- **********************************************************************/ +- +- #ifdef MS_DOS +- #include <dos.h> +- #endif +- #include <stdlib.h> +- #include "common.h" +- #include "encoder.h" +- #include "l3psy.h" +- #include "mdct.h" +- #include "loop.h" +- #include "l3bitstream.h" +- #include <assert.h> +- +- /* Global variable definitions for "musicin.c" */ +- +- FILE *musicin; +- Bit_stream_struc bs; +- char *programName; +- +- /* Implementations */ +- +- /************************************************************************ +- * +- * obtain_parameters +- * +- * PURPOSE: Prompts for and reads user input for encoding parameters +- * +- * SEMANTICS: The parameters read are: +- * - input and output filenames +- * - sampling frequency (if AIFF file, will read from the AIFF file header) +- * - layer number +- * - mode (stereo, joint stereo, dual channel or mono) +- * - psychoacoustic model (I or II) +- * - total bitrate, irrespective of the mode +- * - de-emphasis, error protection, copyright and original or copy flags +- * +- ************************************************************************/ +- +- void +- obtain_parameters(fr_ps,psy,num_samples,original_file_name,encoded_file_name) +- frame_params *fr_ps; +- int *psy; +- unsigned long *num_samples; +- char original_file_name[MAX_NAME_SIZE]; +- char encoded_file_name[MAX_NAME_SIZE]; +- { +- int j; +- long int freq; +- int model, brt; +- char t[50]; +- IFF_AIFF pcm_aiff_data; +- layer *info = fr_ps->header; +- long soundPosition; +- #ifdef MS_DOS +- char temp_str[MAX_NAME_SIZE]; +- #endif +- +- do { +- printf("Enter PCM input file name <required>: "); +- gets(original_file_name); +- if (original_file_name[0] == NULL_CHAR) +- printf("PCM input file name is required.\n"); +- } while (original_file_name[0] == NULL_CHAR); +- printf(">>> PCM input file name is: %s\n", original_file_name); +- +- if ((musicin = fopen(original_file_name, "rb")) == NULL) { +- printf("Could not find \"%s\".\n", original_file_name); +- exit(1); +- } +- +- #ifdef MS_DOS +- /* replace old extension with new one, 1992-08-19, 1995-06-12 shn */ +- new_ext(original_file_name, DFLT_EXT, temp_str); +- printf("Enter MPEG encoded output file name <%s>: ", +- temp_str); +- #else +- printf("Enter MPEG encoded output file name <%s%s>: ", +- original_file_name, DFLT_EXT); +- #endif +- +- gets(encoded_file_name); +- +- if (encoded_file_name[0] == NULL_CHAR) { +- #ifdef MS_DOS +- strcpy(encoded_file_name, temp_str); +- #else +- strcat(strcpy(encoded_file_name, original_file_name), DFLT_EXT); +- #endif +- } +- +- +- printf(">>> MPEG encoded output file name is: %s\n", encoded_file_name); +- +- open_bit_stream_w(&bs, encoded_file_name, BUFFER_SIZE); +- +- if ((soundPosition = aiff_read_headers(musicin, &pcm_aiff_data)) != -1) { +- +- printf(">>> Using Audio IFF sound file headers\n"); +- +- aiff_check(original_file_name, &pcm_aiff_data, &info->version); +- +- if (fseek(musicin, soundPosition, SEEK_SET) != 0) { +- printf("Could not seek to PCM sound data in \"%s\".\n", +- original_file_name); +- exit(1); +- } +- +- info->sampling_frequency = +- SmpFrqIndex((long)pcm_aiff_data.sampleRate, &info->version); +- printf(">>> %f Hz sampling frequency selected\n", +- pcm_aiff_data.sampleRate); +- +- /* Determine number of samples in sound file */ +- #ifndef MS_DOS +- *num_samples = pcm_aiff_data.numChannels * +- pcm_aiff_data.numSampleFrames; +- #else +- *num_samples = (long)(pcm_aiff_data.numChannels) * +- (long)(pcm_aiff_data.numSampleFrames); +- #endif +- +- } +- else { /* Not using Audio IFF sound file headers. */ +- +- printf("What is the sampling frequency? <44100>[Hz]: "); +- gets(t); +- freq = atol(t); +- switch (freq) { +- case 48000 : info->sampling_frequency = 1; +- printf(">>> %ld Hz sampling freq selected\n", freq); +- break; +- case 44100 : info->sampling_frequency = 0; +- printf(">>> %ld Hz sampling freq selected\n", freq); +- break; +- case 32000 : info->sampling_frequency = 2; +- printf(">>> %ld Hz sampling freq selected\n", freq); +- break; +- case 24000 : info->sampling_frequency = 1; +- info->version = MPEG_PHASE2_LSF; +- printf(">>> %ld Hz sampling freq selected\n", freq); +- break; +- case 22050 : info->sampling_frequency = 0; +- info->version = MPEG_PHASE2_LSF; +- printf(">>> %ld Hz sampling freq selected\n", freq); +- break; +- case 16000 : info->sampling_frequency = 2; +- info->version = MPEG_PHASE2_LSF; +- printf(">>> %ld Hz sampling freq selected\n", freq); +- break; +- default: info->sampling_frequency = 0; +- printf(">>> Default 44.1 kHz samp freq selected\n"); +- } +- printf(">>> Encoding algorithm is %s\n", version_names[info->version]); +- if (fseek(musicin, 0, SEEK_SET) != 0) { +- printf("Could not seek to PCM sound data in \"%s\".\n", +- original_file_name); +- exit(1); +- } +- +- /* Declare sound file to have "infinite" number of samples. */ +- *num_samples = MAX_U_32_NUM; +- +- } +- +- printf("Which layer do you want to use?\n"); +- printf("Available: Layer (1), Layer (<2>), Layer (3): "); +- gets(t); +- switch(*t){ +- case '1': info->lay = 1; printf(">>> Using Layer %s\n",t); break; +- case '2': info->lay = 2; printf(">>> Using Layer %s\n",t); break; +- case '3': info->lay = 3; printf(">>> Using Layer %s\n",t); break; +- default: info->lay = 2; printf(">>> Using default Layer 2\n"); break; +- } +- +- printf("Which mode do you want?\n"); +- printf("Available: (<s>)tereo, (j)oint stereo, "); +- printf("(d)ual channel, s(i)ngle Channel: "); +- gets(t); +- switch(*t){ +- case 's': +- case 'S': +- info->mode = MPG_MD_STEREO; info->mode_ext = 0; +- printf(">>> Using mode %s\n",t); +- break; +- case 'j': +- case 'J': +- info->mode = MPG_MD_JOINT_STEREO; +- printf(">>> Using mode %s\n",t); +- break; +- case 'd': +- case 'D': +- info->mode = MPG_MD_DUAL_CHANNEL; info->mode_ext = 0; +- printf(">>> Using mode %s\n",t); +- break; +- case 'i': +- case 'I': +- info->mode = MPG_MD_MONO; info->mode_ext = 0; +- printf(">>> Using mode %s\n",t); +- break; +- default: +- info->mode = MPG_MD_STEREO; info->mode_ext = 0; +- printf(">>> Using default stereo mode\n"); +- break; +- } +- +- printf("Which psychoacoustic model do you want to use? <1>: "); +- gets(t); +- model = atoi(t); +- if (model > 2 || model < 1) { +- printf(">>> Default model 1 selected\n"); +- *psy = 1; +- } +- else { +- *psy = model; +- printf(">>> Using psychoacoustic model %d\n", model); +- } +- +- /* set default bitrate to highest allowed, which is index 14 */ +- brt = bitrate[info->version][info->lay-1][14]; +- printf( "What is the total bitrate? <%u>[kbps]: ", brt ); +- gets( t ); +- brt = atoi( t ); +- if ( brt == 0 ) +- j = 15; +- else +- j = 0; +- while ( j < 15 ) +- { +- if ( bitrate[info->version][info->lay-1][j] == brt ) +- break; +- j++; +- } +- if ( j == 15 ) +- { +- brt = bitrate[info->version][info->lay-1][14]; +- printf( ">>> Using default %u kbps\n", brt ); +- info->bitrate_index = 14; +- } +- else +- { +- info->bitrate_index = j; +- printf( ">>> Bitrate = %d kbps\n", bitrate[info->version][info->lay-1][j] ); +- } +- +- printf("What type of de-emphasis should the decoder use?\n"); +- printf("Available: (<n>)one, (5)0/15 microseconds, (c)citt j.17: "); +- gets(t); +- if (*t != 'n' && *t != '5' && *t != 'c') { +- printf(">>> Using default no de-emphasis\n"); +- info->emphasis = 0; +- } +- else { +- if (*t == 'n') info->emphasis = 0; +- else if (*t == '5') info->emphasis = 1; +- else if (*t == 'c') info->emphasis = 3; +- printf(">>> Using de-emphasis %s\n",t); +- } +- +- /* Start 2. Part changes for CD Ver 3.2; jsp; 22-Aug-1991 */ +- +- printf("Do you want to set the private bit? (y/<n>): "); +- gets(t); +- if (*t == 'y' || *t == 'Y') info->extension = 1; +- else info->extension = 0; +- if(info->extension) printf(">>> Private bit set\n"); +- else printf(">>> Private bit not set\n"); +- +- /* End changes for CD Ver 3.2; jsp; 22-Aug-1991 */ +- +- printf("Do you want error protection? (y/<n>): "); +- gets(t); +- if (*t == 'y' || *t == 'Y') info->error_protection = TRUE; +- else info->error_protection = FALSE; +- if(info->error_protection) printf(">>> Error protection used\n"); +- else printf(">>> Error protection not used\n"); +- +- printf("Is the material copyrighted? (y/<n>): "); +- gets(t); +- if (*t == 'y' || *t == 'Y') info->copyright = 1; +- else info->copyright = 0; +- if(info->copyright) printf(">>> Copyrighted material\n"); +- else printf(">>> Material not copyrighted\n"); +- +- printf("Is this the original? (y/<n>): "); +- gets(t); +- if (*t == 'y' || *t == 'Y') info->original = 1; +- else info->original = 0; +- if(info->original) printf(">>> Original material\n"); +- else printf(">>> Material not original\n"); +- +- printf("Do you wish to exit (last chance before encoding)? (y/<n>): "); +- gets(t); +- if (*t == 'y' || *t == 'Y') exit(0); +- } +- +- /************************************************************************ +- * +- * parse_args +- * +- * PURPOSE: Sets encoding parameters to the specifications of the +- * command line. Default settings are used for parameters +- * not specified in the command line. +- * +- * SEMANTICS: The command line is parsed according to the following +- * syntax: +- * +- * -l is followed by the layer number +- * -m is followed by the mode +- * -p is followed by the psychoacoustic model number +- * -s is followed by the sampling rate +- * -b is followed by the total bitrate, irrespective of the mode +- * -d is followed by the emphasis flag +- * -c is followed by the copyright/no_copyright flag +- * -o is followed by the original/not_original flag +- * -e is followed by the error_protection on/off flag +- * +- * If the input file is in AIFF format, the sampling frequency is read +- * from the AIFF header. +- * +- * The input and output filenames are read into #inpath# and #outpath#. +- * +- ************************************************************************/ +- +- void +- parse_args(argc, argv, fr_ps, psy, num_samples, inPath, outPath) +- int argc; +- char **argv; +- frame_params *fr_ps; +- int *psy; +- unsigned long *num_samples; +- char inPath[MAX_NAME_SIZE]; +- char outPath[MAX_NAME_SIZE]; +- { +- FLOAT srate; +- int brate; +- layer *info = fr_ps->header; +- int err = 0, i = 0; +- IFF_AIFF pcm_aiff_data; +- long samplerate; +- long soundPosition; +- +- /* preset defaults */ +- inPath[0] = '\0'; outPath[0] = '\0'; +- info->lay = DFLT_LAY; +- switch(DFLT_MOD) { +- case 's': info->mode = MPG_MD_STEREO; info->mode_ext = 0; break; +- case 'd': info->mode = MPG_MD_DUAL_CHANNEL; info->mode_ext=0; break; +- case 'j': info->mode = MPG_MD_JOINT_STEREO; break; +- case 'm': info->mode = MPG_MD_MONO; info->mode_ext = 0; break; +- default: +- fprintf(stderr, "%s: Bad mode dflt %c\n", programName, DFLT_MOD); +- abort(); +- } +- *psy = DFLT_PSY; +- if((info->sampling_frequency = SmpFrqIndex((long)(1000*DFLT_SFQ), &info->version)) < 0) { +- fprintf(stderr, "%s: bad sfrq default %.2f\n", programName, DFLT_SFQ); +- abort(); +- } +- info->bitrate_index = 14; +- brate = 0; +- switch(DFLT_EMP) { +- case 'n': info->emphasis = 0; break; +- case '5': info->emphasis = 1; break; +- case 'c': info->emphasis = 3; break; +- default: +- fprintf(stderr, "%s: Bad emph dflt %c\n", programName, DFLT_EMP); +- abort(); +- } +- info->copyright = 0; info->original = 0; info->error_protection = FALSE; +- +- /* process args */ +- while(++i<argc && err == 0) { +- char c, *token, *arg, *nextArg; +- int argUsed; +- +- token = argv[i]; +- if(*token++ == '-') { +- if(i+1 < argc) nextArg = argv[i+1]; +- else nextArg = ""; +- argUsed = 0; +- while( (c = *token++) ) { +- if(*token /* NumericQ(token) */) arg = token; +- else arg = nextArg; +- switch(c) { +- case 'l': info->lay = atoi(arg); argUsed = 1; +- if(info->lay<1 || info->lay>3) { +- fprintf(stderr,"%s: -l layer must be 1, 2, or 3, not %s\n", +- programName, arg); +- err = 1; +- } +- break; +- case 'm': argUsed = 1; +- if (*arg == 's') +- { info->mode = MPG_MD_STEREO; info->mode_ext = 0; } +- else if (*arg == 'd') +- { info->mode = MPG_MD_DUAL_CHANNEL; info->mode_ext=0; } +- else if (*arg == 'j') +- { info->mode = MPG_MD_JOINT_STEREO; } +- else if (*arg == 'm') +- { info->mode = MPG_MD_MONO; info->mode_ext = 0; } +- else { +- fprintf(stderr,"%s: -m mode must be s/d/j/m not %s\n", +- programName, arg); +- err = 1; +- } +- break; +- case 'p': *psy = atoi(arg); argUsed = 1; +- if(*psy<1 || *psy>2) { +- fprintf(stderr,"%s: -p model must be 1 or 2, not %s\n", +- programName, arg); +- err = 1; +- } +- break; +- +- case 's': +- argUsed = 1; +- srate = atof( arg ); +- /* samplerate = rint( 1000.0 * srate ); $A */ +- samplerate = (long) (( 1000.0 * srate ) + 0.5); +- if( (info->sampling_frequency = +- SmpFrqIndex((long) samplerate, &info->version)) < 0 ) +- err = 1; +- break; +- +- case 'b': +- argUsed = 1; +- brate = atoi(arg); +- break; +- case 'd': argUsed = 1; +- if (*arg == 'n') info->emphasis = 0; +- else if (*arg == '5') info->emphasis = 1; +- else if (*arg == 'c') info->emphasis = 3; +- else { +- fprintf(stderr,"%s: -d emp must be n/5/c not %s\n", +- programName, arg); +- err = 1; +- } +- break; +- case 'c': info->copyright = 1; break; +- case 'o': info->original = 1; break; +- case 'e': info->error_protection = TRUE; break; +- default: fprintf(stderr,"%s: unrec option %c\n", +- programName, c); +- err = 1; break; +- } +- if(argUsed) { +- if(arg == token) token = ""; /* no more from token */ +- else ++i; /* skip arg we used */ +- arg = ""; argUsed = 0; +- } +- } +- } +- else { +- if(inPath[0] == '\0') strcpy(inPath, argv[i]); +- else if(outPath[0] == '\0') strcpy(outPath, argv[i]); +- else { +- fprintf(stderr,"%s: excess arg %s\n", programName, argv[i]); +- err = 1; +- } +- } +- } +- +- if(err || inPath[0] == '\0') usage(); /* never returns */ +- +- if(outPath[0] == '\0') { +- #ifdef MS_DOS +- /* replace old extension with new one, 1992-08-19, 1995-06-12 shn */ +- new_ext(inPath, DFLT_EXT, outPath); +- #else +- strcpy(outPath, inPath); +- strcat(outPath, DFLT_EXT); +- #endif +- } +- +- if ((musicin = fopen(inPath, "rb")) == NULL) { +- printf("Could not find \"%s\".\n", inPath); +- exit(1); +- } +- +- open_bit_stream_w(&bs, outPath, BUFFER_SIZE); +- +- if ((soundPosition = aiff_read_headers(musicin, &pcm_aiff_data)) != -1) { +- +- printf(">>> Using Audio IFF sound file headers\n"); +- +- aiff_check(inPath, &pcm_aiff_data, &info->version); +- +- if (fseek(musicin, soundPosition, SEEK_SET) != 0) { +- printf("Could not seek to PCM sound data in \"%s\".\n", inPath); +- exit(1); +- } +- +- info->sampling_frequency = SmpFrqIndex((long)pcm_aiff_data.sampleRate, &info->version); +- printf(">>> %f Hz sampling frequency selected\n", +- pcm_aiff_data.sampleRate); +- +- /* Determine number of samples in sound file */ +- #ifndef MS_DOS +- *num_samples = pcm_aiff_data.numChannels * +- pcm_aiff_data.numSampleFrames; +- #else +- *num_samples = (long)(pcm_aiff_data.numChannels) * +- (long)(pcm_aiff_data.numSampleFrames); +- #endif +- if ( pcm_aiff_data.numChannels == 1 ) { +- info->mode = MPG_MD_MONO; +- info->mode_ext = 0; +- } +- } +- else { /* Not using Audio IFF sound file headers. */ +- +- if (fseek(musicin, 0, SEEK_SET) != 0) { +- printf("Could not seek to PCM sound data in \"%s\".\n", inPath); +- exit(1); +- } +- +- /* Declare sound file to have "infinite" number of samples. */ +- *num_samples = MAX_U_32_NUM; +- +- } +- if ( brate == 0 ) +- brate = bitrate[info->version][info->lay-1][14]; +- if( (info->bitrate_index = BitrateIndex(info->lay, brate, info->version)) < 0) err=1; +- if(err || inPath[0] == '\0') usage(); /* never returns */ +- +- } +- +- /************************************************************************ +- * +- * print_config +- * +- * PURPOSE: Prints the encoding parameters used +- * +- ************************************************************************/ +- +- void print_config( frame_params *fr_ps, int *psy, char *inPath, char *outPath) +- { +- layer *info = fr_ps->header; +- +- printf("Encoding configuration:\n"); +- printf("Algorithm=%s\n", version_names[info->version]); +- if(info->mode != MPG_MD_JOINT_STEREO) +- printf("Layer=%s mode=%s extn=%d psy model=%d\n", +- layer_names[info->lay-1], mode_names[info->mode], +- info->mode_ext, *psy); +- else printf("Layer=%s mode=%s extn=data dependant psy model=%d\n", +- layer_names[info->lay-1], mode_names[info->mode], *psy); +- printf("samp frq=%.1f kHz total bitrate=%d kbps\n", +- s_freq[info->version][info->sampling_frequency], +- bitrate[info->version][info->lay-1][info->bitrate_index]); +- printf("de-emph=%d c/right=%d orig=%d errprot=%s\n", +- info->emphasis, info->copyright, info->original, +- ((info->error_protection) ? "on" : "off")); +- printf("input file: '%s' output file: '%s'\n", inPath, outPath); +- } +- +- +- +- +- +- /************************************************************************ +- * +- * main +- * +- * PURPOSE: MPEG I Encoder supporting layers 1 and 2, and 3, with +- * psychoacoustic models 1 (MUSICAM) and 2 (AT&T) +- * +- * SEMANTICS: One overlapping frame of audio of up to 2 channels are +- * processed at a time in the following order: +- * (associated routines are in parentheses) +- * +- * 1. Filter sliding window of data to get 32 subband +- * samples per channel. +- * (window_subband,filter_subband) +- * +- * 2. If joint stereo mode, combine left and right channels +- * for subbands above #jsbound#. +- * (*_combine_LR) +- * +- * 3. Calculate scalefactors for the frame, and if layer 2, +- * also calculate scalefactor select information. +- * (*_scale_factor_calc) +- * +- * 4. Calculate psychoacoustic masking levels using selected +- * psychoacoustic model. +- * (*_Psycho_One, psycho_anal) +- * +- * 5. Perform iterative bit allocation for subbands with low +- * mask_to_noise ratios using masking levels from step 4. +- * (*_main_bit_allocation) +- * +- * 6. If error protection flag is active, add redundancy for +- * error protection. +- * (*_CRC_calc) +- * +- * 7. Pack bit allocation, scalefactors, and scalefactor select +- * information (layer 2) onto bitstream. +- * (*_encode_bit_alloc,*_encode_scale,II_transmission_pattern) +- * +- * 8. Quantize subbands and pack them into bitstream +- * (*_subband_quantization, *_sample_encoding) +- * +- ************************************************************************/ +- +- int frameNum=0; +- +- void main(argc, argv) +- int argc; +- char **argv; +- { +- typedef double SBS[2][3][SCALE_BLOCK][SBLIMIT]; +- SBS FAR *sb_sample; +- L3SBS FAR *l3_sb_sample; +- typedef double JSBS[3][SCALE_BLOCK][SBLIMIT]; +- JSBS FAR *j_sample; +- typedef double IN[2][HAN_SIZE]; +- IN FAR *win_que; +- typedef unsigned int SUB[2][3][SCALE_BLOCK][SBLIMIT]; +- SUB FAR *subband; +- +- frame_params fr_ps; +- layer info; +- char original_file_name[MAX_NAME_SIZE]; +- char encoded_file_name[MAX_NAME_SIZE]; +- short FAR **win_buf; +- static short FAR buffer[2][1152]; +- static unsigned int bit_alloc[2][SBLIMIT], scfsi[2][SBLIMIT]; +- static unsigned int scalar[2][3][SBLIMIT], j_scale[3][SBLIMIT]; +- static double FAR ltmin[2][SBLIMIT], lgmin[2][SBLIMIT], max_sc[2][SBLIMIT]; +- FLOAT snr32[32]; +- short sam[2][1344]; /* was [1056]; */ +- int whole_SpF, extra_slot = 0; +- double avg_slots_per_frame, frac_SpF, slot_lag; +- int model, stereo, error_protection; +- static unsigned int crc; +- int i, j, k, adb; +- unsigned long bitsPerSlot, samplesPerFrame; +- unsigned long frameBits, sentBits = 0; +- unsigned long num_samples; +- +- #ifdef MACINTOSH +- argc = ccommand( &argv ); +- #endif +- +- +- /* Most large variables are declared dynamically to ensure +- compatibility with smaller machines */ +- +- sb_sample = (SBS FAR *) mem_alloc(sizeof(SBS), "sb_sample"); +- l3_sb_sample = (L3SBS FAR *) mem_alloc(sizeof(SBS), "l3_sb_sample"); +- j_sample = (JSBS FAR *) mem_alloc(sizeof(JSBS), "j_sample"); +- win_que = (IN FAR *) mem_alloc(sizeof(IN), "Win_que"); +- subband = (SUB FAR *) mem_alloc(sizeof(SUB),"subband"); +- win_buf = (short FAR **) mem_alloc(sizeof(short *)*2, "win_buf"); +- +- /* clear buffers */ +- memset((char *) buffer, 0, sizeof(buffer)); +- memset((char *) bit_alloc, 0, sizeof(bit_alloc)); +- memset((char *) scalar, 0, sizeof(scalar)); +- memset((char *) j_scale, 0, sizeof(j_scale)); +- memset((char *) scfsi, 0, sizeof(scfsi)); +- memset((char *) ltmin, 0, sizeof(ltmin)); +- memset((char *) lgmin, 0, sizeof(lgmin)); +- memset((char *) max_sc, 0, sizeof(max_sc)); +- memset((char *) snr32, 0, sizeof(snr32)); +- memset((char *) sam, 0, sizeof(sam)); +- +- fr_ps.header = &info; +- fr_ps.tab_num = -1; /* no table loaded */ +- fr_ps.alloc = NULL; +- info.version = MPEG_AUDIO_ID; /* Default: MPEG-1 */ +- +- programName = argv[0]; +- if(argc==1) /* no command-line args */ +- obtain_parameters(&fr_ps, &model, &num_samples, +- original_file_name, encoded_file_name); +- else +- parse_args(argc, argv, &fr_ps, &model, &num_samples, +- original_file_name, encoded_file_name); +- print_config(&fr_ps, &model, +- original_file_name, encoded_file_name); +- +- hdr_to_frps(&fr_ps); +- stereo = fr_ps.stereo; +- error_protection = info.error_protection; +- +- if (info.lay == 1) +- { bitsPerSlot = 32; samplesPerFrame = 384; } +- else +- if ( info.lay == 2 ) +- { bitsPerSlot = 8; samplesPerFrame = 1152; } +- else +- { /* layer 3 */ +- bitsPerSlot = 8; +- samplesPerFrame = info.version == 1 ? 1152 : 576; +- +- /* Apologize for missing features */ +- if ( info.mode == MPG_MD_JOINT_STEREO ) +- { +- fprintf( stderr, "Sorry, joint stereo not yet available for layer3\n" ); +- exit( 1 ); +- } +- #if 0 +- if ( info.version != MPEG_AUDIO_ID ) +- { +- fprintf( stderr, "Sorry, MPEG2-LSF not yet available for layer3\n" ); +- exit( 1 ); +- } +- #endif +- if ( model != 2 ) +- { +- fprintf( stderr, "Sorry, psycho model 1 not available for layer3\n" ); +- exit( 1 ); +- } +- } +- /* Figure average number of 'slots' per frame. */ +- /* Bitrate means TOTAL for both channels, not per side. */ +- avg_slots_per_frame = ((double)samplesPerFrame / +- s_freq[info.version][info.sampling_frequency]) * +- ((double)bitrate[info.version][info.lay-1][info.bitrate_index] / +- (double)bitsPerSlot); +- whole_SpF = (int) avg_slots_per_frame; +- printf("slots/frame = %d\n",whole_SpF); +- frac_SpF = avg_slots_per_frame - (double)whole_SpF; +- slot_lag = -frac_SpF; +- printf("frac SpF=%.3f, tot bitrate=%d kbps, s freq=%.1f kHz\n", +- frac_SpF, bitrate[info.version][info.lay-1][info.bitrate_index], +- s_freq[info.version][info.sampling_frequency]); +- +- if (frac_SpF != 0) +- printf("Fractional number of slots, padding required\n"); +- else info.padding = 0; +- +- while ( get_audio(musicin, buffer, num_samples, stereo, &info) > 0 ) +- { +- fprintf(stderr, "{%4lu}", frameNum++); fflush(stderr); +- win_buf[0] = &buffer[0][0]; +- win_buf[1] = &buffer[1][0]; +- if (frac_SpF != 0) { +- if (slot_lag > (frac_SpF-1.0) ) { +- slot_lag -= frac_SpF; +- extra_slot = 0; +- info.padding = 0; +- /* printf("No padding for this frame\n"); */ +- } +- else { +- extra_slot = 1; +- info.padding = 1; +- slot_lag += (1-frac_SpF); +- /* printf("Padding for this frame\n"); */ +- } +- } +- adb = (whole_SpF+extra_slot) * bitsPerSlot; +- +- switch (info.lay) +- { +- +- /***************************** Layer I **********************************/ +- +- case 1 : +- for (j=0;j<SCALE_BLOCK;j++) +- for (k=0;k<stereo;k++) { +- window_subband(&win_buf[k], &(*win_que)[k][0], k); +- filter_subband(&(*win_que)[k][0], &(*sb_sample)[k][0][j][0]); +- } +- +- I_scale_factor_calc(*sb_sample, scalar, stereo); +- if(fr_ps.actual_mode == MPG_MD_JOINT_STEREO) { +- I_combine_LR(*sb_sample, *j_sample); +- I_scale_factor_calc(j_sample, &j_scale, 1); +- } +- +- put_scale(scalar, &fr_ps, max_sc); +- +- if (model == 1) I_Psycho_One(buffer, max_sc, ltmin, &fr_ps); +- else { +- for (k=0;k<stereo;k++) { +- psycho_anal(&buffer[k][0],&sam[k][0], k, info.lay, snr32, +- (FLOAT)s_freq[info.version][info.sampling_frequency]*1000); +- for (i=0;i<SBLIMIT;i++) ltmin[k][i] = (double) snr32[i]; +- } +- } +- +- I_main_bit_allocation(ltmin, bit_alloc, &adb, &fr_ps); +- +- if (error_protection) I_CRC_calc(&fr_ps, bit_alloc, &crc); +- +- encode_info(&fr_ps, &bs); +- +- if (error_protection) encode_CRC(crc, &bs); +- +- I_encode_bit_alloc(bit_alloc, &fr_ps, &bs); +- I_encode_scale(scalar, bit_alloc, &fr_ps, &bs); +- I_subband_quantization(scalar, *sb_sample, j_scale, *j_sample, +- bit_alloc, *subband, &fr_ps); +- I_sample_encoding(*subband, bit_alloc, &fr_ps, &bs); +- for (i=0;i<adb;i++) put1bit(&bs, 0); +- break; +- +- /***************************** Layer 2 **********************************/ +- +- case 2 : +- for (i=0;i<3;i++) for (j=0;j<SCALE_BLOCK;j++) +- for (k=0;k<stereo;k++) { +- window_subband(&win_buf[k], &(*win_que)[k][0], k); +- filter_subband(&(*win_que)[k][0], &(*sb_sample)[k][i][j][0]); +- } +- +- II_scale_factor_calc(*sb_sample, scalar, stereo, fr_ps.sblimit); +- pick_scale(scalar, &fr_ps, max_sc); +- if(fr_ps.actual_mode == MPG_MD_JOINT_STEREO) { +- II_combine_LR(*sb_sample, *j_sample, fr_ps.sblimit); +- II_scale_factor_calc(j_sample, &j_scale, 1, fr_ps.sblimit); +- } /* this way we calculate more mono than we need */ +- /* but it is cheap */ +- +- if (model == 1) II_Psycho_One(buffer, max_sc, ltmin, &fr_ps); +- else { +- for (k=0;k<stereo;k++) { +- psycho_anal(&buffer[k][0],&sam[k][0], k, +- info.lay, snr32, +- (FLOAT)s_freq[info.version][info.sampling_frequency]*1000); +- for (i=0;i<SBLIMIT;i++) ltmin[k][i] = (double) snr32[i]; +- } +- } +- +- II_transmission_pattern(scalar, scfsi, &fr_ps); +- II_main_bit_allocation(ltmin, scfsi, bit_alloc, &adb, &fr_ps); +- +- if (error_protection) +- II_CRC_calc(&fr_ps, bit_alloc, scfsi, &crc); +- +- encode_info(&fr_ps, &bs); +- +- if (error_protection) encode_CRC(crc, &bs); +- +- II_encode_bit_alloc(bit_alloc, &fr_ps, &bs); +- II_encode_scale(bit_alloc, scfsi, scalar, &fr_ps, &bs); +- II_subband_quantization(scalar, *sb_sample, j_scale, +- *j_sample, bit_alloc, *subband, &fr_ps); +- II_sample_encoding(*subband, bit_alloc, &fr_ps, &bs); +- for (i=0;i<adb;i++) put1bit(&bs, 0); +- break; +- +- /***************************** Layer 3 **********************************/ +- +- case 3: +- { +- /* +- large "auto" vars are static due to the Macintosh linker +- */ +- static double xr[2][2][576]; +- static double xr_dec[2][2][576]; +- static double pe[2][2]; +- static int l3_enc[2][2][576]; +- static III_psy_ratio ratio; +- static III_side_info_t l3_side; +- static III_scalefac_t scalefac; +- int gr, mode_gr, ch; +- int mean_bits, sideinfo_len; +- +- int bitsPerFrame = 8 * whole_SpF + (info.padding * 8); +- mode_gr = (info.version == 1) ? 2 : 1; +- +- /* +- determine the mean bitrate for main data +- */ +- sideinfo_len = 32; +- if ( info.version == 1 ) +- { /* MPEG 1 */ +- if ( stereo == 1 ) +- sideinfo_len += 136; +- else +- sideinfo_len += 256; +- } +- else +- { /* MPEG 2 */ +- if ( stereo == 1 ) +- sideinfo_len += 72; +- else +- sideinfo_len += 136; +- } +- if ( info.error_protection ) +- sideinfo_len += 16; +- mean_bits = (bitsPerFrame - sideinfo_len) / mode_gr; +- +- /* +- psychoacoustic model +- */ +- for ( gr = 0; gr < mode_gr; gr++ ) +- for ( ch = 0; ch < stereo; ch++ ) +- { +- L3psycho_anal( &buffer[ch][gr*576], &sam[ch][0], ch, info.lay, +- snr32, s_freq[info.version][info.sampling_frequency] * 1000.0, +- &ratio.l[gr][ch][0], &ratio.s[gr][ch][0], +- &pe[gr][ch], &l3_side.gr[gr].ch[ch].tt ); +- } +- +- /* +- polyphase filtering +- */ +- for( gr = 0; gr < mode_gr; gr++ ) +- for ( ch = 0; ch < stereo; ch++ ) +- for ( j = 0; j < 18; j++ ) +- { +- window_subband( &win_buf[ch], &(*win_que)[ch][0], ch ); +- filter_subband( &(*win_que)[ch][0], &(*l3_sb_sample)[ch][gr+1][j][0] ); +- } +- +- /* +- apply mdct to the polyphase outputs +- */ +- mdct_sub( l3_sb_sample, xr, stereo, &l3_side, mode_gr ); +- #if 0 +- delay( xr, stereo ); +- #endif +- /* +- bit and noise allocation +- */ +- iteration_loop( pe, xr, &ratio, &l3_side, l3_enc, mean_bits, +- stereo, xr_dec, &scalefac, &fr_ps, 0, bitsPerFrame ); +- +- /* +- write the frame to the bitstream +- */ +- III_format_bitstream( bitsPerFrame, &fr_ps, l3_enc, &l3_side, &scalefac, &bs, +- xr, NULL, 0 ); +- } +- break; /* end of layer 3 */ +- +- +- } /* end switch */ +- +- frameBits = sstell( &bs ) - sentBits; +- if ( frameBits % bitsPerSlot ) /* a program failure */ +- fprintf( stderr, "Sent %ld bits = %ld slots plus %ld\n", +- frameBits, frameBits/bitsPerSlot, +- frameBits%bitsPerSlot ); +- sentBits += frameBits; +- +- } +- +- if ( info.lay == 3 ) +- III_FlushBitstream(); +- +- close_bit_stream_w( &bs ); +- +- printf("Avg slots/frame = %.3f; b/smp = %.2f; br = %.3f kbps\n", +- (FLOAT) sentBits / (frameNum * bitsPerSlot), +- (FLOAT) sentBits / (frameNum * samplesPerFrame), +- (FLOAT) sentBits / (frameNum * samplesPerFrame) * +- s_freq[info.version][info.sampling_frequency]); +- +- if (fclose(musicin) != 0){ +- printf("Could not close \"%s\".\n", original_file_name); +- exit(2); +- } +- +- #ifdef MACINTOSH +- set_mac_file_attr( encoded_file_name, VOL_REF_NUM, CREATOR_ENCODE, +- FILETYPE_ENCODE ); +- #endif +- +- printf("Encoding of \"%s\" with psychoacoustic model %d is finished\n", +- original_file_name, model); +- printf("The MPEG encoded output file name is \"%s\"\n", +- encoded_file_name); +- exit(0); +- } +- +- /************************************************************************ +- * +- * usage +- * +- * PURPOSE: Writes command line syntax to the file specified by #stderr# +- * +- ************************************************************************/ +- +- void usage() /* print syntax & exit */ +- { +- fprintf(stderr, +- "usage: %s queries for all arguments, or\n", +- programName); +- fprintf(stderr, +- " %s [-l lay][-m mode][-p psy][-s sfrq][-b br][-d emp]\n", +- programName); +- fprintf(stderr, +- " [-c][-o][-e] inputPCM [outBS]\n"); +- fprintf(stderr,"where\n"); +- fprintf(stderr," -l lay use layer <lay> coding (dflt %4u)\n",DFLT_LAY); +- fprintf(stderr," -m mode channel mode : s/d/j/m (dflt %4c)\n",DFLT_MOD); +- fprintf(stderr," -p psy psychoacoustic model 1/2 (dflt %4u)\n",DFLT_PSY); +- fprintf(stderr," -s sfrq input smpl rate in kHz (dflt %4.1f)\n",DFLT_SFQ); +- fprintf(stderr," -b br total bitrate in kbps (dflt highest)\n"); +- fprintf(stderr," -d emp de-emphasis n/5/c (dflt %4c)\n",DFLT_EMP); +- fprintf(stderr," -c mark as copyright\n"); +- fprintf(stderr," -o mark as original\n"); +- fprintf(stderr," -e add error protection\n"); +- fprintf(stderr," inputPCM input PCM sound file (standard or AIFF)\n"); +- fprintf(stderr," outBS output bit stream of encoded audio (dflt inName+%s)\n", +- DFLT_EXT); +- exit(1); +- } +- +- /************************************************************************ +- * +- * aiff_check +- * +- * PURPOSE: Checks AIFF header information to make sure it is valid. +- * Exits if not. +- * +- ************************************************************************/ +- +- void aiff_check( char *file_name, IFF_AIFF *pcm_aiff_data, int *version) +- { +- if (pcm_aiff_data->sampleType != IFF_ID_SSND) { +- printf("Sound data is not PCM in \"%s\".\n", file_name); +- exit(1); +- } +- +- if(SmpFrqIndex((long)pcm_aiff_data->sampleRate, version) < 0) { +- printf("in \"%s\".\n", file_name); +- exit(1); +- } +- +- if (pcm_aiff_data->sampleSize != sizeof(short) * BITS_IN_A_BYTE) { +- printf("Sound data is not %d bits in \"%s\".\n", +- sizeof(short) * BITS_IN_A_BYTE, file_name); +- exit(1); +- } +- +- if (pcm_aiff_data->numChannels != MONO && +- pcm_aiff_data->numChannels != STEREO) { +- printf("Sound data is not mono or stereo in \"%s\".\n", file_name); +- exit(1); +- } +- +- if (pcm_aiff_data->blkAlgn.blockSize != 0) { +- printf("Block size is not %d bytes in \"%s\".\n", 0, file_name); +- exit(1); +- } +- +- if (pcm_aiff_data->blkAlgn.offset != 0) { +- printf("Block offset is not %d bytes in \"%s\".\n", 0, file_name); +- exit(1); +- } +- } +--- 0 ---- +diff -r -c -N encoder/portableio.c lame3.50/portableio.c +*** encoder/portableio.c Wed Jan 22 02:43:12 1997 +--- lame3.50/portableio.c Mon Oct 11 14:28:20 1999 +*************** +*** 59,66 **** + + + int +! ReadByte(fp) +! FILE *fp; + { + int result; + +--- 59,65 ---- + + + int +! ReadByte(FILE *fp) + { + int result; + +*************** +*** 72,79 **** + + + int +! Read16BitsLowHigh(fp) +! FILE *fp; + { + int first, second, result; + +--- 71,77 ---- + + + int +! Read16BitsLowHigh(FILE *fp) + { + int first, second, result; + +*************** +*** 90,97 **** + + + int +! Read16BitsHighLow(fp) +! FILE *fp; + { + int first, second, result; + +--- 88,94 ---- + + + int +! Read16BitsHighLow(FILE *fp) + { + int first, second, result; + +*************** +*** 108,125 **** + + + void +! Write8Bits(fp, i) +! FILE *fp; +! int i; + { + putc(i&0xff,fp); + } + + + void +! Write16BitsLowHigh(fp, i) +! FILE *fp; +! int i; + { + putc(i&0xff,fp); + putc((i>>8)&0xff,fp); +--- 105,118 ---- + + + void +! Write8Bits(FILE *fp, int i) + { + putc(i&0xff,fp); + } + + + void +! Write16BitsLowHigh(FILE *fp, int i) + { + putc(i&0xff,fp); + putc((i>>8)&0xff,fp); +*************** +*** 127,135 **** + + + void +! Write16BitsHighLow(fp, i) +! FILE *fp; +! int i; + { + putc((i>>8)&0xff,fp); + putc(i&0xff,fp); +--- 120,126 ---- + + + void +! Write16BitsHighLow(FILE *fp, int i) + { + putc((i>>8)&0xff,fp); + putc(i&0xff,fp); +*************** +*** 137,144 **** + + + int +! Read24BitsHighLow(fp) +! FILE *fp; + { + int first, second, third; + int result; +--- 128,134 ---- + + + int +! Read24BitsHighLow(FILE *fp) + { + int first, second, third; + int result; +*************** +*** 157,164 **** + + + int +! Read32Bits(fp) +! FILE *fp; + { + int first, second, result; + +--- 147,153 ---- + + + int +! Read32Bits(FILE *fp) + { + int first, second, result; + +*************** +*** 175,182 **** + + + int +! Read32BitsHighLow(fp) +! FILE *fp; + { + int first, second, result; + +--- 164,170 ---- + + + int +! Read32BitsHighLow(FILE *fp) + { + int first, second, result; + +*************** +*** 193,201 **** + + + void +! Write32Bits(fp, i) +! FILE *fp; +! int i; + { + Write16BitsLowHigh(fp,(int)(i&0xffffL)); + Write16BitsLowHigh(fp,(int)((i>>16)&0xffffL)); +--- 181,187 ---- + + + void +! Write32Bits(FILE *fp, int i) + { + Write16BitsLowHigh(fp,(int)(i&0xffffL)); + Write16BitsLowHigh(fp,(int)((i>>16)&0xffffL)); +*************** +*** 203,211 **** + + + void +! Write32BitsLowHigh(fp, i) +! FILE *fp; +! int i; + { + Write16BitsLowHigh(fp,(int)(i&0xffffL)); + Write16BitsLowHigh(fp,(int)((i>>16)&0xffffL)); +--- 189,195 ---- + + + void +! Write32BitsLowHigh(FILE *fp, int i) + { + Write16BitsLowHigh(fp,(int)(i&0xffffL)); + Write16BitsLowHigh(fp,(int)((i>>16)&0xffffL)); +*************** +*** 213,243 **** + + + void +! Write32BitsHighLow(fp, i) +! FILE *fp; +! int i; + { + Write16BitsHighLow(fp,(int)((i>>16)&0xffffL)); + Write16BitsHighLow(fp,(int)(i&0xffffL)); + } + +! void ReadBytes(fp, p, n) +! FILE *fp; +! char *p; +! int n; + { +! while (!feof(fp) & n-- > 0) + *p++ = getc(fp); + } + +! void ReadBytesSwapped(fp, p, n) +! FILE *fp; +! char *p; +! int n; + { + register char *q = p; + +! while (!feof(fp) & n-- > 0) + *q++ = getc(fp); + + for (q--; p < q; p++, q--){ +--- 197,219 ---- + + + void +! Write32BitsHighLow(FILE *fp, int i) + { + Write16BitsHighLow(fp,(int)((i>>16)&0xffffL)); + Write16BitsHighLow(fp,(int)(i&0xffffL)); + } + +! void ReadBytes(FILE *fp, char *p, int n) + { +! while (!feof(fp) & (n-- > 0)) + *p++ = getc(fp); + } + +! void ReadBytesSwapped(FILE *fp, char *p, int n) + { + register char *q = p; + +! while (!feof(fp) & (n-- > 0)) + *q++ = getc(fp); + + for (q--; p < q; p++, q--){ +*************** +*** 247,265 **** + } + } + +! void WriteBytes(fp, p, n) +! FILE *fp; +! char *p; +! int n; + { + while (n-- > 0) + putc(*p++, fp); + } + +! void WriteBytesSwapped(fp, p, n) +! FILE *fp; +! char *p; +! int n; + { + p += n-1; + while (n-- > 0) +--- 223,235 ---- + } + } + +! void WriteBytes(FILE *fp, char *p, int n) + { + while (n-- > 0) + putc(*p++, fp); + } + +! void WriteBytesSwapped(FILE *fp, char *p, int n) + { + p += n-1; + while (n-- > 0) +*************** +*** 267,274 **** + } + + defdouble +! ReadIeeeFloatHighLow(fp) +! FILE *fp; + { + char bits[kFloatLength]; + +--- 237,243 ---- + } + + defdouble +! ReadIeeeFloatHighLow(FILE *fp) + { + char bits[kFloatLength]; + +*************** +*** 277,284 **** + } + + defdouble +! ReadIeeeFloatLowHigh(fp) +! FILE *fp; + { + char bits[kFloatLength]; + +--- 246,252 ---- + } + + defdouble +! ReadIeeeFloatLowHigh(FILE *fp) + { + char bits[kFloatLength]; + +*************** +*** 287,294 **** + } + + defdouble +! ReadIeeeDoubleHighLow(fp) +! FILE *fp; + { + char bits[kDoubleLength]; + +--- 255,261 ---- + } + + defdouble +! ReadIeeeDoubleHighLow(FILE *fp) + { + char bits[kDoubleLength]; + +*************** +*** 297,304 **** + } + + defdouble +! ReadIeeeDoubleLowHigh(fp) +! FILE *fp; + { + char bits[kDoubleLength]; + +--- 264,270 ---- + } + + defdouble +! ReadIeeeDoubleLowHigh(FILE *fp) + { + char bits[kDoubleLength]; + +*************** +*** 307,314 **** + } + + defdouble +! ReadIeeeExtendedHighLow(fp) +! FILE *fp; + { + char bits[kExtendedLength]; + +--- 273,279 ---- + } + + defdouble +! ReadIeeeExtendedHighLow(FILE *fp) + { + char bits[kExtendedLength]; + +*************** +*** 317,324 **** + } + + defdouble +! ReadIeeeExtendedLowHigh(fp) +! FILE *fp; + { + char bits[kExtendedLength]; + +--- 282,288 ---- + } + + defdouble +! ReadIeeeExtendedLowHigh(FILE *fp) + { + char bits[kExtendedLength]; + +*************** +*** 327,335 **** + } + + void +! WriteIeeeFloatLowHigh(fp, num) +! FILE *fp; +! defdouble num; + { + char bits[kFloatLength]; + +--- 291,297 ---- + } + + void +! WriteIeeeFloatLowHigh(FILE *fp, defdouble num) + { + char bits[kFloatLength]; + +*************** +*** 338,346 **** + } + + void +! WriteIeeeFloatHighLow(fp, num) +! FILE *fp; +! defdouble num; + { + char bits[kFloatLength]; + +--- 300,306 ---- + } + + void +! WriteIeeeFloatHighLow(FILE *fp, defdouble num) + { + char bits[kFloatLength]; + +*************** +*** 349,357 **** + } + + void +! WriteIeeeDoubleLowHigh(fp, num) +! FILE *fp; +! defdouble num; + { + char bits[kDoubleLength]; + +--- 309,315 ---- + } + + void +! WriteIeeeDoubleLowHigh(FILE *fp, defdouble num) + { + char bits[kDoubleLength]; + +*************** +*** 360,368 **** + } + + void +! WriteIeeeDoubleHighLow(fp, num) +! FILE *fp; +! defdouble num; + { + char bits[kDoubleLength]; + +--- 318,324 ---- + } + + void +! WriteIeeeDoubleHighLow(FILE *fp, defdouble num) + { + char bits[kDoubleLength]; + +*************** +*** 371,379 **** + } + + void +! WriteIeeeExtendedLowHigh(fp, num) +! FILE *fp; +! defdouble num; + { + char bits[kExtendedLength]; + +--- 327,333 ---- + } + + void +! WriteIeeeExtendedLowHigh(FILE *fp, defdouble num) + { + char bits[kExtendedLength]; + +*************** +*** 383,391 **** + + + void +! WriteIeeeExtendedHighLow(fp, num) +! FILE *fp; +! defdouble num; + { + char bits[kExtendedLength]; + +--- 337,343 ---- + + + void +! WriteIeeeExtendedHighLow(FILE *fp, defdouble num) + { + char bits[kExtendedLength]; + +diff -r -c -N encoder/psy.c lame3.50/psy.c +*** encoder/psy.c Wed Jan 22 02:43:18 1997 +--- lame3.50/psy.c Wed Dec 31 17:00:00 1969 +*************** +*** 1,454 **** +- /********************************************************************** +- * ISO MPEG Audio Subgroup Software Simulation Group (1996) +- * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- * +- * $Id: psy.c,v 1.1 1996/02/14 04:04:23 rowlands Exp $ +- * +- * $Log: psy.c,v $ +- * Revision 1.1 1996/02/14 04:04:23 rowlands +- * Initial revision +- * +- * Received from Mike Coleman +- **********************************************************************/ +- /********************************************************************** +- * date programmers comment * +- * 2/25/91 Davis Pan start of version 1.0 records * +- * 5/10/91 W. Joseph Carter Ported to Macintosh and Unix. * +- * 7/10/91 Earle Jennings Ported to MsDos. * +- * replace of floats with FLOAT * +- * 2/11/92 W. Joseph Carter Fixed mem_alloc() arg for "absthr". * +- * 7/24/92 M. Iwadare HANN window coefficients modified. * +- * 7/27/92 Masahiro Iwadare Bug fix, FFT modification for Layer 3 * +- * 7/27/92 Masahiro Iwadare Bug fix, "new", "old", and "oldest" * +- * updates * +- * 8/07/92 Mike Coleman Bug fix, read_absthr() * +- * 95/3/21 Jon Rowlands Removed extra debug statements * +- **********************************************************************/ +- +- #include "common.h" +- #include "encoder.h" +- FILE *fpo; /* file pointer */ +- void psycho_anal(buffer,savebuf,chn,lay,snr32,sfreq) +- short int *buffer; +- short int savebuf[1056]; +- int chn, lay; +- FLOAT snr32[32]; +- double sfreq; /* to match prototype : float args are always double */ +- { +- unsigned int i, j, k; +- FLOAT r_prime, phi_prime; +- FLOAT freq_mult, bval_lo, minthres, sum_energy; +- double tb, temp1, temp2, temp3; +- +- /* The static variables "r", "phi_sav", "new", "old" and "oldest" have */ +- /* to be remembered for the unpredictability measure. For "r" and */ +- /* "phi_sav", the first index from the left is the channel select and */ +- /* the second index is the "age" of the data. */ +- +- static int new = 0, old = 1, oldest = 0; +- static int init = 0, flush, sync_flush, syncsize, sfreq_idx; +- +- /* The following static variables are constants. */ +- +- static double nmt = 5.5; +- +- static FLOAT crit_band[27] = {0, 100, 200, 300, 400, 510, 630, 770, +- 920, 1080, 1270,1480,1720,2000,2320, 2700, +- 3150, 3700, 4400,5300,6400,7700,9500,12000, +- 15500,25000,30000}; +- +- static FLOAT bmax[27] = {20.0, 20.0, 20.0, 20.0, 20.0, 17.0, 15.0, +- 10.0, 7.0, 4.4, 4.5, 4.5, 4.5, 4.5, +- 4.5, 4.5, 4.5, 4.5, 4.5, 4.5, 4.5, +- 4.5, 4.5, 4.5, 3.5, 3.5, 3.5}; +- +- /* The following pointer variables point to large areas of memory */ +- /* dynamically allocated by the mem_alloc() function. Dynamic memory */ +- /* allocation is used in order to avoid stack frame or data area */ +- /* overflow errors that otherwise would have occurred at compile time */ +- /* on the Macintosh computer. */ +- +- FLOAT *grouped_c, *grouped_e, *nb, *cb, *ecb, *bc; +- FLOAT *wsamp_r, *wsamp_i, *phi, *energy; +- FLOAT *c, *fthr; +- F32 *snrtmp; +- +- static int *numlines; +- static int *partition; +- static FLOAT *cbval, *rnorm; +- static FLOAT *window; +- static FLOAT *absthr; +- static double *tmn; +- static FCB *s; +- static FHBLK *lthr; +- static F2HBLK *r, *phi_sav; +- +- /* These dynamic memory allocations simulate "automatic" variables */ +- /* placed on the stack. For each mem_alloc() call here, there must be */ +- /* a corresponding mem_free() call at the end of this function. */ +- +- grouped_c = (FLOAT *) mem_alloc(sizeof(FCB), "grouped_c"); +- grouped_e = (FLOAT *) mem_alloc(sizeof(FCB), "grouped_e"); +- nb = (FLOAT *) mem_alloc(sizeof(FCB), "nb"); +- cb = (FLOAT *) mem_alloc(sizeof(FCB), "cb"); +- ecb = (FLOAT *) mem_alloc(sizeof(FCB), "ecb"); +- bc = (FLOAT *) mem_alloc(sizeof(FCB), "bc"); +- wsamp_r = (FLOAT *) mem_alloc(sizeof(FBLK), "wsamp_r"); +- wsamp_i = (FLOAT *) mem_alloc(sizeof(FBLK), "wsamp_i"); +- phi = (FLOAT *) mem_alloc(sizeof(FBLK), "phi"); +- energy = (FLOAT *) mem_alloc(sizeof(FBLK), "energy"); +- c = (FLOAT *) mem_alloc(sizeof(FHBLK), "c"); +- fthr = (FLOAT *) mem_alloc(sizeof(FHBLK), "fthr"); +- snrtmp = (F32 *) mem_alloc(sizeof(F2_32), "snrtmp"); +- +- if(init==0){ +- +- /* These dynamic memory allocations simulate "static" variables placed */ +- /* in the data space. Each mem_alloc() call here occurs only once at */ +- /* initialization time. The mem_free() function must not be called. */ +- +- numlines = (int *) mem_alloc(sizeof(ICB), "numlines"); +- partition = (int *) mem_alloc(sizeof(IHBLK), "partition"); +- fpo = fopen("out.dat", "wb"); +- if(fpo==NULL) { +- puts("\t The attempt to open the output file failed.\n"); +- exit(-1);} +- cbval = (FLOAT *) mem_alloc(sizeof(FCB), "cbval"); +- rnorm = (FLOAT *) mem_alloc(sizeof(FCB), "rnorm"); +- window = (FLOAT *) mem_alloc(sizeof(FBLK), "window"); +- absthr = (FLOAT *) mem_alloc(sizeof(FHBLK), "absthr"); +- tmn = (double *) mem_alloc(sizeof(DCB), "tmn"); +- s = (FCB *) mem_alloc(sizeof(FCBCB), "s"); +- lthr = (FHBLK *) mem_alloc(sizeof(F2HBLK), "lthr"); +- r = (F2HBLK *) mem_alloc(sizeof(F22HBLK), "r"); +- phi_sav = (F2HBLK *) mem_alloc(sizeof(F22HBLK), "phi_sav"); +- +- i = sfreq + 0.5; +- switch(i){ +- case 32000: sfreq_idx = 0; break; +- case 44100: sfreq_idx = 1; break; +- case 48000: sfreq_idx = 2; break; +- default: printf("error, invalid sampling frequency: %d Hz\n",i); +- exit(-1); +- } +- printf("absthr[][] sampling frequency index: %d\n",sfreq_idx); +- read_absthr(absthr, sfreq_idx); +- if(lay==1){ +- flush = 384; +- syncsize = 1024; +- sync_flush = 576; +- } +- else { +- flush = 384*3.0/2.0; +- syncsize = 1056; +- sync_flush = syncsize - flush; +- } +- /* calculate HANN window coefficients */ +- /* for(i=0;i<BLKSIZE;i++)window[i]=0.5*(1-cos(2.0*PI*i/(BLKSIZE-1.0))); */ +- for(i=0;i<BLKSIZE;i++)window[i]=0.5*(1-cos(2.0*PI*(i-0.5)/BLKSIZE)); +- /* reset states used in unpredictability measure */ +- for(i=0;i<HBLKSIZE;i++){ +- r[0][0][i]=r[1][0][i]=r[0][1][i]=r[1][1][i]=0; +- phi_sav[0][0][i]=phi_sav[1][0][i]=0; +- phi_sav[0][1][i]=phi_sav[1][1][i]=0; +- lthr[0][i] = 60802371420160.0; +- lthr[1][i] = 60802371420160.0; +- } +- /***************************************************************************** +- * Initialization: Compute the following constants for use later * +- * partition[HBLKSIZE] = the partition number associated with each * +- * frequency line * +- * cbval[CBANDS] = the center (average) bark value of each * +- * partition * +- * numlines[CBANDS] = the number of frequency lines in each partition * +- * tmn[CBANDS] = tone masking noise * +- *****************************************************************************/ +- /* compute fft frequency multiplicand */ +- freq_mult = sfreq/BLKSIZE; +- +- /* calculate fft frequency, then bval of each line (use fthr[] as tmp storage)*/ +- for(i=0;i<HBLKSIZE;i++){ +- temp1 = i*freq_mult; +- j = 1; +- while(temp1>crit_band[j])j++; +- fthr[i]=j-1+(temp1-crit_band[j-1])/(crit_band[j]-crit_band[j-1]); +- } +- partition[0] = 0; +- /* temp2 is the counter of the number of frequency lines in each partition */ +- temp2 = 1; +- cbval[0]=fthr[0]; +- bval_lo=fthr[0]; +- for(i=1;i<HBLKSIZE;i++){ +- if((fthr[i]-bval_lo)>0.33){ +- partition[i]=partition[i-1]+1; +- cbval[partition[i-1]] = cbval[partition[i-1]]/temp2; +- cbval[partition[i]] = fthr[i]; +- bval_lo = fthr[i]; +- numlines[partition[i-1]] = temp2; +- temp2 = 1; +- } +- else { +- partition[i]=partition[i-1]; +- cbval[partition[i]] += fthr[i]; +- temp2++; +- } +- } +- numlines[partition[i-1]] = temp2; +- cbval[partition[i-1]] = cbval[partition[i-1]]/temp2; +- +- /************************************************************************ +- * Now compute the spreading function, s[j][i], the value of the spread-* +- * ing function, centered at band j, for band i, store for later use * +- ************************************************************************/ +- for(j=0;j<CBANDS;j++){ +- for(i=0;i<CBANDS;i++){ +- temp1 = (cbval[i] - cbval[j])*1.05; +- if(temp1>=0.5 && temp1<=2.5){ +- temp2 = temp1 - 0.5; +- temp2 = 8.0 * (temp2*temp2 - 2.0 * temp2); +- } +- else temp2 = 0; +- temp1 += 0.474; +- temp3 = 15.811389+7.5*temp1-17.5*sqrt((double) (1.0+temp1*temp1)); +- if(temp3 <= -100) s[i][j] = 0; +- else { +- temp3 = (temp2 + temp3)*LN_TO_LOG10; +- s[i][j] = exp(temp3); +- } +- } +- } +- +- /* Calculate Tone Masking Noise values */ +- for(j=0;j<CBANDS;j++){ +- temp1 = 15.5 + cbval[j]; +- tmn[j] = (temp1>24.5) ? temp1 : 24.5; +- /* Calculate normalization factors for the net spreading functions */ +- rnorm[j] = 0; +- for(i=0;i<CBANDS;i++){ +- rnorm[j] += s[j][i]; +- } +- } +- init++; +- } +- +- /************************* End of Initialization *****************************/ +- switch(lay) { +- case 1: +- case 2: +- for(i=0; i<lay; i++){ +- /***************************************************************************** +- * Net offset is 480 samples (1056-576) for layer 2; this is because one must* +- * stagger input data by 256 samples to synchronize psychoacoustic model with* +- * filter bank outputs, then stagger so that center of 1024 FFT window lines * +- * up with center of 576 "new" audio samples. * +- * * +- * For layer 1, the input data still needs to be staggered by 256 samples, * +- * then it must be staggered again so that the 384 "new" samples are centered* +- * in the 1024 FFT window. The net offset is then 576 and you need 448 "new"* +- * samples for each iteration to keep the 384 samples of interest centered * +- *****************************************************************************/ +- for(j=0; j<syncsize; j++){ +- if(j<(sync_flush))savebuf[j] = savebuf[j+flush]; +- else savebuf[j] = *buffer++; +- if(j<BLKSIZE){ +- /**window data with HANN window***********************************************/ +- wsamp_r[j] = window[j]*((FLOAT) savebuf[j]); +- wsamp_i[j] = 0; +- } +- } +- /**Compute FFT****************************************************************/ +- fft(wsamp_r,wsamp_i,energy,phi,1024); +- /***************************************************************************** +- * calculate the unpredictability measure, given energy[f] and phi[f] * +- *****************************************************************************/ +- /*only update data "age" pointers after you are done with both channels */ +- /*for layer 1 computations, for the layer 2 double computations, the pointers*/ +- /*are reset automatically on the second pass */ +- if(lay==2 || (lay==1 && chn==0) ){ +- if(new==0){new = 1; oldest = 1;} +- else {new = 0; oldest = 0;} +- if(old==0)old = 1; else old = 0; +- } +- for(j=0; j<HBLKSIZE; j++){ +- r_prime = 2.0 * r[chn][old][j] - r[chn][oldest][j]; +- phi_prime = 2.0 * phi_sav[chn][old][j] - phi_sav[chn][oldest][j]; +- r[chn][new][j] = sqrt((double) energy[j]); +- phi_sav[chn][new][j] = phi[j]; +- temp1=r[chn][new][j] * cos((double) phi[j]) - r_prime * cos((double) phi_prime); +- temp2=r[chn][new][j] * sin((double) phi[j]) - r_prime * sin((double) phi_prime); +- temp3=r[chn][new][j] + fabs((double)r_prime); +- if(temp3 != 0)c[j]=sqrt(temp1*temp1+temp2*temp2)/temp3; +- else c[j] = 0; +- } +- /***************************************************************************** +- * Calculate the grouped, energy-weighted, unpredictability measure, * +- * grouped_c[], and the grouped energy. grouped_e[] * +- *****************************************************************************/ +- for(j=1;j<CBANDS;j++){ +- grouped_e[j] = 0; +- grouped_c[j] = 0; +- } +- grouped_e[0] = energy[0]; +- grouped_c[0] = energy[0]*c[0]; +- for(j=1;j<HBLKSIZE;j++){ +- grouped_e[partition[j]] += energy[j]; +- grouped_c[partition[j]] += energy[j]*c[j]; +- } +- +- /***************************************************************************** +- * convolve the grouped energy-weighted unpredictability measure * +- * and the grouped energy with the spreading function, s[j][k] * +- *****************************************************************************/ +- for(j=0;j<CBANDS;j++){ +- ecb[j] = 0; +- cb[j] = 0; +- for(k=0;k<CBANDS;k++){ +- if(s[j][k] != 0.0){ +- ecb[j] += s[j][k]*grouped_e[k]; +- cb[j] += s[j][k]*grouped_c[k]; +- } +- } +- if(ecb[j] !=0)cb[j] = cb[j]/ecb[j]; +- else cb[j] = 0; +- } +- +- /***************************************************************************** +- * Calculate the required SNR for each of the frequency partitions * +- * this whole section can be accomplished by a table lookup * +- *****************************************************************************/ +- for(j=0;j<CBANDS;j++){ +- if(cb[j]<.05)cb[j]=0.05; +- else if(cb[j]>.5)cb[j]=0.5; +- tb = -0.434294482*log((double) cb[j])-0.301029996; +- cb[j]=tb; +- bc[j] = tmn[j]*tb + nmt*(1.0-tb); +- k = cbval[j] + 0.5; +- bc[j] = (bc[j] > bmax[k]) ? bc[j] : bmax[k]; +- bc[j] = exp((double) -bc[j]*LN_TO_LOG10); +- } +- +- /***************************************************************************** +- * Calculate the permissible noise energy level in each of the frequency * +- * partitions. Include absolute threshold and pre-echo controls * +- * this whole section can be accomplished by a table lookup * +- *****************************************************************************/ +- for(j=0;j<CBANDS;j++) +- if(rnorm[j] && numlines[j]) +- nb[j] = ecb[j]*bc[j]/(rnorm[j]*numlines[j]); +- else nb[j] = 0; +- for(j=0;j<HBLKSIZE;j++){ +- /*temp1 is the preliminary threshold */ +- temp1=nb[partition[j]]; +- temp1=(temp1>absthr[j])?temp1:absthr[j]; +- /*do not use pre-echo control for layer 2 because it may do bad things to the*/ +- /* MUSICAM bit allocation algorithm */ +- if(lay==1){ +- fthr[j] = (temp1 < lthr[chn][j]) ? temp1 : lthr[chn][j]; +- temp2 = temp1 * 0.00316; +- fthr[j] = (temp2 > fthr[j]) ? temp2 : fthr[j]; +- } +- else fthr[j] = temp1; +- lthr[chn][j] = LXMIN*temp1; +- } +- +- /***************************************************************************** +- * Translate the 512 threshold values to the 32 filter bands of the coder * +- *****************************************************************************/ +- for(j=0;j<193;j += 16){ +- minthres = 60802371420160.0; +- sum_energy = 0.0; +- for(k=0;k<17;k++){ +- if(minthres>fthr[j+k])minthres = fthr[j+k]; +- sum_energy += energy[j+k]; +- } +- snrtmp[i][j/16] = sum_energy/(minthres * 17.0); +- snrtmp[i][j/16] = 4.342944819 * log((double)snrtmp[i][j/16]); +- } +- for(j=208;j<(HBLKSIZE-1);j += 16){ +- minthres = 0.0; +- sum_energy = 0.0; +- for(k=0;k<17;k++){ +- minthres += fthr[j+k]; +- sum_energy += energy[j+k]; +- } +- snrtmp[i][j/16] = sum_energy/minthres; +- snrtmp[i][j/16] = 4.342944819 * log((double)snrtmp[i][j/16]); +- } +- /***************************************************************************** +- * End of Psychoacuostic calculation loop * +- *****************************************************************************/ +- } +- for(i=0; i<32; i++){ +- if(lay==2) +- snr32[i]=(snrtmp[0][i]>snrtmp[1][i])?snrtmp[0][i]:snrtmp[1][i]; +- else snr32[i]=snrtmp[0][i]; +- } +- break; +- case 3: +- printf("layer 3 is not currently supported\n"); +- break; +- default: +- printf("error, invalid MPEG/audio coding layer: %d\n",lay); +- } +- +- /* These mem_free() calls must correspond with the mem_alloc() calls */ +- /* used at the beginning of this function to simulate "automatic" */ +- /* variables placed on the stack. */ +- +- mem_free((void **) &grouped_c); +- mem_free((void **) &grouped_e); +- mem_free((void **) &nb); +- mem_free((void **) &cb); +- mem_free((void **) &ecb); +- mem_free((void **) &bc); +- mem_free((void **) &wsamp_r); +- mem_free((void **) &wsamp_i); +- mem_free((void **) &phi); +- mem_free((void **) &energy); +- mem_free((void **) &c); +- mem_free((void **) &fthr); +- mem_free((void **) &snrtmp); +- } +- +- /****************************************************************************** +- routine to read in absthr table from a file. +- ******************************************************************************/ +- +- void read_absthr(absthr, table) +- FLOAT *absthr; +- int table; +- { +- FILE *fp; +- long j,index; +- float a; +- char t[80]; +- char ta[16]; +- +- strcpy( ta, "absthr_0" ); +- +- switch(table){ +- case 0 : ta[7] = '0'; +- break; +- case 1 : ta[7] = '1'; +- break; +- case 2 : ta[7] = '2'; +- break; +- default : printf("absthr table: Not valid table number\n"); +- } +- if(!(fp = OpenTableFile(ta) ) ){ +- printf("Please check %s table\n", ta); +- exit(1); +- } +- fgets(t, 150, fp); +- sscanf(t, "table %ld", &index); +- if(index != table){ +- printf("error in absthr table %s",ta); +- exit(1); +- } +- for(j=0; j<HBLKSIZE; j++){ +- fgets(t,80,fp); +- sscanf(t,"%f", &a); +- absthr[j] = a; +- } +- fclose(fp); +- } +--- 0 ---- +diff -r -c -N encoder/psymodel.c lame3.50/psymodel.c +*** encoder/psymodel.c Wed Dec 31 17:00:00 1969 +--- lame3.50/psymodel.c Mon Nov 1 18:19:12 1999 +*************** +*** 0 **** +--- 1,1176 ---- ++ /********************************************************************** ++ * ISO MPEG Audio Subgroup Software Simulation Group (1996) ++ * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension ++ * ++ * $Id: l3psy.c,v 1.2 1998/10/05 17:06:48 larsi Exp $ ++ * ++ * $Log: l3psy.c,v $ ++ * Revision 1.2 1998/10/05 17:06:48 larsi ++ * *** empty log message *** ++ * ++ * Revision 1.1.1.1 1998/10/05 14:47:18 larsi ++ * ++ * Revision 1.2 1997/01/19 22:28:29 rowlands ++ * Layer 3 bug fixes from Seymour Shlien ++ * ++ * Revision 1.1 1996/02/14 04:04:23 rowlands ++ * Initial revision ++ * ++ * Received from Mike Coleman ++ **********************************************************************/ ++ /********************************************************************** ++ * date programmers comment * ++ * 2/25/91 Davis Pan start of version 1.0 records * ++ * 5/10/91 W. Joseph Carter Ported to Macintosh and Unix. * ++ * 7/10/91 Earle Jennings Ported to MsDos. * ++ * replace of floats with FLOAT * ++ * 2/11/92 W. Joseph Carter Fixed mem_alloc() arg for "absthr". * ++ * 3/16/92 Masahiro Iwadare Modification for Layer III * ++ * 17/4/93 Masahiro Iwadare Updated for IS Modification * ++ **********************************************************************/ ++ ++ #include "util.h" ++ #include "globalflags.h" ++ #include "encoder.h" ++ #include "psymodel.h" ++ #include "l3side.h" ++ #include <assert.h> ++ #include "gtkanal.h" ++ #include "tables.h" ++ #include "fft.h" ++ ++ #ifdef M_LN10 ++ #define LN_TO_LOG10 (M_LN10/10) ++ #else ++ #define LN_TO_LOG10 0.2302585093 ++ #endif ++ ++ #define maximum(x,y) ( (x>y) ? x : y ) ++ #define minimum(x,y) ( (x<y) ? x : y ) ++ ++ /* some different types of adaptive window switching */ ++ ++ ++ /* This mode will turn on short_blocks if there is a localized surge in energy */ ++ #define ENER_AWS ++ ++ static int switch_pe=1800; ++ ++ static double s3_l[CBANDS][CBANDS]; /* needed global static by sprdngfs */ ++ ++ void L3para_read( double sfreq, int numlines[CBANDS],int numlines_s[CBANDS], int partition_l[HBLKSIZE], ++ double minval[CBANDS], double qthr_l[CBANDS], double norm_l[CBANDS], ++ double s3_l[CBANDS][CBANDS], double s3_s[CBANDS][CBANDS], ++ int partition_s[HBLKSIZE_s], double qthr_s[CBANDS], ++ double norm_s[CBANDS], double SNR_s[CBANDS], ++ int cbw_l[SBMAX_l], int bu_l[SBMAX_l], int bo_l[SBMAX_l], ++ double w1_l[SBMAX_l], double w2_l[SBMAX_l], ++ int cbw_s[SBMAX_s], int bu_s[SBMAX_s], int bo_s[SBMAX_s], ++ double w1_s[SBMAX_s], double w2_s[SBMAX_s] ); ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ void L3psycho_energy( short int *buffer, ++ FLOAT energy[HBLKSIZE], ++ FLOAT ax[HBLKSIZE], FLOAT bx[HBLKSIZE], ++ FLOAT energy_s[3][HBLKSIZE_s], ++ FLOAT ax_s[3][HBLKSIZE_s], FLOAT bx_s[3][HBLKSIZE_s], ++ int chn,int gr_out , layer * info) ++ { ++ static short int savebuffer[2][WINDELAY+576]; ++ static int sync_flush,flush,syncsize; ++ static FLOAT scalefac; ++ static FLOAT window_s[BLKSIZE_s]; ++ static FLOAT window[BLKSIZE]; ++ static int firstcall=1; ++ ++ #ifdef HAVEGTK ++ static FLOAT energy_save[4][HBLKSIZE]; ++ #endif ++ ++ int i,j,k,sblock; ++ ++ FLOAT wsamp_r[BLKSIZE]; ++ FLOAT wsamp_rs[256]; ++ ++ if(firstcall) { ++ firstcall=0; ++ memset((char *) savebuffer, 0, sizeof(savebuffer)); ++ sync_flush=WINDELAY; flush=576; syncsize=WINDELAY+576; ++ ++ scalefac=1.0; ++ if (force_ms) scalefac=SQRT2; ++ ++ /* calculate HANN window coefficients */ ++ /* note: ISO DOCS use i-.5 because their i starts at 1, not 0 */ ++ for(i=0;i<BLKSIZE;i++) window[i] =0.5*(1-cos(2.0*PI*(i+0.5)/BLKSIZE)); ++ for(i=0;i<BLKSIZE_s;i++)window_s[i]=0.5*(1-cos(2.0*PI*(i+0.5)/BLKSIZE_s)); ++ ++ } ++ ++ /********************************************************************** ++ * compute FFTs ++ **********************************************************************/ ++ if (chn<2) { /* the most common option */ ++ for ( j = 0; j < sync_flush; j++ ) /* for long window samples */ ++ savebuffer[chn][j] = savebuffer[chn][j+flush]; ++ for ( j = sync_flush; j < syncsize; j++ ) ++ savebuffer[chn][j] = buffer[j-sync_flush]; ++ ++ for ( j = 0; j < BLKSIZE; j++ ) ++ wsamp_r[j] = window[j] * savebuffer[chn][j] * scalefac; ++ ++ } else ++ if (chn==2) { ++ for ( j = 0; j < BLKSIZE; j++ ) ++ wsamp_r[j] = window[j] * (savebuffer[0][j] + savebuffer[1][j] )/SQRT2; ++ } else /* if (chn==3) */ ++ { ++ for ( j = 0; j < BLKSIZE; j++ ) ++ wsamp_r[j] = window[j] * (savebuffer[0][j] - savebuffer[1][j] )/SQRT2; ++ } ++ ++ fft( wsamp_r, energy, ax, bx, 1024 ); ++ /* mt 7/99 ++ Note: fft_side() can be used to compute energy, ax & bx for the ++ mid and side channels (chn=2,3) without calling additional FFTs. ++ But it requires wsamp_r to be saved from channels 0 and 1. ++ My tests show that the FFT is so fast that this gives no savings. ++ Probably the extra memory hurts the cache performance. ++ */ ++ ++ ++ #ifdef HAVEGTK ++ if(gtkflag) { ++ for (j=0; j<HBLKSIZE ; j++) { ++ pinfo->energy[gr_out][chn][j]=energy_save[chn][j]; ++ energy_save[chn][j]=energy[j]; ++ } ++ } ++ #endif ++ for ( sblock = 0; sblock < 3; sblock++ ) { ++ int shlen = 192; ++ int shoff = 1; ++ if (chn<2) ++ for ( j = 0, k = shlen * (shoff + sblock); j < 256; j++, k++ ) ++ wsamp_rs[j] = window_s[j]* savebuffer[chn][k]*scalefac; ++ if (chn==2) ++ for ( j = 0, k = shlen * (shoff + sblock); j < 256; j++, k++ ) ++ wsamp_rs[j] = window_s[j] * ++ ((savebuffer[0][k]+savebuffer[1][k])/2)*SQRT2; ++ if (chn==3) ++ for ( j = 0, k = shlen * (shoff + sblock); j < 256; j++, k++ ) ++ wsamp_rs[j] = window_s[j] * ++ ((savebuffer[0][k]-savebuffer[1][k])/2)*SQRT2; ++ ++ fft( wsamp_rs, energy_s[sblock], ax_s[sblock], bx_s[sblock], 256 ); ++ } ++ ++ ++ } ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ void L3psycho_anal( short int *buffer[2], int stereo, ++ int gr_out , layer * info, ++ double sfreq, ++ int check_ms_stereo, double *ms_ener_ratio, ++ double ratio_d[4][21], double ratio_ds[4][12][3], ++ double percep_energy[4], int blocktype_d[2]) ++ { ++ static double pe[4]={0,0,0,0}; ++ static double ms_ratio_s_old=0,ms_ratio_l_old=0; ++ ++ static double ratio[4][SBMAX_l]; ++ static double ratio_s[4][SBMAX_s][3]; ++ #ifdef HAVEGTK ++ static double pe_save[4]; ++ static double ers_save[4]; ++ #endif ++ static double thm_save[4][SBMAX_l]; ++ static double en_save[4][SBMAX_l]; ++ static double thm_s_save[4][SBMAX_s][3]; ++ static double en_s_save[4][SBMAX_s][3]; ++ ++ ++ int blocktype[2],uselongblock[2],chn; ++ int numchn; ++ unsigned int b, i, j, k; ++ double ms_ratio_l=0,ms_ratio_s=0; ++ double estot[4][3]; ++ ++ ++ double thr[CBANDS]; ++ FLOAT ax[HBLKSIZE], bx[HBLKSIZE]; ++ FLOAT energy[HBLKSIZE]; ++ FLOAT energy_s[3][HBLKSIZE_s]; ++ FLOAT ax_s[3][HBLKSIZE_s], bx_s[3][HBLKSIZE_s]; /* 256 samples not 129. */ ++ ++ static float mld_l[SBMAX_l],mld_s[SBMAX_s]; ++ ++ ++ ++ ++ /* The static variables "r", "phi_sav", "new", "old" and "oldest" have */ ++ /* to be remembered for the unpredictability measure. For "r" and */ ++ /* "phi_sav", the first index from the left is the channel select and */ ++ /* the second index is the "age" of the data. */ ++ static int sfreq_idx; ++ static double cw[HBLKSIZE], eb[CBANDS]; ++ static double ctb[CBANDS]; ++ static double SNR_l[CBANDS], SNR_s[CBANDS]; ++ static double minval[CBANDS],qthr_l[CBANDS],norm_l[CBANDS]; ++ static double qthr_s[CBANDS],norm_s[CBANDS]; ++ static double nb_1[4][CBANDS], nb_2[4][CBANDS]; ++ static double s3_s[CBANDS][CBANDS]; ++ ++ /* Scale Factor Bands */ ++ static int cbw_l[SBMAX_l],bu_l[SBMAX_l],bo_l[SBMAX_l] ; ++ static int cbw_s[SBMAX_s],bu_s[SBMAX_s],bo_s[SBMAX_s] ; ++ static double w1_l[SBMAX_l], w2_l[SBMAX_l]; ++ static double w1_s[SBMAX_s], w2_s[SBMAX_s]; ++ static double en[SBMAX_l], thm[SBMAX_l] ; ++ static int blocktype_old[2] ; ++ int sb,sblock; ++ static int partition_l[HBLKSIZE],partition_s[HBLKSIZE_s]; ++ static int numCBANDS,numCBANDS_s; ++ static int num_part,num_part_s; ++ ++ static int s3ind[CBANDS][2]; ++ static int s3ind_s[CBANDS][2]; ++ ++ ++ static FLOAT nb[CBANDS], cb[CBANDS], ecb[CBANDS]; ++ static int numlines_s[CBANDS] ; ++ static int numlines_l[CBANDS]; ++ static FLOAT ax_sav[4][2][HBLKSIZE], bx_sav[4][2][HBLKSIZE],rx_sav[4][2][HBLKSIZE]; ++ ++ if((frameNum==0) && (gr_out==0)){ ++ ++ i = sfreq + 0.5; ++ switch(i){ ++ case 32000: sfreq_idx = 0; break; ++ case 44100: sfreq_idx = 1; break; ++ case 48000: sfreq_idx = 2; break; ++ case 16000: sfreq_idx = 3; break; ++ case 22050: sfreq_idx = 4; break; ++ case 24000: sfreq_idx = 5; break; ++ default: fprintf(stderr,"error, invalid sampling frequency: %d Hz\n",i); ++ exit(-1); ++ } ++ ++ /* reset states used in unpredictability measure */ ++ memset (rx_sav,0, sizeof(rx_sav)); ++ memset (ax_sav,0, sizeof(ax_sav)); ++ memset (bx_sav,0, sizeof(bx_sav)); ++ ++ ++ ++ /* setup stereo demasking thresholds */ ++ /* formula reverse enginerred from plot in paper */ ++ for ( sb = 0; sb < SBMAX_s; sb++ ) { ++ double mld = 1.25*(1-cos(3.14159*sb/SBMAX_s))-2.5; ++ mld_s[sb] = pow(10.0,mld); ++ } ++ for ( sb = 0; sb < SBMAX_l; sb++ ) { ++ double mld = 1.25*(1-cos(3.14159*sb/SBMAX_l))-2.5; ++ mld_l[sb] = pow(10.0,mld); ++ } ++ ++ for (i=0;i<HBLKSIZE;i++) partition_l[i]=-1; ++ for (i=0;i<HBLKSIZE_s;i++) partition_s[i]=-1; ++ ++ ++ L3para_read( sfreq,numlines_l,numlines_s,partition_l,minval,qthr_l,norm_l,s3_l,s3_s, ++ partition_s,qthr_s,norm_s,SNR_s, ++ cbw_l,bu_l,bo_l,w1_l,w2_l, cbw_s,bu_s,bo_s,w1_s,w2_s ); ++ ++ ++ /* num_part = number of partition bands before convolution */ ++ /* numCBANDS = number of partition bands after convolution */ ++ num_part=0; num_part_s=0; ++ for (i=0;i<HBLKSIZE;i++) ++ if (partition_l[i]>num_part) num_part=partition_l[i]; ++ for (i=0;i<HBLKSIZE_s;i++) ++ if (partition_s[i]>num_part_s) num_part_s=partition_s[i]; ++ num_part++; ++ num_part_s++; ++ ++ numCBANDS=bo_l[SBMAX_l-1]+1; ++ numCBANDS_s=bo_s[SBMAX_s-1]+1; ++ ++ for (i=0; i<num_part; i++) { ++ for (j = 0; j < num_part; j++) { ++ if (s3_l[i][j] != 0.0) ++ break; ++ } ++ s3ind[i][0] = j; ++ ++ for (j = num_part; j > 0; j--) { ++ if (s3_l[i][j] != 0.0) ++ break; ++ } ++ s3ind[i][1] = j; ++ } ++ ++ ++ for (i=0; i<num_part_s; i++) { ++ for (j = 0; j < num_part_s; j++) { ++ if (s3_s[i][j] != 0.0) ++ break; ++ } ++ s3ind_s[i][0] = j; ++ ++ for (j = num_part_s; j > 0; j--) { ++ if (s3_s[i][j] != 0.0) ++ break; ++ } ++ s3ind_s[i][1] = j; ++ } ++ ++ #define NEWS3XX ++ #ifdef NEWS3 ++ // compute norm_l, norm_s instead of relying on table data ++ for ( b = 0;b < numCBANDS; b++ ) { ++ double norm=0; ++ for ( k = s3ind[b][0]; k <= s3ind[b][1]; k++ ) { ++ norm += s3_l[b][k]; ++ } ++ norm_l[b] = 1/norm; ++ //printf("%i norm=%f norm_l=%f \n",b,1/norm,norm_l[b]); ++ } ++ for ( b = 0;b < numCBANDS_s; b++ ) { ++ double norm=0; ++ for ( k = s3ind_s[b][0]; k <= s3ind_s[b][1]; k++ ) { ++ norm += s3_s[b][k]; ++ } ++ norm_s[b] = 1/norm; ++ //printf("%i norm=%f norm_s=%f \n",b,1/norm,norm_l[b]); ++ } ++ #endif ++ ++ /* MPEG1 SNR_s data is given in db, convert to energy */ ++ if (info->version == MPEG_AUDIO_ID) { ++ for ( b = 0;b < numCBANDS_s; b++ ) { ++ SNR_s[b]=exp( (double) SNR_s[b] * LN_TO_LOG10 ); ++ } ++ } ++ } ++ /************************* End of Initialization *****************************/ ++ ++ ++ ++ ++ ++ numchn=stereo; ++ if (highq && (info->mode == MPG_MD_JOINT_STEREO)) numchn=4; ++ for (chn=0; chn<numchn; chn++) { ++ ++ for ( j = 0; j < 21; j++ ) ++ ratio_d[chn][j] = ratio[chn][j]; ++ for ( j = 0; j < 12; j++ ) ++ for ( i = 0; i < 3; i++ ) ++ ratio_ds[chn][j][i] = ratio_s[chn][j][i]; ++ percep_energy[chn] = pe[chn]; ++ ++ /********************************************************************** ++ * compute FFTs ++ **********************************************************************/ ++ L3psycho_energy( buffer[chn%2], energy, ax, bx, energy_s, ax_s, bx_s, ++ chn,gr_out,info); ++ ++ ++ /********************************************************************** ++ * compute unpredicatability of first six spectral lines * ++ **********************************************************************/ ++ for ( j = 0; j < 6; j++ ) ++ { /* calculate unpredictability measure cw */ ++ double an, a1, a2; ++ double bn, b1, b2; ++ double rn, r1, r2; ++ double numre, numim, den; ++ ++ a2 = ax_sav[chn][1][j]; ++ b2 = bx_sav[chn][1][j]; ++ r2 = rx_sav[chn][1][j]; ++ a1 = ax_sav[chn][1][j] = ax_sav[chn][0][j]; ++ b1 = bx_sav[chn][1][j] = bx_sav[chn][0][j]; ++ r1 = rx_sav[chn][1][j] = rx_sav[chn][0][j]; ++ an = ax_sav[chn][0][j] = ax[j]; ++ bn = bx_sav[chn][0][j] = bx[j]; ++ rn = rx_sav[chn][0][j] = sqrt(energy[j]); ++ ++ { /* square (x1,y1) */ ++ if( r1 != 0.0 ) { ++ numre = (a1*b1); ++ numim = (a1*a1-b1*b1)*0.5; ++ den = r1*r1; ++ } else { ++ numre = 1.0; ++ numim = 0.0; ++ den = 1.0; ++ } ++ } ++ ++ { /* multiply by (x2,-y2) */ ++ if( r2 != 0.0 ) { ++ double tmp2 = (numim+numre)*(a2+b2)*0.5; ++ double tmp1 = -a2*numre+tmp2; ++ numre = -b2*numim+tmp2; ++ numim = tmp1; ++ den *= r2; ++ } else { ++ /* do nothing */ ++ } ++ } ++ ++ { /* r-prime factor */ ++ double tmp = (2.0*r1-r2)/den; ++ numre *= tmp; ++ numim *= tmp; ++ } ++ ++ if( (den=rn+fabs(2.0*r1-r2)) != 0.0 ) { ++ numre = (an+bn)/2.0-numre; ++ numim = (an-bn)/2.0-numim; ++ cw[j] = sqrt(numre*numre+numim*numim)/den; ++ } else { ++ cw[j] = 0.0; ++ } ++ ++ } ++ ++ /********************************************************************** ++ * compute unpredicatibility of next 200 spectral lines * ++ **********************************************************************/ ++ for ( j = 6; j < 206; j += 4 ) ++ {/* calculate unpredictability measure cw */ ++ double rn, r1, r2; ++ double numre, numim, den; ++ ++ k = (j+2) / 4; ++ ++ { /* square (x1,y1) */ ++ r1 = sqrt((double)energy_s[0][k]); ++ if( r1 != 0.0 ) { ++ double a1 = ax_s[0][k]; ++ double b1 = bx_s[0][k]; ++ numre = (a1*b1); ++ numim = (a1*a1-b1*b1)*0.5; ++ den = r1*r1; ++ } else { ++ numre = 1.0; ++ numim = 0.0; ++ den = 1.0; ++ } ++ } ++ ++ ++ { /* multiply by (x2,-y2) */ ++ r2 = sqrt((double)energy_s[2][k]); ++ if( r2 != 0.0 ) { ++ double a2 = ax_s[2][k]; ++ double b2 = bx_s[2][k]; ++ ++ double tmp2 = (numim+numre)*(a2+b2)*0.5; ++ double tmp1 = -a2*numre+tmp2; ++ numre = -b2*numim+tmp2; ++ numim = tmp1; ++ ++ den *= r2; ++ } else { ++ /* do nothing */ ++ } ++ } ++ ++ { /* r-prime factor */ ++ double tmp = (2.0*r1-r2)/den; ++ numre *= tmp; ++ numim *= tmp; ++ } ++ ++ rn = sqrt((double)energy_s[1][k]); ++ if( (den=rn+fabs(2.0*r1-r2)) != 0.0 ) { ++ double an = ax_s[1][k]; ++ double bn = bx_s[1][k]; ++ numre = (an+bn)/2.0-numre; ++ numim = (an-bn)/2.0-numim; ++ cw[j] = sqrt(numre*numre+numim*numim)/den; ++ } else { ++ cw[j] = 0.0; ++ } ++ ++ cw[j+1] = cw[j+2] = cw[j+3] = cw[j]; ++ } ++ ++ ++ ++ ++ ++ /********************************************************************** ++ * Set unpredicatiblility of remaining spectral lines to 0.4 206..513 * ++ **********************************************************************/ ++ for ( j = 206; j < HBLKSIZE; j++ ) ++ cw[j] = 0.4; ++ ++ ++ ++ #if 0 ++ for ( j = 14; j < HBLKSIZE-4; j += 4 ) ++ {/* calculate energy from short ffts */ ++ double tot,ave; ++ k = (j+2) / 4; ++ for (tot=0, sblock=0; sblock < 3; sblock++) ++ tot+=energy_s[sblock][k]; ++ ave = energy[j+1]+ energy[j+2]+ energy[j+3]+ energy[j]; ++ ave /= 4.; ++ /* ++ printf("energy / tot %i %5.2f %e %e\n",j,ave/(tot*16./3.), ++ ave,tot*16./3.); ++ */ ++ energy[j+1] = energy[j+2] = energy[j+3] = energy[j]=tot; ++ } ++ #endif ++ ++ ++ ++ ++ ++ ++ ++ ++ /********************************************************************** ++ * Calculate the energy and the unpredictability in the threshold * ++ * calculation partitions * ++ **********************************************************************/ ++ for ( b = 0; b < CBANDS; b++ ) ++ { ++ eb[b] = 0.0; ++ cb[b] = 0.0; ++ } ++ for ( j = 0; j < HBLKSIZE; j++ ) ++ { ++ int tp = partition_l[j]; ++ if ( tp >= 0 ) ++ { ++ eb[tp] += energy[j]; ++ cb[tp] += cw[j] * energy[j]; ++ } ++ } ++ ++ ++ /********************************************************************** ++ * convolve the partitioned energy and unpredictability * ++ * with the spreading function, s3_l[b][k] * ++ ******************************************************************** */ ++ for ( b = 0; b < CBANDS; b++ ) ++ { ++ ecb[b] = 0.0; ++ ctb[b] = 0.0; ++ } ++ for ( b = 0;b < numCBANDS; b++ ) ++ for ( k = s3ind[b][0]; k <= s3ind[b][1]; k++ ) ++ { ++ ecb[b] += s3_l[b][k] * eb[k]; /* sprdngf for Layer III */ ++ ctb[b] += s3_l[b][k] * cb[k]; ++ } ++ ++ ++ /* calculate the tonality of each threshold calculation partition */ ++ /* calculate the SNR in each threshhold calculation partition */ ++ ++ for ( b = 0; b < numCBANDS; b++ ) ++ { ++ double cbb,tbb; ++ if (ecb[b] != 0.0 ) ++ { ++ cbb = ctb[b]/ecb[b]; ++ if (cbb <0.01) cbb = 0.01; ++ cbb = log( cbb); ++ } ++ else ++ cbb = 0.0 ; ++ tbb = -0.299 - 0.43*cbb; /* conv1=-0.299, conv2=-0.43 */ ++ tbb = minimum( 1.0, maximum( 0.0, tbb) ) ; /* 0<tbb<1 */ ++ SNR_l[b] = maximum( minval[b], 29.0*tbb+6.0*(1.0-tbb) ); ++ ++ ++ } /* TMN=29.0,NMT=6.0 for all calculation partitions */ ++ ++ for ( b = 0; b < numCBANDS; b++ ) /* calculate the threshold for each partition */ ++ nb[b] = ecb[b] * norm_l[b] * exp( -SNR_l[b] * LN_TO_LOG10 ); ++ ++ ++ ++ for ( b = 0; b < numCBANDS; b++ ) ++ { /* pre-echo control */ ++ double temp_1; /* BUG of IS */ ++ int rpelev=2; int rpelev2=16; ++ temp_1 = minimum( nb[b], minimum(rpelev*nb_1[chn][b],rpelev2*nb_2[chn][b]) ); ++ thr[b] = maximum( qthr_l[b], temp_1 );/* rpelev=2.0, rpelev2=16.0 */ ++ nb_2[chn][b] = nb_1[chn][b]; ++ nb_1[chn][b] = nb[b]; ++ } ++ ++ /* note: all surges in PE are because of the above pre-echo formula ++ * for temp_1. it this is not used, PE is always around 600 ++ */ ++ ++ pe[chn] = 0.0; /* calculate percetual entropy */ ++ for ( b = 0; b < numCBANDS; b++ ) ++ { ++ double tp = log((thr[b]+1.0) / (eb[b]+1.0) ); ++ tp = minimum( 0.0, tp ) ; /*not log*/ ++ pe[chn] -= numlines_l[b] * tp ; ++ ++ ++ } /* thr[b] -> thr[b]+1.0 : for non sound portition */ ++ ++ ++ ++ /*************************************************************** ++ * Check to see if we also need to compute long block thresholds ++ ***************************************************************/ ++ if (voice_mode){ ++ uselongblock[chn]=1; ++ } else { ++ double mn,mx; ++ uselongblock[chn] = (pe[chn] < switch_pe); ++ ++ for (sblock=0; sblock < 3; sblock++) ++ estot[chn][sblock]=0; ++ for ( j = HBLKSIZE_s/2; j < HBLKSIZE_s; j ++) ++ for (sblock=0; sblock < 3; sblock++) ++ estot[chn][sblock]+=energy_s[sblock][j]; ++ mn = minimum(estot[chn][0],estot[chn][1]); ++ mn = minimum(mn,estot[chn][2]); ++ mx = maximum(estot[chn][0],estot[chn][1]); ++ mx = maximum(mx,estot[chn][2]); ++ #ifdef HAVEGTK ++ if (gtkflag) { ++ pinfo->ers[gr_out][chn]=ers_save[chn]; ++ ers_save[chn]=mx/(1e-12+mn); ++ pinfo->pe[gr_out][chn]=pe_save[chn]; ++ pe_save[chn]=pe[chn]; ++ } ++ #endif ++ #ifdef ENER_AWS ++ if (chn<2) { ++ uselongblock[chn] = 1; ++ ++ /* tuned for t1.wav. doesnt effect most other samples */ ++ if (pe[chn] > 3000) uselongblock[chn]=0; ++ ++ /* big surge of energy - always use short blocks */ ++ if ( mx > 30*mn) uselongblock[chn] = 0; ++ ++ /* medium surge, medium pe - use short blocks */ ++ if ((mx > 10*mn) && (pe[chn] > 1000)) uselongblock[chn] = 0; ++ #endif ++ } ++ } ++ ++ ++ ++ /*************************************************************** ++ * compute masking thresholds for both short and long blocks ++ ***************************************************************/ ++ /* threshold calculation (part 2) */ ++ for ( sb = 0; sb < SBMAX_l; sb++ ) ++ { ++ en[sb] = w1_l[sb] * eb[bu_l[sb]] + w2_l[sb] * eb[bo_l[sb]]; ++ thm[sb] = w1_l[sb] *thr[bu_l[sb]] + w2_l[sb] * thr[bo_l[sb]]; ++ for ( b = bu_l[sb]+1; b < bo_l[sb]; b++ ) ++ { ++ en[sb] += eb[b]; ++ thm[sb] += thr[b]; ++ } ++ if ( en[sb] != 0.0 ) ++ ratio[chn][sb] = thm[sb]/en[sb]; ++ else ++ ratio[chn][sb] = 0.0; ++ } ++ #ifdef HAVEGTK ++ if (gtkflag) { ++ for (sb=0; sb< SBMAX_l; sb ++ ) { ++ pinfo->thr[gr_out][chn][sb]=thm_save[chn][sb]; ++ pinfo->en[gr_out][chn][sb]=en_save[chn][sb]; ++ thm_save[chn][sb]=thm[sb]; ++ en_save[chn][sb]=en[sb]; ++ } ++ } ++ #endif ++ for (sb=0; sb< SBMAX_l; sb ++ ) { ++ thm_save[chn][sb]=thm[sb]; ++ en_save[chn][sb]=en[sb]; ++ } ++ ++ /* threshold calculation for short blocks */ ++ for ( sblock = 0; sblock < 3; sblock++ ) { ++ for ( b = 0; b < CBANDS; b++ ) ++ { ++ eb[b] = 0.0; ++ ecb[b] = 0.0; ++ } ++ for ( j = 0; j < HBLKSIZE_s; j++ ) { ++ if (partition_s[j]>=0) eb[partition_s[j]] += energy_s[sblock][j]; ++ } ++ for ( b = 0; b < numCBANDS_s; b++ ) ++ for ( k = s3ind_s[b][0]; k <= s3ind_s[b][1]; k++ ) ++ ecb[b] += s3_s[b][k] * eb[k]; ++ ++ ++ for ( b = 0; b < numCBANDS_s; b++ ) ++ { ++ // nb[b] = ecb[b] * norm_s[b] * exp( (double) SNR_s[b] * LN_TO_LOG10 ); ++ nb[b] = ecb[b] * norm_s[b] * SNR_s[b]; ++ thr[b] = maximum (qthr_s[b],nb[b]); ++ } ++ ++ for ( sb = 0; sb < SBMAX_s; sb++ ) ++ { ++ en[sb] = w1_s[sb] * eb[bu_s[sb]] + w2_s[sb] * eb[bo_s[sb]]; ++ thm[sb] = w1_s[sb] *thr[bu_s[sb]] + w2_s[sb] * thr[bo_s[sb]]; ++ for ( b = bu_s[sb]+1; b < bo_s[sb]; b++ ) ++ { ++ en[sb] += eb[b]; ++ thm[sb] += thr[b]; ++ } ++ if ( en[sb] != 0.0 ) ++ ratio_s[chn][sb][sblock] = thm[sb]/en[sb]; ++ else ++ ratio_s[chn][sb][sblock] = 0.0; ++ #ifdef HAVEGTK ++ if (gtkflag) { ++ pinfo->thr_s[gr_out][chn][3*sb+sblock]=thm_s_save[chn][sb][sblock]; ++ pinfo->en_s[gr_out][chn][3*sb+sblock]=en_s_save[chn][sb][sblock]; ++ } ++ #endif ++ thm_s_save[chn][sb][sblock]=thm[sb]; ++ en_s_save[chn][sb][sblock]=en[sb]; ++ ++ ++ } ++ ++ } ++ ++ ++ /* compute M/S thresholds from Johnston & Ferreira 1992 ICASSP paper */ ++ #define JOHNSTON ++ #ifdef JOHNSTON ++ if ((force_ms && (chn==1)) || (chn==3)) { ++ double rside,rmid,mld; ++ int ch0,ch1; ++ ch0=chn-1; ++ ch1=chn; ++ ++ for ( sb = 0; sb < SBMAX_l; sb++ ) { ++ /* use this fix if L & R masking differs by 2db or less */ ++ /* if db = 10*log10(x2/x1) < 2 */ ++ /* if (x2 < 1.58*x1) { */ ++ if ((chn==1) || (ratio[0][sb] <= 1.58*ratio[1][sb]) ) { ++ mld = mld_l[sb]; ++ rmid = Max(ratio[ch0][sb],Min(ratio[ch1][sb],mld)); ++ rside = Max(ratio[ch1][sb],Min(ratio[ch0][sb],mld)); ++ ratio[ch0][sb]=rmid; ++ ratio[ch1][sb]=rside; ++ } ++ } ++ for ( sblock = 0; sblock < 3; sblock++ ){ ++ for ( sb = 0; sb < SBMAX_s; sb++ ) { ++ if ((chn==1) || (ratio_s[0][sb][sblock] <= 1.58*ratio_s[1][sb][sblock]) ) { ++ mld = mld_s[sb]; ++ rmid = Max(ratio_s[ch0][sb][sblock],Min(ratio_s[ch1][sb][sblock],mld)); ++ rside = Max(ratio_s[ch1][sb][sblock],Min(ratio_s[ch0][sb][sblock],mld)); ++ ratio_s[ch0][sb][sblock]=rmid; ++ ratio_s[ch1][sb][sblock]=rside; ++ } ++ } ++ } ++ } ++ #endif ++ } /* end loop over chn */ ++ ++ ++ ++ if (check_ms_stereo) { ++ /* determin ms_ratio from masking thresholds*/ ++ /* use ms_stereo (ms_ratio < .35) if average thresh. diff < 5 db */ ++ { double db,x1,x2,sidetot=0,tot=0; ++ for (sb= SBMAX_l/4 ; sb< SBMAX_l; sb ++ ) { ++ x1 = minimum(thm_save[0][sb],thm_save[1][sb]); ++ x2 = maximum(thm_save[0][sb],thm_save[1][sb]); ++ /* thresholds difference in db */ ++ if (x2 >= 1000*x1) db=30; ++ else db = 10*log10(x2/x1); ++ sidetot += db; ++ tot++; ++ } ++ ms_ratio_l= .35*(sidetot/tot)/5.0; ++ ++ sidetot=0; tot=0; ++ for ( sblock = 0; sblock < 3; sblock++ ) ++ for ( sb = SBMAX_s/4; sb < SBMAX_s; sb++ ) { ++ x1 = minimum(thm_s_save[0][sb][sblock],thm_s_save[1][sb][sblock]); ++ x2 = maximum(thm_s_save[0][sb][sblock],thm_s_save[1][sb][sblock]); ++ /* thresholds difference in db */ ++ if (x2 >= 1000*x1) db=30; ++ else db = 10*log10(x2/x1); ++ sidetot += db; ++ tot++; ++ } ++ ms_ratio_s = .35*(sidetot/tot)/5.0; ++ } ++ } ++ ++ ++ ++ /*************************************************************** ++ * determin final block type ++ ***************************************************************/ ++ if (force_ms) { ++ /* Forced ms_stereo mode. */ ++ /* ch=0 (mid) blocktype determines ch=1 (side) blocktype */ ++ uselongblock[1] = uselongblock[0]; ++ } ++ ++ ++ for (chn=0; chn<stereo; chn++) { ++ blocktype[chn] = NORM_TYPE; ++ } ++ ++ if (!allow_diff_short) ++ if (info->mode==MPG_MD_JOINT_STEREO) { ++ /* force both channels to use the same block type */ ++ /* this is necessary if the frame is to be encoded in ms_stereo. */ ++ /* But even without ms_stereo, FhG does this */ ++ int bothlong= (uselongblock[0] && uselongblock[1]); ++ if (!bothlong) { ++ uselongblock[0]=0; ++ uselongblock[1]=0; ++ } ++ } ++ ++ ++ ++ ++ /* update the blocktype of the previous granule, since it depends on what ++ * happend in this granule */ ++ for (chn=0; chn<stereo; chn++) { ++ if ( uselongblock[chn]) ++ { /* no attack : use long blocks */ ++ switch( blocktype_old[chn] ) ++ { ++ case NORM_TYPE: ++ case STOP_TYPE: ++ blocktype[chn] = NORM_TYPE; ++ break; ++ case SHORT_TYPE: ++ blocktype[chn] = STOP_TYPE; ++ break; ++ case START_TYPE: ++ fprintf( stderr, "Error in block selecting\n" ); ++ abort(); ++ break; /* problem */ ++ } ++ } else { ++ /* attack : use short blocks */ ++ blocktype[chn] = SHORT_TYPE; ++ if ( blocktype_old[chn] == NORM_TYPE ) { ++ blocktype_old[chn] = START_TYPE; ++ } ++ if ( blocktype_old[chn] == STOP_TYPE ) { ++ blocktype_old[chn] = SHORT_TYPE ; ++ } ++ } ++ ++ ++ blocktype_d[chn] = blocktype_old[chn]; /* value returned to calling program */ ++ blocktype_old[chn] = blocktype[chn]; /* save for next call to l3psy_anal */ ++ } ++ ++ if (blocktype_d[0]==2) ++ *ms_ener_ratio = ms_ratio_s_old; ++ else ++ *ms_ener_ratio = ms_ratio_l_old; ++ ms_ratio_s_old = ms_ratio_s; ++ ms_ratio_l_old = ms_ratio_l; ++ ++ ++ ++ } ++ ++ ++ ++ ++ ++ ++ void L3para_read(double sfreq, int *numlines_l,int *numlines_s, int *partition_l, double *minval, ++ double *qthr_l, double *norm_l, double (*s3_l)[63], double s3_s[CBANDS][CBANDS], ++ int *partition_s, double *qthr_s, double *norm_s, double *SNR, ++ int *cbw_l, int *bu_l, int *bo_l, double *w1_l, double *w2_l, ++ int *cbw_s, int *bu_s, int *bo_s, double *w1_s, double *w2_s) ++ { ++ double freq_tp; ++ static double bval_l[CBANDS], bval_s[CBANDS]; ++ int cbmax=0, cbmax_tp; ++ // static double s3_s[CBANDS][CBANDS]; ++ double *p = psy_data; ++ ++ int sbmax ; ++ int i,j,k,k2,loop, part_max ; ++ ++ /* Read long block data */ ++ ++ for(loop=0;loop<6;loop++) ++ { ++ freq_tp = *p++; ++ cbmax_tp = (int) *p++; ++ cbmax_tp++; ++ ++ if (sfreq == freq_tp ) ++ { ++ cbmax = cbmax_tp; ++ for(i=0,k2=0;i<cbmax_tp;i++) ++ { ++ j = (int) *p++; ++ numlines_l[i] = (int) *p++; ++ minval[i] = *p++; ++ qthr_l[i] = *p++; ++ norm_l[i] = *p++; ++ bval_l[i] = *p++; ++ if (j!=i) ++ { ++ fprintf(stderr,"1. please check \"psy_data\""); ++ exit(-1); ++ } ++ for(k=0;k<numlines_l[i];k++) ++ partition_l[k2++] = i ; ++ } ++ } ++ else ++ p += cbmax_tp * 6; ++ } ++ ++ #define NEWBARKXX ++ #ifdef NEWBARK ++ /* compute bark values of each critical band */ ++ for(i=0;i<cbmax;i++) { ++ for (j=0;(i != partition_l[j]);j++); ++ { double ji = j + (numlines_l[i]-1)/2.0; ++ double freq = sfreq*ji/1024000.0; ++ double bark = 13*atan(.76*freq) + 3.5*atan(freq*freq/(7.5*7.5)); ++ printf("%i %i bval_l = %f %f %f \n",i,j,bval_l[i],freq,bark); ++ bval_l[i]=bark; ++ } ++ } ++ #endif ++ ++ /************************************************************************ ++ * Now compute the spreading function, s[j][i], the value of the spread-* ++ * ing function, centered at band j, for band i, store for later use * ++ ************************************************************************/ ++ part_max = cbmax ; ++ for(i=0;i<part_max;i++) ++ { ++ double tempx,x,tempy,temp; ++ for(j=0;j<part_max;j++) ++ { ++ //tempx = (bval_l[i] - bval_l[j])*1.05; ++ if (j>=i) tempx = (bval_l[i] - bval_l[j])*3.0; ++ else tempx = (bval_l[i] - bval_l[j])*1.5; ++ /* if (j>=i) tempx = (bval_l[j] - bval_l[i])*3.0; ++ else tempx = (bval_l[j] - bval_l[i])*1.5; */ ++ ++ ++ if(tempx>=0.5 && tempx<=2.5) ++ { ++ temp = tempx - 0.5; ++ x = 8.0 * (temp*temp - 2.0 * temp); ++ } ++ else x = 0.0; ++ tempx += 0.474; ++ tempy = 15.811389 + 7.5*tempx - 17.5*sqrt(1.0+tempx*tempx); ++ ++ #ifdef NEWS3 ++ if (j>=i) tempy = (bval_l[j] - bval_l[i])*(-15); ++ else tempy = (bval_l[j] - bval_l[i])*25; ++ x=0; ++ #endif ++ /* ++ if ((i==part_max/2) && (fabs(bval_l[j] - bval_l[i])) < 3) { ++ printf("bark=%f x+tempy = %f \n",bval_l[j] - bval_l[i],x+tempy); ++ //printf("i,j %i %i bark=%f %f \n",i,j,bval_l[j],j/(.001+bval_l[j])); ++ } ++ */ ++ ++ if (tempy <= -60.0) s3_l[i][j] = 0.0; ++ else s3_l[i][j] = exp( (x + tempy)*LN_TO_LOG10 ); ++ } ++ } ++ ++ /* Read short block data */ ++ ++ for(loop=0;loop<6;loop++) ++ { ++ freq_tp = *p++; ++ cbmax_tp = (int) *p++; ++ cbmax_tp++; ++ ++ if (sfreq == freq_tp ) ++ { ++ cbmax = cbmax_tp; ++ for(i=0,k2=0;i<cbmax_tp;i++) ++ { ++ j = (int) *p++; ++ numlines_s[i] = (int) *p++; ++ qthr_s[i] = *p++; ++ norm_s[i] = *p++; ++ SNR[i] = *p++; ++ bval_s[i] = *p++; ++ if (j!=i) ++ { ++ fprintf(stderr,"3. please check \"psy_data\""); ++ exit(-1); ++ } ++ for(k=0;k<numlines_s[i];k++) ++ partition_s[k2++] = i ; ++ } ++ } ++ else ++ p += cbmax_tp * 6; ++ } ++ ++ ++ #ifdef NEWBARK ++ /* compute bark values of each critical band */ ++ for(i=0;i<cbmax;i++) { ++ for (j=0;(i != partition_s[j]);j++); ++ { double ji = j + (numlines_s[i]-1)/2.0; ++ double freq = sfreq*ji/256000.0; ++ double bark = 13*atan(.76*freq) + 3.5*atan(freq*freq/(7.5*7.5)); ++ printf("%i %i bval_s = %f %f %f \n",i,j,bval_s[i],freq,bark); ++ bval_s[i]=bark; ++ } ++ } ++ #endif ++ ++ ++ ++ /************************************************************************ ++ * Now compute the spreading function, s[j][i], the value of the spread-* ++ * ing function, centered at band j, for band i, store for later use * ++ ************************************************************************/ ++ part_max = cbmax ; ++ for(i=0;i<part_max;i++) ++ { ++ double tempx,x,tempy,temp; ++ for(j=0;j<part_max;j++) ++ { ++ // tempx = (bval_s[i] - bval_s[j])*1.05; ++ if (j>=i) tempx = (bval_s[i] - bval_s[j])*3.0; ++ else tempx = (bval_s[i] - bval_s[j])*1.5; ++ if(tempx>=0.5 && tempx<=2.5) ++ { ++ temp = tempx - 0.5; ++ x = 8.0 * (temp*temp - 2.0 * temp); ++ } ++ else x = 0.0; ++ tempx += 0.474; ++ tempy = 15.811389 + 7.5*tempx - 17.5*sqrt(1.0+tempx*tempx); ++ #ifdef NEWS3 ++ if (j>=i) tempy = (bval_s[j] - bval_s[i])*(-15); ++ else tempy = (bval_s[j] - bval_s[i])*25; ++ x=0; ++ #endif ++ if (tempy <= -60.0) s3_s[i][j] = 0.0; ++ else s3_s[i][j] = exp( (x + tempy)*LN_TO_LOG10 ); ++ } ++ } ++ /* Read long block data for converting threshold calculation ++ partitions to scale factor bands */ ++ ++ for(loop=0;loop<6;loop++) ++ { ++ freq_tp = *p++; ++ sbmax = (int) *p++; ++ sbmax++; ++ ++ if (sfreq == freq_tp) ++ { ++ for(i=0;i<sbmax;i++) ++ { ++ j = (int) *p++; ++ cbw_l[i] = (int) *p++; ++ bu_l[i] = (int) *p++; ++ bo_l[i] = (int) *p++; ++ w1_l[i] = (double) *p++; ++ w2_l[i] = (double) *p++; ++ if (j!=i) ++ { fprintf(stderr,"30:please check \"psy_data\"\n"); ++ exit(-1); ++ } ++ ++ if (i!=0) ++ if ( (bo_l[i] != (bu_l[i]+cbw_l[i])) || ++ (fabs(1.0-w1_l[i]-w2_l[i-1]) > 0.01 ) ) ++ { ++ fprintf(stderr,"31l: please check \"psy_data.\"\n"); ++ exit(-1); ++ } ++ } ++ } ++ else ++ p += sbmax * 6; ++ } ++ ++ /* Read short block data for converting threshold calculation ++ partitions to scale factor bands */ ++ ++ for(loop=0;loop<6;loop++) ++ { ++ freq_tp = *p++; ++ sbmax = (int) *p++; ++ sbmax++; ++ ++ if (sfreq == freq_tp) ++ { ++ for(i=0;i<sbmax;i++) ++ { ++ j = (int) *p++; ++ cbw_s[i] = (int) *p++; ++ bu_s[i] = (int) *p++; ++ bo_s[i] = (int) *p++; ++ w1_s[i] = *p++; ++ w2_s[i] = *p++; ++ if (j!=i) ++ { fprintf(stderr,"30:please check \"psy_data\"\n"); ++ exit(-1); ++ } ++ ++ if (i!=0) ++ if ( (bo_s[i] != (bu_s[i]+cbw_s[i])) || ++ (fabs(1.0-w1_s[i]-w2_s[i-1]) > 0.01 ) ) ++ { fprintf(stderr,"31s: please check \"psy_data.\"\n"); ++ exit(-1); ++ } ++ } ++ } ++ else ++ p += sbmax * 6; ++ } ++ ++ } +diff -r -c -N encoder/psymodel.h lame3.50/psymodel.h +*** encoder/psymodel.h Wed Dec 31 17:00:00 1969 +--- lame3.50/psymodel.h Tue Oct 5 18:18:16 1999 +*************** +*** 0 **** +--- 1,24 ---- ++ #ifndef L3PSY_DOT_H_ ++ #define L3PSY_DOT_H_ ++ #define BLKSIZE_s 256 ++ #define HBLKSIZE_s 129 ++ #define SBMAX_l 21 ++ #define SBMAX_s 12 ++ ++ ++ /* #define switch_pe 1800 */ ++ #define NORM_TYPE 0 ++ #define START_TYPE 1 ++ #define SHORT_TYPE 2 ++ #define STOP_TYPE 3 ++ ++ /* l3psy.c */ ++ #include "l3side.h" ++ void L3psycho_anal( short int *buffer[2], int chn, ++ int gr , layer *info, ++ double sfreq, ++ int check_ms, double *ms_ener_ratio, ++ double ratio_d[2][21], double ratio_ds[2][12][3], ++ double pe[2], ++ int blocktype_d[2]); ++ #endif +diff -r -c -N encoder/quantize-pvt.h lame3.50/quantize-pvt.h +*** encoder/quantize-pvt.h Wed Dec 31 17:00:00 1969 +--- lame3.50/quantize-pvt.h Sun Oct 3 16:14:46 1999 +*************** +*** 0 **** +--- 1,178 ---- ++ /********************************************************************** ++ * ISO MPEG Audio Subgroup Software Simulation Group (1996) ++ * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension ++ * ++ * $Id: loop-pvt.h,v 1.1 1996/02/14 04:04:23 rowlands Exp $ ++ * ++ * Private interface declarations for loop.c ++ * ++ * $Log: loop-pvt.h,v $ ++ * Revision 1.1 1996/02/14 04:04:23 rowlands ++ * Initial revision ++ * ++ * Received from Mike Coleman ++ **********************************************************************/ ++ ++ #ifndef LOOP_PVT_H ++ #define LOOP_PVT_H ++ ++ /* ++ Revision History: ++ ++ Date Programmer Comment ++ ========== ========================= =============================== ++ 1995/10/01 mc@fivebats.com created ++ ++ */ ++ ++ void outer_loop( double xr[2][2][576], /*vector of the magnitudees of the spectral values */ ++ int bits, ++ double noise[4], ++ double targ_noise[4], /* VBR target noise info */ ++ int sloppy, /* 1=quit as soon as noise < targ_noise */ ++ III_psy_xmin *l3_xmin, /* the allowed distortion of the scalefactor */ ++ int l3_enc[2][2][576], /* vector of quantized values ix(0..575) */ ++ frame_params *fr_ps, ++ III_scalefac_t *scalefac, /* scalefactors */ ++ int gr, ++ int stereo, ++ III_side_info_t *l3_side, ++ III_psy_ratio *ratio, ++ double ms_ratio, ++ int ch); ++ ++ ++ void outer_loop_dual( double xr[2][2][576], /*vector of the magnitudees of the spectral values */ ++ double xr_org[2][2][576], ++ int mean_bits, ++ int VBRbits[2][2], ++ int bit_rate, ++ int best_over[2], ++ III_psy_xmin *l3_xmin, /* the allowed distortion of the scalefactor */ ++ int l3_enc[2][2][576], /* vector of quantized values ix(0..575) */ ++ frame_params *fr_ps, ++ III_scalefac_t *scalefac, /* scalefactors */ ++ int gr, ++ int ch, ++ III_side_info_t *l3_side, ++ III_psy_ratio *ratio, ++ double pe[2][2], ++ double ms_ratio[2]); ++ ++ ++ ++ int part2_length( III_scalefac_t *scalefac, ++ int version, gr_info *cod_info); ++ ++ int quantanf_init( double xr[576] ); ++ ++ int inner_loop( double xr[2][2][576], double xrpow[576], ++ int l3_enc[2][2][576], ++ int max_bits, ++ gr_info *cod_info, ++ int gr, ++ int ch ); ++ void calc_xmin( double xr[2][2][576], ++ III_psy_ratio *ratio, ++ gr_info *cod_info, ++ III_psy_xmin *l3_xmin, ++ int gr, ++ int ch ); ++ double xr_max( double xr[576], ++ unsigned int begin, ++ unsigned int end ); ++ ++ void calc_scfsi( double xr[576], ++ III_side_info_t *l3_side, ++ III_psy_xmin *l3_xmin, ++ int ch, ++ int gr ); ++ ++ void gr_deco( gr_info *cod_info ); ++ ++ ++ int count_bit( int ix[576], unsigned int start, unsigned int end, unsigned int table); ++ int bigv_bitcount( int ix[576], gr_info *cod_info ); ++ int choose_table( int max); ++ void bigv_tab_select( int ix[576], gr_info *cod_info ); ++ void subdivide( gr_info *cod_info ); ++ int count1_bitcount( int ix[576], gr_info *cod_info ); ++ void calc_runlen( int ix[576], ++ gr_info *cod_info ); ++ int scale_bitcount( III_scalefac_t *scalefac, ++ gr_info *cod_info, ++ int gr, ++ int ch ); ++ int scale_bitcount_lsf( III_scalefac_t *scalefac, gr_info *cod_info, ++ int gr, int ch ); ++ int calc_noise1( double xr[576], ++ int ix[576], ++ gr_info *cod_info, ++ double xfsf[4][CBLIMIT], ++ double distort[4][CBLIMIT], ++ III_psy_xmin *l3_xmin, ++ int gr, int ch, ++ double *noise, double *tot_noise, double *max_noise); ++ ++ void calc_noise2( double xr[2][576], ++ int ix[2][576], ++ gr_info *cod_info[2], ++ double xfsf[2][4][CBLIMIT], ++ double distort[2][4][CBLIMIT], ++ III_psy_xmin *l3_xmin, ++ int gr, int ch, int over[2], ++ double noise[2], double tot_noise[2], double max_noise[2]); ++ ++ ++ int loop_break( III_scalefac_t *scalefac, ++ gr_info *cod_info, ++ int gr, ++ int ch ); ++ int preemphasis( double xr[576], double xrpow[576], ++ III_psy_xmin *l3_xmin, ++ int gr, ++ int ch, ++ III_side_info_t *l3_side, ++ double distort[4][CBLIMIT] ); ++ int amp_scalefac_bands( double xr[576], double xrpow[576], ++ III_psy_xmin *l3_xmin, ++ III_side_info_t *l3_side, ++ III_scalefac_t *scalefac, ++ int gr, ++ int ch, ++ int iteration, ++ double distort[4][CBLIMIT]); ++ int quantize( double xr[576], ++ int ix[576], ++ gr_info *cod_info ); ++ void quantize_xrpow( double xr[576], ++ int ix[576], ++ gr_info *cod_info ); ++ int ix_max( int ix[576], ++ unsigned int begin, ++ unsigned int end ); ++ ++ ++ int ++ new_choose_table( int ix[576], ++ unsigned int begin, ++ unsigned int end, int * s ); ++ ++ /* New SS 20-12-96 */ ++ int bin_search_StepSize(int desired_rate, double start, int bot, int ix[576], ++ double xrs[576], double xrspow[576], gr_info * cod_info); ++ int bin_search_StepSize2(int desired_rate, double start, int bot, int ix[576], ++ double xrs[576], double xrspow[576], gr_info * cod_info); ++ int count_bits(int *ix,gr_info *cod_info); ++ ++ ++ int quant_compare( ++ int best_over,double best_tot_noise,double best_over_noise,double best_max_over, ++ int over,double tot_noise, double over_noise,double max_noise); ++ ++ int VBR_compare( ++ int best_over,double best_tot_noise,double best_over_noise,double best_max_over, ++ int over,double tot_noise, double over_noise,double max_noise); ++ ++ ++ #endif +diff -r -c -N encoder/quantize.c lame3.50/quantize.c +*** encoder/quantize.c Wed Dec 31 17:00:00 1969 +--- lame3.50/quantize.c Fri Oct 29 00:06:26 1999 +*************** +*** 0 **** +--- 1,2046 ---- ++ /* ++ * MP3 quantization ++ * ++ * Copyright (c) 1999 Mark Taylor ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License as published by ++ * the Free Software Foundation; either version 2, or (at your option) ++ * any later version. ++ * ++ * This program is distributed in the hope that it will be useful, ++ * but WITHOUT ANY WARRANTY; without even the implied warranty of ++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the ++ * GNU General Public License for more details. ++ * ++ * You should have received a copy of the GNU General Public License ++ * along with this program; see the file COPYING. If not, write to ++ * the Free Software Foundation, 675 Mass Ave, Cambridge, MA 02139, USA. ++ */ ++ ++ ++ #include <assert.h> ++ #include "globalflags.h" ++ #include "util.h" ++ #include "l3side.h" ++ #include "quantize.h" ++ #include "l3bitstream.h" ++ #include "reservoir.h" ++ #include "quantize-pvt.h" ++ #include "gtkanal.h" ++ ++ #define PRECALC_SIZE 8206 /* 8191+15. should never be outside this. see count_bits() */ ++ ++ static double masking_lower=1; ++ ++ ++ static int convert_mdct, convert_psy, reduce_sidechannel; ++ /* ++ mt 5/99. These global flags denote 4 possibilities: ++ mode l3_xmin ++ 1 MDCT input L/R, quantize L/R, psy-model thresholds: L/R -m s either ++ 2 MDCT input L/R, quantize M/S, psy-model thresholds: L/R -m j orig ++ 3 MDCT input M/S, quantize M/S, psy-model thresholds: M/S -m f either ++ 4 MDCT input L/R, quantize M/S, psy-model thresholds: M/S -m j -h m/s ++ ++ 1: convert_mdct = 0, convert_psy=0, reduce_sidechannel=0 ++ 2: convert_mdct = 1, convert_psy=1, reduce_sidechannel=1 ++ 3: convert_mdct = 0, convert_psy=0, reduce_sidechannel=1 ++ 4: convert_mdct = 1, convert_psy=0, reduce_sidechannel=1 ++ ++ if (convert_mdct), then iteration_loop will quantize M/S data from ++ the L/R input MDCT coefficients. ++ ++ if (convert_psy), then calc_noise will compute the noise for the L/R ++ channels from M/S MDCT data and L/R psy-model threshold information. ++ Distortion in ether L or R channel will be marked as distortion in ++ both Mid and Side channels. ++ ++ if (reduce_sidechannel) then outer_loop will allocate less bits ++ to the side channel and more bits to the mid channel based on relative ++ energies. ++ */ ++ ++ ++ ++ ++ /* ++ Here are MPEG1 Table B.8 and MPEG2 Table B.1 ++ -- Layer III scalefactor bands. ++ Index into this using a method such as: ++ idx = fr_ps->header->sampling_frequency ++ + (fr_ps->header->version * 3) ++ */ ++ ++ struct scalefac_struct sfBandIndex[6] = ++ { ++ ++ { /* Table B.2.b: 22.05 kHz */ ++ {0,6,12,18,24,30,36,44,54,66,80,96,116,140,168,200,238,284,336,396,464,522,576}, ++ {0,4,8,12,18,24,32,42,56,74,100,132,174,192} ++ }, ++ { /* Table B.2.c: 24 kHz */ ++ {0,6,12,18,24,30,36,44,54,66,80,96,114,136,162,194,232,278,332,394,464,540,576}, ++ {0,4,8,12,18,26,36,48,62,80,104,136,180,192} ++ }, ++ { /* Table B.2.a: 16 kHz */ ++ {0,6,12,18,24,30,36,44,54,66,80,96,116,140,168,200,238,284,336,396,464,522,576}, ++ {0,4,8,12,18,26,36,48,62,80,104,134,174,192} ++ }, ++ { /* Table B.8.b: 44.1 kHz */ ++ {0,4,8,12,16,20,24,30,36,44,52,62,74,90,110,134,162,196,238,288,342,418,576}, ++ {0,4,8,12,16,22,30,40,52,66,84,106,136,192} ++ }, ++ { /* Table B.8.c: 48 kHz */ ++ {0,4,8,12,16,20,24,30,36,42,50,60,72,88,106,128,156,190,230,276,330,384,576}, ++ {0,4,8,12,16,22,28,38,50,64,80,100,126,192} ++ }, ++ { /* Table B.8.a: 32 kHz */ ++ {0,4,8,12,16,20,24,30,36,44,54,66,82,102,126,156,194,240,296,364,448,550,576}, ++ {0,4,8,12,16,22,30,42,58,78,104,138,180,192} ++ } ++ }; ++ ++ /* ++ The following table is used to implement the scalefactor ++ partitioning for MPEG2 as described in section ++ 2.4.3.2 of the IS. The indexing corresponds to the ++ way the tables are presented in the IS: ++ ++ [table_number][row_in_table][column of nr_of_sfb] ++ */ ++ static unsigned nr_of_sfb_block[6][3][4] = ++ { ++ { ++ {6, 5, 5, 5}, ++ {9, 9, 9, 9}, ++ {6, 9, 9, 9} ++ }, ++ { ++ {6, 5, 7, 3}, ++ {9, 9, 12, 6}, ++ {6, 9, 12, 6} ++ }, ++ { ++ {11, 10, 0, 0}, ++ {18, 18, 0, 0}, ++ {15,18,0,0} ++ }, ++ { ++ {7, 7, 7, 0}, ++ {12, 12, 12, 0}, ++ {6, 15, 12, 0} ++ }, ++ { ++ {6, 6, 6, 3}, ++ {12, 9, 9, 6}, ++ {6, 12, 9, 6} ++ }, ++ { ++ {8, 8, 5, 0}, ++ {15,12,9,0}, ++ {6,18,9,0} ++ } ++ }; ++ ++ /* ++ table of largest scalefactors for MPEG2 ++ */ ++ static unsigned max_sfac_tab[6][4] = ++ { ++ {4, 4, 3, 3}, ++ {4, 4, 3, 0}, ++ {3, 2, 0, 0}, ++ {4, 5, 5, 0}, ++ {3, 3, 3, 0}, ++ {2, 2, 0, 0} ++ }; ++ ++ /* Table B.6: layer3 preemphasis */ ++ int pretab[21] = ++ { ++ 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 1, 1, 1, 1, 2, 2, 3, 3, 3, 2 ++ }; ++ ++ /* This is the scfsi_band table from 2.4.2.7 of the IS */ ++ int scfsi_band_long[5] = { 0, 6, 11, 16, 21 }; ++ ++ int *scalefac_band_long = &sfBandIndex[3].l[0]; ++ int *scalefac_band_short = &sfBandIndex[3].s[0]; ++ ++ ++ ++ /* ++ compute the ATH for each scalefactor band ++ cd range: 0..96db ++ ++ Input: 3.3kHz signal 32767 amplitude (3.3kHz is where ATH is smallest = -5db) ++ longblocks: sfb=12 en0/bw=-11db max_en0 = 1.3db ++ shortblocks: sfb=5 -9db 0db ++ ++ Input: 1 1 1 1 1 1 1 -1 -1 -1 -1 -1 -1 -1 (repeated) ++ longblocks: amp=1 sfb=12 en0/bw=-103 db max_en0 = -92db ++ amp=32767 sfb=12 -12 db -1.4db ++ ++ Input: 1 1 1 1 1 1 1 -1 -1 -1 -1 -1 -1 -1 (repeated) ++ shortblocks: amp=1 sfb=5 en0/bw= -99 -86 ++ amp=32767 sfb=5 -9 db 4db ++ ++ ++ MAX energy of largest wave at 3.3kHz = 1db ++ AVE energy of largest wave at 3.3kHz = -11db ++ Let's take AVE: -11db = maximum signal in sfb=12. ++ Dynamic range of CD: 96db. Therefor energy of smallest audible wave ++ in sfb=12 = -11 - 96 = -107db = ATH at 3.3kHz. ++ ++ ATH formula for this wave: -5db. To adjust to LAME scaling, we need ++ ATH = ATH_formula - 103 (db) ++ ATH = ATH * 2.5e-10 (ener) ++ ++ */ ++ static double ATH_l[CBLIMIT]; ++ static double ATH_s[CBLIMIT]; ++ ++ double ATHformula(double f) ++ { ++ double ath; ++ f = Max(0.02, f); ++ /* from Painter & Spanias, 1997 */ ++ /* minimum: (i=77) 3.3kHz = -5db */ ++ ath=(3.640 * pow(f,-0.8) ++ - 6.500 * exp(-0.6*pow(f-3.3,2.0)) ++ + 0.001 * pow(f,4.0)); ++ ++ /* convert to energy */ ++ ath -= 114; /* MDCT scaling. From tests by macik and MUS420 code */ ++ //ath -= 200; /* disabled */ ++ ath = pow( 10.0, ath/10.0 ); ++ return ath; ++ } ++ ++ ++ void compute_ath(layer *info,double ATH_l[CBLIMIT],double ATH_s[CBLIMIT]) ++ { ++ int sfb,i,start,end; ++ double ATH_f; ++ double samp_freq = s_freq[info->version][info->sampling_frequency]; ++ ++ /* last sfb is not used */ ++ for ( sfb = 0; sfb < SFB_LMAX-1; sfb++ ) { ++ start = scalefac_band_long[ sfb ]; ++ end = scalefac_band_long[ sfb+1 ]; ++ ATH_l[sfb]=1e99; ++ for (i=start ; i < end; i++) { ++ ATH_f = ATHformula(samp_freq*i/(2*576)); /* freq in kHz */ ++ ATH_l[sfb]=Min(ATH_l[sfb],ATH_f); ++ } ++ /* ++ printf("sfb=%i %f ATH=%f %f %f \n",sfb,samp_freq*start/(2*576), ++ 10*log10(ATH_l[sfb]), ++ 10*log10( ATHformula(samp_freq*start/(2*576))) , ++ 10*log10(ATHformula(samp_freq*end/(2*576)))); ++ */ ++ } ++ ++ for ( sfb = 0; sfb < SFB_SMAX - 1; sfb++ ){ ++ start = scalefac_band_short[ sfb ]; ++ end = scalefac_band_short[ sfb+1 ]; ++ ATH_s[sfb]=1e99; ++ for (i=start ; i < end; i++) { ++ ATH_f = ATHformula(samp_freq*i/(2*192)); /* freq in kHz */ ++ ATH_s[sfb]=Min(ATH_s[sfb],ATH_f); ++ } ++ } ++ } ++ ++ ++ ++ ++ ++ /* convert from L/R <-> Mid/Side */ ++ void ms_convert(double xr[2][576],double xr_org[2][576]) ++ { ++ int i; ++ for ( i = 0; i < 576; i++ ) { ++ xr[0][i] = (xr_org[0][i]+xr_org[1][i])/SQRT2; ++ xr[1][i] = (xr_org[0][i]-xr_org[1][i])/SQRT2; ++ } ++ } ++ ++ ++ ++ /************************************************************************ ++ * allocate bits among 2 channels based on PE ++ * mt 6/99 ++ ************************************************************************/ ++ void on_pe(double pe[2][2],III_side_info_t *l3_side, ++ int targ_bits[2],int mean_bits,int stereo, int gr) ++ { ++ gr_info *cod_info; ++ int extra_bits,tbits,bits; ++ int add_bits[2]; ++ int ch; ++ ++ /* allocate targ_bits for granule */ ++ ResvMaxBits2( mean_bits, &tbits, &extra_bits, gr); ++ ++ ++ for (ch=0 ; ch < stereo ; ch ++) { ++ /****************************************************************** ++ * allocate bits for each channel ++ ******************************************************************/ ++ cod_info = &l3_side->gr[gr].ch[ch].tt; ++ ++ targ_bits[ch]=tbits/stereo; ++ ++ // allocate extra bits from reservoir based on PE ++ bits=0; ++ ++ /* extra bits based on PE > 700 */ ++ add_bits[ch]=(pe[gr][ch]-750)/1.4; /* 2.0; */ ++ ++ /* short blocks need extra, no matter what the pe */ ++ if (cod_info->block_type==2) ++ if (add_bits[ch]<500) add_bits[ch]=500; ++ ++ if (add_bits[ch] < 0) add_bits[ch]=0; ++ bits += add_bits[ch]; ++ ++ if (bits > extra_bits) add_bits[ch] = (extra_bits*add_bits[ch])/bits; ++ if ((targ_bits[ch]+add_bits[ch]) > 4095) ++ add_bits[ch]=4095-targ_bits[ch]; ++ ++ targ_bits[ch] = targ_bits[ch] + add_bits[ch]; ++ extra_bits -= add_bits[ch]; ++ } ++ } ++ ++ void reduce_side(int targ_bits[2],double ms_ener_ratio,int mean_bits) ++ { ++ int ch; ++ int stereo=2; ++ /* ms_ener_ratio = 0: allocate 66/33 mid/side fac=.33 ++ * ms_ener_ratio =.5: allocate 50/50 mid/side fac= 0 */ ++ /* 75/25 split is fac=.5 */ ++ /* float fac = .50*(.5-ms_ener_ratio[gr])/.5;*/ ++ float fac = .33*(.5-ms_ener_ratio)/.5; ++ if (fac<0) fac=0; ++ ++ if (targ_bits[1] >= 125) { ++ /* dont reduce side channel below 125 bits */ ++ if (targ_bits[1]-targ_bits[1]*fac > 125) { ++ targ_bits[0] += targ_bits[1]*fac; ++ targ_bits[1] -= targ_bits[1]*fac; ++ } else { ++ targ_bits[0] += targ_bits[1] - 125; ++ targ_bits[1] = 125; ++ } ++ } ++ ++ /* dont allow to many bits per channel */ ++ for (ch=0; ch<stereo; ch++) { ++ int max_bits = Min(4095,mean_bits/2 + 1200); ++ if (targ_bits[ch] > max_bits) { ++ targ_bits[ch] = max_bits; ++ } ++ } ++ ++ } ++ ++ ++ ++ ++ /************************************************************************/ ++ /* initialization for iteration_loop */ ++ /************************************************************************/ ++ void ++ iteration_init( double xr_org[2][2][576], ++ III_side_info_t *l3_side, int l3_enc[2][2][576], ++ frame_params *fr_ps, III_psy_xmin *l3_xmin) ++ { ++ gr_info *cod_info; ++ layer *info = fr_ps->header; ++ int stereo = fr_ps->stereo; ++ int ch, gr, i, mode_gr; ++ // int bit_rate,samp_freq; ++ ++ ++ if ( frameNum==0 ) { ++ l3_side->main_data_begin = 0; ++ memset((char *) &l3_xmin, 0, sizeof(l3_xmin)); ++ compute_ath(info,ATH_l,ATH_s); ++ } ++ ++ l3_side->resvDrain = 0; ++ mode_gr = (info->version == 1) ? 2 : 1; ++ //bit_rate = bitrate[info->version][info->lay-1][info->bitrate_index]; ++ //samp_freq = s_freq[info->version][info->sampling_frequency]*1000 +.5; ++ ++ scalefac_band_long = &sfBandIndex[info->sampling_frequency + (info->version * 3)].l[0]; ++ scalefac_band_short = &sfBandIndex[info->sampling_frequency + (info->version * 3)].s[0]; ++ ++ ++ convert_mdct=0; ++ convert_psy=0; ++ reduce_sidechannel=0; ++ if (info->mode_ext==MPG_MD_MS_LR) { ++ if (highq) { ++ convert_mdct = 1; ++ convert_psy = 0; ++ reduce_sidechannel=1; ++ }else{ ++ convert_mdct = 1; ++ convert_psy = 1; ++ reduce_sidechannel=1; ++ } ++ } ++ if (force_ms) { ++ convert_mdct = 0; ++ convert_psy = 0; ++ reduce_sidechannel=1; ++ } ++ if (convert_psy) memset(l3_enc,0,sizeof(int)*2*2*576); ++ ++ ++ /* voice mode disables short blocks */ ++ if (voice_mode) { ++ for ( gr = 0; gr < mode_gr; gr++ ) { ++ for (ch =0 ; ch < stereo ; ch++) { ++ int start = scalefac_band_long[ SFB_LMAX-3 ]; ++ for ( i = start; i < 576; i++ ) ++ xr_org[gr][ch][i]=0; ++ } ++ for (ch =0 ; ch < stereo ; ch++) { ++ for ( i = 0; i < 2; i++ ) ++ xr_org[gr][ch][i]=0; ++ } ++ } ++ } ++ ++ if (sfb21) { ++ for ( gr = 0; gr < mode_gr; gr++ ) { ++ for (ch =0 ; ch < stereo ; ch++) { ++ int shortblock = (l3_side->gr[gr].ch[0].tt.block_type==2); ++ if (shortblock) { ++ int j; ++ for (j=0; j<3; j++) { ++ int start = scalefac_band_short[ SFB_SMAX-1 ]; ++ for ( i = start; i < 192; i++ ) { ++ int i0 = 3*i+j; ++ xr_org[gr][ch][i0]=0; ++ } ++ } ++ }else{ ++ int start = scalefac_band_long[ SFB_LMAX-1 ]; ++ for ( i = start; i < 576; i++ ) xr_org[gr][ch][i]=0; ++ } ++ } ++ } ++ } ++ ++ ++ ++ /* some intializations. */ ++ for ( gr = 0; gr < mode_gr; gr++ ){ ++ for ( ch = 0; ch < stereo; ch++ ){ ++ cod_info = (gr_info *) &(l3_side->gr[gr].ch[ch]); ++ gr_deco(cod_info); ++ } ++ } ++ ++ ++ /* dont bother with scfsi. */ ++ for ( ch = 0; ch < stereo; ch++ ) ++ for ( i = 0; i < 4; i++ ) ++ l3_side->scfsi[ch][i] = 0; ++ ++ ++ ++ ++ } ++ ++ ++ ++ ++ /************************************************************************/ ++ /* iteration_loop() */ ++ /************************************************************************/ ++ void ++ iteration_loop( double pe[2][2], double ms_ener_ratio[2], ++ double xr_org[2][2][576], III_psy_ratio *ratio, ++ III_side_info_t *l3_side, int l3_enc[2][2][576], ++ III_scalefac_t *scalefac, frame_params *fr_ps) ++ { ++ III_psy_xmin l3_xmin; ++ gr_info *cod_info; ++ layer *info; ++ int VBRbits[2][2]; ++ int over[2]; ++ double noise[4]; /* over,max_noise,over_noise,tot_noise; */ ++ double targ_noise[4]; /* over,max_noise,over_noise,tot_noise; */ ++ int bitsPerFrame; ++ int mean_bits; ++ int stereo = fr_ps->stereo; ++ int ch, gr, i, mode_gr, bit_rate; ++ double xr[2][2][576]; ++ ++ ++ iteration_init(xr_org,l3_side,l3_enc,fr_ps,&l3_xmin); ++ info = fr_ps->header; ++ mode_gr = (info->version == 1) ? 2 : 1; ++ bit_rate = bitrate[info->version][info->lay-1][info->bitrate_index]; ++ ++ ++ getframebits(info,stereo,&bitsPerFrame,&mean_bits); ++ ResvFrameBegin( fr_ps, l3_side, mean_bits, bitsPerFrame ); ++ ++ /* quantize! */ ++ ++ for ( gr = 0; gr < mode_gr; gr++ ) { ++ if (convert_psy) { ++ /* dual channel version can quantize Mid/Side channels with L/R ++ * maskings (by constantly reconstructing L/R data). Used before we ++ * we had proper mid/side maskings. */ ++ outer_loop_dual( xr, xr_org, mean_bits, VBRbits, bit_rate, over, ++ &l3_xmin,l3_enc, fr_ps, ++ scalefac,gr,stereo, l3_side, ratio, pe, ms_ener_ratio); ++ } else { ++ int targ_bits[2]; ++ /* copy data to be quantized into xr */ ++ if (convert_mdct) ms_convert(xr[gr],xr_org[gr]); ++ else memcpy(xr[gr],xr_org[gr],sizeof(double)*2*576); ++ ++ for (ch=0 ; ch < stereo ; ch ++) targ_bits[ch]=VBRbits[gr][ch]; ++ on_pe(pe,l3_side,targ_bits,mean_bits,stereo,gr); ++ if (reduce_sidechannel) ++ reduce_side(targ_bits,ms_ener_ratio[gr],mean_bits); ++ ++ for (ch=0 ; ch < stereo ; ch ++) { ++ outer_loop( xr, targ_bits[ch], noise, targ_noise, 0, &l3_xmin,l3_enc, ++ fr_ps, scalefac,gr,stereo, l3_side, ratio, ms_ener_ratio[gr],ch); ++ cod_info = &l3_side->gr[gr].ch[ch].tt; ++ ResvAdjust( fr_ps, cod_info, l3_side, mean_bits ); ++ } ++ } ++ } ++ ++ ++ /* set the sign of l3_enc */ ++ for ( gr = 0; gr < mode_gr; gr++ ) { ++ for ( ch = 0; ch < stereo; ch++ ) { ++ int *pi = &l3_enc[gr][ch][0]; ++ for ( i = 0; i < 576; i++) { ++ double pr = xr[gr][ch][i]; ++ if ( (pr < 0) && (pi[i] > 0) ) pi[i] *= -1; ++ } ++ } ++ } ++ ResvFrameEnd( fr_ps, l3_side, mean_bits ); ++ } ++ ++ ++ /************************************************************************ ++ * ++ * VBR_iteration_loop() ++ * ++ * tries to find out how many bits are needed for each granule and channel ++ * to get an acceptable quantization. An appropriate bitrate will then be ++ * choosed for quantization. rh 8/99 ++ * ++ ************************************************************************/ ++ void ++ VBR_iteration_loop (double pe[2][2], double ms_ener_ratio[2], ++ double xr_org[2][2][576], III_psy_ratio * ratio, ++ III_side_info_t * l3_side, int l3_enc[2][2][576], ++ III_scalefac_t * scalefac, frame_params * fr_ps) ++ { ++ III_psy_xmin l3_xmin; ++ gr_info *cod_info = NULL; ++ layer *info; ++ int save_bits[2][2]; ++ double noise[4]; /* over,max_noise,over_noise,tot_noise; */ ++ double targ_noise[4]; /* over,max_noise,over_noise,tot_noise; */ ++ int this_bits, dbits; ++ int used_bits=0; ++ int min_bits,max_bits,min_mean_bits=0; ++ int frameBits[15]; ++ int bitsPerFrame; ++ int bits; ++ int mean_bits; ++ int stereo = fr_ps->stereo; ++ int i,ch, gr, mode_gr; ++ double xr[2][2][576]; ++ double xr_save[576]; ++ double masking_lower_db; ++ ++ iteration_init(xr_org,l3_side,l3_enc,fr_ps,&l3_xmin); ++ info = fr_ps->header; ++ mode_gr = (info->version == 1) ? 2 : 1; ++ ++ ++ /******************************************************************* ++ * how many bits are available for each bitrate? ++ *******************************************************************/ ++ for( info->bitrate_index = VBR_min_bitrate; ++ info->bitrate_index <= VBR_max_bitrate; ++ info->bitrate_index++ ) { ++ getframebits (info, stereo, &bitsPerFrame, &mean_bits); ++ if (info->bitrate_index == VBR_min_bitrate) { ++ /* always use at least this many bits per granule per channel */ ++ min_mean_bits=mean_bits/stereo; ++ } ++ frameBits[info->bitrate_index]= ++ ResvFrameBegin (fr_ps, l3_side, mean_bits, bitsPerFrame); ++ //printf("framebits = %i \n",frameBits[info->bitrate_index]); ++ } ++ ++ info->bitrate_index=VBR_max_bitrate; ++ ++ ++ /******************************************************************* ++ * how many bits would we use of it? ++ *******************************************************************/ ++ for (gr = 0; gr < mode_gr; gr++) { ++ int num_chan=stereo; ++ /* determine quality based on mid channel only */ ++ if (reduce_sidechannel) num_chan=1; ++ ++ /* copy data to be quantized into xr */ ++ if (convert_mdct) ms_convert(xr[gr],xr_org[gr]); ++ else memcpy(xr[gr],xr_org[gr],sizeof(double)*2*576); ++ ++ ++ ++ ++ ++ for (ch = 0; ch < num_chan; ch++) { ++ /****************************************************************** ++ * find smallest number of bits for an allowable quantization ++ ******************************************************************/ ++ memcpy(xr_save,xr[gr][ch],sizeof(double)*576); ++ cod_info = &l3_side->gr[gr].ch[ch].tt; ++ min_bits = Max(125,min_mean_bits); ++ if (cod_info->block_type ==2) { ++ min_bits += Max(1100,pe[gr][ch]); ++ min_bits=Min(min_bits,1800); ++ } ++ ++ max_bits = 1200 + frameBits[VBR_max_bitrate]/(stereo*mode_gr); ++ max_bits=Min(max_bits,2500); ++ max_bits=Max(max_bits,min_bits); ++ ++ ++ /** in the case we will not find any better, we allocate max_bits ++ ****************************************************************/ ++ save_bits[gr][ch] = max_bits; ++ ++ dbits = (max_bits-min_bits)/4; ++ this_bits = (max_bits+min_bits)/2; ++ /* bin search to within +/- 10 bits of optimal */ ++ do { ++ int better; ++ double fac; ++ assert(this_bits>=min_bits); ++ assert(this_bits<=max_bits); ++ ++ /* quality setting */ ++ /* Adjust allowed masking based on quality setting */ ++ /* db_lower varies from -10 to +8 db */ ++ masking_lower_db = -10 + 2*VBR_q; ++ /* adjust by -6(min)..0(max) depending on bitrate */ ++ fac=(double)(this_bits-125)/(double)(2500-125); ++ fac = 6*(fac-1); ++ //fac = 3*(fac-1); ++ ++ masking_lower_db += fac; ++ masking_lower = pow(10.0,masking_lower_db/10); ++ ++ ++ ++ ++ /* VBR will look for a quantization which has better values ++ * then those specified below.*/ ++ targ_noise[0]=0; /* over */ ++ targ_noise[1]=0; /* max_noise */ ++ targ_noise[2]=0; /* over_noise */ ++ targ_noise[3]=-2; /* tot_noise */ ++ ++ targ_noise[0]=Max(0,targ_noise[0]); ++ targ_noise[2]=Max(0,targ_noise[2]); ++ ++ ++ /* restore xr */ ++ memcpy(xr[gr][ch],xr_save,sizeof(double)*576); ++ outer_loop( xr, this_bits, noise, targ_noise, 1,&l3_xmin,l3_enc, fr_ps, ++ scalefac,gr,stereo, l3_side, ratio, ms_ener_ratio[gr], ch); ++ ++ /* is quantization as good as we are looking for ? */ ++ better=VBR_compare((int)targ_noise[0],targ_noise[3],targ_noise[2], ++ targ_noise[1],(int)noise[0],noise[3],noise[2],noise[1]); ++ ++ if (better) { ++ save_bits[gr][ch] = this_bits; ++ this_bits -= dbits; ++ } else { ++ this_bits += dbits; ++ } ++ dbits /= 2; ++ } while (dbits>10) ; ++ used_bits += save_bits[gr][ch]; ++ ++ } /* for */ ++ } /* for */ ++ ++ ++ if (reduce_sidechannel) { ++ /* number of bits needed was found for MID channel above. Use formula ++ * (fixed bitrate code) to set the side channel bits */ ++ for (gr = 0; gr < mode_gr; gr++) { ++ double fac = .33*(.5-ms_ener_ratio[gr])/.5; ++ save_bits[gr][1]=((1-fac)/(1+fac))*save_bits[gr][0]; ++ used_bits += save_bits[gr][1]; ++ } ++ } ++ ++ ++ /****************************************************************** ++ * find lowest bitrate able to hold used bits ++ ******************************************************************/ ++ for( info->bitrate_index = VBR_min_bitrate; ++ info->bitrate_index < VBR_max_bitrate; ++ info->bitrate_index++ ) ++ if( used_bits <= frameBits[info->bitrate_index] ) break; ++ ++ ++ /******************************************************************* ++ * calculate quantization for this bitrate ++ *******************************************************************/ ++ getframebits (info, stereo, &bitsPerFrame, &mean_bits); ++ bits=ResvFrameBegin (fr_ps, l3_side, mean_bits, bitsPerFrame); ++ ++ /* repartion available bits in same proportion */ ++ if (used_bits > bits ) { ++ for( gr = 0; gr < mode_gr; gr++) { ++ for(ch = 0; ch < stereo; ch++) { ++ save_bits[gr][ch]=(save_bits[gr][ch]*frameBits[info->bitrate_index])/used_bits; ++ } ++ } ++ used_bits=0; ++ for( gr = 0; gr < mode_gr; gr++) { ++ for(ch = 0; ch < stereo; ch++) { ++ used_bits += save_bits[gr][ch]; ++ } ++ } ++ } ++ assert(used_bits <= bits); ++ ++ for(gr = 0; gr < mode_gr; gr++) { ++ /* copy data to be quantized into xr */ ++ if (convert_mdct) ms_convert(xr[gr],xr_org[gr]); ++ else memcpy(xr[gr],xr_org[gr],sizeof(double)*2*576); ++ for(ch = 0; ch < stereo; ch++) { ++ outer_loop( xr, save_bits[gr][ch], noise,targ_noise,0, ++ &l3_xmin,l3_enc, fr_ps, ++ scalefac,gr,stereo, l3_side, ratio, ms_ener_ratio[gr], ch); ++ } ++ } ++ ++ /******************************************************************* ++ * update reservoir status after FINAL quantization/bitrate ++ *******************************************************************/ ++ for (gr = 0; gr < mode_gr; gr++) ++ for (ch = 0; ch < stereo; ch++) { ++ cod_info = &l3_side->gr[gr].ch[ch].tt; ++ ResvAdjust (fr_ps, cod_info, l3_side, mean_bits); ++ } ++ ++ /******************************************************************* ++ * set the sign of l3_enc ++ *******************************************************************/ ++ for (gr = 0; gr < mode_gr; gr++) ++ for (ch = 0; ch < stereo; ch++) { ++ int *pi = &l3_enc[gr][ch][0]; ++ ++ for (i = 0; i < 576; i++) { ++ double pr = xr[gr][ch][i]; ++ ++ if ((pr < 0) && (pi[i] > 0)) ++ pi[i] *= -1; ++ } ++ } ++ ++ ResvFrameEnd (fr_ps, l3_side, mean_bits); ++ } ++ ++ ++ ++ ++ /************************************************************************/ ++ /* init_outer_loop mt 6/99 */ ++ /************************************************************************/ ++ void init_outer_loop( ++ double xr[2][2][576], /* could be L/R OR MID/SIDE */ ++ III_psy_xmin *l3_xmin, /* the allowed distortion of the scalefactor */ ++ III_scalefac_t *scalefac, /* scalefactors */ ++ int gr, int stereo, III_side_info_t *l3_side, ++ III_psy_ratio *ratio, int ch) ++ { ++ int sfb,i; ++ gr_info *cod_info; ++ cod_info = &l3_side->gr[gr].ch[ch].tt; ++ ++ /* compute max allowed distortion */ ++ calc_xmin( xr, ratio, cod_info, l3_xmin, gr, ch ); ++ ++ /* if ( info->version == 1 ) ++ calc_scfsi( xr[gr][ch], l3_side, &l3_xmin, ch, gr ); ++ */ ++ ++ ++ /* reset of iteration variables */ ++ ++ for ( sfb = 0; sfb < SFB_LMAX; sfb++ ) ++ scalefac->l[gr][ch][sfb] = 0; ++ for ( sfb = 0; sfb < SFB_SMAX; sfb++ ) ++ for ( i = 0; i < 3; i++ ) ++ scalefac->s[gr][ch][sfb][i] = 0; ++ ++ for ( i = 0; i < 4; i++ ) ++ cod_info->slen[i] = 0; ++ cod_info->sfb_partition_table = &nr_of_sfb_block[0][0][0]; ++ ++ cod_info->part2_3_length = 0; ++ cod_info->big_values = ((cod_info->block_type==2)?288:0); ++ cod_info->count1 = 0; ++ cod_info->scalefac_compress = 0; ++ cod_info->table_select[0] = 0; ++ cod_info->table_select[1] = 0; ++ cod_info->table_select[2] = 0; ++ cod_info->subblock_gain[0] = 0; ++ cod_info->subblock_gain[1] = 0; ++ cod_info->subblock_gain[2] = 0; ++ cod_info->region0_count = 0; ++ cod_info->region1_count = 0; ++ cod_info->part2_length = 0; ++ cod_info->preflag = 0; ++ cod_info->scalefac_scale = 0; ++ cod_info->quantizerStepSize = 0.0; ++ cod_info->count1table_select= 0; ++ cod_info->address1 = 0; ++ cod_info->address2 = 0; ++ cod_info->address3 = 0; ++ ++ ++ if (experimentalZ) { ++ /* compute subblock gains */ ++ int j,b; double en[3],mx; ++ if ((cod_info->block_type ==2) ) { ++ /* estimate energy within each subblock */ ++ for (b=0; b<3; b++) en[b]=0; ++ for ( i=0,j = 0; j < 192; j++ ) { ++ for (b=0; b<3; b++) { ++ en[b]+=xr[gr][ch][i]*xr[gr][ch][i]; ++ i++; ++ } ++ } ++ mx = 1e-12; ++ for (b=0; b<3; b++) mx=Max(mx,en[b]); ++ for (b=0; b<3; b++) en[b] = Max(en[b],1e-12)/mx; ++ // printf("ener = %4.2f %4.2f %4.2f \n",en[0],en[1],en[2]); ++ /* pick gain so that 2^(2gain)*en[0] = 1 */ ++ /* gain = .5* log( 1/en[0] )/log(2) = -.5*log(en[])/log(2) */ ++ for (b=0; b<3; b++) { ++ cod_info->subblock_gain[b]=nint2(-.5*log(en[b])/log(2.0)); ++ if (cod_info->subblock_gain[b] > 2) ++ cod_info->subblock_gain[b]=2; ++ if (cod_info->subblock_gain[b] < 0) ++ cod_info->subblock_gain[b]=0; ++ } ++ } ++ } ++ } ++ ++ ++ /* old outer_loop routines which will eventually be removed are in: */ ++ #include "loopold.c" ++ ++ ++ /************************************************************************/ ++ /* outer_loop */ ++ /************************************************************************/ ++ /* Function: The outer iteration loop controls the masking conditions */ ++ /* of all scalefactorbands. It computes the best scalefac and */ ++ /* global gain. This module calls the inner iteration loop ++ * ++ * mt 5/99 completely rewritten to allow for bit reservoir control, ++ * mid/side channels with L/R or mid/side masking thresholds, ++ * and chooses best quantization instead of last quantization when ++ * no distortion free quantization can be found. ++ * ++ * added VBR support mt 5/99 ++ ************************************************************************/ ++ void outer_loop( ++ double xr[2][2][576], /* could be L/R OR MID/SIDE */ ++ int targ_bits, ++ double best_noise[4], ++ double targ_noise[4], ++ int sloppy, ++ III_psy_xmin *l3_xmin, /* the allowed distortion of the scalefactor */ ++ int l3_enc[2][2][576], /* vector of quantized values ix(0..575) */ ++ frame_params *fr_ps, ++ III_scalefac_t *scalefac, /* scalefactors */ ++ int gr, int stereo, III_side_info_t *l3_side, ++ III_psy_ratio *ratio, double ms_ener_ratio, ++ int ch) ++ { ++ ++ int i, iteration; ++ int status,count=0,bits_found=0; ++ int real_bits=0; ++ int scalesave_l[CBLIMIT], scalesave_s[CBLIMIT][3]; ++ int sfb, huff_bits; ++ double xfsf[4][CBLIMIT]; ++ double xrpow[576],temp; ++ double distort[4][CBLIMIT]; ++ int save_l3_enc[576]; ++ int save_real_bits=0; ++ int save_preflag=0, save_compress=0; ++ int better; ++ int over=0; ++ double max_noise; ++ double over_noise; ++ double tot_noise; ++ int best_over=0; ++ double best_max_noise=0; ++ double best_over_noise=0; ++ double best_tot_noise=0; ++ gr_info save_cod_info; ++ gr_info *cod_info; ++ double xr_save[576]; ++ ++ int compute_stepsize=1; ++ int pre_just_turned_on; ++ int notdone=1; ++ ++ ++ ++ ++ if (experimentalY) memcpy(xr_save,xr[gr][ch],sizeof(double)*576); ++ cod_info = &l3_side->gr[gr].ch[ch].tt; ++ init_outer_loop(xr,l3_xmin,scalefac,gr,stereo,l3_side,ratio,ch); ++ best_over = 100; ++ count=0; ++ for (i=0; i<576; i++) { ++ if ( fabs(xr[gr][ch][i]) > 0 ) count++; ++ } ++ if (count==0) { ++ best_over=0; ++ notdone = 0; ++ } ++ ++ ++ ++ /* BEGIN MAIN LOOP */ ++ iteration = 0; ++ while ( notdone ) { ++ int try_scale=0; ++ iteration ++; ++ ++ if (compute_stepsize) { ++ /* init and compute initial quantization step */ ++ compute_stepsize=0; ++ for(i=0;i<576;i++) { ++ temp=fabs(xr[gr][ch][i]); ++ xrpow[i]=sqrt(sqrt(temp)*temp); ++ } ++ bits_found=bin_search_StepSize2(targ_bits,-211.0,46, ++ l3_enc[gr][ch],xr[gr][ch],xrpow,cod_info); ++ } ++ ++ ++ ++ /* inner_loop starts with the initial quantization step computed above ++ * and slowly increases until the bits < huff_bits. ++ * Thus is it important not to start with too large of an inital ++ * quantization step. Too small is ok, but inner_loop will take longer ++ */ ++ huff_bits = targ_bits - cod_info->part2_length; ++ if (huff_bits < 0) { ++ if (iteration==1) { ++ fprintf(stderr,"ERROR: outer_loop(): huff_bits < 0. \n"); ++ exit(-5); ++ }else{ ++ /* scale factors too large, not enough bits. use previous quantizaton */ ++ notdone=0; ++ } ++ }else{ ++ /* if this is the first iteration, see if we can reuse the quantization ++ * computed in bin_search_StepSize above */ ++ if (iteration==1) { ++ if(bits_found>huff_bits) { ++ cod_info->quantizerStepSize+=1.0; ++ real_bits = inner_loop( xr, xrpow, l3_enc, huff_bits, cod_info, gr, ch ); ++ } else real_bits=bits_found; ++ } ++ else ++ real_bits=inner_loop( xr, xrpow, l3_enc, huff_bits, cod_info, gr, ch ); ++ } ++ ++ ++ if (notdone) { ++ /* compute the distortion in this quantization */ ++ if (fast_mode) { ++ over=0; ++ }else{ ++ /* coefficients and thresholds both l/r (or both mid/side) */ ++ over=calc_noise1( xr[gr][ch], l3_enc[gr][ch], cod_info, ++ xfsf,distort, l3_xmin,gr,ch, &over_noise, ++ &tot_noise, &max_noise); ++ } ++ ++ /* check if this quantization is better the our saved quantization */ ++ if (iteration == 1) better=1; ++ else ++ better=quant_compare( ++ best_over,best_tot_noise,best_over_noise,best_max_noise, ++ over, tot_noise, over_noise, max_noise); ++ ++ ++ /* save data so we can restore this quantization later */ ++ if (better) { ++ best_over=over; ++ best_max_noise=max_noise; ++ best_over_noise=over_noise; ++ best_tot_noise=tot_noise; ++ ++ /* we need not save the quantization when in sloppy mode, so skip ++ */ ++ if (!sloppy) { ++ for ( sfb = 0; sfb < SFB_LMAX-1; sfb++ ) /* save scaling factors */ ++ scalesave_l[sfb] = scalefac->l[gr][ch][sfb]; ++ ++ for ( sfb = 0; sfb < SFB_SMAX-1; sfb++ ) ++ for ( i = 0; i < 3; i++ ) ++ scalesave_s[sfb][i] = scalefac->s[gr][ch][sfb][i]; ++ ++ save_preflag = cod_info->preflag; ++ save_compress = cod_info->scalefac_compress; ++ ++ memcpy(save_l3_enc,l3_enc[gr][ch],sizeof(l3_enc[gr][ch])); ++ memcpy(&save_cod_info,cod_info,sizeof(save_cod_info)); ++ save_real_bits=real_bits; ++ ++ #ifdef HAVEGTK ++ if (gtkflag) { ++ int l,start,end,bw; ++ double en0; ++ D192_3 *xr_s = (D192_3 *)xr[gr][ch]; ++ ++ for ( i = 0; i < 3; i++ ) { ++ for ( sfb = cod_info->sfb_smax; sfb < 12; sfb++ ) { ++ start = scalefac_band_short[ sfb ]; ++ end = scalefac_band_short[ sfb + 1 ]; ++ bw = end - start; ++ for ( en0 = 0.0, l = start; l < end; l++ ) ++ en0 += (*xr_s)[l][i] * (*xr_s)[l][i]; ++ en0=Max(en0/bw,1e-20); ++ /* ++ pinfo->xfsf_s[gr][ch][3*sfb+i] = ++ pinfo->thr_s[gr][ch][3*sfb+i]*xfsf[i+1][sfb]/ ++ l3_xmin->s[gr][ch][sfb][i]; ++ */ ++ pinfo->xfsf_s[gr][ch][3*sfb+i] = ++ pinfo->en_s[gr][ch][3*sfb+i]*xfsf[i+1][sfb]/en0; ++ pinfo->thr_s[gr][ch][3*sfb+i] = ++ pinfo->en_s[gr][ch][3*sfb+i]*l3_xmin->s[gr][ch][sfb][i]/en0; ++ ++ } ++ } ++ for ( sfb = 0; sfb < cod_info->sfb_lmax; sfb++ ) { ++ start = scalefac_band_long[ sfb ]; ++ end = scalefac_band_long[ sfb+1 ]; ++ bw = end - start; ++ for ( en0 = 0.0, l = start; l < end; l++ ) ++ en0 += xr[gr][ch][l] * xr[gr][ch][l]; ++ en0=Max(en0/bw,1e-20); ++ /* ++ pinfo->xfsf[gr][ch][sfb] = ++ pinfo->thr[gr][ch][sfb]*xfsf[0][sfb]/ ++ l3_xmin->l[gr][ch][sfb]; ++ */ ++ pinfo->xfsf[gr][ch][sfb] = ++ pinfo->en[gr][ch][sfb]*xfsf[0][sfb]/en0; ++ pinfo->thr[gr][ch][sfb] = ++ pinfo->en[gr][ch][sfb]*l3_xmin->l[gr][ch][sfb]/en0; ++ ++ ++ } ++ pinfo->over[gr][ch]=over; ++ pinfo->max_noise[gr][ch]=max_noise; ++ pinfo->tot_noise[gr][ch]=tot_noise; ++ pinfo->over_noise[gr][ch]=over_noise; ++ } ++ #endif ++ } /* end of sloppy skipping */ ++ } ++ } ++ ++ /* if no bands with distortion, we are done */ ++ if (experimentalX==0) ++ if (over==0) notdone=0; ++ ++ /* in sloppy mode, as soon as we know we can do better than targ_noise, ++ quit. This is used for the inital VBR bin search. Turn it off for ++ final (optimal) quantization */ ++ if (sloppy && notdone) notdone = ++ !VBR_compare((int)targ_noise[0],targ_noise[3],targ_noise[2], ++ targ_noise[1],over,tot_noise,over_noise,max_noise); ++ ++ ++ ++ ++ ++ if (notdone) { ++ /* see if we should apply preemphasis */ ++ pre_just_turned_on= ++ preemphasis(xr[gr][ch],xrpow,l3_xmin,gr,ch,l3_side,distort); ++ ++ /* if we didn't just apply pre-emph, let us see if we should ++ * amplify some scale factor bands */ ++ if (!pre_just_turned_on) { ++ notdone = amp_scalefac_bands( xr[gr][ch], xrpow, l3_xmin, ++ l3_side, scalefac, gr, ch, iteration,distort); ++ } ++ } ++ ++ ++ if (notdone) { ++ /* check to make sure we have not amplified too much */ ++ if ( (status = loop_break(scalefac, cod_info, gr, ch)) == 0 ) { ++ if ( fr_ps->header->version == 1 ) { ++ status = scale_bitcount( scalefac, cod_info, gr, ch ); ++ if (status && (cod_info->scalefac_scale==0)) try_scale=1; ++ }else{ ++ status = scale_bitcount_lsf( scalefac, cod_info, gr, ch ); ++ if (status && (cod_info->scalefac_scale==0)) try_scale=1; ++ } ++ } ++ notdone = !status; ++ } ++ ++ ++ ++ if (try_scale && experimentalY) { ++ memcpy(xr[gr][ch],xr_save,sizeof(double)*576); ++ init_outer_loop(xr,l3_xmin,scalefac,gr,stereo,l3_side,ratio,ch); ++ compute_stepsize=1; /* compute a new global gain */ ++ notdone=1; ++ cod_info->scalefac_scale=1; ++ } ++ } /* done with main iteration */ ++ ++ ++ /* in sloppy mode we don�t need to restore the quantization, ++ * cos we didn�t saved it and we don�t need it, as we only want ++ * to know if we can do better ++ */ ++ if (!sloppy) ++ /* restore some data */ ++ if (count ) { ++ cod_info->preflag = save_preflag; ++ cod_info->scalefac_compress = save_compress; ++ ++ for ( sfb = 0; sfb < SFB_LMAX-1; sfb++ ) { ++ scalefac->l[gr][ch][sfb] = scalesave_l[sfb]; ++ } ++ ++ for ( i = 0; i < 3; i++ ) ++ for ( sfb = 0; sfb < SFB_SMAX-1; sfb++ ) { ++ scalefac->s[gr][ch][sfb][i] = scalesave_s[sfb][i]; ++ } ++ ++ { ++ real_bits=save_real_bits; ++ memcpy(l3_enc[gr][ch],save_l3_enc,sizeof(l3_enc[gr][ch])); ++ memcpy(cod_info,&save_cod_info,sizeof(save_cod_info)); ++ ++ if ( fr_ps->header->version == 1 ) { ++ status = scale_bitcount( scalefac, cod_info, gr, ch ); ++ } else { ++ status = scale_bitcount_lsf( scalefac, cod_info, gr, ch ); ++ } ++ if (status) { ++ fprintf(stderr,"Error recomputing scalefac_compress...this should not happen"); ++ exit(-10); ++ } ++ } ++ cod_info->part2_3_length = cod_info->part2_length + real_bits; ++ ++ #ifdef HAVEGTK ++ if (gtkflag) ++ pinfo->LAMEmainbits[gr][ch]=cod_info->part2_3_length; ++ #endif ++ } ++ ++ ++ /* finish up */ ++ cod_info->global_gain = nint2( cod_info->quantizerStepSize + 210.0 ); ++ assert( cod_info->global_gain < 256 ); ++ ++ best_noise[0]=best_over; ++ best_noise[1]=best_max_noise; ++ best_noise[2]=best_over_noise; ++ best_noise[3]=best_tot_noise; ++ } ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ /*************************************************************************** ++ * inner_loop * ++ *************************************************************************** ++ * The code selects the best quantizerStepSize for a particular set ++ * of scalefacs */ ++ ++ int ++ inner_loop( double xr[2][2][576], double xrpow[576], ++ int l3_enc[2][2][576], int max_bits, ++ gr_info *cod_info, int gr, int ch ) ++ { ++ int bits; ++ assert( max_bits >= 0 ); ++ cod_info->quantizerStepSize -= 1.0;; ++ do ++ { ++ cod_info->quantizerStepSize += 1.0; ++ quantize_xrpow( xrpow, l3_enc[gr][ch], cod_info ); ++ bits = count_bits(l3_enc[gr][ch],cod_info); ++ } ++ while ( bits > max_bits ); ++ return bits; ++ } ++ ++ ++ ++ /*************************************************************************/ ++ /* scale_bitcount */ ++ /*************************************************************************/ ++ ++ /* Also calculates the number of bits necessary to code the scalefactors. */ ++ ++ int scale_bitcount( III_scalefac_t *scalefac, gr_info *cod_info, ++ int gr, int ch ) ++ { ++ int i, k, sfb, max_slen1 = 0, max_slen2 = 0, /*a, b, */ ep = 2; ++ ++ // static int slen1[16] = { 0, 0, 0, 0, 3, 1, 1, 1, 2, 2, 2, 3, 3, 3, 4, 4 }; ++ // static int slen2[16] = { 0, 1, 2, 3, 0, 1, 2, 3, 1, 2, 3, 1, 2, 3, 2, 3 }; ++ static int slen1[16] = { 1, 1, 1, 1, 8, 2, 2, 2, 4, 4, 4, 8, 8, 8,16,16 }; ++ static int slen2[16] = { 1, 2, 4, 8, 1, 2, 4, 8, 2, 4, 8, 2, 4, 8, 4, 8 }; ++ ++ static int slen1_tab[16] = {0, ++ 18, 36, 54, 54, 36, 54, 72, 54, 72, 90, 72, 90,108,108,126 ++ }; ++ static int slen2_tab[16] = {0, ++ 10, 20, 30, 33, 21, 31, 41, 32, 42, 52, 43, 53, 63, 64, 74 ++ }; ++ int *tab; ++ ++ ++ if ( cod_info->block_type == 2 ) ++ { ++ tab = slen1_tab; ++ /* a = 18; b = 18; */ ++ for ( i = 0; i < 3; i++ ) ++ { ++ for ( sfb = 0; sfb < 6; sfb++ ) ++ if ( scalefac->s[gr][ch][sfb][i] > max_slen1 ) ++ max_slen1 = scalefac->s[gr][ch][sfb][i]; ++ for (sfb = 6; sfb < 12; sfb++ ) ++ if ( scalefac->s[gr][ch][sfb][i] > max_slen2 ) ++ max_slen2 = scalefac->s[gr][ch][sfb][i]; ++ } ++ } ++ else ++ { /* block_type == 1,2,or 3 */ ++ tab = slen2_tab; ++ /* a = 11; b = 10; */ ++ for ( sfb = 0; sfb < 11; sfb++ ) ++ if ( scalefac->l[gr][ch][sfb] > max_slen1 ) ++ max_slen1 = scalefac->l[gr][ch][sfb]; ++ for ( sfb = 11; sfb < 21; sfb++ ) ++ if ( scalefac->l[gr][ch][sfb] > max_slen2 ) ++ max_slen2 = scalefac->l[gr][ch][sfb]; ++ } ++ ++ ++ ++ /* from Takehiro TOMINAGA <tominaga@isoternet.org> 10/99 ++ * loop over *all* posible values of scalefac_compress to find the ++ * one which uses the smallest number of bits. ISO would stop ++ * at first valid index */ ++ cod_info->part2_length=10000; ++ for ( k = 0; k < 16; k++ ) ++ { ++ if ( (max_slen1 < slen1[k]) && (max_slen2 < slen2[k]) && ++ (cod_info->part2_length > tab[k])) { ++ cod_info->part2_length=tab[k]; ++ cod_info->scalefac_compress=k; ++ ep=0; /* we found a suitable scalefac_compress */ ++ //break; ++ } ++ } ++ ++ // if ( ep == 0 ) cod_info->scalefac_compress = k; ++ return ep; ++ } ++ ++ ++ ++ ++ /*************************************************************************/ ++ /* scale_bitcount_lsf */ ++ /*************************************************************************/ ++ ++ /* Also counts the number of bits to encode the scalefacs but for MPEG 2 */ ++ /* Lower sampling frequencies (24, 22.05 and 16 kHz.) */ ++ ++ /* This is reverse-engineered from section 2.4.3.2 of the MPEG2 IS, */ ++ /* "Audio Decoding Layer III" */ ++ ++ int scale_bitcount_lsf( III_scalefac_t *scalefac, gr_info *cod_info, ++ int gr, int ch ) ++ { ++ int table_number, row_in_table, partition, nr_sfb, window, over; ++ int i, sfb, max_sfac[ 4 ]; ++ unsigned *partition_table; ++ ++ /* ++ Set partition table. Note that should try to use table one, ++ but do not yet... ++ */ ++ if ( cod_info->preflag ) ++ table_number = 2; ++ else ++ table_number = 0; ++ ++ for ( i = 0; i < 4; i++ ) ++ max_sfac[i] = 0; ++ ++ if ( cod_info->window_switching_flag != 0 && cod_info->block_type == 2 ) ++ { ++ row_in_table = 1; ++ partition_table = &nr_of_sfb_block[table_number][row_in_table][0]; ++ for ( sfb = 0, partition = 0; partition < 4; partition++ ) ++ { ++ nr_sfb = partition_table[ partition ] / 3; ++ for ( i = 0; i < nr_sfb; i++, sfb++ ) ++ for ( window = 0; window < 3; window++ ) ++ if ( scalefac->s[gr][ch][sfb][window] > max_sfac[partition] ) ++ max_sfac[partition] = scalefac->s[gr][ch][sfb][window]; ++ } ++ } ++ else ++ { ++ row_in_table = 0; ++ partition_table = &nr_of_sfb_block[table_number][row_in_table][0]; ++ for ( sfb = 0, partition = 0; partition < 4; partition++ ) ++ { ++ nr_sfb = partition_table[ partition ]; ++ for ( i = 0; i < nr_sfb; i++, sfb++ ) ++ if ( scalefac->l[gr][ch][sfb] > max_sfac[partition] ) ++ max_sfac[partition] = scalefac->l[gr][ch][sfb]; ++ } ++ } ++ ++ for ( over = 0, partition = 0; partition < 4; partition++ ) ++ { ++ if ( max_sfac[partition] > max_sfac_tab[table_number][partition] ) ++ over++; ++ } ++ if ( !over ) ++ { ++ /* ++ Since no bands have been over-amplified, we can set scalefac_compress ++ and slen[] for the formatter ++ */ ++ static int log2tab[] = { 0, 1, 2, 2, 3, 3, 3, 3, 4, 4, 4, 4, 4, 4, 4, 4 }; ++ ++ unsigned slen1, slen2, slen3, slen4; ++ ++ cod_info->sfb_partition_table = &nr_of_sfb_block[table_number][row_in_table][0]; ++ for ( partition = 0; partition < 4; partition++ ) ++ cod_info->slen[partition] = log2tab[max_sfac[partition]]; ++ ++ /* set scalefac_compress */ ++ slen1 = cod_info->slen[ 0 ]; ++ slen2 = cod_info->slen[ 1 ]; ++ slen3 = cod_info->slen[ 2 ]; ++ slen4 = cod_info->slen[ 3 ]; ++ ++ switch ( table_number ) ++ { ++ case 0: ++ cod_info->scalefac_compress = (((slen1 * 5) + slen2) << 4) ++ + (slen3 << 2) ++ + slen4; ++ break; ++ ++ case 1: ++ cod_info->scalefac_compress = 400 ++ + (((slen1 * 5) + slen2) << 2) ++ + slen3; ++ break; ++ ++ case 2: ++ cod_info->scalefac_compress = 500 + (slen1 * 3) + slen2; ++ break; ++ ++ default: ++ fprintf( stderr, "intensity stereo not implemented yet\n" ); ++ exit( EXIT_FAILURE ); ++ break; ++ } ++ } ++ #ifdef DEBUG ++ if ( over ) ++ printf( "---WARNING !! Amplification of some bands over limits\n" ); ++ #endif ++ if (!over) { ++ assert( cod_info->sfb_partition_table ); ++ cod_info->part2_length=0; ++ for ( partition = 0; partition < 4; partition++ ) ++ cod_info->part2_length += cod_info->slen[partition] * cod_info->sfb_partition_table[partition]; ++ } ++ return over; ++ } ++ ++ ++ ++ ++ ++ /*************************************************************************/ ++ /* calc_noise */ ++ /*************************************************************************/ ++ /* mt 5/99: Function: Improved calc_noise for a single channel */ ++ int calc_noise1( double xr[576], int ix[576], gr_info *cod_info, ++ double xfsf[4][CBLIMIT], double distort[4][CBLIMIT], ++ III_psy_xmin *l3_xmin,int gr, int ch, double *over_noise, ++ double *tot_noise, double *max_noise) ++ ++ { ++ int start, end, sfb, l, i, over=0; ++ double sum,step,bw; ++ ++ D192_3 *xr_s; ++ I192_3 *ix_s; ++ ++ static double pow43[PRECALC_SIZE]; ++ static int init=0; ++ int count=0; ++ double noise; ++ *over_noise=0; ++ *tot_noise=0; ++ *max_noise=-999; ++ ++ if (init==0) { ++ init++; ++ for(i=0;i<PRECALC_SIZE;i++) ++ pow43[i] = pow((double)i, 4.0/3.0); ++ } ++ ++ xr_s = (D192_3 *) xr; ++ ix_s = (I192_3 *) ix; ++ ++ step = pow( 2.0, (cod_info->quantizerStepSize) * 0.25 ); ++ for ( sfb = 0; sfb < cod_info->sfb_lmax; sfb++ ) ++ { ++ start = scalefac_band_long[ sfb ]; ++ end = scalefac_band_long[ sfb+1 ]; ++ bw = end - start; ++ ++ for ( sum = 0.0, l = start; l < end; l++ ) ++ { ++ double temp; ++ temp = fabs(xr[l]) - pow43[ix[l]] * step; ++ sum += temp * temp; ++ } ++ xfsf[0][sfb] = sum / bw; ++ // max -30db noise below threshold ++ noise = 10*log10(Max(.001,xfsf[0][sfb] / l3_xmin->l[gr][ch][sfb])); ++ distort[0][sfb] = noise; ++ if (noise>0) { ++ over++; ++ //*over_noise += xfsf[0][sfb] - l3_xmin->l[gr][ch][sfb]; ++ *over_noise += noise; ++ } ++ *tot_noise += noise; ++ *max_noise=Max(*max_noise,noise); ++ count++; ++ ++ } ++ ++ ++ for ( i = 0; i < 3; i++ ) ++ { ++ step = pow( 2.0, (cod_info->quantizerStepSize) * 0.25 ); /* ++ subblock_gain ? */ ++ if (cod_info->subblock_gain[i] ) ++ step *= pow(2.0,-2.0*cod_info->subblock_gain[i]); ++ ++ for ( sfb = cod_info->sfb_smax; sfb < 12; sfb++ ) ++ { ++ start = scalefac_band_short[ sfb ]; ++ end = scalefac_band_short[ sfb+1 ]; ++ bw = end - start; ++ ++ for ( sum = 0.0, l = start; l < end; l++ ) ++ { ++ double temp; ++ temp = fabs((*xr_s)[l][i]) - pow43[(*ix_s)[l][i]] * step; ++ sum += temp * temp; ++ } ++ xfsf[i+1][sfb] = sum / bw; ++ // max -30db noise below threshold ++ noise = 10*log10(Max(.001,xfsf[i+1][sfb] / l3_xmin->s[gr][ch][sfb][i] )); ++ distort[i+1][sfb] = noise; ++ if (noise > 0) { ++ over++; ++ //*over_noise +=xfsf[i+1][sfb]-l3_xmin->s[gr][ch][sfb][i]; ++ *over_noise += noise; ++ } ++ *tot_noise += noise; ++ *max_noise=Max(*max_noise,noise); ++ count++; ++ } ++ } ++ ++ if (count>1) *tot_noise /= count; ++ if (over>1) *over_noise /= over; ++ return over; ++ } ++ ++ ++ ++ ++ /*************************************************************************/ ++ /* calc_xmin */ ++ /*************************************************************************/ ++ ++ /* ++ Calculate the allowed distortion for each scalefactor band, ++ as determined by the psychoacoustic model. ++ xmin(sb) = ratio(sb) * en(sb) / bw(sb) ++ */ ++ void calc_xmin( double xr[2][2][576], III_psy_ratio *ratio, ++ gr_info *cod_info, III_psy_xmin *l3_xmin, ++ int gr, int ch ) ++ { ++ int start, end, sfb, l, b; ++ double en0, bw, ener; ++ ++ D192_3 *xr_s; ++ xr_s = (D192_3 *) xr[gr][ch] ; ++ ++ if (ATHonly) { ++ for ( sfb = cod_info->sfb_smax; sfb < SFB_SMAX - 1; sfb++ ) ++ for ( b = 0; b < 3; b++ ) ++ l3_xmin->s[gr][ch][sfb][b]=ATH_s[sfb]; ++ for ( sfb = 0; sfb < cod_info->sfb_lmax; sfb++ ) ++ l3_xmin->l[gr][ch][sfb]=ATH_l[sfb]; ++ ++ }else{ ++ ++ for ( sfb = cod_info->sfb_smax; sfb < SFB_SMAX - 1; sfb++ ){ ++ start = scalefac_band_short[ sfb ]; ++ end = scalefac_band_short[ sfb + 1 ]; ++ bw = end - start; ++ for ( b = 0; b < 3; b++ ) { ++ for ( en0 = 0.0, l = start; l < end; l++ ) { ++ ener = (*xr_s)[l][b] * (*xr_s)[l][b]; ++ en0 += ener; ++ } ++ l3_xmin->s[gr][ch][sfb][b] = masking_lower*ratio->s[gr][ch][sfb][b] * en0/bw; ++ l3_xmin->s[gr][ch][sfb][b]=Max(ATH_s[sfb],l3_xmin->s[gr][ch][sfb][b]); ++ } ++ } ++ ++ ++ for ( sfb = 0; sfb < cod_info->sfb_lmax; sfb++ ){ ++ start = scalefac_band_long[ sfb ]; ++ end = scalefac_band_long[ sfb+1 ]; ++ bw = end - start; ++ ++ for ( en0 = 0.0, l = start; l < end; l++ ) { ++ ener = xr[gr][ch][l] * xr[gr][ch][l]; ++ en0 += ener; ++ } ++ ++ l3_xmin->l[gr][ch][sfb] =masking_lower*ratio->l[gr][ch][sfb] * en0/bw; ++ l3_xmin->l[gr][ch][sfb]=Max(ATH_l[sfb],l3_xmin->l[gr][ch][sfb]); ++ } ++ } ++ } ++ ++ ++ ++ /*************************************************************************/ ++ /* loop_break */ ++ /*************************************************************************/ ++ ++ /* Function: Returns zero if there is a scalefac which has not been ++ amplified. Otherwise it returns one. ++ */ ++ ++ int loop_break( III_scalefac_t *scalefac, gr_info *cod_info, ++ int gr, int ch ) ++ { ++ int i, sfb; ++ ++ for ( sfb = 0; sfb < cod_info->sfb_lmax; sfb++ ) ++ if ( scalefac->l[gr][ch][sfb] == 0 ) ++ return 0; ++ ++ for ( sfb = cod_info->sfb_smax; sfb < 12; sfb++ ) ++ for ( i = 0; i < 3; i++ ) ++ if ( scalefac->s[gr][ch][sfb][i] == 0 ) ++ return 0; ++ ++ return 1; ++ } ++ ++ ++ ++ ++ /*************************************************************************/ ++ /* preemphasis */ ++ /*************************************************************************/ ++ ++ /* ++ See ISO 11172-3 section C.1.5.4.3.4 ++ */ ++ int preemphasis( double xr[576], double xrpow[576], ++ III_psy_xmin *l3_xmin, ++ int gr, int ch, III_side_info_t *l3_side, double distort[4][CBLIMIT] ) ++ { ++ int i, sfb, start, end, over; ++ double ifqstep; ++ gr_info *cod_info = &l3_side->gr[gr].ch[ch].tt; ++ ++ ++ /* ++ Preemphasis is switched on if in all the upper four scalefactor ++ bands the actual distortion exceeds the threshold after the ++ first call of the inner loop ++ */ ++ over = 0; ++ if ( cod_info->block_type != 2 && cod_info->preflag == 0 ) ++ { ++ for ( sfb = 17; sfb < 21; sfb++ ) ++ if ( distort[0][sfb]>0 ) over++; ++ ++ if (over == 4 ) ++ { ++ double t,t34; ++ cod_info->preflag = 1; ++ ifqstep = ( cod_info->scalefac_scale == 0 ) ? SQRT2 : 2.0; ++ for ( sfb = 11; sfb < cod_info->sfb_lmax; sfb++ ) ++ { ++ t=pow( ifqstep, (double) pretab[sfb] ); ++ t34=sqrt(sqrt(t)*t); ++ l3_xmin->l[gr][ch][sfb] *= t*t; ++ start = scalefac_band_long[ sfb ]; ++ end = scalefac_band_long[ sfb+1 ]; ++ for( i = start; i < end; i++ ) xr[i]*=t; ++ for( i = start; i < end; i++ ) xrpow[i]*=t34; ++ } ++ } ++ } ++ return (over == 4); ++ } ++ ++ ++ ++ ++ /*************************************************************************/ ++ /* amp_scalefac_bands */ ++ /*************************************************************************/ ++ ++ /* ++ Amplify the scalefactor bands that violate the masking threshold. ++ See ISO 11172-3 Section C.1.5.4.3.5 ++ */ ++ int amp_scalefac_bands( double xr[576], double xrpow[576], ++ III_psy_xmin *l3_xmin, III_side_info_t *l3_side, ++ III_scalefac_t *scalefac, ++ int gr, int ch, int iteration, double distort[4][CBLIMIT]) ++ { ++ int start, end, l, sfb, i, over = 0; ++ double ifqstep, ifqstep2, ifqstep34; ++ double distort_thresh; ++ D192_3 *xr_s; ++ D192_3 *xrpow_s; ++ gr_info *cod_info; ++ cod_info = &l3_side->gr[gr].ch[ch].tt; ++ ++ xr_s = (D192_3 *) xr; ++ xrpow_s = (D192_3 *) xrpow; ++ ++ ++ if ( cod_info->scalefac_scale == 0 ) ++ ifqstep = SQRT2; ++ else ++ ifqstep = 2.0; ++ ++ ++ ifqstep2 = ifqstep * ifqstep; ++ ifqstep34=sqrt(sqrt(ifqstep)*ifqstep); ++ ++ /* distort_thresh = 0, unless all bands have distortion ++ * less than masking. In that case, just amplify bands with distortion ++ * within 95% of largest distortion/masking ratio */ ++ distort_thresh = -900; ++ for ( sfb = 0; sfb < cod_info->sfb_lmax; sfb++ ) ++ distort_thresh = Max(1.05*distort[0][sfb],distort_thresh); ++ distort_thresh=Min(distort_thresh,0.0); ++ ++ ++ for ( sfb = 0; sfb < cod_info->sfb_lmax; sfb++ ) ++ { ++ if ( distort[0][sfb]>distort_thresh ) ++ { ++ over++; ++ l3_xmin->l[gr][ch][sfb] *= ifqstep2; ++ scalefac->l[gr][ch][sfb]++; ++ start = scalefac_band_long[sfb]; ++ end = scalefac_band_long[sfb+1]; ++ for ( l = start; l < end; l++ ) xr[l] *= ifqstep; ++ for ( l = start; l < end; l++ ) xrpow[l] *= ifqstep34; ++ } ++ } ++ ++ ++ ++ /* ++ Note that scfsi is not enabled for frames containing ++ short blocks ++ */ ++ ++ distort_thresh = -900; ++ for ( i = 0; i < 3; i++ ) ++ for ( sfb = cod_info->sfb_smax; sfb < 12; sfb++ ) { ++ distort_thresh = Max(1.05*distort[i+1][sfb],distort_thresh); ++ } ++ distort_thresh=Min(distort_thresh,0.0); ++ ++ ++ for ( i = 0; i < 3; i++ ) ++ for ( sfb = cod_info->sfb_smax; sfb < 12; sfb++ ) { ++ if ( distort[i+1][sfb]>distort_thresh) ++ { ++ over++; ++ l3_xmin->s[gr][ch][sfb][i] *= ifqstep2; ++ scalefac->s[gr][ch][sfb][i]++; ++ start = scalefac_band_short[sfb]; ++ end = scalefac_band_short[sfb+1]; ++ for ( l = start; l < end; l++ ) (*xr_s)[l][i] *= ifqstep; ++ for ( l = start; l < end; l++ ) (*xrpow_s)[l][i] *= ifqstep34; ++ } ++ } ++ return over; ++ } ++ ++ ++ ++ ++ #ifdef _MSC_VER ++ #define MSVC_XRPOW_ASM ++ #ifdef MSVC_XRPOW_ASM ++ # define MSVC_FTOL(src, dest) do { \ ++ double src_ = (src); \ ++ int dest_; \ ++ { \ ++ __asm fld src_ \ ++ __asm fistp dest_ \ ++ } \ ++ (dest) = dest_; \ ++ } while (0) ++ # endif ++ #endif ++ ++ void quantize_xrpow( double xr[576], int ix[576], gr_info *cod_info ) ++ { ++ /* quantize on xr^(3/4) instead of xr */ ++ register int j; ++ double quantizerStepSize; ++ double istep_l,istep0,istep1,istep2; ++ #if defined(__GNUC__) && defined(__i386__) ++ #elif defined(MSVC_XRPOW_ASM) ++ #else ++ double compareval0; ++ #endif ++ ++ quantizerStepSize = cod_info->quantizerStepSize; ++ ++ istep_l = pow ( 2.0, quantizerStepSize * -0.1875 ); ++ ++ if ((cod_info->block_type==2)) ++ { ++ istep0 = istep_l * pow(2.0,1.5* (double) cod_info->subblock_gain[0]); ++ istep1 = istep_l * pow(2.0,1.5* (double) cod_info->subblock_gain[1]); ++ istep2 = istep_l * pow(2.0,1.5* (double) cod_info->subblock_gain[2]); ++ for (j=192;j>0;j--) ++ { ++ #if defined(__GNUC__) && defined(__i386__) ++ asm ("fistpl %0 ": "=m"(*(ix++)): "t"(istep0*(*(xr++)) - 0.0946): "st"); ++ asm ("fistpl %0 ": "=m"(*(ix++)): "t"(istep1*(*(xr++)) - 0.0946): "st"); ++ asm ("fistpl %0 ": "=m"(*(ix++)): "t"(istep2*(*(xr++)) - 0.0946): "st"); ++ #elif defined(MSVC_XRPOW_ASM) ++ MSVC_FTOL((istep0*(*(xr++)) - 0.0946), *(ix++)); ++ MSVC_FTOL((istep1*(*(xr++)) - 0.0946), *(ix++)); ++ MSVC_FTOL((istep2*(*(xr++)) - 0.0946), *(ix++)); ++ #else ++ *(ix++) = (int)( istep0*(*(xr++)) + 0.4054); ++ *(ix++) = (int)( istep1*(*(xr++)) + 0.4054); ++ *(ix++) = (int)( istep2*(*(xr++)) + 0.4054); ++ #endif ++ } ++ } ++ else ++ { ++ #if defined(__GNUC__) && defined(__i386__) ++ for (j=576;j>0;j--) ++ asm ("fistpl %0 ": "=m"(*(ix++)): "t"(istep_l*(*(xr++)) - 0.0946): "st"); ++ #elif defined(MSVC_XRPOW_ASM) ++ for (j=576;j>0;j--) { ++ MSVC_FTOL((istep_l*(*(xr++)) - 0.0946), *(ix++)); ++ } ++ #else ++ compareval0 = (1.0 - 0.4054)/istep_l; ++ /* depending on architecture, it may be worth calculating a few more compareval's. ++ eg. compareval1 = (2.0 - 0.4054/istep_l); ++ .. and then after the first compare do this ... ++ if compareval1>*xr then ix = 1; ++ On a pentium166, it's only worth doing the one compare (as done here), as the second ++ compare becomes more expensive than just calculating the value. Architectures with ++ slow FP operations may want to add some more comparevals. try it and send your diffs ++ statistically speaking ++ 73% of all xr*istep_l values give ix=0 ++ 16% will give 1 ++ 4% will give 2 ++ */ ++ for (j=576;j>0;j--) ++ { ++ if (compareval0 > *xr) { ++ *(ix++) = 0; ++ xr++; ++ } else ++ *(ix++) = (int)( istep_l*(*(xr++)) + 0.4054); ++ } ++ #endif ++ } ++ } ++ ++ ++ ++ ++ ++ ++ /* ++ Seymour's comment: Jan 8 1995 ++ When mixed_block_flag is set, the low subbands 0-1 undergo the long ++ window transform and are each split into 18 frequency lines, while ++ the remaining 30 subbands undergo the short window transform and are ++ each split into 6 frequency lines. A problem now arises, as neither ++ the short or long scale factor bands apply to this mixed spectrum. ++ The standard resolves this situation by using the first 8 long scale ++ factor bands for the low spectrum and the short scale factor bands ++ in the range of 3 to 11 (inclusive) for the remaining frequency lines. ++ These scale factor bands do not match exactly to the 0-1 subbands ++ for all sampling frequencies (32,44.1 and 48 kHz); however they ++ were designed so that there would not be a frequency gap or overlap ++ at the switch over point. (Note multiply short frequency lines by 3 ++ to account for wider frequency line.) ++ */ ++ ++ /* mt 4/99: ISO code cannot produces mixed blocks, ++ * Fhg Code also never seems to use them, so no need to add them ++ * to this code ++ */ ++ /*************************************************************************/ ++ /* gr_deco */ ++ /*************************************************************************/ ++ ++ void gr_deco( gr_info *cod_info ) ++ { ++ if ( cod_info->window_switching_flag != 0 && cod_info->block_type == 2 ) ++ { ++ cod_info->sfb_lmax = 0; /* No sb*/ ++ cod_info->sfb_smax = 0; ++ } ++ else ++ { ++ /* MPEG 1 doesnt use last scalefactor band? */ ++ cod_info->sfb_lmax = SFB_LMAX - 1; ++ cod_info->sfb_smax = SFB_SMAX - 1; /* No sb */ ++ } ++ } ++ ++ ++ ++ ++ ++ ++ ++ static int OldValue = -30; /* guess it or so. */ ++ ++ typedef enum { ++ BINSEARCH_NONE, ++ BINSEARCH_UP, ++ BINSEARCH_DOWN ++ } binsearchDirection_t; ++ ++ /*-------------------------------------------------------------------------*/ ++ int ++ bin_search_StepSize2(int desired_rate, ++ double start, ++ int bot, ++ int *ix, ++ double xrs[576], ++ double xrspow[576], ++ gr_info *cod_info) ++ /*-------------------------------------------------------------------------*/ ++ { ++ int flag_GoneOver = 0; ++ int CurrentStep = 4; ++ int nBits; ++ int StepSize = OldValue; ++ binsearchDirection_t Direction = BINSEARCH_NONE; ++ do ++ { ++ cod_info->quantizerStepSize = StepSize; ++ quantize_xrpow(xrspow, ix, cod_info); ++ nBits = count_bits(ix,cod_info); ++ ++ if (CurrentStep == 1 ) ++ { ++ break; /* nothing to adjust anymore */ ++ } ++ if (flag_GoneOver) ++ { ++ CurrentStep /= 2; ++ } ++ if (nBits > desired_rate) /* increase Quantize_StepSize */ ++ { ++ if (Direction == BINSEARCH_DOWN && !flag_GoneOver) ++ { ++ flag_GoneOver = 1; ++ CurrentStep /= 2; /* late adjust */ ++ } ++ Direction = BINSEARCH_UP; ++ StepSize += CurrentStep; ++ } ++ else if (nBits < desired_rate) ++ { ++ if (Direction == BINSEARCH_UP && !flag_GoneOver) ++ { ++ flag_GoneOver = 1; ++ CurrentStep /= 2; /* late adjust */ ++ } ++ Direction = BINSEARCH_DOWN; ++ StepSize -= CurrentStep; ++ } ++ else break; /* nBits == desired_rate;; most unlikely to happen. ++ */ ++ } while (1); /* For-ever, break is adjusted. */ ++ OldValue = StepSize; ++ return nBits; ++ } ++ ++ ++ ++ ++ ++ ++ ++ int quant_compare( ++ int best_over,double best_tot_noise,double best_over_noise,double best_max_noise, ++ int over,double tot_noise, double over_noise, double max_noise) ++ { ++ /* ++ noise is given in decibals (db) relative to masking thesholds. ++ ++ over_noise: sum of quantization noise > masking ++ tot_noise: sum of all quantization noise ++ max_noise: max quantization noise ++ ++ */ ++ int better=0; ++ ++ if (experimentalX==0) { ++ better = ((over < best_over) || ++ ((over==best_over) && (over_noise<best_over_noise)) ) ; ++ } ++ ++ if (experimentalX==1) ++ better = max_noise < best_max_noise; ++ ++ if (experimentalX==2) { ++ better = tot_noise < best_tot_noise; ++ } ++ if (experimentalX==3) { ++ better = (tot_noise < best_tot_noise) && ++ (max_noise < best_max_noise + 2); ++ } ++ if (experimentalX==4) { ++ better = ( ( (0>=max_noise) && (best_max_noise>2)) || ++ ( (0>=max_noise) && (best_max_noise<0) && ((best_max_noise+2)>max_noise) && (tot_noise<best_tot_noise) ) || ++ ( (0>=max_noise) && (best_max_noise>0) && ((best_max_noise+2)>max_noise) && (tot_noise<(best_tot_noise+best_over_noise)) ) || ++ ( (0<max_noise) && (best_max_noise>-0.5) && ((best_max_noise+1)>max_noise) && ((tot_noise+over_noise)<(best_tot_noise+best_over_noise)) ) || ++ ( (0<max_noise) && (best_max_noise>-1) && ((best_max_noise+1.5)>max_noise) && ((tot_noise+over_noise+over_noise)<(best_tot_noise+best_over_noise+best_over_noise)) ) ); ++ } ++ if (experimentalX==5) { ++ better = (over_noise < best_over_noise) ++ || ((over_noise == best_over_noise)&&(tot_noise < best_tot_noise)); ++ } ++ ++ return better; ++ } ++ ++ ++ int VBR_compare( ++ int best_over,double best_tot_noise,double best_over_noise,double best_max_noise, ++ int over,double tot_noise, double over_noise, double max_noise) ++ { ++ /* ++ noise is given in decibals (db) relative to masking thesholds. ++ ++ over_noise: sum of quantization noise > masking ++ tot_noise: sum of all quantization noise ++ max_noise: max quantization noise ++ ++ */ ++ int better=0; ++ ++ better = ((over <= best_over) && ++ (over_noise<=best_over_noise) && ++ (tot_noise<=best_tot_noise) && ++ (max_noise<=best_max_noise)); ++ return better; ++ } ++ ++ ++ ++ ++ ++ ++ ++ +diff -r -c -N encoder/quantize.h lame3.50/quantize.h +*** encoder/quantize.h Wed Dec 31 17:00:00 1969 +--- lame3.50/quantize.h Tue Oct 12 14:55:53 1999 +*************** +*** 0 **** +--- 1,65 ---- ++ /********************************************************************** ++ * ISO MPEG Audio Subgroup Software Simulation Group (1996) ++ * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension ++ * ++ * $Id: loop.h,v 1.1 1996/02/14 04:04:23 rowlands Exp $ ++ * ++ * $Log: loop.h,v $ ++ * Revision 1.1 1996/02/14 04:04:23 rowlands ++ * Initial revision ++ * ++ * Received from Mike Coleman ++ **********************************************************************/ ++ ++ #ifndef LOOP_DOT_H ++ #define LOOP_DOT_H ++ #include "util.h" ++ #include "l3side.h" ++ ++ /********************************************************************** ++ * date programmers comment * ++ * 25. 6.92 Toshiyuki Ishino Ver 1.0 * ++ * 29.10.92 Masahiro Iwadare Ver 2.0 * ++ * 17. 4.93 Masahiro Iwadare Updated for IS Modification * ++ * * ++ *********************************************************************/ ++ ++ extern int cont_flag; ++ ++ #define CBLIMIT 21 ++ ++ #define SFB_LMAX 22 ++ #define SFB_SMAX 13 ++ ++ extern int pretab[]; ++ ++ struct scalefac_struct ++ { ++ int l[23]; ++ int s[14]; ++ }; ++ ++ extern struct scalefac_struct sfBandIndex[]; /* Table B.8 -- in loop.c */ ++ ++ ++ void iteration_loop( double pe[2][2], double ms_ratio[2], ++ double xr_org[2][2][576], III_psy_ratio *ratio, ++ III_side_info_t *l3_side, int l3_enc[2][2][576], ++ III_scalefac_t *scalefac, frame_params *fr_ps); ++ ++ void VBR_iteration_loop( double pe[2][2], double ms_ratio[2], ++ double xr_org[2][2][576], III_psy_ratio *ratio, ++ III_side_info_t *l3_side, int l3_enc[2][2][576], ++ III_scalefac_t *scalefac, frame_params *fr_ps); ++ ++ ++ ++ ++ #define maximum(A,B) ( (A) > (B) ? (A) : (B) ) ++ #define minimum(A,B) ( (A) < (B) ? (A) : (B) ) ++ #define signum( A ) ( (A) > 0 ? 1 : -1 ) ++ ++ ++ extern int bit_buffer[50000]; ++ ++ #endif +diff -r -c -N encoder/reservoir.c lame3.50/reservoir.c +*** encoder/reservoir.c Wed Jan 22 02:43:18 1997 +--- lame3.50/reservoir.c Mon Oct 25 12:12:22 1999 +*************** +*** 23,32 **** + #include <math.h> + #include <assert.h> + #include "l3side.h" +! #include "loop.h" + #include "huffman.h" + #include "l3bitstream.h" + #include "reservoir.h" + + /* + Layer3 bit reservoir: +--- 23,34 ---- + #include <math.h> + #include <assert.h> + #include "l3side.h" +! #include "quantize.h" + #include "huffman.h" + #include "l3bitstream.h" + #include "reservoir.h" ++ #include "gtkanal.h" ++ #include "globalflags.h" + + /* + Layer3 bit reservoir: +*************** +*** 36,48 **** + static int ResvSize = 0; /* in bits */ + static int ResvMax = 0; /* in bits */ + + /* + ResvFrameBegin: + Called at the beginning of a frame. Updates the maximum + size of the reservoir, and checks to make sure main_data_begin + was set properly by the formatter + */ +! void + ResvFrameBegin( frame_params *fr_ps, III_side_info_t *l3_side, int mean_bits, int frameLength ) + { + layer *info; +--- 38,56 ---- + static int ResvSize = 0; /* in bits */ + static int ResvMax = 0; /* in bits */ + ++ void InitReservoir(void) ++ { ++ ResvSize = 0; ++ ResvMax = 0; ++ } ++ + /* + ResvFrameBegin: + Called at the beginning of a frame. Updates the maximum + size of the reservoir, and checks to make sure main_data_begin + was set properly by the formatter + */ +! int + ResvFrameBegin( frame_params *fr_ps, III_side_info_t *l3_side, int mean_bits, int frameLength ) + { + layer *info; +*************** +*** 71,78 **** + fprintf( stderr, ">>> ResvSize = %d\n", ResvSize ); + #endif + assert( expectedResvSize == ResvSize ); +! +! fullFrameBits = mean_bits * mode_gr; + + /* + determine maximum size of reservoir: +--- 79,85 ---- + fprintf( stderr, ">>> ResvSize = %d\n", ResvSize ); + #endif + assert( expectedResvSize == ResvSize ); +! fullFrameBits = mean_bits * mode_gr + ResvSize; + + /* + determine maximum size of reservoir: +*************** +*** 90,136 **** + */ + if ( ResvMax > resvLimit ) + ResvMax = resvLimit; + } + + /* +! ResvMaxBits: +! Called at the beginning of each granule to get the max bit +! allowance for the current granule based on reservoir size +! and perceptual entropy. + */ +! int +! ResvMaxBits( frame_params *fr_ps, III_side_info_t *l3_side, double *pe, int mean_bits ) + { +! int more_bits, max_bits, add_bits, over_bits; +! +! mean_bits /= fr_ps->stereo; +! max_bits = mean_bits; + +- if ( max_bits > 4095 ) +- max_bits = 4095; + +! if ( ResvMax == 0 ) +! return max_bits; + +! more_bits = *pe * 3.1 - mean_bits; +! add_bits = 0; +! if ( more_bits > 100 ) +! { +! int frac = (ResvSize * 6) / 10; +! +! if ( frac < more_bits ) +! add_bits = frac; +! else +! add_bits = more_bits; +! } +! over_bits = ResvSize - ((ResvMax * 8) / 10) - add_bits; +! if ( over_bits > 0 ) +! add_bits += over_bits; +! +! max_bits += add_bits; +! if ( max_bits > 4095 ) +! max_bits = 4095; +! return max_bits; + } + + /* +--- 97,146 ---- + */ + if ( ResvMax > resvLimit ) + ResvMax = resvLimit; ++ ++ #ifdef HAVEGTK ++ if (gtkflag){ ++ pinfo->mean_bits=mean_bits/2; /* expected bits per channel per granule */ ++ pinfo->resvsize=ResvSize; ++ } ++ #endif ++ ++ return fullFrameBits; + } + ++ + /* +! ResvMaxBits2: +! As above, but now it *really* is bits per granule (both channels). +! Mark Taylor 4/99 + */ +! void ResvMaxBits2(int mean_bits, int *targ_bits, int *extra_bits, int gr) + { +! int add_bits; + + +! *targ_bits = mean_bits ; +! /* extra bits if the reservoir is almost full */ +! if (ResvSize > ((ResvMax * 9) / 10)) { +! add_bits= ResvSize-((ResvMax * 9) / 10); +! *targ_bits += add_bits; +! }else { +! add_bits =0 ; +! /* build up reservoir. this builds the reservoir a little slower +! * than FhG. It could simple be mean_bits/15, but this was rigged +! * to always produce 100 (the old value) at 128kbs */ +! *targ_bits -= (int) (mean_bits/15.2); +! } +! +! +! /* amount from the reservoir we are allowed to use. ISO says 6/10 */ +! *extra_bits = +! (ResvSize < (ResvMax*6)/10 ? ResvSize : (ResvMax*6)/10); +! *extra_bits -= add_bits; +! +! if (*extra_bits < 0) *extra_bits=0; + +! + } + + /* +*************** +*** 144,149 **** +--- 154,160 ---- + ResvSize += (mean_bits / fr_ps->stereo) - gi->part2_3_length; + } + ++ + /* + ResvFrameEnd: + Called after all granules in a frame have been allocated. Makes sure +*************** +*** 155,168 **** + void + ResvFrameEnd( frame_params *fr_ps, III_side_info_t *l3_side, int mean_bits ) + { +! layer *info; +! gr_info *gi; +! int mode_gr, gr, ch, stereo, ancillary_pad, stuffingBits; + int over_bits; + +- info = fr_ps->header; + stereo = fr_ps->stereo; +- mode_gr = (info->version == 1) ? 2 : 1; + ancillary_pad = 0; + + #if 1 +--- 166,175 ---- + void + ResvFrameEnd( frame_params *fr_ps, III_side_info_t *l3_side, int mean_bits ) + { +! int stereo, ancillary_pad, stuffingBits; + int over_bits; + + stereo = fr_ps->stereo; + ancillary_pad = 0; + + #if 1 +*************** +*** 185,228 **** + ResvSize -= over_bits; + } + +! if ( stuffingBits ) +! { +! /* +! plan a: put all into the first granule +! This was preferred by someone designing a +! real-time decoder... +! */ +! gi = (gr_info *) &(l3_side->gr[0].ch[0]); +! +! if ( gi->part2_3_length + stuffingBits < 4095 ) +! gi->part2_3_length += stuffingBits; +! else +! { +! /* plan b: distribute throughout the granules */ +! for (gr = 0; gr < mode_gr; gr++ ) +! for (ch = 0; ch < stereo; ch++ ) +! { +! int extraBits, bitsThisGr; +! gr_info *gi = (gr_info *) &(l3_side->gr[gr].ch[ch]); +! if ( stuffingBits == 0 ) +! break; +! extraBits = 4095 - gi->part2_3_length; +! bitsThisGr = extraBits < stuffingBits ? extraBits : stuffingBits; +! gi->part2_3_length += bitsThisGr; +! stuffingBits -= bitsThisGr; +! } +! /* +! If any stuffing bits remain, we elect to spill them +! into ancillary data. The bitstream formatter will do this if +! l3side->resvDrain is set +! */ +! #ifdef DEBUG +! if ( stuffingBits ) +! fprintf( stderr, "spilling %d stuffing bits into ancillary data\n", stuffingBits ); +! #endif +! l3_side->resvDrain = stuffingBits; +! } +! } + } + + +--- 192,201 ---- + ResvSize -= over_bits; + } + +! +! l3_side->resvDrain = stuffingBits; +! return; +! + } + + +diff -r -c -N encoder/reservoir.h lame3.50/reservoir.h +*** encoder/reservoir.h Wed Jan 22 02:43:18 1997 +--- lame3.50/reservoir.h Tue Oct 5 09:36:52 1999 +*************** +*** 22,30 **** + #ifndef RESERVOIR_H + #define RESERVOIR_H + +! void ResvFrameBegin( frame_params *fr_ps, III_side_info_t *l3_side, int mean_bits, int frameLength ); +! int ResvMaxBits( frame_params *fr_ps, III_side_info_t *l3_side, double *pe, int mean_bits ); + void ResvAdjust( frame_params *fr_ps, gr_info *gi, III_side_info_t *l3_side, int mean_bits ); + void ResvFrameEnd( frame_params *fr_ps, III_side_info_t *l3_side, int mean_bits ); + + #endif +--- 22,31 ---- + #ifndef RESERVOIR_H + #define RESERVOIR_H + +! int ResvFrameBegin( frame_params *fr_ps, III_side_info_t *l3_side, int mean_bits, int frameLength ); +! void ResvMaxBits2( int mean_bits, int *targ_bits, int *max_bits, int gr); + void ResvAdjust( frame_params *fr_ps, gr_info *gi, III_side_info_t *l3_side, int mean_bits ); + void ResvFrameEnd( frame_params *fr_ps, III_side_info_t *l3_side, int mean_bits ); ++ void InitReservoir(void); + + #endif +diff -r -c -N encoder/subs.c lame3.50/subs.c +*** encoder/subs.c Wed Jan 22 02:43:18 1997 +--- lame3.50/subs.c Wed Dec 31 17:00:00 1969 +*************** +*** 1,163 **** +- /********************************************************************** +- * ISO MPEG Audio Subgroup Software Simulation Group (1996) +- * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- * +- * $Id: subs.c,v 1.1 1996/02/14 04:04:23 rowlands Exp $ +- * +- * $Log: subs.c,v $ +- * Revision 1.1 1996/02/14 04:04:23 rowlands +- * Initial revision +- * +- * Received from Mike Coleman +- **********************************************************************/ +- /********************************************************************** +- * date programmers comment * +- * 2/25/91 Davis Pan start of version 1.0 records * +- * 5/10/91 W. Joseph Carter Ported to Macintosh and Unix. * +- * 7/10/91 Earle Jennings Ported to MsDos from Macintosh * +- * Replacement of one float with FLOAT * +- * 2/11/92 W. Joseph Carter Added type casting to memset() args. * +- * 4/27/92 Masahiro Iwadare Added 256 point version for Layer III * +- **********************************************************************/ +- +- #include "common.h" +- #include "encoder.h" +- +- /***************************************************************************** +- ************************** Start of Subroutines ***************************** +- *****************************************************************************/ +- +- /***************************************************************************** +- * FFT computes fast fourier transform of BLKSIZE samples of data * +- * uses decimation-in-frequency algorithm described in "Digital * +- * Signal Processing" by Oppenheim and Schafer, refer to pages 304 * +- * (flow graph) and 330-332 (Fortran program in problem 5) * +- * to get the inverse fft, change line 20 from * +- * w_imag[L] = -sin(PI/le1); * +- * to * +- * w_imag[L] = sin(PI/le1); * +- * * +- * required constants: * +- * #define PI 3.14159265358979 * +- * #define BLKSIZE 1024 * +- * #define LOGBLKSIZE 10 * +- * #define BLKSIZE_S 256 * +- * #define LOGBLKSIZE_S 8 * +- * * +- *****************************************************************************/ +- #define BLKSIZE_S 256 +- #define LOGBLKSIZE_S 8 +- +- void fft(x_real,x_imag, energy, phi, N) +- FLOAT x_real[BLKSIZE], x_imag[BLKSIZE], energy[BLKSIZE], phi[BLKSIZE]; +- int N; +- { +- int M,MM1; +- static int init=0; +- int NV2, NM1, MP; +- static double w_real[2][LOGBLKSIZE], w_imag[2][LOGBLKSIZE]; +- int i,j,k,L; +- int ip, le,le1; +- double t_real, t_imag, u_real, u_imag; +- +- if(init==0) { +- memset((char *) w_real, 0, sizeof(w_real)); /* preset statics to 0 */ +- memset((char *) w_imag, 0, sizeof(w_imag)); /* preset statics to 0 */ +- M = LOGBLKSIZE; +- for(L=0; L<M; L++){ +- le = 1 << (M-L); +- le1 = le >> 1; +- w_real[0][L] = cos(PI/le1); +- w_imag[0][L] = -sin(PI/le1); +- } +- M = LOGBLKSIZE_S; +- for(L=0; L<M; L++){ +- le = 1 << (M-L); +- le1 = le >> 1; +- w_real[1][L] = cos(PI/le1); +- w_imag[1][L] = -sin(PI/le1); +- } +- init++; +- } +- switch(N) { +- case BLKSIZE: +- M = LOGBLKSIZE; +- MP = 0; +- break; +- case BLKSIZE_S: +- M = LOGBLKSIZE_S; +- MP = 1; +- break; +- default: printf("Error: Bad FFT Size in subs.c\n"); +- exit(-1); +- } +- MM1 = M-1; +- NV2 = N >> 1; +- NM1 = N - 1; +- for(L=0; L<MM1; L++){ +- le = 1 << (M-L); +- le1 = le >> 1; +- u_real = 1; +- u_imag = 0; +- for(j=0; j<le1; j++){ +- for(i=j; i<N; i+=le){ +- ip = i + le1; +- t_real = x_real[i] + x_real[ip]; +- t_imag = x_imag[i] + x_imag[ip]; +- x_real[ip] = x_real[i] - x_real[ip]; +- x_imag[ip] = x_imag[i] - x_imag[ip]; +- x_real[i] = t_real; +- x_imag[i] = t_imag; +- t_real = x_real[ip]; +- x_real[ip] = x_real[ip]*u_real - x_imag[ip]*u_imag; +- x_imag[ip] = x_imag[ip]*u_real + t_real*u_imag; +- } +- t_real = u_real; +- u_real = u_real*w_real[MP][L] - u_imag*w_imag[MP][L]; +- u_imag = u_imag*w_real[MP][L] + t_real*w_imag[MP][L]; +- } +- } +- /* special case: L = M-1; all Wn = 1 */ +- for(i=0; i<N; i+=2){ +- ip = i + 1; +- t_real = x_real[i] + x_real[ip]; +- t_imag = x_imag[i] + x_imag[ip]; +- x_real[ip] = x_real[i] - x_real[ip]; +- x_imag[ip] = x_imag[i] - x_imag[ip]; +- x_real[i] = t_real; +- x_imag[i] = t_imag; +- energy[i] = x_real[i]*x_real[i] + x_imag[i]*x_imag[i]; +- if(energy[i] <= 0.0005){phi[i] = 0;energy[i] = 0.0005;} +- else phi[i] = atan2((double) x_imag[i],(double) x_real[i]); +- energy[ip] = x_real[ip]*x_real[ip] + x_imag[ip]*x_imag[ip]; +- if(energy[ip] == 0)phi[ip] = 0; +- else phi[ip] = atan2((double) x_imag[ip],(double) x_real[ip]); +- } +- /* this section reorders the data to the correct ordering */ +- j = 0; +- for(i=0; i<NM1; i++){ +- if(i<j){ +- /* use this section only if you need the FFT in complex number form * +- * (and in the correct ordering) */ +- t_real = x_real[j]; +- t_imag = x_imag[j]; +- x_real[j] = x_real[i]; +- x_imag[j] = x_imag[i]; +- x_real[i] = t_real; +- x_imag[i] = t_imag; +- /* reorder the energy and phase, phi */ +- t_real = energy[j]; +- energy[j] = energy[i]; +- energy[i] = t_real; +- t_real = phi[j]; +- phi[j] = phi[i]; +- phi[i] = t_real; +- } +- k=NV2; +- while(k<=j){ +- j = j-k; +- k = k >> 1; +- } +- j = j+k; +- } +- } +--- 0 ---- +diff -r -c -N encoder/tables.c lame3.50/tables.c +*** encoder/tables.c Wed Dec 31 17:00:00 1969 +--- lame3.50/tables.c Tue Oct 12 14:53:14 1999 +*************** +*** 0 **** +--- 1,1091 ---- ++ #include "util.h" ++ #include "huffman.h" ++ ++ ++ ++ unsigned int hs = sizeof(HUFFBITS)*8; ++ ++ static HUFFBITS t1HB[] = {1, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 1, 0}; ++ static HUFFBITS t2HB[] = {1, 2, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 3, 1, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 3, 2, 0}; ++ static HUFFBITS t3HB[] = {3, 2, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 1, 1, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 3, 2, 0}; ++ static HUFFBITS t5HB[] = {1, 2, 6, 5, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 3, 1, 4, 4, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 7, 5, 7, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 6, 1, 1, 0}; ++ static HUFFBITS t6HB[] = {7, 3, 5, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 6, 2, 3, 2, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 5, 4, 4, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 3, 3, 2, 0}; ++ static HUFFBITS t7HB[] = {1, 2, 10, 19, 16, 10, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 3, 3, 7, 10, 5, 3, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 11, 4, 13, 17, 8, 4, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 12, 11, 18, 15, 11, 2, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 7, 6, 9, 14, 3, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 6, 4, 5, 3, 2, 0}; ++ static HUFFBITS t8HB[] = {3, 4, 6, 18, 12, 5, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 5, 1, 2, 16, 9, 3, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 7, 3, 5, 14, 7, 3, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 19, 17, 15, 13, 10, 4, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 13, 5, 8, 11, 5, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 12, 4, 4, 1, 1, 0}; ++ static HUFFBITS t9HB[] = {7, 5, 9, 14, 15, 7, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 6, 4, 5, 5, 6, 7, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 7, 6, 8, 8, 8, 5, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 15, 6, 9, 10, 5, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 11, 7, 9, 6, 4, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 14, 4, 6, 2, 6, 0}; ++ static HUFFBITS t10HB[] = {1, 2, 10, 23, 35, 30, 12, 17, 0, 0, 0, 0, 0, 0, 0, 0, 3, 3, 8, 12, 18, 21, 12, 7, 0, 0, 0, 0, 0, 0, 0, 0, 11, 9, 15, 21, 32, 40, 19, 6, 0, 0, 0, 0, 0, 0, 0, 0, 14, 13, 22, 34, 46, 23, 18, 7, 0, 0, 0, 0, 0, 0, 0, 0, 20, 19, 33, 47, 27, 22, 9, 3, 0, 0, 0, 0, 0, 0, 0, 0, 31, 22, 41, 26, 21, 20, 5, 3, 0, 0, 0, 0, 0, 0, 0, 0, 14, 13, 10, 11, 16, 6, 5, 1, 0, 0, 0, 0, 0, 0, 0, 0, 9, 8, 7, 8, 4 , 4, 2, 0}; ++ static HUFFBITS t11HB[] = {3, 4, 10, 24, 34, 33, 21, 15, 0, 0, 0, 0, 0, 0, 0, 0, 5, 3, 4, 10, 32, 17, 11, 10, 0, 0, 0, 0, 0, 0, 0, 0, 11, 7, 13, 18, 30, 31, 20, 5, 0, 0, 0, 0, 0, 0, 0, 0, 25, 11, 19, 59, 27, 18, 12, 5, 0, 0, 0, 0, 0, 0, 0, 0, 35, 33, 31, 58, 30, 16, 7, 5, 0, 0, 0, 0, 0, 0, 0, 0, 28, 26, 32, 19, 17, 15, 8, 14, 0, 0, 0, 0, 0, 0, 0, 0, 14, 12, 9, 13, 14, 9, 4, 1, 0, 0, 0, 0, 0, 0, 0, 0, 11, 4, 6, 6, 6, 3, 2, 0}; ++ static HUFFBITS t12HB[] = {9, 6, 16, 33, 41, 39, 38, 26, 0, 0, 0, 0, 0, 0, 0, 0, 7, 5, 6, 9, 23, 16, 26, 11, 0, 0, 0, 0, 0, 0, 0, 0, 17, 7, 11, 14, 21, 30, 10, 7, 0, 0, 0, 0, 0, 0, 0, 0, 17, 10, 15, 12, 18, 28, 14, 5, 0, 0, 0, 0, 0, 0, 0, 0, 32, 13, 22, 19, 18, 16, 9, 5, 0, 0, 0, 0, 0, 0, 0, 0, 40, 17, 31, 29, 17, 13, 4, 2, 0, 0, 0, 0, 0, 0, 0, 0, 27, 12, 11, 15, 10, 7, 4, 1, 0, 0, 0, 0, 0, 0, 0, 0, 27, 12, 8, 12 , 6, 3, 1, 0}; ++ static HUFFBITS t13HB[] = {1, 5, 14, 21, 34, 51, 46, 71, 42, 52, 68, 52, 67, 44, 43, 19, 3, 4, 12, 19, 31, 26, 44, 33, 31, 24, 32, 24, 31, 35, 22, 14, 15, 13, 23, 36, 59, 49, 77, 65, 29, 40, 30, 40, 27, 33, 42, 16, 22, ++ 20, 37, 61, 56, 79, 73, 64, 43, 76, 56, 37, 26, 31, 25, 14, 35, 16, 60, 57, 97, 75, 114, 91, 54, 73, 55, 41, 48, 53, 23, 24, 58, 27, 50, 96, 76, 70, 93, 84, 77, 58, 79, 29, 74, 49, 41, 17, 47, ++ 45, 78, 74, 115, 94, 90, 79, 69, 83, 71, 50, 59, 38, 36, 15, 72, 34, 56, 95, 92, 85, 91, 90, 86, 73, 77, 65, 51, 44, 43, 42, 43, 20, 30, 44, 55, 78, 72, 87, 78, 61, 46, 54, 37, 30, 20, 16, 53, ++ 25, 41, 37, 44, 59, 54, 81, 66, 76, 57, 54, 37, 18, 39, 11, 35, 33, 31, 57, 42, 82, 72, 80, 47, 58, 55, 21, 22, 26, 38, 22, 53, 25, 23, 38, 70, 60, 51, 36, 55, 26, 34, 23, 27, 14, 9, 7, 34, 32, ++ 28, 39, 49, 75, 30, 52, 48, 40, 52, 28, 18, 17, 9, 5, 45, 21, 34, 64, 56, 50, 49, 45, 31, 19, 12, 15, 10, 7, 6, 3, 48, 23, 20, 39, 36, 35, 53, 21, 16, 23, 13, 10, 6, 1, 4, 2, 16, 15, 17, 27, 25, ++ 20, 29, 11, 17, 12, 16, 8, 1, 1, 0, 1}; ++ static HUFFBITS t15HB[] = {7, 12, 18, 53, 47, 76, 124, 108, 89, 123, 108, 119, 107, 81, 122, 63, 13, 5, 16, 27, 46, 36, 61, 51, 42, 70, 52, 83, 65, 41, 59, 36, 19, 17, 15, 24, 41, 34, 59, 48, 40, 64, 50, 78, 62, 80, 56, ++ 33, 29, 28, 25, 43, 39, 63, 55, 93, 76, 59, 93, 72, 54, 75, 50, 29, 52, 22, 42, 40, 67, 57, 95, 79, 72, 57, 89, 69, 49, 66, 46, 27, 77, 37, 35, 66, 58, 52, 91, 74, 62, 48, 79, 63, 90, 62, 40, 38, ++ 125, 32, 60, 56, 50, 92, 78, 65, 55, 87, 71, 51, 73, 51, 70, 30, 109, 53, 49, 94, 88, 75, 66, 122, 91, 73, 56, 42, 64, 44, 21, 25, 90, 43, 41, 77, 73, 63, 56, 92, 77, 66, 47, 67, 48, 53, 36, 20, ++ 71, 34, 67, 60, 58, 49, 88, 76, 67, 106, 71, 54, 38, 39, 23, 15, 109, 53, 51, 47, 90, 82, 58, 57, 48, 72, 57, 41, 23, 27, 62, 9, 86, 42, 40, 37, 70, 64, 52, 43, 70, 55, 42, 25, 29, 18, 11, 11, ++ 118, 68, 30, 55, 50, 46, 74, 65, 49, 39, 24, 16, 22, 13, 14, 7, 91, 44, 39, 38, 34, 63, 52, 45, 31, 52, 28, 19, 14, 8, 9, 3, 123, 60, 58, 53, 47, 43, 32, 22, 37, 24, 17, 12, 15, 10, 2, 1, 71, ++ 37, 34, 30, 28, 20, 17, 26, 21, 16, 10, 6, 8, 6, 2, 0}; ++ static HUFFBITS t16HB[] = {1, 5, 14, 44, 74, 63, 110, 93, 172, 149, 138, 242, 225, 195, 376, 17, 3, 4, 12, 20, 35, 62, 53, 47, 83, 75, 68, 119, 201, 107, 207, 9, 15, 13, 23, 38, 67, 58, 103, 90, 161, 72, 127, 117, ++ 110, 209, 206, 16, 45, 21, 39, 69, 64, 114, 99, 87, 158, 140, 252, 212, 199, 387, 365, 26, 75, 36, 68, 65, 115, 101, 179, 164, 155, 264, 246, 226, 395, 382, 362, 9, 66, 30, 59, 56, 102, ++ 185, 173, 265, 142, 253, 232, 400, 388, 378, 445, 16, 111, 54, 52, 100, 184, 178, 160, 133, 257, 244, 228, 217, 385, 366, 715, 10, 98, 48, 91, 88, 165, 157, 148, 261, 248, 407, 397, 372, ++ 380, 889, 884, 8, 85, 84, 81, 159, 156, 143, 260, 249, 427, 401, 392, 383, 727, 713, 708, 7, 154, 76, 73, 141, 131, 256, 245, 426, 406, 394, 384, 735, 359, 710, 352, 11, 139, 129, 67, 125, ++ 247, 233, 229, 219, 393, 743, 737, 720, 885, 882, 439, 4, 243, 120, 118, 115, 227, 223, 396, 746, 742, 736, 721, 712, 706, 223, 436, 6, 202, 224, 222, 218, 216, 389, 386, 381, 364, 888, ++ 443, 707, 440, 437, 1728, 4, 747, 211, 210, 208, 370, 379, 734, 723, 714, 1735, 883, 877, 876, 3459, 865, 2, 377, 369, 102, 187, 726, 722, 358, 711, 709, 866, 1734, 871, 3458, 870, 434, ++ 0, 12, 10, 7, 11, 10, 17, 11, 9, 13, 12, 10, 7, 5, 3, 1, 3}; ++ static HUFFBITS t24HB[] = {15, 13, 46, 80, 146, 262, 248, 434, 426, 669, 653, 649, 621, 517, 1032, 88, 14, 12, 21, 38, 71, 130, 122, 216, 209, 198, 327, 345, 319, 297, 279, 42, 47, 22, 41, 74, 68, 128, 120, 221, ++ 207, 194, 182, 340, 315, 295, 541, 18, 81, 39, 75, 70, 134, 125, 116, 220, 204, 190, 178, 325, 311, 293, 271, 16, 147, 72, 69, 135, 127, 118, 112, 210, 200, 188, 352, 323, 306, 285, ++ 540, 14, 263, 66, 129, 126, 119, 114, 214, 202, 192, 180, 341, 317, 301, 281, 262, 12, 249, 123, 121, 117, 113, 215, 206, 195, 185, 347, 330, 308, 291, 272, 520, 10, 435, 115, 111, ++ 109, 211, 203, 196, 187, 353, 332, 313, 298, 283, 531, 381, 17, 427, 212, 208, 205, 201, 193, 186, 177, 169, 320, 303, 286, 268, 514, 377, 16, 335, 199, 197, 191, 189, 181, 174, 333, ++ 321, 305, 289, 275, 521, 379, 371, 11, 668, 184, 183, 179, 175, 344, 331, 314, 304, 290, 277, 530, 383, 373, 366, 10, 652, 346, 171, 168, 164, 318, 309, 299, 287, 276, 263, 513, 375, ++ 368, 362, 6, 648, 322, 316, 312, 307, 302, 292, 284, 269, 261, 512, 376, 370, 364, 359, 4, 620, 300, 296, 294, 288, 282, 273, 266, 515, 380, 374, 369, 365, 361, 357, 2, 1033, 280, 278, ++ 274, 267, 264, 259, 382, 378, 372, 367, 363, 360, 358, 356, 0, 43, 20, 19, 17, 15, 13, 11, 9, 7, 6, 4, 7, 5, 3, 1, 3}; ++ static HUFFBITS t32HB[] = {1, 5, 4, 5, 6, 5, 4, 4, 7, 3, 6, 0, 7, 2, 3, 1}; ++ static HUFFBITS t33HB[] = {15, 14, 13, 12, 11, 10, 9, 8, 7, 6, 5, 4, 3, 2, 1, 0}; ++ ++ static unsigned char t1l[] = {1, 3, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 2, 3}; ++ static unsigned char t2l[] = {1, 3, 6, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 3, 3, 5, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 5, 5, 6}; ++ static unsigned char t3l[] = {2, 2, 6, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 3, 2, 5, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 5, 5, 6}; ++ static unsigned char t5l[] = {1, 3, 6, 7, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 3, 3, 6, 7, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 6, 6, 7, 8, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 7, 6, 7, 8}; ++ static unsigned char t6l[] = {3, 3, 5, 7, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 3, 2, 4, 5, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 4, 4, 5, 6, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 6, 5, 6, 7}; ++ static unsigned char t7l[] = {1, 3, 6, 8, 8, 9, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 3, 4, 6, 7, 7, 8, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 6, 5, 7, 8, 8, 9, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 7, 7, 8, 9, 9, 9, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 7, 7, 8, 9, 9, 10, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 8, 8, 9, 10, 10, 10}; ++ static unsigned char t8l[] = {2, 3, 6, 8, 8, 9, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 3, 2, 4, 8, 8, 8, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 6, 4, 6, 8, 8, 9, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 8, 8, 8, 9, 9, 10, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 8, 7, 8, 9, 10, 10, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 9, 8, 9, 9, 11, 11}; ++ static unsigned char t9l[] = {3, 3, 5, 6, 8, 9, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 3, 3, 4, 5, 6, 8, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 4, 4, 5, 6, 7, 8, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 6, 5, 6, 7, 7, 8, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 7, 6, 7, 7, 8, 9, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 8, 7, 8, 8, 9, 9}; ++ static unsigned char t10l[] = {1, 3, 6, 8, 9, 9, 9, 10, 0, 0, 0, 0, 0, 0, 0, 0, 3, 4, 6, 7, 8, 9, 8, 8, 0, 0, 0, 0, 0, 0, 0, 0, 6, 6, 7, 8, 9, 10, 9, 9, 0, 0, 0, 0, 0, 0, 0, 0, 7, 7, 8, 9, 10, 10, 9, 10, 0, 0, 0, 0, 0, 0, 0, 0, 8, 8, 9, 10, 10, 10, 10, 10, 0, 0, 0, 0, 0, 0, 0, 0, 9, 9, 10, 10, 11, 11, 10, 11, 0, 0, 0, 0, 0, 0, 0, 0, 8, 8, 9, 10, 10, 10, 11, 11, 0, 0, 0, 0, 0, 0, 0, 0, 9, 8, 9, 10, 10, 11, 11, 11}; ++ static unsigned char t11l[] = {2, 3, 5, 7, 8, 9, 8, 9, 0, 0, 0, 0, 0, 0, 0, 0, 3, 3, 4, 6, 8, 8, 7, 8, 0, 0, 0, 0, 0, 0, 0, 0, 5, 5, 6, 7, 8, 9, 8, 8, 0, 0, 0, 0, 0, 0, 0, 0, 7, 6, 7, 9, 8, 10, 8, 9, 0, 0, 0, 0, 0, 0, 0, 0, 8, 8, 8, 9, 9, 10, 9, 10, 0, 0, 0, 0, 0, 0, 0, 0, 8, 8, 9, 10, 10, 11, 10, 11, 0, 0, 0, 0, 0, 0, 0, 0, 8, 7, 7, 8, 9, 10, 10, 10, 0, 0, 0, 0, 0, 0, 0, 0, 8, 7, 8, 9, 10, 10, 10, 10}; ++ static unsigned char t12l[] = {4, 3, 5, 7, 8, 9, 9, 9, 0, 0, 0, 0, 0, 0, 0, 0, 3, 3, 4, 5, 7, 7, 8, 8, 0, 0, 0, 0, 0, 0, 0, 0, 5, 4, 5, 6, 7, 8, 7, 8, 0, 0, 0, 0, 0, 0, 0, 0, 6, 5, 6, 6, 7, 8, 8, 8, 0, 0, 0, 0, 0, 0, 0, 0, 7, 6, 7, 7, 8, 8, 8, 9, 0, 0, 0, 0, 0, 0, 0, 0, 8, 7, 8, 8, 8, 9, 8, 9, 0, 0, 0, 0, 0, 0, 0, 0, 8, 7, 7, 8, 8, 9, 9, 10, 0, 0, 0, 0, 0, 0, 0, 0, 9, 8, 8, 9, 9, 9, 9, 10}; ++ static unsigned char t13l[] = {1, 4, 6, 7, 8, 9, 9, 10, 9, 10, 11, 11, 12, 12, 13, 13, 3, 4, 6, 7, 8, 8, 9, 9, 9, 9, 10, 10, 11, 12, 12, 12, 6, 6, 7, 8, 9, 9, 10, 10, 9, 10, 10, 11, 11, 12, 13, 13, 7, 7, 8, 9, 9, 10, 10, 10, 10, 11, 11, 11, 11, 12, 13, 13, ++ 8, 7, 9, 9, 10, 10, 11, 11, 10, 11, 11, 12, 12, 13, 13, 14, 9, 8, 9, 10, 10, 10, 11, 11, 11, 11, 12, 11, 13, 13, 14, 14, 9, 9, 10, 10, 11, 11, 11, 11, 11, 12, 12, 12, 13, 13, 14, 14, 10, 9, 10, 11, 11, 11, 12, 12, 12, 12, 13, 13, 13, 14, 16, 16, 9, 8, 9, 10, ++ 10, 11, 11, 12, 12, 12, 12, 13, 13, 14, 15, 15, 10, 9, 10, 10, 11, 11, 11, 13, 12, 13, 13, 14, 14, 14, 16, 15, 10, 10, 10, 11, 11, 12, 12, 13, 12, 13, 14, 13, 14, 15, 16, 17, 11, 10, 10, 11, 12, 12, 12, 12, 13, 13, 13, 14, 15, 15, 15, 16, 11, 11, 11, 12, 12, ++ 13, 12, 13, 14, 14, 15, 15, 15, 16, 16, 16, 12, 11, 12, 13, 13, 13, 14, 14, 14, 14, 14, 15, 16, 15, 16, 16, 13, 12, 12, 13, 13, 13, 15, 14, 14, 17, 15, 15, 15, 17, 16, 16, 12, 12, 13, 14, 14, 14, 15, 14, 15, 15, 16, 16, 19, 18, 19, 16}; ++ static unsigned char t15l[] = {3, 4, 5, 7, 7, 8, 9, 9, 9, 10, 10, 11, 11, 11, 12, 13, 4, 3, 5, 6, 7, 7, 8, 8, 8, 9, 9, 10, 10, 10, 11, 11, 5, 5, 5, 6, 7, 7, 8, 8, 8, 9, 9, 10, 10, 11, 11, 11, 6, 6, 6, 7, 7, 8, 8, 9, 9, 9, 10, 10, 10, 11, 11, 11, 7, 6, 7, ++ 7, 8, 8, 9, 9, 9, 9, 10, 10, 10, 11, 11, 11, 8, 7, 7, 8, 8, 8, 9, 9, 9, 9, 10, 10, 11, 11, 11, 12, 9, 7, 8, 8, 8, 9, 9, 9, 9, 10, 10, 10, 11, 11, 12, 12, 9, 8, 8, 9, 9, 9, 9, 10, 10, 10, 10, 10, 11, 11, 11, 12, 9, 8, 8, 9, 9, 9, 9, 10, 10, 10, 10, 11, 11, ++ 12, 12, 12, 9, 8, 9, 9, 9, 9, 10, 10, 10, 11, 11, 11, 11, 12, 12, 12, 10, 9, 9, 9, 10, 10, 10, 10, 10, 11, 11, 11, 11, 12, 13, 12, 10, 9, 9, 9, 10, 10, 10, 10, 11, 11, 11, 11, 12, 12, 12, 13, 11, 10, 9, 10, 10, 10, 11, 11, 11, 11, 11, 11, 12, 12, 13, 13, ++ 11, 10, 10, 10, 10, 11, 11, 11, 11, 12, 12, 12, 12, 12, 13, 13, 12, 11, 11, 11, 11, 11, 11, 11, 12, 12, 12, 12, 13, 13, 12, 13, 12, 11, 11, 11, 11, 11, 11, 12, 12, 12, 12, 12, 13, 13, 13, 13}; ++ static unsigned char t16l[] = {1, 4, 6, 8, 9, 9, 10, 10, 11, 11, 11, 12, 12, 12, 13, 9, 3, 4, 6, 7, 8, 9, 9, 9, 10, 10, 10, 11, 12, 11, 12, 8, 6, 6, 7, 8, 9, 9, 10, 10, 11, 10, 11, 11, 11, 12, 12, 9, 8, 7, 8, 9, 9, 10, 10, 10, 11, 11, 12, 12, 12, 13, 13, ++ 10, 9, 8, 9, 9, 10, 10, 11, 11, 11, 12, 12, 12, 13, 13, 13, 9, 9, 8, 9, 9, 10, 11, 11, 12, 11, 12, 12, 13, 13, 13, 14, 10, 10, 9, 9, 10, 11, 11, 11, 11, 12, 12, 12, 12, 13, 13, 14, 10, 10, 9, 10, 10, 11, 11, 11, 12, 12, 13, 13, 13, 13, 15, 15, 10, 10, 10, ++ 10, 11, 11, 11, 12, 12, 13, 13, 13, 13, 14, 14, 14, 10, 11, 10, 10, 11, 11, 12, 12, 13, 13, 13, 13, 14, 13, 14, 13, 11, 11, 11, 10, 11, 12, 12, 12, 12, 13, 14, 14, 14, 15, 15, 14, 10, 12, 11, 11, 11, 12, 12, 13, 14, 14, 14, 14, 14, 14, 13, 14, 11, 12, 12, ++ 12, 12, 12, 13, 13, 13, 13, 15, 14, 14, 14, 14, 16, 11, 14, 12, 12, 12, 13, 13, 14, 14, 14, 16, 15, 15, 15, 17, 15, 11, 13, 13, 11, 12, 14, 14, 13, 14, 14, 15, 16, 15, 17, 15, 14, 11, 9, 8, 8, 9, 9, 10, 10, 10, 11, 11, 11, 11, 11, 11, 11, 8}; ++ static unsigned char t24l[] = {4, 4, 6, 7, 8, 9, 9, 10, 10, 11, 11, 11, 11, 11, 12, 9, 4, 4, 5, 6, 7, 8, 8, 9, 9, 9, 10, 10, 10, 10, 10, 8, 6, 5, 6, 7, 7, 8, 8, 9, 9, 9, 9, 10, 10, 10, 11, 7, 7, 6, 7, 7, 8, 8, 8, 9, 9, 9, 9, 10, 10, 10, 10, 7, 8, 7, 7, 8, ++ 8, 8, 8, 9, 9, 9, 10, 10, 10, 10, 11, 7, 9, 7, 8, 8, 8, 8, 9, 9, 9, 9, 10, 10, 10, 10, 10, 7, 9, 8, 8, 8, 8, 9, 9, 9, 9, 10, 10, 10, 10, 10, 11, 7, 10, 8, 8, 8, 9, 9, 9, 9, 10, 10, 10, 10, 10, 11, 11, 8, 10, 9, 9, 9, 9, 9, 9, 9, 9, 10, 10, 10, 10, 11, 11, ++ 8, 10, 9, 9, 9, 9, 9, 9, 10, 10, 10, 10, 10, 11, 11, 11, 8, 11, 9, 9, 9, 9, 10, 10, 10, 10, 10, 10, 11, 11, 11, 11, 8, 11, 10, 9, 9, 9, 10, 10, 10, 10, 10, 10, 11, 11, 11, 11, 8, 11, 10, 10, 10, 10, 10, 10, 10, 10, 10, 11, 11, 11, 11, 11, 8, 11, 10, 10, ++ 10, 10, 10, 10, 10, 11, 11, 11, 11, 11, 11, 11, 8, 12, 10, 10, 10, 10, 10, 10, 11, 11, 11, 11, 11, 11, 11, 11, 8, 8, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 8, 8, 8, 8, 4}; ++ static unsigned char t32l[] = {1, 4, 4, 5, 4, 6, 5, 6, 4, 5, 5, 6, 5, 6, 6, 6}; ++ static unsigned char t33l[] = {4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4}; ++ ++ #define NOREF -1 ++ struct huffcodetab ht[HTN] = ++ { ++ { 0, 0, 0, 0,NULL,NULL}, ++ { 2,16, 0, 0,t1HB, t1l}, ++ { 3,16, 0, 0,t2HB, t2l}, ++ { 3,16, 0, 0,t3HB, t3l}, ++ { 0, 0, 0, 0,NULL,NULL},/* Apparently not used */ ++ { 4,16, 0, 0,t5HB, t5l}, ++ { 4,16, 0, 0,t6HB, t6l}, ++ { 6,16, 0, 0,t7HB, t7l}, ++ { 6,16, 0, 0,t8HB, t8l}, ++ { 6,16, 0, 0,t9HB, t9l}, ++ { 8,16, 0, 0,t10HB, t10l}, ++ { 8,16, 0, 0,t11HB, t11l}, ++ { 8,16, 0, 0,t12HB, t12l}, ++ {16,16, 0, 0,t13HB, t13l}, ++ { 0, 0, 0, 0,NULL,NULL},/* Apparently not used */ ++ {16,16, 0, 0,t15HB, t15l}, ++ {16,16, 1, 1,t16HB, t16l}, ++ {16,16, 2, 3,t16HB, t16l}, ++ {16,16, 3, 7,t16HB, t16l}, ++ {16,16, 4, 15,t16HB, t16l}, ++ {16,16, 6, 63,t16HB, t16l}, ++ {16,16, 8, 255,t16HB, t16l}, ++ {16,16,10,1023,t16HB, t16l}, ++ {16,16,13,8191,t16HB, t16l}, ++ {16,16, 4, 15,t24HB, t24l}, ++ {16,16, 5, 31,t24HB, t24l}, ++ {16,16, 6, 63,t24HB, t24l}, ++ {16,16, 7, 127,t24HB, t24l}, ++ {16,16, 8, 255,t24HB, t24l}, ++ {16,16, 9, 511,t24HB, t24l}, ++ {16,16,11,2047,t24HB, t24l}, ++ {16,16,13,8191,t24HB, t24l}, ++ { 1,16, 0, 0,t32HB, t32l}, ++ { 1,16, 0, 0,t33HB, t33l}, ++ }; ++ ++ ++ ++ ++ ++ double psy_data[] = ++ {48000.0, 61, ++ 0, 1, 24.5, 4.532, 0.970, 0.000, ++ 1, 1, 24.5, 4.532, 0.755, 0.469, ++ 2, 1, 24.5, 4.532, 0.738, 0.938, ++ 3, 1, 24.5, 0.904, 0.730, 1.406, ++ 4, 1, 24.5, 0.904, 0.724, 1.875, ++ 5, 1, 20.0, 0.090, 0.723, 2.344, ++ 6, 1, 20.0, 0.090, 0.723, 2.813, ++ 7, 1, 20.0, 0.029, 0.723, 3.281, ++ 8, 1, 20.0, 0.029, 0.718, 3.750, ++ 9, 1, 20.0, 0.009, 0.690, 4.199, ++ 10, 1, 20.0, 0.009, 0.660, 4.625, ++ 11, 1, 18.0, 0.009, 0.641, 5.047, ++ 12, 1, 18.0, 0.009, 0.600, 5.438, ++ 13, 1, 18.0, 0.009, 0.584, 5.828, ++ 14, 1, 12.0, 0.009, 0.532, 6.188, ++ 15, 1, 12.0, 0.009, 0.537, 6.522, ++ 16, 2, 6.0, 0.018, 0.857, 7.174, ++ 17, 2, 6.0, 0.018, 0.858, 7.801, ++ 18, 2, 3.0, 0.018, 0.853, 8.402, ++ 19, 2, 3.0, 0.018, 0.824, 8.966, ++ 20, 2, 3.0, 0.018, 0.778, 9.484, ++ 21, 2, 3.0, 0.018, 0.740, 9.966, ++ 22, 2, 0.0, 0.018, 0.709, 10.426, ++ 23, 2, 0.0, 0.018, 0.676, 10.866, ++ 24, 2, 0.0, 0.018, 0.632, 11.279, ++ 25, 2, 0.0, 0.018, 0.592, 11.669, ++ 26, 2, 0.0, 0.018, 0.553, 12.042, ++ 27, 2, 0.0, 0.018, 0.510, 12.386, ++ 28, 2, 0.0, 0.018, 0.513, 12.721, ++ 29, 3, 0.0, 0.027, 0.608, 13.115, ++ 30, 3, 0.0, 0.027, 0.673, 13.562, ++ 31, 3, 0.0, 0.027, 0.637, 13.984, ++ 32, 3, 0.0, 0.027, 0.586, 14.371, ++ 33, 3, 0.0, 0.027, 0.571, 14.741, ++ 34, 4, 0.0, 0.036, 0.616, 15.140, ++ 35, 4, 0.0, 0.036, 0.640, 15.563, ++ 36, 4, 0.0, 0.036, 0.598, 15.962, ++ 37, 4, 0.0, 0.036, 0.538, 16.324, ++ 38, 4, 0.0, 0.036, 0.512, 16.665, ++ 39, 5, 0.0, 0.045, 0.528, 17.020, ++ 40, 5, 0.0, 0.045, 0.517, 17.373, ++ 41, 5, 0.0, 0.045, 0.493, 17.708, ++ 42, 6, 0.0, 0.054, 0.499, 18.045, ++ 43, 7, 0.0, 0.063, 0.525, 18.398, ++ 44, 7, 0.0, 0.063, 0.541, 18.762, ++ 45, 8, 0.0, 0.072, 0.528, 19.120, ++ 46, 8, 0.0, 0.072, 0.510, 19.466, ++ 47, 8, 0.0, 0.072, 0.506, 19.807, ++ 48, 10, 0.0, 0.180, 0.525, 20.159, ++ 49, 10, 0.0, 0.180, 0.536, 20.522, ++ 50, 10, 0.0, 0.180, 0.518, 20.874, ++ 51, 13, 0.0, 0.372, 0.501, 21.214, ++ 52, 13, 0.0, 0.372, 0.497, 21.553, ++ 53, 14, 0.0, 0.400, 0.497, 21.892, ++ 54, 18, 0.0, 1.627, 0.495, 22.231, ++ 55, 18, 0.0, 1.627, 0.494, 22.569, ++ 56, 20, 0.0, 1.808, 0.497, 22.909, ++ 57, 25, 0.0, 22.607, 0.494, 23.248, ++ 58, 25, 0.0, 22.607, 0.487, 23.583, ++ 59, 35, 0.0, 31.650, 0.483, 23.915, ++ 60, 67, 0.0, 605.867, 0.482, 24.246, ++ 61, 67, 0.0, 605.867, 0.524, 24.576, ++ 44100.0, 62, ++ 0, 1, 24.5, 4.532, 0.951, 0.000, ++ 1, 1, 24.5, 4.532, 0.700, 0.431, ++ 2, 1, 24.5, 4.532, 0.681, 0.861, ++ 3, 1, 24.5, 0.904, 0.675, 1.292, ++ 4, 1, 24.5, 0.904, 0.667, 1.723, ++ 5, 1, 20.0, 0.090, 0.665, 2.153, ++ 6, 1, 20.0, 0.090, 0.664, 2.584, ++ 7, 1, 20.0, 0.029, 0.664, 3.015, ++ 8, 1, 20.0, 0.029, 0.664, 3.445, ++ 9, 1, 20.0, 0.029, 0.655, 3.876, ++ 10, 1, 20.0, 0.009, 0.616, 4.279, ++ 11, 1, 20.0, 0.009, 0.597, 4.670, ++ 12, 1, 18.0, 0.009, 0.578, 5.057, ++ 13, 1, 18.0, 0.009, 0.541, 5.416, ++ 14, 1, 18.0, 0.009, 0.575, 5.774, ++ 15, 2, 12.0, 0.018, 0.856, 6.422, ++ 16, 2, 6.0, 0.018, 0.846, 7.026, ++ 17, 2, 6.0, 0.018, 0.840, 7.609, ++ 18, 2, 3.0, 0.018, 0.822, 8.168, ++ 19, 2, 3.0, 0.018, 0.800, 8.710, ++ 20, 2, 3.0, 0.018, 0.753, 9.207, ++ 21, 2, 3.0, 0.018, 0.704, 9.662, ++ 22, 2, 0.0, 0.018, 0.674, 10.099, ++ 23, 2, 0.0, 0.018, 0.640, 10.515, ++ 24, 2, 0.0, 0.018, 0.609, 10.917, ++ 25, 2, 0.0, 0.018, 0.566, 11.293, ++ 26, 2, 0.0, 0.018, 0.535, 11.652, ++ 27, 2, 0.0, 0.018, 0.531, 11.997, ++ 28, 3, 0.0, 0.027, 0.615, 12.394, ++ 29, 3, 0.0, 0.027, 0.686, 12.850, ++ 30, 3, 0.0, 0.027, 0.650, 13.277, ++ 31, 3, 0.0, 0.027, 0.612, 13.681, ++ 32, 3, 0.0, 0.027, 0.567, 14.062, ++ 33, 3, 0.0, 0.027, 0.520, 14.411, ++ 34, 3, 0.0, 0.027, 0.513, 14.751, ++ 35, 4, 0.0, 0.036, 0.557, 15.119, ++ 36, 4, 0.0, 0.036, 0.584, 15.508, ++ 37, 4, 0.0, 0.036, 0.570, 15.883, ++ 38, 5, 0.0, 0.045, 0.579, 16.263, ++ 39, 5, 0.0, 0.045, 0.585, 16.654, ++ 40, 5, 0.0, 0.045, 0.548, 17.020, ++ 41, 6, 0.0, 0.054, 0.536, 17.374, ++ 42, 6, 0.0, 0.054, 0.550, 17.744, ++ 43, 7, 0.0, 0.063, 0.532, 18.104, ++ 44, 7, 0.0, 0.063, 0.504, 18.447, ++ 45, 7, 0.0, 0.063, 0.496, 18.782, ++ 46, 9, 0.0, 0.081, 0.517, 19.130, ++ 47, 9, 0.0, 0.081, 0.527, 19.487, ++ 48, 9, 0.0, 0.081, 0.516, 19.838, ++ 49, 10, 0.0, 0.180, 0.497, 20.179, ++ 50, 10, 0.0, 0.180, 0.489, 20.510, ++ 51, 11, 0.0, 0.198, 0.502, 20.852, ++ 52, 14, 0.0, 0.400, 0.501, 21.196, ++ 53, 14, 0.0, 0.400, 0.491, 21.531, ++ 54, 15, 0.0, 0.429, 0.497, 21.870, ++ 55, 20, 0.0, 1.808, 0.504, 22.214, ++ 56, 20, 0.0, 1.808, 0.504, 22.558, ++ 57, 21, 0.0, 1.898, 0.495, 22.898, ++ 58, 27, 0.0, 24.416, 0.486, 23.232, ++ 59, 27, 0.0, 24.416, 0.484, 23.564, ++ 60, 36, 0.0, 32.554, 0.483, 23.897, ++ 61, 73, 0.0, 660.124, 0.475, 24.229, ++ 62, 18, 0.0, 162.770, 0.515, 24.442, ++ 32000.0, 58, ++ 0, 2, 24.5, 4.532, 0.997, 0.313, ++ 1, 2, 24.5, 4.532, 0.893, 0.938, ++ 2, 2, 24.5, 1.809, 0.881, 1.563, ++ 3, 2, 20.0, 0.181, 0.873, 2.188, ++ 4, 2, 20.0, 0.181, 0.872, 2.813, ++ 5, 2, 20.0, 0.057, 0.871, 3.438, ++ 6, 2, 20.0, 0.018, 0.860, 4.045, ++ 7, 2, 20.0, 0.018, 0.839, 4.625, ++ 8, 2, 18.0, 0.018, 0.812, 5.173, ++ 9, 2, 18.0, 0.018, 0.784, 5.698, ++ 10, 2, 12.0, 0.018, 0.741, 6.185, ++ 11, 2, 12.0, 0.018, 0.697, 6.634, ++ 12, 2, 6.0, 0.018, 0.674, 7.070, ++ 13, 2, 6.0, 0.018, 0.651, 7.492, ++ 14, 2, 6.0, 0.018, 0.633, 7.905, ++ 15, 2, 3.0, 0.018, 0.611, 8.305, ++ 16, 2, 3.0, 0.018, 0.589, 8.695, ++ 17, 2, 3.0, 0.018, 0.575, 9.064, ++ 18, 3, 3.0, 0.027, 0.654, 9.484, ++ 19, 3, 3.0, 0.027, 0.724, 9.966, ++ 20, 3, 0.0, 0.027, 0.701, 10.426, ++ 21, 3, 0.0, 0.027, 0.673, 10.866, ++ 22, 3, 0.0, 0.027, 0.631, 11.279, ++ 23, 3, 0.0, 0.027, 0.592, 11.669, ++ 24, 3, 0.0, 0.027, 0.553, 12.042, ++ 25, 3, 0.0, 0.027, 0.510, 12.386, ++ 26, 3, 0.0, 0.027, 0.506, 12.721, ++ 27, 4, 0.0, 0.036, 0.562, 13.091, ++ 28, 4, 0.0, 0.036, 0.598, 13.488, ++ 29, 4, 0.0, 0.036, 0.589, 13.873, ++ 30, 5, 0.0, 0.045, 0.607, 14.268, ++ 31, 5, 0.0, 0.045, 0.620, 14.679, ++ 32, 5, 0.0, 0.045, 0.580, 15.067, ++ 33, 5, 0.0, 0.045, 0.532, 15.424, ++ 34, 5, 0.0, 0.045, 0.517, 15.771, ++ 35, 6, 0.0, 0.054, 0.517, 16.120, ++ 36, 6, 0.0, 0.054, 0.509, 16.466, ++ 37, 6, 0.0, 0.054, 0.506, 16.807, ++ 38, 8, 0.0, 0.072, 0.522, 17.158, ++ 39, 8, 0.0, 0.072, 0.531, 17.518, ++ 40, 8, 0.0, 0.072, 0.519, 17.869, ++ 41, 10, 0.0, 0.090, 0.512, 18.215, ++ 42, 10, 0.0, 0.090, 0.509, 18.563, ++ 43, 10, 0.0, 0.090, 0.498, 18.902, ++ 44, 12, 0.0, 0.109, 0.494, 19.239, ++ 45, 12, 0.0, 0.109, 0.501, 19.580, ++ 46, 13, 0.0, 0.118, 0.508, 19.925, ++ 47, 14, 0.0, 0.252, 0.502, 20.269, ++ 48, 14, 0.0, 0.252, 0.493, 20.606, ++ 49, 16, 0.0, 0.288, 0.497, 20.944, ++ 50, 20, 0.0, 0.572, 0.506, 21.288, ++ 51, 20, 0.0, 0.572, 0.510, 21.635, ++ 52, 23, 0.0, 0.658, 0.504, 21.980, ++ 53, 27, 0.0, 2.441, 0.496, 22.319, ++ 54, 27, 0.0, 2.441, 0.493, 22.656, ++ 55, 32, 0.0, 2.893, 0.490, 22.993, ++ 56, 37, 0.0, 33.458, 0.482, 23.326, ++ 57, 37, 0.0, 33.458, 0.458, 23.656, ++ 58, 12, 0.0, 10.851, 0.500, 23.937, ++ 24000, 59, ++ 0, 2, 15, 17.8250179, 0.697374165, 0.236874461, ++ 1, 2, 15, 17.8250179, 0.455024809, 0.71016103, ++ 2, 2, 15, 1.78250182, 0.431440443, 1.18193281, ++ 3, 2, 15, 1.78250182, 0.42391625, 1.65102732, ++ 4, 2, 13, 0.178250194, 0.418206781, 2.11632562, ++ 5, 2, 13, 0.178250194, 0.41158545, 2.57676744, ++ 6, 2, 13, 0.0563676581, 0.405409157, 3.03136396, ++ 7, 2, 13, 0.0563676581, 0.399695486, 3.47920918, ++ 8, 2, 13, 0.0563676581, 0.393753231, 3.91948748, ++ 9, 2, 12, 0.0178250186, 0.387357473, 4.35147953, ++ 10, 2, 12, 0.0178250186, 0.38045457, 4.77456427, ++ 11, 2, 10, 0.0178250186, 0.373053908, 5.18822002, ++ 12, 2, 10, 0.0178250186, 0.365188122, 5.59202194, ++ 13, 2, 10, 0.0178250186, 0.356897771, 5.98564005, ++ 14, 2, 9, 0.0178250186, 0.348700613, 6.36883163, ++ 15, 2, 9, 0.0178250186, 0.340260029, 6.74143791, ++ 16, 2, 6, 0.0178250186, 0.332341045, 7.10337448, ++ 17, 2, 6, 0.0178250186, 0.330462843, 7.45462418, ++ 18, 2, 6, 0.0178250186, 0.345568359, 7.79523182, ++ 19, 3, 3, 0.0267375279, 0.377859652, 8.20455742, ++ 20, 3, 3, 0.0267375279, 0.396689415, 8.67640114, ++ 21, 3, 3, 0.0267375279, 0.391237885, 9.12561035, ++ 22, 3, 3, 0.0267375279, 0.37761277, 9.55298138, ++ 23, 3, 3, 0.0267375279, 0.362836808, 9.95940971, ++ 24, 3, 0, 0.0267375279, 0.349010617, 10.3458519, ++ 25, 3, 0, 0.0267375279, 0.339673489, 10.7132998, ++ 26, 3, 0, 0.0267375279, 0.343845725, 11.0627575, ++ 27, 4, 0, 0.0356500372, 0.355822682, 11.447506, ++ 28, 4, 0, 0.0356500372, 0.358104348, 11.8627586, ++ 29, 4, 0, 0.0356500372, 0.34745428, 12.2520256, ++ 30, 4, 0, 0.0356500372, 0.334927917, 12.6173973, ++ 31, 4, 0, 0.0356500372, 0.331643254, 12.9608269, ++ 32, 5, 0, 0.0445625484, 0.333368897, 13.3219252, ++ 33, 5, 0, 0.0445625484, 0.332313001, 13.6976833, ++ 34, 5, 0, 0.0445625484, 0.3314417, 14.047802, ++ 35, 6, 0, 0.0534750558, 0.330947191, 14.405302, ++ 36, 6, 0, 0.0534750558, 0.332477689, 14.7684803, ++ 37, 7, 0, 0.062387567, 0.332647532, 15.1315956, ++ 38, 7, 0, 0.062387567, 0.330841452, 15.4940481, ++ 39, 8, 0, 0.0713000745, 0.327769846, 15.8516159, ++ 40, 8, 0, 0.0713000745, 0.324572712, 16.204628, ++ 41, 9, 0, 0.0802125856, 0.323825002, 16.5502281, ++ 42, 10, 0, 0.0891250968, 0.321414798, 16.9067478, ++ 43, 10, 0, 0.0891250968, 0.318189293, 17.2537231, ++ 44, 11, 0, 0.0980376005, 0.315934151, 17.5901108, ++ 45, 12, 0, 0.106950112, 0.315639287, 17.931406, ++ 46, 13, 0, 0.115862623, 0.316569835, 18.2750721, ++ 47, 14, 0, 0.124775134, 0.31656, 18.6191597, ++ 48, 15, 0, 0.133687645, 0.315465957, 18.9621754, ++ 49, 16, 0, 0.142600149, 0.313576341, 19.3029613, ++ 50, 17, 0, 0.151512653, 0.311635971, 19.6405869, ++ 51, 18, 0, 0.160425171, 0.311066717, 19.9742699, ++ 52, 20, 0, 0.355655879, 0.311465651, 20.3115921, ++ 53, 21, 0, 0.373438686, 0.311872005, 20.6507797, ++ 54, 23, 0, 0.409004271, 0.311015964, 20.9890823, ++ 55, 24, 0, 0.676411927, 0.309207708, 21.3251152, ++ 56, 26, 0, 0.732779562, 0.3081128, 21.6565971, ++ 57, 28, 0, 0.789147198, 0.310006589, 21.9881554, ++ 58, 31, 0, 2.76287794, 0.327113092, 22.3222847, ++ 59, 34, 0, 3.03025317, 0.416082352, 22.6605186, ++ 22050, 59, ++ 0, 2, 15, 17.8250179, 0.658683598, 0.217637643, ++ 1, 2, 15, 17.8250179, 0.432554901, 0.652563453, ++ 2, 2, 15, 1.78250182, 0.405113578, 1.08633137, ++ 3, 2, 15, 1.78250182, 0.397231787, 1.51803517, ++ 4, 2, 15, 1.78250182, 0.392088681, 1.94679713, ++ 5, 2, 13, 0.178250194, 0.386788279, 2.37177849, ++ 6, 2, 13, 0.178250194, 0.380574644, 2.79218864, ++ 7, 2, 13, 0.0563676581, 0.375309765, 3.20729256, ++ 8, 2, 13, 0.0563676581, 0.370087624, 3.61641645, ++ 9, 2, 12, 0.0178250186, 0.364568561, 4.01895428, ++ 10, 2, 12, 0.0178250186, 0.358959526, 4.4143672, ++ 11, 2, 12, 0.0178250186, 0.352938265, 4.80218887, ++ 12, 2, 10, 0.0178250186, 0.3465029, 5.18202305, ++ 13, 2, 10, 0.0178250186, 0.33968094, 5.55354261, ++ 14, 2, 10, 0.0178250186, 0.332571507, 5.91648674, ++ 15, 2, 9, 0.0178250186, 0.326015651, 6.27065945, ++ 16, 2, 9, 0.0178250186, 0.325442046, 6.61592293, ++ 17, 2, 9, 0.0178250186, 0.341315031, 6.95219517, ++ 18, 3, 6, 0.0267375279, 0.374984443, 7.3584404, ++ 19, 3, 6, 0.0267375279, 0.396138102, 7.8290925, ++ 20, 3, 3, 0.0267375279, 0.39271906, 8.27975655, ++ 21, 3, 3, 0.0267375279, 0.380755007, 8.71083069, ++ 22, 3, 3, 0.0267375279, 0.367386311, 9.12284088, ++ 23, 3, 3, 0.0267375279, 0.354351997, 9.51640987, ++ 24, 3, 3, 0.0267375279, 0.341508389, 9.89222908, ++ 25, 3, 0, 0.0267375279, 0.333577901, 10.2510386, ++ 26, 3, 0, 0.0267375279, 0.338108748, 10.5936022, ++ 27, 4, 0, 0.0356500372, 0.350744486, 10.9723492, ++ 28, 4, 0, 0.0356500372, 0.354519457, 11.38272, ++ 29, 4, 0, 0.0356500372, 0.345274031, 11.7689981, ++ 30, 4, 0, 0.0356500372, 0.333828837, 12.1329184, ++ 31, 4, 0, 0.0356500372, 0.331436664, 12.4761295, ++ 32, 5, 0, 0.0445625484, 0.334172577, 12.8381901, ++ 33, 5, 0, 0.0445625484, 0.334024847, 13.2160273, ++ 34, 5, 0, 0.0445625484, 0.33392629, 13.5690479, ++ 35, 6, 0, 0.0534750558, 0.334218502, 13.9303951, ++ 36, 6, 0, 0.0534750558, 0.336405039, 14.298193, ++ 37, 7, 0, 0.062387567, 0.337080389, 14.666563, ++ 38, 7, 0, 0.062387567, 0.335603535, 15.0346909, ++ 39, 8, 0, 0.0713000745, 0.332515866, 15.398139, ++ 40, 8, 0, 0.0713000745, 0.327727586, 15.7570457, ++ 41, 9, 0, 0.0802125856, 0.322346836, 16.1083431, ++ 42, 9, 0, 0.0802125856, 0.317575186, 16.4528522, ++ 43, 10, 0, 0.0891250968, 0.31632933, 16.7886105, ++ 44, 11, 0, 0.0980376005, 0.317602783, 17.132, ++ 45, 12, 0, 0.106950112, 0.319945186, 17.4796028, ++ 46, 13, 0, 0.115862623, 0.320881754, 17.8287659, ++ 47, 14, 0, 0.124775134, 0.320346534, 18.1774921, ++ 48, 15, 0, 0.133687645, 0.318628669, 18.5243168, ++ 49, 16, 0, 0.142600149, 0.316125751, 18.8681736, ++ 50, 17, 0, 0.151512653, 0.313746184, 19.2082729, ++ 51, 18, 0, 0.160425171, 0.312971771, 19.5440025, ++ 52, 20, 0, 0.178250194, 0.313278913, 19.8831882, ++ 53, 21, 0, 0.373438686, 0.313735574, 20.224247, ++ 54, 23, 0, 0.409004271, 0.31308493, 20.5646286, ++ 55, 24, 0, 0.426787049, 0.31156227, 20.903141, ++ 56, 26, 0, 0.732779562, 0.310435742, 21.2376747, ++ 57, 28, 0, 0.789147198, 0.31132248, 21.5730591, ++ 58, 30, 0, 0.845514894, 0.32730341, 21.9066811, ++ 59, 33, 0, 2.94112802, 0.414659739, 22.2411156, ++ 16000, 55, ++ 0, 3, 15, 26.7375278, 0.697374165, 0.236874461, ++ 1, 3, 15, 26.7375278, 0.455024809, 0.71016103, ++ 2, 3, 15, 2.67375278, 0.431440443, 1.18193281, ++ 3, 3, 15, 2.67375278, 0.42391625, 1.65102732, ++ 4, 3, 13, 0.26737529, 0.418206781, 2.11632562, ++ 5, 3, 13, 0.26737529, 0.41158545, 2.57676744, ++ 6, 3, 13, 0.0845514908, 0.405409157, 3.03136396, ++ 7, 3, 13, 0.0845514908, 0.399695486, 3.47920918, ++ 8, 3, 13, 0.0845514908, 0.393753231, 3.91948748, ++ 9, 3, 12, 0.0267375279, 0.387357473, 4.35147953, ++ 10, 3, 12, 0.0267375279, 0.38045457, 4.77456427, ++ 11, 3, 10, 0.0267375279, 0.373053908, 5.18822002, ++ 12, 3, 10, 0.0267375279, 0.365188122, 5.59202194, ++ 13, 3, 10, 0.0267375279, 0.356897742, 5.98564005, ++ 14, 3, 9, 0.0267375279, 0.34869957, 6.36883163, ++ 15, 3, 9, 0.0267375279, 0.340241522, 6.74143791, ++ 16, 3, 6, 0.0267375279, 0.332089454, 7.10337448, ++ 17, 3, 6, 0.0267375279, 0.328292668, 7.45462418, ++ 18, 3, 6, 0.0267375279, 0.336574793, 7.79523182, ++ 19, 4, 3, 0.0356500372, 0.354600489, 8.17827797, ++ 20, 4, 3, 0.0356500372, 0.364343345, 8.59994984, ++ 21, 4, 3, 0.0356500372, 0.359369367, 9.00363636, ++ 22, 4, 3, 0.0356500372, 0.347775847, 9.38988018, ++ 23, 4, 3, 0.0356500372, 0.335562587, 9.7592926, ++ 24, 4, 0, 0.0356500372, 0.326988578, 10.1125278, ++ 25, 4, 0, 0.0356500372, 0.327966213, 10.4502735, ++ 26, 5, 0, 0.0445625484, 0.334450752, 10.811614, ++ 27, 5, 0, 0.0445625484, 0.335228145, 11.1935263, ++ 28, 5, 0, 0.0445625484, 0.329595625, 11.5549288, ++ 29, 5, 0, 0.0445625484, 0.326683223, 11.8971443, ++ 30, 6, 0, 0.0534750558, 0.326986551, 12.2520256, ++ 31, 6, 0, 0.0534750558, 0.325072199, 12.6173973, ++ 32, 6, 0, 0.0534750558, 0.323560268, 12.9608269, ++ 33, 7, 0, 0.062387567, 0.322494298, 13.3093863, ++ 34, 7, 0, 0.062387567, 0.323403448, 13.6617231, ++ 35, 8, 0, 0.0713000745, 0.323232353, 14.0134668, ++ 36, 8, 0, 0.0713000745, 0.322662383, 14.3639784, ++ 37, 9, 0, 0.0802125856, 0.324054241, 14.7098465, ++ 38, 10, 0, 0.0891250968, 0.323228806, 15.0686541, ++ 39, 10, 0, 0.0891250968, 0.320751846, 15.4191036, ++ 40, 11, 0, 0.0980376005, 0.318823338, 15.7594051, ++ 41, 12, 0, 0.106950112, 0.318418682, 16.104557, ++ 42, 13, 0, 0.115862623, 0.318762124, 16.451416, ++ 43, 14, 0, 0.124775134, 0.317806393, 16.7975388, ++ 44, 15, 0, 0.133687645, 0.315653771, 17.1411018, ++ 45, 16, 0, 0.142600149, 0.313369036, 17.4808159, ++ 46, 17, 0, 0.151512653, 0.312513858, 17.8158207, ++ 47, 19, 0, 0.169337675, 0.312785119, 18.1543369, ++ 48, 20, 0, 0.178250194, 0.31343773, 18.4948578, ++ 49, 22, 0, 0.196075201, 0.313258767, 18.8350143, ++ 50, 23, 0, 0.20498772, 0.312570423, 19.1740704, ++ 51, 25, 0, 0.222812727, 0.312572777, 19.5104179, ++ 52, 27, 0, 0.240637749, 0.313047856, 19.8497677, ++ 53, 29, 0, 0.515701056, 0.315029174, 20.1900635, ++ 54, 31, 0, 0.551266611, 0.330613613, 20.5294952, ++ 55, 33, 0, 0.586832225, 0.41819948, 20.8664398, ++ 48000.0, 37, ++ 0, 1, 4.532, 1.000, -8.240, 0.000, ++ 1, 1, 0.904, 0.989, -8.240, 1.875, ++ 2, 1, 0.029, 0.989, -8.240, 3.750, ++ 3, 1, 0.009, 0.981, -8.240, 5.438, ++ 4, 1, 0.009, 0.985, -8.240, 6.857, ++ 5, 1, 0.009, 0.984, -8.240, 8.109, ++ 6, 1, 0.009, 0.980, -8.240, 9.237, ++ 7, 1, 0.009, 0.968, -8.240, 10.202, ++ 8, 1, 0.009, 0.954, -8.240, 11.083, ++ 9, 1, 0.009, 0.929, -8.240, 11.865, ++ 10, 1, 0.009, 0.906, -7.447, 12.554, ++ 11, 1, 0.009, 0.883, -7.447, 13.195, ++ 12, 1, 0.009, 0.844, -7.447, 13.781, ++ 13, 1, 0.009, 0.792, -7.447, 14.309, ++ 14, 1, 0.009, 0.747, -7.447, 14.803, ++ 15, 1, 0.009, 0.689, -7.447, 15.250, ++ 16, 1, 0.009, 0.644, -7.447, 15.667, ++ 17, 1, 0.009, 0.592, -7.447, 16.068, ++ 18, 1, 0.009, 0.553, -7.447, 16.409, ++ 19, 2, 0.018, 0.850, -7.447, 17.045, ++ 20, 2, 0.018, 0.811, -6.990, 17.607, ++ 21, 2, 0.018, 0.736, -6.990, 18.097, ++ 22, 2, 0.018, 0.665, -6.990, 18.528, ++ 23, 2, 0.018, 0.610, -6.990, 18.931, ++ 24, 2, 0.018, 0.544, -6.990, 19.295, ++ 25, 2, 0.018, 0.528, -6.990, 19.636, ++ 26, 3, 0.054, 0.621, -6.990, 20.038, ++ 27, 3, 0.054, 0.673, -6.990, 20.486, ++ 28, 3, 0.054, 0.635, -6.990, 20.900, ++ 29, 4, 0.114, 0.626, -6.990, 21.306, ++ 30, 4, 0.114, 0.636, -6.020, 21.722, ++ 31, 5, 0.452, 0.615, -6.020, 22.128, ++ 32, 5, 0.452, 0.579, -6.020, 22.513, ++ 33, 5, 0.452, 0.551, -6.020, 22.877, ++ 34, 7, 6.330, 0.552, -5.229, 23.241, ++ 35, 7, 6.330, 0.559, -5.229, 23.616, ++ 36, 11, 9.947, 0.528, -5.229, 23.974, ++ 37, 17, 153.727, 0.479, -5.229, 24.313, ++ 44100.0, 38, ++ 0, 1, 4.532, 1.000, -8.240, 0.000, ++ 1, 1, 0.904, 0.983, -8.240, 1.723, ++ 2, 1, 0.029, 0.983, -8.240, 3.445, ++ 3, 1, 0.009, 0.982, -8.240, 5.057, ++ 4, 1, 0.009, 0.985, -8.240, 6.422, ++ 5, 1, 0.009, 0.983, -8.240, 7.609, ++ 6, 1, 0.009, 0.978, -8.240, 8.710, ++ 7, 1, 0.009, 0.967, -8.240, 9.662, ++ 8, 1, 0.009, 0.948, -8.240, 10.515, ++ 9, 1, 0.009, 0.930, -8.240, 11.293, ++ 10, 1, 0.009, 0.914, -7.447, 12.009, ++ 11, 1, 0.009, 0.870, -7.447, 12.625, ++ 12, 1, 0.009, 0.845, -7.447, 13.210, ++ 13, 1, 0.009, 0.800, -7.447, 13.748, ++ 14, 1, 0.009, 0.749, -7.447, 14.241, ++ 15, 1, 0.009, 0.701, -7.447, 14.695, ++ 16, 1, 0.009, 0.653, -7.447, 15.125, ++ 17, 1, 0.009, 0.590, -7.447, 15.508, ++ 18, 1, 0.009, 0.616, -7.447, 15.891, ++ 19, 2, 0.018, 0.860, -7.447, 16.537, ++ 20, 2, 0.018, 0.823, -6.990, 17.112, ++ 21, 2, 0.018, 0.762, -6.990, 17.621, ++ 22, 2, 0.018, 0.688, -6.990, 18.073, ++ 23, 2, 0.018, 0.612, -6.990, 18.470, ++ 24, 2, 0.018, 0.594, -6.990, 18.849, ++ 25, 3, 0.027, 0.658, -6.990, 19.271, ++ 26, 3, 0.027, 0.706, -6.990, 19.741, ++ 27, 3, 0.054, 0.660, -6.990, 20.177, ++ 28, 3, 0.054, 0.606, -6.990, 20.576, ++ 29, 3, 0.054, 0.565, -6.990, 20.950, ++ 30, 4, 0.114, 0.560, -6.020, 21.316, ++ 31, 4, 0.114, 0.579, -6.020, 21.699, ++ 32, 5, 0.452, 0.567, -6.020, 22.078, ++ 33, 5, 0.452, 0.534, -6.020, 22.438, ++ 34, 5, 0.452, 0.514, -5.229, 22.782, ++ 35, 7, 6.330, 0.520, -5.229, 23.133, ++ 36, 7, 6.330, 0.518, -5.229, 23.484, ++ 37, 7, 6.330, 0.507, -5.229, 23.828, ++ 38, 19, 171.813, 0.447, -4.559, 24.173, ++ 32000.0, 41, ++ 0, 1, 4.532, 1.000, -8.240, 0.000, ++ 1, 1, 0.904, 0.985, -8.240, 1.250, ++ 2, 1, 0.090, 0.983, -8.240, 2.500, ++ 3, 1, 0.029, 0.983, -8.240, 3.750, ++ 4, 1, 0.009, 0.981, -8.240, 4.909, ++ 5, 1, 0.009, 0.975, -8.240, 5.958, ++ 6, 1, 0.009, 0.959, -8.240, 6.857, ++ 7, 1, 0.009, 0.944, -8.240, 7.700, ++ 8, 1, 0.009, 0.933, -8.240, 8.500, ++ 9, 1, 0.009, 0.920, -8.240, 9.237, ++ 10, 1, 0.009, 0.892, -7.447, 9.895, ++ 11, 1, 0.009, 0.863, -7.447, 10.500, ++ 12, 1, 0.009, 0.839, -7.447, 11.083, ++ 13, 1, 0.009, 0.786, -7.447, 11.604, ++ 14, 1, 0.009, 0.755, -7.447, 12.107, ++ 15, 1, 0.009, 0.698, -7.447, 12.554, ++ 16, 1, 0.009, 0.673, -7.447, 13.000, ++ 17, 1, 0.009, 0.605, -7.447, 13.391, ++ 18, 1, 0.009, 0.629, -7.447, 13.781, ++ 19, 2, 0.018, 0.883, -7.447, 14.474, ++ 20, 2, 0.018, 0.858, -6.990, 15.096, ++ 21, 2, 0.018, 0.829, -6.990, 15.667, ++ 22, 2, 0.018, 0.767, -6.990, 16.177, ++ 23, 2, 0.018, 0.705, -6.990, 16.636, ++ 24, 2, 0.018, 0.637, -6.990, 17.057, ++ 25, 2, 0.018, 0.564, -6.990, 17.429, ++ 26, 2, 0.018, 0.550, -6.990, 17.786, ++ 27, 3, 0.027, 0.603, -6.990, 18.177, ++ 28, 3, 0.027, 0.635, -6.990, 18.597, ++ 29, 3, 0.027, 0.592, -6.990, 18.994, ++ 30, 3, 0.027, 0.533, -6.020, 19.352, ++ 31, 3, 0.027, 0.518, -6.020, 19.693, ++ 32, 4, 0.072, 0.568, -6.020, 20.066, ++ 33, 4, 0.072, 0.594, -6.020, 20.462, ++ 34, 4, 0.072, 0.568, -5.229, 20.841, ++ 35, 5, 0.143, 0.536, -5.229, 21.201, ++ 36, 5, 0.143, 0.522, -5.229, 21.549, ++ 37, 6, 0.172, 0.542, -5.229, 21.911, ++ 38, 7, 0.633, 0.539, -4.559, 22.275, ++ 39, 7, 0.633, 0.519, -4.559, 22.625, ++ 40, 8, 0.723, 0.514, -3.980, 22.971, ++ 41, 10, 9.043, 0.518, -3.980, 23.321, ++ 24000, 44, ++ 0, 1, 8.91250896, 0.971850038, 0.150000006, 0, ++ 1, 1, 8.91250896, 0.874727964, 0.150000006, 0.946573138, ++ 2, 1, 0.891250908, 0.85779953, 0.150000006, 1.88476217, ++ 3, 1, 0.0891250968, 0.839743853, 0.150000006, 2.8056457, ++ 4, 1, 0.028183829, 0.82260257, 0.150000006, 3.70133615, ++ 5, 1, 0.00891250931, 0.80018574, 0.150000006, 4.56532001, ++ 6, 1, 0.00891250931, 0.771475196, 0.150000006, 5.39263105, ++ 7, 1, 0.00891250931, 0.737389982, 0.150000006, 6.17986727, ++ 8, 1, 0.00891250931, 0.701111019, 0.150000006, 6.92507982, ++ 9, 1, 0.00891250931, 0.65977633, 0.150000006, 7.62757969, ++ 10, 1, 0.00891250931, 0.615037441, 0.150000006, 8.28770351, ++ 11, 1, 0.00891250931, 0.568658054, 0.150000006, 8.90657234, ++ 12, 1, 0.00891250931, 0.522260666, 0.180000007, 9.48587132, ++ 13, 1, 0.00891250931, 0.478903115, 0.180000007, 10.0276566, ++ 14, 1, 0.00891250931, 0.43808648, 0.180000007, 10.5341988, ++ 15, 1, 0.00891250931, 0.412505627, 0.180000007, 11.0078659, ++ 16, 1, 0.00891250931, 0.39070797, 0.180000007, 11.4510288, ++ 17, 1, 0.00891250931, 0.371887118, 0.180000007, 11.866004, ++ 18, 1, 0.00891250931, 0.367617637, 0.180000007, 12.2550087, ++ 19, 1, 0.00891250931, 0.422220588, 0.180000007, 12.6201363, ++ 20, 2, 0.0178250186, 0.564990044, 0.180000007, 13.2772083, ++ 21, 2, 0.0178250186, 0.519700944, 0.180000007, 13.871047, ++ 22, 2, 0.0178250186, 0.455360681, 0.200000003, 14.4024391, ++ 23, 2, 0.0178250186, 0.408867925, 0.200000003, 14.8811684, ++ 24, 2, 0.0178250186, 0.381538749, 0.200000003, 15.3153324, ++ 25, 2, 0.0178250186, 0.362357527, 0.200000003, 15.7116165, ++ 26, 2, 0.0178250186, 0.365735918, 0.200000003, 16.0755405, ++ 27, 3, 0.0267375279, 0.38064, 0.200000003, 16.4882088, ++ 28, 3, 0.0267375279, 0.379183382, 0.200000003, 16.9410992, ++ 29, 3, 0.0267375279, 0.360672712, 0.200000003, 17.3513336, ++ 30, 3, 0.0267375279, 0.343065977, 0.200000003, 17.7264423, ++ 31, 3, 0.0267375279, 0.339290261, 0.200000003, 18.0722466, ++ 32, 4, 0.0356500372, 0.342963994, 0.200000003, 18.4426575, ++ 33, 4, 0.0356500372, 0.343128443, 0.200000003, 18.8344078, ++ 34, 4, 0.0356500372, 0.343988508, 0.25, 19.1955795, ++ 35, 5, 0.0445625484, 0.343928397, 0.25, 19.5697021, ++ 36, 5, 0.0445625484, 0.339527696, 0.25, 19.9551182, ++ 37, 5, 0.0889139697, 0.336541563, 0.280000001, 20.3115921, ++ 38, 6, 0.106696762, 0.334955156, 0.280000001, 20.6737747, ++ 39, 6, 0.169102982, 0.335601568, 0.300000012, 21.0404968, ++ 40, 7, 0.1972868, 0.334716886, 0.300000012, 21.4060211, ++ 41, 7, 0.1972868, 0.331676662, 0.300000012, 21.7696877, ++ 42, 8, 0.713000774, 0.328550965, 0.400000006, 22.1267223, ++ 43, 8, 0.713000774, 0.339241952, 0.400000006, 22.4769249, ++ 44, 9, 0.802125871, 0.425207615, 0.400000006, 22.8164864, ++ 22050, 44, ++ 0, 1, 8.91250896, 0.954045713, 0.150000006, 0, ++ 1, 1, 8.91250896, 0.833381653, 0.150000006, 0.869851649, ++ 2, 1, 0.891250908, 0.815945923, 0.150000006, 1.73325908, ++ 3, 1, 0.0891250968, 0.794244766, 0.150000006, 2.58322191, ++ 4, 1, 0.028183829, 0.776486695, 0.150000006, 3.4134295, ++ 5, 1, 0.00891250931, 0.755260408, 0.150000006, 4.21850443, ++ 6, 1, 0.00891250931, 0.731070817, 0.150000006, 4.99414825, ++ 7, 1, 0.00891250931, 0.701775849, 0.150000006, 5.73718691, ++ 8, 1, 0.00891250931, 0.667876124, 0.150000006, 6.44553185, ++ 9, 1, 0.00891250931, 0.630284071, 0.150000006, 7.11807632, ++ 10, 1, 0.00891250931, 0.590170324, 0.150000006, 7.75455618, ++ 11, 1, 0.00891250931, 0.548788548, 0.150000006, 8.3553915, ++ 12, 1, 0.00891250931, 0.507795513, 0.150000006, 8.92152882, ++ 13, 1, 0.00891250931, 0.469515711, 0.180000007, 9.45430183, ++ 14, 1, 0.00891250931, 0.432291716, 0.180000007, 9.95530319, ++ 15, 1, 0.00891250931, 0.411131173, 0.180000007, 10.4262848, ++ 16, 1, 0.00891250931, 0.390771538, 0.180000007, 10.8690758, ++ 17, 1, 0.00891250931, 0.373318017, 0.180000007, 11.2855215, ++ 18, 1, 0.00891250931, 0.36956048, 0.180000007, 11.6774378, ++ 19, 1, 0.00891250931, 0.42595759, 0.180000007, 12.0465794, ++ 20, 2, 0.0178250186, 0.576900065, 0.180000007, 12.7141209, ++ 21, 2, 0.0178250186, 0.533114731, 0.180000007, 13.3197365, ++ 22, 2, 0.0178250186, 0.469967514, 0.180000007, 13.8634901, ++ 23, 2, 0.0178250186, 0.417268544, 0.200000003, 14.3544445, ++ 24, 2, 0.0178250186, 0.389299124, 0.200000003, 14.8002586, ++ 25, 2, 0.0178250186, 0.362824857, 0.200000003, 15.2073727, ++ 26, 2, 0.0178250186, 0.346801281, 0.200000003, 15.5811834, ++ 27, 2, 0.0178250186, 0.349400043, 0.200000003, 15.926218, ++ 28, 3, 0.0267375279, 0.364026934, 0.200000003, 16.3194923, ++ 29, 3, 0.0267375279, 0.36560446, 0.200000003, 16.752903, ++ 30, 3, 0.0267375279, 0.354275256, 0.200000003, 17.1470814, ++ 31, 3, 0.0267375279, 0.351219416, 0.200000003, 17.5086212, ++ 32, 4, 0.0356500372, 0.354364097, 0.200000003, 17.8938141, ++ 33, 4, 0.0356500372, 0.348915905, 0.200000003, 18.2992878, ++ 34, 4, 0.0356500372, 0.337649345, 0.200000003, 18.6713982, ++ 35, 4, 0.0356500372, 0.332076877, 0.25, 19.015646, ++ 36, 5, 0.0445625484, 0.330793113, 0.25, 19.3734016, ++ 37, 5, 0.0445625484, 0.327528268, 0.25, 19.7430382, ++ 38, 5, 0.0889139697, 0.32551071, 0.280000001, 20.0859604, ++ 39, 6, 0.106696762, 0.324436843, 0.280000001, 20.4354992, ++ 40, 6, 0.106696762, 0.325835049, 0.280000001, 20.7905579, ++ 41, 7, 0.1972868, 0.326221824, 0.300000012, 21.1458054, ++ 42, 7, 0.1972868, 0.325960994, 0.300000012, 21.5005951, ++ 43, 8, 0.225470632, 0.339019388, 0.300000012, 21.8504524, ++ 44, 8, 0.713000774, 0.426850349, 0.400000006, 22.1951065, ++ 16000, 45, ++ 0, 1, 8.91250896, 0.834739447, 0.150000006, 0, ++ 1, 1, 8.91250896, 0.623757005, 0.150000006, 0.631518543, ++ 2, 1, 0.891250908, 0.60420388, 0.150000006, 1.2606914, ++ 3, 1, 0.891250908, 0.591974258, 0.150000006, 1.88476217, ++ 4, 1, 0.0891250968, 0.575301588, 0.150000006, 2.50111985, ++ 5, 1, 0.028183829, 0.561547697, 0.150000006, 3.1073606, ++ 6, 1, 0.028183829, 0.546665847, 0.150000006, 3.70133615, ++ 7, 1, 0.00891250931, 0.52986443, 0.150000006, 4.28118753, ++ 8, 1, 0.00891250931, 0.511183441, 0.150000006, 4.84536505, ++ 9, 1, 0.00891250931, 0.490902334, 0.150000006, 5.39263105, ++ 10, 1, 0.00891250931, 0.46938166, 0.150000006, 5.92205667, ++ 11, 1, 0.00891250931, 0.447003782, 0.150000006, 6.43299866, ++ 12, 1, 0.00891250931, 0.428170592, 0.150000006, 6.92507982, ++ 13, 1, 0.00891250931, 0.414536625, 0.150000006, 7.39815664, ++ 14, 1, 0.00891250931, 0.401033074, 0.150000006, 7.85228777, ++ 15, 1, 0.00891250931, 0.38779071, 0.150000006, 8.28770351, ++ 16, 1, 0.00891250931, 0.374230444, 0.150000006, 8.704772, ++ 17, 1, 0.00891250931, 0.360547513, 0.180000007, 9.10397339, ++ 18, 1, 0.00891250931, 0.348256677, 0.180000007, 9.48587132, ++ 19, 1, 0.00891250931, 0.350327015, 0.180000007, 9.85109234, ++ 20, 1, 0.00891250931, 0.406330824, 0.180000007, 10.200304, ++ 21, 2, 0.0178250186, 0.554098248, 0.180000007, 10.846529, ++ 22, 2, 0.0178250186, 0.528312504, 0.180000007, 11.4447651, ++ 23, 2, 0.0178250186, 0.476527005, 0.180000007, 11.9928398, ++ 24, 2, 0.0178250186, 0.428205669, 0.180000007, 12.495945, ++ 25, 2, 0.0178250186, 0.402271926, 0.180000007, 12.9588718, ++ 26, 2, 0.0178250186, 0.378024429, 0.180000007, 13.3859692, ++ 27, 2, 0.0178250186, 0.36254698, 0.180000007, 13.7811394, ++ 28, 2, 0.0178250186, 0.368058592, 0.200000003, 14.1478529, ++ 29, 3, 0.0267375279, 0.385963261, 0.200000003, 14.5674343, ++ 30, 3, 0.0267375279, 0.38640517, 0.200000003, 15.0304852, ++ 31, 3, 0.0267375279, 0.367834061, 0.200000003, 15.4513416, ++ 32, 3, 0.0267375279, 0.349686563, 0.200000003, 15.836277, ++ 33, 3, 0.0267375279, 0.345709383, 0.200000003, 16.1904697, ++ 34, 4, 0.0356500372, 0.34871915, 0.200000003, 16.5683517, ++ 35, 4, 0.0356500372, 0.347054332, 0.200000003, 16.9660263, ++ 36, 4, 0.0356500372, 0.346329987, 0.200000003, 17.3304482, ++ 37, 5, 0.0445625484, 0.344658494, 0.200000003, 17.7055588, ++ 38, 5, 0.0445625484, 0.338779271, 0.200000003, 18.0899811, ++ 39, 5, 0.0445625484, 0.334878683, 0.200000003, 18.4440536, ++ 40, 6, 0.0534750558, 0.332811534, 0.200000003, 18.8030052, ++ 41, 6, 0.0534750558, 0.333717585, 0.25, 19.1665268, ++ 42, 7, 0.062387567, 0.333986402, 0.25, 19.5299358, ++ 43, 7, 0.062387567, 0.334142625, 0.25, 19.8934898, ++ 44, 8, 0.142262354, 0.34677428, 0.280000001, 20.2535706, ++ 45, 8, 0.142262354, 0.436254472, 0.280000001, 20.610569, ++ 48000.0, 20, ++ 0, 3, 0, 4, 1.000, 0.056, ++ 1, 3, 4, 7, 0.944, 0.611, ++ 2, 4, 7, 11, 0.389, 0.167, ++ 3, 3, 11, 14, 0.833, 0.722, ++ 4, 3, 14, 17, 0.278, 0.639, ++ 5, 2, 17, 19, 0.361, 0.417, ++ 6, 3, 19, 22, 0.583, 0.083, ++ 7, 2, 22, 24, 0.917, 0.750, ++ 8, 3, 24, 27, 0.250, 0.417, ++ 9, 3, 27, 30, 0.583, 0.648, ++ 10, 3, 30, 33, 0.352, 0.611, ++ 11, 3, 33, 36, 0.389, 0.625, ++ 12, 4, 36, 40, 0.375, 0.144, ++ 13, 3, 40, 43, 0.856, 0.389, ++ 14, 3, 43, 46, 0.611, 0.160, ++ 15, 3, 46, 49, 0.840, 0.217, ++ 16, 3, 49, 52, 0.783, 0.184, ++ 17, 2, 52, 54, 0.816, 0.886, ++ 18, 3, 54, 57, 0.114, 0.313, ++ 19, 2, 57, 59, 0.687, 0.452, ++ 20, 1, 59, 60, 0.548, 0.908, ++ 44100.0, 20, ++ 0, 3, 0, 4, 1.000, 0.056, ++ 1, 3, 4, 7, 0.944, 0.611, ++ 2, 4, 7, 11, 0.389, 0.167, ++ 3, 3, 11, 14, 0.833, 0.722, ++ 4, 3, 14, 17, 0.278, 0.139, ++ 5, 1, 17, 18, 0.861, 0.917, ++ 6, 3, 18, 21, 0.083, 0.583, ++ 7, 3, 21, 24, 0.417, 0.250, ++ 8, 3, 24, 27, 0.750, 0.805, ++ 9, 3, 27, 30, 0.194, 0.574, ++ 10, 3, 30, 33, 0.426, 0.537, ++ 11, 3, 33, 36, 0.463, 0.819, ++ 12, 4, 36, 40, 0.180, 0.100, ++ 13, 3, 40, 43, 0.900, 0.468, ++ 14, 3, 43, 46, 0.532, 0.623, ++ 15, 3, 46, 49, 0.376, 0.450, ++ 16, 3, 49, 52, 0.550, 0.552, ++ 17, 3, 52, 55, 0.448, 0.403, ++ 18, 2, 55, 57, 0.597, 0.643, ++ 19, 2, 57, 59, 0.357, 0.722, ++ 20, 2, 59, 61, 0.278, 0.960, ++ 32000, 20, ++ 0, 1, 0, 2, 1.000, 0.528, ++ 1, 2, 2, 4, 0.472, 0.305, ++ 2, 2, 4, 6, 0.694, 0.083, ++ 3, 1, 6, 7, 0.917, 0.861, ++ 4, 2, 7, 9, 0.139, 0.639, ++ 5, 2, 9, 11, 0.361, 0.417, ++ 6, 3, 11, 14, 0.583, 0.083, ++ 7, 2, 14, 16, 0.917, 0.750, ++ 8, 3, 16, 19, 0.250, 0.870, ++ 9, 3, 19, 22, 0.130, 0.833, ++ 10, 4, 22, 26, 0.167, 0.389, ++ 11, 4, 26, 30, 0.611, 0.478, ++ 12, 4, 30, 34, 0.522, 0.033, ++ 13, 3, 34, 37, 0.967, 0.917, ++ 14, 4, 37, 41, 0.083, 0.617, ++ 15, 3, 41, 44, 0.383, 0.995, ++ 16, 4, 44, 48, 0.005, 0.274, ++ 17, 3, 48, 51, 0.726, 0.480, ++ 18, 3, 51, 54, 0.519, 0.261, ++ 19, 2, 54, 56, 0.739, 0.884, ++ 20, 2, 56, 58, 0.116, 1.000, ++ 24000, 20, ++ 0, 2, 0, 3, 1, 0.916666746, ++ 1, 3, 3, 6, 0.0833332539, 0.583333492, ++ 2, 3, 6, 9, 0.416666508, 0.25, ++ 3, 2, 9, 11, 0.75, 0.916666985, ++ 4, 3, 11, 14, 0.0833330154, 0.583333969, ++ 5, 3, 14, 17, 0.416666031, 0.25, ++ 6, 3, 17, 20, 0.75, 0.537036896, ++ 7, 3, 20, 23, 0.462963104, 0.5, ++ 8, 4, 23, 27, 0.5, 0.0555559993, ++ 9, 3, 27, 30, 0.944444001, 0.402778625, ++ 10, 3, 30, 33, 0.597221375, 0.766667187, ++ 11, 3, 33, 36, 0.233332828, 0.805555999, ++ 12, 3, 36, 39, 0.194444016, 0.769841909, ++ 13, 3, 39, 42, 0.23015812, 0.611111104, ++ 14, 3, 42, 45, 0.388888896, 0.449494779, ++ 15, 3, 45, 48, 0.550505221, 0.194444954, ++ 16, 2, 48, 50, 0.805555046, 0.913194656, ++ 17, 3, 50, 53, 0.0868053436, 0.580555737, ++ 18, 3, 53, 56, 0.419444263, 0.113426208, ++ 19, 2, 56, 58, 0.886573792, 0.533730626, ++ 20, 2, 58, 60, 0.466269344, 0.691176474, ++ 22050, 20, ++ 0, 2, 0, 3, 1, 0.916666746, ++ 1, 3, 3, 6, 0.0833332539, 0.583333492, ++ 2, 3, 6, 9, 0.416666508, 0.25, ++ 3, 2, 9, 11, 0.75, 0.916666985, ++ 4, 3, 11, 14, 0.0833330154, 0.583333969, ++ 5, 3, 14, 17, 0.416666031, 0.25, ++ 6, 3, 17, 20, 0.75, 0.203703582, ++ 7, 3, 20, 23, 0.796296418, 0.166666687, ++ 8, 3, 23, 26, 0.833333313, 0.722222686, ++ 9, 4, 26, 30, 0.277777344, 0.152778625, ++ 10, 3, 30, 33, 0.847221375, 0.566667199, ++ 11, 3, 33, 36, 0.433332831, 0.93518573, ++ 12, 4, 36, 40, 0.0648142472, 0.118056297, ++ 13, 3, 40, 43, 0.881943703, 0.0925937295, ++ 14, 2, 43, 45, 0.907406271, 0.934344172, ++ 15, 3, 45, 48, 0.0656557977, 0.575398028, ++ 16, 3, 48, 51, 0.424601972, 0.232026935, ++ 17, 2, 51, 53, 0.767973065, 0.758334339, ++ 18, 3, 53, 56, 0.241665646, 0.187501252, ++ 19, 2, 56, 58, 0.812498748, 0.533731699, ++ 20, 2, 58, 60, 0.466268271, 0.257577598, ++ 16000, 20, ++ 0, 1, 0, 2, 1, 0.944444478, ++ 1, 2, 2, 4, 0.0555555038, 0.722222328, ++ 2, 2, 4, 6, 0.277777672, 0.5, ++ 3, 2, 6, 8, 0.5, 0.27777797, ++ 4, 2, 8, 10, 0.72222203, 0.0555559993, ++ 5, 1, 10, 11, 0.944444001, 0.833333313, ++ 6, 3, 11, 14, 0.166666672, 0.203703582, ++ 7, 3, 14, 17, 0.796296418, 0.166666687, ++ 8, 3, 17, 20, 0.833333313, 0.54166698, ++ 9, 3, 20, 23, 0.458333015, 0.652778625, ++ 10, 4, 23, 27, 0.347221375, 0.166667163, ++ 11, 3, 27, 30, 0.833332837, 0.722222924, ++ 12, 4, 30, 34, 0.277777106, 0.277778625, ++ 13, 3, 34, 37, 0.722221375, 0.604167938, ++ 14, 3, 37, 40, 0.395832062, 0.627778649, ++ 15, 3, 40, 43, 0.37222138, 0.542736351, ++ 16, 3, 43, 46, 0.457263649, 0.371528625, ++ 17, 3, 46, 49, 0.628471375, 0.00833433867, ++ 18, 2, 49, 51, 0.991665661, 0.500001311, ++ 19, 2, 51, 53, 0.499998659, 0.886832893, ++ 20, 2, 53, 55, 0.113167092, 0.629034221, ++ 48000.0, 11, ++ 0, 2, 0, 2, 1.000, 0.167, ++ 1, 2, 3, 5, 0.833, 0.833, ++ 2, 3, 5, 8, 0.167, 0.500, ++ 3, 3, 8, 11, 0.500, 0.167, ++ 4, 4, 11, 15, 0.833, 0.167, ++ 5, 4, 15, 19, 0.833, 0.583, ++ 6, 3, 19, 22, 0.417, 0.917, ++ 7, 4, 22, 26, 0.083, 0.944, ++ 8, 4, 26, 30, 0.055, 0.042, ++ 9, 2, 30, 32, 0.958, 0.567, ++ 10, 3, 32, 35, 0.433, 0.167, ++ 11, 2, 35, 37, 0.833, 0.618, ++ 44100.0, 11, ++ 0, 2, 0, 2, 1.000, 0.167, ++ 1, 2, 3, 5, 0.833, 0.833, ++ 2, 3, 5, 8, 0.167, 0.500, ++ 3, 3, 8, 11, 0.500, 0.167, ++ 4, 4, 11, 15, 0.833, 0.167, ++ 5, 5, 15, 20, 0.833, 0.250, ++ 6, 3, 20, 23, 0.750, 0.583, ++ 7, 4, 23, 27, 0.417, 0.055, ++ 8, 3, 27, 30, 0.944, 0.375, ++ 9, 3, 30, 33, 0.625, 0.300, ++ 10, 3, 33, 36, 0.700, 0.167, ++ 11, 2, 36, 38, 0.833, 1.000, ++ 32000, 11, ++ 0, 2, 0, 2, 1.000, 0.167, ++ 1, 2, 3, 5, 0.833, 0.833, ++ 2, 3, 5, 8, 0.167, 0.500, ++ 3, 3, 8, 11, 0.500, 0.167, ++ 4, 4, 11, 15, 0.833, 0.167, ++ 5, 5, 15, 20, 0.833, 0.250, ++ 6, 4, 20, 24, 0.750, 0.250, ++ 7, 5, 24, 29, 0.750, 0.055, ++ 8, 4, 29, 33, 0.944, 0.375, ++ 9, 4, 33, 37, 0.625, 0.472, ++ 10, 3, 37, 40, 0.528, 0.937, ++ 11, 1, 40, 41, 0.062, 1.000, ++ 24000, 11, ++ 0, 3, 0, 4, 1, 0.166666746, ++ 1, 2, 4, 6, 0.833333254, 0.833333492, ++ 2, 3, 6, 9, 0.166666508, 0.5, ++ 3, 4, 9, 13, 0.5, 0.5, ++ 4, 5, 13, 18, 0.5, 0.833333969, ++ 5, 5, 18, 23, 0.166666031, 0.25, ++ 6, 4, 23, 27, 0.75, 0.25, ++ 7, 3, 27, 30, 0.75, 0.611111999, ++ 8, 4, 30, 34, 0.388888031, 0.208333969, ++ 9, 3, 34, 37, 0.791666031, 0.766667187, ++ 10, 4, 37, 41, 0.233332828, 0.45238167, ++ 11, 4, 41, 45, 0.54761833, 0.277778625, ++ 22050, 11, ++ 0, 3, 0, 4, 1, 0.166666746, ++ 1, 2, 4, 6, 0.833333254, 0.833333492, ++ 2, 3, 6, 9, 0.166666508, 0.5, ++ 3, 4, 9, 13, 0.5, 0.5, ++ 4, 4, 13, 17, 0.5, 0.5, ++ 5, 4, 17, 21, 0.5, 0.916666985, ++ 6, 4, 21, 25, 0.0833330154, 0.25, ++ 7, 4, 25, 29, 0.75, 0.611111999, ++ 8, 4, 29, 33, 0.388888031, 0.458333969, ++ 9, 4, 33, 37, 0.541666031, 0.633334339, ++ 10, 4, 37, 41, 0.366665661, 0.583334565, ++ 11, 4, 41, 45, 0.416665405, 0.437500954, ++ 16000, 11, ++ 0, 3, 0, 4, 1, 0.166666746, ++ 1, 2, 4, 6, 0.833333254, 0.833333492, ++ 2, 3, 6, 9, 0.166666508, 0.5, ++ 3, 4, 9, 13, 0.5, 0.5, ++ 4, 5, 13, 18, 0.5, 0.833333969, ++ 5, 5, 18, 23, 0.166666031, 0.75, ++ 6, 4, 23, 27, 0.25, 0.75, ++ 7, 4, 27, 31, 0.25, 0.611111999, ++ 8, 4, 31, 35, 0.388888031, 0.458333969, ++ 9, 4, 35, 39, 0.541666031, 0.166667163, ++ 10, 3, 39, 42, 0.833332837, 0.805555999, ++ 11, 4, 42, 46, 0.194444016, 0.4375}; ++ ++ double enwindow[] = ++ {0.000000000, -0.000000477, -0.000000477, -0.000000477, ++ -0.000000477, -0.000000477, -0.000000477, -0.000000954, ++ -0.000000954, -0.000000954, -0.000000954, -0.000001431, ++ -0.000001431, -0.000001907, -0.000001907, -0.000002384, ++ -0.000002384, -0.000002861, -0.000003338, -0.000003338, ++ -0.000003815, -0.000004292, -0.000004768, -0.000005245, ++ -0.000006199, -0.000006676, -0.000007629, -0.000008106, ++ -0.000009060, -0.000010014, -0.000011444, -0.000012398, ++ -0.000013828, -0.000014782, -0.000016689, -0.000018120, ++ -0.000019550, -0.000021458, -0.000023365, -0.000025272, ++ -0.000027657, -0.000030041, -0.000032425, -0.000034809, ++ -0.000037670, -0.000040531, -0.000043392, -0.000046253, ++ -0.000049591, -0.000052929, -0.000055790, -0.000059605, ++ -0.000062943, -0.000066280, -0.000070095, -0.000073433, ++ -0.000076771, -0.000080585, -0.000083923, -0.000087261, ++ -0.000090599, -0.000093460, -0.000096321, -0.000099182, ++ 0.000101566, 0.000103951, 0.000105858, 0.000107288, ++ 0.000108242, 0.000108719, 0.000108719, 0.000108242, ++ 0.000106812, 0.000105381, 0.000102520, 0.000099182, ++ 0.000095367, 0.000090122, 0.000084400, 0.000077724, ++ 0.000069618, 0.000060558, 0.000050545, 0.000039577, ++ 0.000027180, 0.000013828, -0.000000954, -0.000017166, ++ -0.000034332, -0.000052929, -0.000072956, -0.000093937, ++ -0.000116348, -0.000140190, -0.000165462, -0.000191212, ++ -0.000218868, -0.000247478, -0.000277042, -0.000307560, ++ -0.000339031, -0.000371456, -0.000404358, -0.000438213, ++ -0.000472546, -0.000507355, -0.000542164, -0.000576973, ++ -0.000611782, -0.000646591, -0.000680923, -0.000714302, ++ -0.000747204, -0.000779152, -0.000809669, -0.000838757, ++ -0.000866413, -0.000891685, -0.000915051, -0.000935555, ++ -0.000954151, -0.000968933, -0.000980854, -0.000989437, ++ -0.000994205, -0.000995159, -0.000991821, -0.000983715, ++ 0.000971317, 0.000953674, 0.000930786, 0.000902653, ++ 0.000868797, 0.000829220, 0.000783920, 0.000731945, ++ 0.000674248, 0.000610352, 0.000539303, 0.000462532, ++ 0.000378609, 0.000288486, 0.000191689, 0.000088215, ++ -0.000021458, -0.000137329, -0.000259876, -0.000388145, ++ -0.000522137, -0.000661850, -0.000806808, -0.000956535, ++ -0.001111031, -0.001269817, -0.001432419, -0.001597881, ++ -0.001766682, -0.001937389, -0.002110004, -0.002283096, ++ -0.002457142, -0.002630711, -0.002803326, -0.002974033, ++ -0.003141880, -0.003306866, -0.003467083, -0.003622532, ++ -0.003771782, -0.003914356, -0.004048824, -0.004174709, ++ -0.004290581, -0.004395962, -0.004489899, -0.004570484, ++ -0.004638195, -0.004691124, -0.004728317, -0.004748821, ++ -0.004752159, -0.004737377, -0.004703045, -0.004649162, ++ -0.004573822, -0.004477024, -0.004357815, -0.004215240, ++ -0.004049301, -0.003858566, -0.003643036, -0.003401756, ++ 0.003134727, 0.002841473, 0.002521515, 0.002174854, ++ 0.001800537, 0.001399517, 0.000971317, 0.000515938, ++ 0.000033379, -0.000475883, -0.001011848, -0.001573563, ++ -0.002161503, -0.002774239, -0.003411293, -0.004072189, ++ -0.004756451, -0.005462170, -0.006189346, -0.006937027, ++ -0.007703304, -0.008487225, -0.009287834, -0.010103703, ++ -0.010933399, -0.011775017, -0.012627602, -0.013489246, ++ -0.014358521, -0.015233517, -0.016112804, -0.016994476, ++ -0.017876148, -0.018756866, -0.019634247, -0.020506859, ++ -0.021372318, -0.022228718, -0.023074150, -0.023907185, ++ -0.024725437, -0.025527000, -0.026310921, -0.027073860, ++ -0.027815342, -0.028532982, -0.029224873, -0.029890060, ++ -0.030526638, -0.031132698, -0.031706810, -0.032248020, ++ -0.032754898, -0.033225536, -0.033659935, -0.034055710, ++ -0.034412861, -0.034730434, -0.035007000, -0.035242081, ++ -0.035435200, -0.035586357, -0.035694122, -0.035758972, ++ 0.035780907, 0.035758972, 0.035694122, 0.035586357, ++ 0.035435200, 0.035242081, 0.035007000, 0.034730434, ++ 0.034412861, 0.034055710, 0.033659935, 0.033225536, ++ 0.032754898, 0.032248020, 0.031706810, 0.031132698, ++ 0.030526638, 0.029890060, 0.029224873, 0.028532982, ++ 0.027815342, 0.027073860, 0.026310921, 0.025527000, ++ 0.024725437, 0.023907185, 0.023074150, 0.022228718, ++ 0.021372318, 0.020506859, 0.019634247, 0.018756866, ++ 0.017876148, 0.016994476, 0.016112804, 0.015233517, ++ 0.014358521, 0.013489246, 0.012627602, 0.011775017, ++ 0.010933399, 0.010103703, 0.009287834, 0.008487225, ++ 0.007703304, 0.006937027, 0.006189346, 0.005462170, ++ 0.004756451, 0.004072189, 0.003411293, 0.002774239, ++ 0.002161503, 0.001573563, 0.001011848, 0.000475883, ++ -0.000033379, -0.000515938, -0.000971317, -0.001399517, ++ -0.001800537, -0.002174854, -0.002521515, -0.002841473, ++ 0.003134727, 0.003401756, 0.003643036, 0.003858566, ++ 0.004049301, 0.004215240, 0.004357815, 0.004477024, ++ 0.004573822, 0.004649162, 0.004703045, 0.004737377, ++ 0.004752159, 0.004748821, 0.004728317, 0.004691124, ++ 0.004638195, 0.004570484, 0.004489899, 0.004395962, ++ 0.004290581, 0.004174709, 0.004048824, 0.003914356, ++ 0.003771782, 0.003622532, 0.003467083, 0.003306866, ++ 0.003141880, 0.002974033, 0.002803326, 0.002630711, ++ 0.002457142, 0.002283096, 0.002110004, 0.001937389, ++ 0.001766682, 0.001597881, 0.001432419, 0.001269817, ++ 0.001111031, 0.000956535, 0.000806808, 0.000661850, ++ 0.000522137, 0.000388145, 0.000259876, 0.000137329, ++ 0.000021458, -0.000088215, -0.000191689, -0.000288486, ++ -0.000378609, -0.000462532, -0.000539303, -0.000610352, ++ -0.000674248, -0.000731945, -0.000783920, -0.000829220, ++ -0.000868797, -0.000902653, -0.000930786, -0.000953674, ++ 0.000971317, 0.000983715, 0.000991821, 0.000995159, ++ 0.000994205, 0.000989437, 0.000980854, 0.000968933, ++ 0.000954151, 0.000935555, 0.000915051, 0.000891685, ++ 0.000866413, 0.000838757, 0.000809669, 0.000779152, ++ 0.000747204, 0.000714302, 0.000680923, 0.000646591, ++ 0.000611782, 0.000576973, 0.000542164, 0.000507355, ++ 0.000472546, 0.000438213, 0.000404358, 0.000371456, ++ 0.000339031, 0.000307560, 0.000277042, 0.000247478, ++ 0.000218868, 0.000191212, 0.000165462, 0.000140190, ++ 0.000116348, 0.000093937, 0.000072956, 0.000052929, ++ 0.000034332, 0.000017166, 0.000000954, -0.000013828, ++ -0.000027180, -0.000039577, -0.000050545, -0.000060558, ++ -0.000069618, -0.000077724, -0.000084400, -0.000090122, ++ -0.000095367, -0.000099182, -0.000102520, -0.000105381, ++ -0.000106812, -0.000108242, -0.000108719, -0.000108719, ++ -0.000108242, -0.000107288, -0.000105858, -0.000103951, ++ 0.000101566, 0.000099182, 0.000096321, 0.000093460, ++ 0.000090599, 0.000087261, 0.000083923, 0.000080585, ++ 0.000076771, 0.000073433, 0.000070095, 0.000066280, ++ 0.000062943, 0.000059605, 0.000055790, 0.000052929, ++ 0.000049591, 0.000046253, 0.000043392, 0.000040531, ++ 0.000037670, 0.000034809, 0.000032425, 0.000030041, ++ 0.000027657, 0.000025272, 0.000023365, 0.000021458, ++ 0.000019550, 0.000018120, 0.000016689, 0.000014782, ++ 0.000013828, 0.000012398, 0.000011444, 0.000010014, ++ 0.000009060, 0.000008106, 0.000007629, 0.000006676, ++ 0.000006199, 0.000005245, 0.000004768, 0.000004292, ++ 0.000003815, 0.000003338, 0.000003338, 0.000002861, ++ 0.000002384, 0.000002384, 0.000001907, 0.000001907, ++ 0.000001431, 0.000001431, 0.000000954, 0.000000954, ++ 0.000000954, 0.000000954, 0.000000477, 0.000000477, ++ 0.000000477, 0.000000477, 0.000000477, 0.000000477}; ++ +diff -r -c -N encoder/tables.h lame3.50/tables.h +*** encoder/tables.h Wed Dec 31 17:00:00 1969 +--- lame3.50/tables.h Fri Sep 10 09:52:07 1999 +*************** +*** 0 **** +--- 1,25 ---- ++ /* ++ ** Copyright (C) 1999 Albert L. Faber ++ ** ++ ** This program is free software; you can redistribute it and/or modify ++ ** it under the terms of the GNU General Public License as published by ++ ** the Free Software Foundation; either version 2 of the License, or ++ ** (at your option) any later version. ++ ** ++ ** This program is distributed in the hope that it will be useful, ++ ** but WITHOUT ANY WARRANTY; without even the implied warranty of ++ ** MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the ++ ** GNU General Public License for more details. ++ ** ++ ** You should have received a copy of the GNU General Public License ++ ** along with this program; if not, write to the Free Software ++ ** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. ++ */ ++ ++ #ifndef TABLES_H_INCLUDED ++ #define TABLES_H_INCLUDED ++ ++ extern double enwindow[]; ++ extern double psy_data[]; ++ ++ #endif +diff -r -c -N encoder/takehiro.c lame3.50/takehiro.c +*** encoder/takehiro.c Wed Dec 31 17:00:00 1969 +--- lame3.50/takehiro.c Sun Oct 24 19:50:53 1999 +*************** +*** 0 **** +--- 1,530 ---- ++ /* ++ * MP3 huffman table selecting and bit counting ++ * ++ * Copyright (c) 1999 Takehiro TOMINAGA ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License as published by ++ * the Free Software Foundation; either version 2, or (at your option) ++ * any later version. ++ * ++ * This program is distributed in the hope that it will be useful, ++ * but WITHOUT ANY WARRANTY; without even the implied warranty of ++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the ++ * GNU General Public License for more details. ++ * ++ * You should have received a copy of the GNU General Public License ++ * along with this program; see the file COPYING. If not, write to ++ * the Free Software Foundation, 675 Mass Ave, Cambridge, MA 02139, USA. ++ */ ++ ++ #include "util.h" ++ #include "l3side.h" ++ #include "huffman.h" ++ #include "quantize.h" ++ ++ ++ extern int *scalefac_band_long; ++ extern int *scalefac_band_short; ++ ++ ++ ++ struct ++ { ++ unsigned region0_count; ++ unsigned region1_count; ++ } subdv_table[ 23 ] = ++ { ++ {0, 0}, /* 0 bands */ ++ {0, 0}, /* 1 bands */ ++ {0, 0}, /* 2 bands */ ++ {0, 0}, /* 3 bands */ ++ {0, 0}, /* 4 bands */ ++ {0, 1}, /* 5 bands */ ++ {1, 1}, /* 6 bands */ ++ {1, 1}, /* 7 bands */ ++ {1, 2}, /* 8 bands */ ++ {2, 2}, /* 9 bands */ ++ {2, 3}, /* 10 bands */ ++ {2, 3}, /* 11 bands */ ++ {3, 4}, /* 12 bands */ ++ {3, 4}, /* 13 bands */ ++ {3, 4}, /* 14 bands */ ++ {4, 5}, /* 15 bands */ ++ {4, 5}, /* 16 bands */ ++ {4, 6}, /* 17 bands */ ++ {5, 6}, /* 18 bands */ ++ {5, 6}, /* 19 bands */ ++ {5, 7}, /* 20 bands */ ++ {6, 7}, /* 21 bands */ ++ {6, 7}, /* 22 bands */ ++ }; ++ ++ ++ /*************************************************************************/ ++ /* ix_max */ ++ /*************************************************************************/ ++ ++ static int ix_max_takehiro(int *ix, int *end) ++ { ++ int max = 0; ++ ++ while (ix < end) { ++ int x = *ix++; ++ if (max < x) ++ max = x; ++ } ++ return max; ++ } ++ ++ ++ /*************************************************************************/ ++ /* count_bit */ ++ /*************************************************************************/ ++ ++ /* ++ Function: Count the number of bits necessary to code the subregion. ++ */ ++ ++ static int count_bit_ESC(int *ix, int *end, unsigned int table) ++ { ++ /* ESC-table is used */ ++ unsigned char *hlen = ht[table].hlen; ++ int linbits = ht[table].linbits; ++ int sum = 0; ++ ++ while (ix < end) { ++ int x = *ix++; ++ int y = *ix++; ++ ++ if (x != 0) { ++ sum++; ++ if (x > 14) { ++ x = 15; ++ sum += linbits; ++ } ++ x *= 16; ++ } ++ ++ if (y != 0) { ++ sum++; ++ if (y > 14) { ++ y = 15; ++ sum += linbits; ++ } ++ x += y; ++ } ++ ++ sum += hlen[x]; ++ } ++ ++ return sum; ++ } ++ ++ ++ ++ static int count_bit_noESC(int *ix, int *end, unsigned int table) ++ { ++ /* No ESC-words */ ++ int sum = 0; ++ unsigned char *hlen = ht[table].hlen; ++ ++ while (ix < end) { ++ int x = *ix++; ++ int y = *ix++; ++ if (x != 0) { ++ sum++; ++ x *= 16; ++ } ++ ++ if (y != 0) { ++ sum++; ++ x += y; ++ } ++ ++ sum += hlen[x]; ++ } ++ ++ return sum; ++ } ++ ++ ++ ++ static int count_bit_short_ESC(int *ix, int *end, unsigned int table) ++ { ++ /* ESC-table is used */ ++ unsigned char *hlen = ht[table].hlen; ++ unsigned int linbits = ht[table].linbits; ++ int sum = 0; ++ ++ while (ix < end) { ++ int i; ++ for (i = 0; i < 3; i++) { ++ int y = *(ix + 3); ++ int x = *ix++; ++ ++ if (x != 0) { ++ sum++; ++ if (x > 14) { ++ x = 15; ++ sum += linbits; ++ } ++ x *= 16; ++ } ++ ++ if (y != 0) { ++ sum++; ++ if (y > 14) { ++ y = 15; ++ sum += linbits; ++ } ++ x += y; ++ } ++ ++ sum += hlen[x]; ++ } ++ ix += 3; ++ } ++ ++ return sum; ++ } ++ ++ ++ ++ static int count_bit_short_noESC(int *ix, int *end, unsigned int table) ++ { ++ /* No ESC-words */ ++ int sum = 0; ++ unsigned char *hlen = ht[table].hlen; ++ ++ while (ix < end) { ++ int i; ++ for (i = 0; i < 3; i++) { ++ int y = *(ix + 3); ++ int x = *ix++; ++ if (x != 0) { ++ sum++; ++ x *= 16; ++ } ++ ++ if (y != 0) { ++ sum++; ++ x += y; ++ } ++ ++ sum += hlen[x]; ++ } ++ ix += 3; ++ } ++ ++ return sum; ++ } ++ ++ ++ ++ /*************************************************************************/ ++ /* new_choose table */ ++ /*************************************************************************/ ++ ++ /* ++ Choose the Huffman table that will encode ix[begin..end] with ++ the fewest bits. ++ ++ Note: This code contains knowledge about the sizes and characteristics ++ of the Huffman tables as defined in the IS (Table B.7), and will not work ++ with any arbitrary tables. ++ */ ++ ++ int choose_table_takehiro(int *ix, int *end, int * s) ++ { ++ int max; ++ int choice0, choice1; ++ int sum0, sum1; ++ ++ max = ix_max_takehiro(ix, end); ++ ++ if (max <= 15) { ++ if (max == 0) { ++ return 0; ++ } ++ /* try tables with no linbits */ ++ for (choice0 = 1; choice0 < 14; choice0++) { ++ if (ht[choice0].xlen > max) { ++ break; ++ } ++ } ++ ++ sum0 = count_bit_noESC(ix, end, choice0); ++ choice1 = choice0; ++ ++ switch (choice0) { ++ case 7: ++ case 10: ++ choice1++; ++ sum1 = count_bit_noESC(ix, end, choice1); ++ if (sum0 > sum1) { ++ sum0 = sum1; ++ choice0 = choice1; ++ } ++ /*fall*/ ++ case 2: ++ case 5: ++ choice1++; ++ sum1 = count_bit_noESC(ix, end, choice1); ++ if (sum0 > sum1) { ++ sum0 = sum1; ++ choice0 = choice1; ++ } ++ break; ++ ++ case 13: ++ choice1 += 2; ++ sum1 = count_bit_noESC(ix, end, choice1); ++ if (sum0 > sum1) { ++ sum0 = sum1; ++ choice0 = choice1; ++ } ++ break; ++ ++ default: ++ break; ++ } ++ } else { ++ /* try tables with linbits */ ++ max -= 15; ++ ++ for (choice0 = 15; choice0 < 24; choice0++) { ++ if (ht[choice0].linmax >= max) { ++ break; ++ } ++ } ++ ++ for (choice1 = 24; choice1 < 32; choice1++) { ++ if (ht[choice1].linmax >= max) { ++ break; ++ } ++ } ++ ++ sum0 = count_bit_ESC(ix, end, choice0); ++ sum1 = count_bit_ESC(ix, end, choice1); ++ ++ if (sum0 > sum1) { ++ sum0 = sum1; ++ choice0 = choice1; ++ } ++ } ++ ++ *s += sum0; ++ return choice0; ++ } ++ ++ ++ int choose_table_takehiro_short(int *ix, int *end, int * s) ++ { ++ int max; ++ int choice0, sum0; ++ #define TAKEHIRO_SHORT ++ #ifdef TAKEHIRO_SHORT ++ int choice1, sum1; ++ #endif ++ ++ max = ix_max_takehiro(ix, end); ++ ++ if (max <= 15) { ++ if (max == 0) { ++ return 0; ++ } ++ /* try tables with no linbits */ ++ for (choice0 = 1; choice0 < 14; choice0++) { ++ if (ht[choice0].xlen > max) { ++ break; ++ } ++ } ++ ++ sum0 = count_bit_short_noESC(ix, end, choice0); ++ ++ #ifdef TAKEHIRO_SHORT ++ choice1 = choice0; ++ ++ switch (choice0) { ++ case 7: ++ case 10: ++ choice1++; ++ sum1 = count_bit_short_noESC(ix, end, choice1); ++ if (sum0 > sum1) { ++ sum0 = sum1; ++ choice0 = choice1; ++ } ++ /*fall*/ ++ case 2: ++ case 5: ++ choice1++; ++ sum1 = count_bit_short_noESC(ix, end, choice1); ++ if (sum0 > sum1) { ++ sum0 = sum1; ++ choice0 = choice1; ++ } ++ break; ++ ++ case 13: ++ choice1 += 2; ++ sum1 = count_bit_short_noESC(ix, end, choice1); ++ if (sum0 > sum1) { ++ sum0 = sum1; ++ choice0 = choice1; ++ } ++ break; ++ ++ default: ++ break; ++ } ++ #endif ++ } else { ++ /* try tables with linbits */ ++ max -= 15; ++ ++ for (choice0 = 15; choice0 < 24; choice0++) { ++ if (ht[choice0].linmax >= max) { ++ break; ++ } ++ } ++ sum0 = count_bit_short_ESC(ix, end, choice0); ++ ++ #ifdef TAKEHIRO_SHORT ++ for (choice1 = 24; choice1 < 32; choice1++) { ++ if (ht[choice1].linmax >= max) { ++ break; ++ } ++ } ++ sum1 = count_bit_short_ESC(ix, end, choice1); ++ ++ if (sum0 > sum1) { ++ sum0 = sum1; ++ choice0 = choice1; ++ } ++ #endif ++ } ++ ++ *s += sum0; ++ return choice0; ++ } ++ ++ ++ static int count_bits_long(int ix[576], gr_info *gi) ++ { ++ int i, a1, a2; ++ int bits = 0; ++ ++ i=576; ++ for (; i > 1; i -= 2) ++ if (ix[i - 1] | ix[i - 2]) ++ break; ++ ++ /* Determines the number of bits to encode the quadruples. */ ++ gi->count1 = i; ++ { ++ int sum0 = 0, sum1 = 0; ++ for (; i > 3; i -= 4) { ++ int p, v; ++ if ((unsigned int)(ix[i-1] | ix[i-2] | ix[i-3] | ix[i-4]) > 1) ++ break; ++ ++ v = ix[i-1]; ++ p = v; ++ bits += v; ++ ++ v = ix[i-2]; ++ if (v != 0) { ++ p += 2; ++ bits++; ++ } ++ ++ v = ix[i-3]; ++ if (v != 0) { ++ p += 4; ++ bits++; ++ } ++ ++ v = ix[i-4]; ++ if (v != 0) { ++ p += 8; ++ bits++; ++ } ++ ++ sum0 += ht[32].hlen[p]; ++ sum1 += ht[33].hlen[p]; ++ } ++ ++ if (sum0 < sum1) { ++ bits += sum0; ++ gi->count1table_select = 0; ++ } else { ++ bits += sum1; ++ gi->count1table_select = 1; ++ } ++ } ++ ++ gi->big_values = i; ++ if (i == 0) ++ return bits; ++ ++ if (gi->block_type == 0) { ++ int index; ++ int scfb_anz = 0; ++ ++ while (scalefac_band_long[++scfb_anz] < i) ++ ; ++ ++ index = subdv_table[scfb_anz].region0_count; ++ while (scalefac_band_long[index + 1] > i) ++ index--; ++ gi->region0_count = index; ++ ++ index = subdv_table[scfb_anz].region1_count; ++ while (scalefac_band_long[index + gi->region0_count + 2] > i) ++ index--; ++ gi->region1_count = index; ++ ++ a1 = scalefac_band_long[gi->region0_count + 1]; ++ a2 = scalefac_band_long[index + gi->region0_count + 2]; ++ gi->table_select[2] = choose_table_takehiro(ix + a2, ix + i, &bits); ++ } else { ++ gi->region0_count = 7; ++ //gi->region1_count = SBMAX_l - 7 - 1; ++ gi->region1_count = SFB_LMAX -1 - 7 - 1; ++ a1 = scalefac_band_long[7 + 1]; ++ a2 = i; ++ if (a1 > a2) { ++ a1 = a2; ++ } ++ } ++ ++ /* Count the number of bits necessary to code the bigvalues region. */ ++ gi->table_select[0] = choose_table_takehiro(ix, ix + a1, &bits); ++ gi->table_select[1] = choose_table_takehiro(ix + a1, ix + a2, &bits); ++ return bits; ++ } ++ ++ ++ ++ ++ int count_bits(int *ix,gr_info *cod_info) ++ { ++ int bits=0,i; ++ for ( i = 0; i < 576; i++ ) { ++ if ( ix[i] > 8191 + 14) return 100000;/* report unsuitable quantizer */ ++ } ++ ++ ++ if (cod_info->block_type==2) { ++ cod_info->table_select[0] = choose_table_takehiro_short(ix, ix + 36, &bits); ++ cod_info->table_select[1] = choose_table_takehiro_short(ix + 36, ix + 576, &bits); ++ }else{ ++ bits=count_bits_long(ix, cod_info); ++ cod_info->count1 = (cod_info->count1 - cod_info->big_values) / 4; ++ cod_info->big_values /= 2; ++ } ++ return bits; ++ ++ } ++ +diff -r -c -N encoder/timestatus.c lame3.50/timestatus.c +*** encoder/timestatus.c Wed Dec 31 17:00:00 1969 +--- lame3.50/timestatus.c Thu Oct 28 22:52:55 1999 +*************** +*** 0 **** +--- 1,150 ---- ++ #include "timestatus.h" ++ #include "util.h" ++ #include <time.h> ++ ++ ++ #if defined(CLOCKS_PER_SEC) ++ /* ANSI/ISO systems */ ++ # define TS_CLOCKS_PER_SEC CLOCKS_PER_SEC ++ #elif defined(CLK_TCK) ++ /* Non-standard systems */ ++ # define TS_CLOCKS_PER_SEC CLK_TCK ++ #elif defined(HZ) ++ /* Older BSD systems */ ++ # define TS_CLOCKS_PER_SEC HZ ++ #else ++ # error no suitable value for TS_CLOCKS_PER_SEC ++ #endif ++ ++ /*********************************************************/ ++ /* ts_real_time: real time elapsed in seconds */ ++ /*********************************************************/ ++ double ts_real_time(long frame) { ++ ++ static time_t initial_time; ++ time_t current_time; ++ ++ time(¤t_time); ++ ++ if (frame==0) { ++ initial_time = current_time; ++ } ++ ++ return difftime(current_time, initial_time); ++ } ++ ++ /*********************************************************/ ++ /* ts_process_time: process time elapsed in seconds */ ++ /*********************************************************/ ++ double ts_process_time(long frame) { ++ static clock_t initial_time; ++ clock_t current_time; ++ ++ #if ( defined(_MSC_VER) || defined(__BORLANDC__) ) ++ ++ { static HANDLE hProcess; ++ FILETIME Ignored1, Ignored2, KernelTime, UserTime; ++ ++ if ( frame==0 ) { ++ hProcess = GetCurrentProcess(); ++ } ++ ++ /* GetProcessTimes() always fails under Win9x */ ++ if (GetProcessTimes(hProcess, &Ignored1, &Ignored2, &KernelTime, &UserTime)) { ++ LARGE_INTEGER Kernel = { KernelTime.dwLowDateTime, KernelTime.dwHighDateTime }; ++ LARGE_INTEGER User = { UserTime.dwLowDateTime, UserTime.dwHighDateTime }; ++ ++ current_time = (clock_t)((double)(Kernel.QuadPart + User.QuadPart) * TS_CLOCKS_PER_SEC / 10000000); ++ } else { ++ current_time = clock(); ++ } ++ } ++ #else ++ current_time = clock(); ++ #endif ++ ++ if (frame==0) { ++ initial_time = current_time; ++ } ++ ++ return (double)(current_time - initial_time) / TS_CLOCKS_PER_SEC; ++ } ++ ++ #undef TS_CLOCKS_PER_SEC ++ ++ typedef struct ts_times { ++ double so_far; ++ double estimated; ++ double speed; ++ double eta; ++ } ts_times; ++ ++ /*********************************************************/ ++ /* ts_calc_times: calculate time info (eta, speed, etc.) */ ++ /*********************************************************/ ++ void ts_calc_times(ts_times *time, int samp_rate, long frame, long frames, ++ int framesize) ++ { ++ if (frame > 0) { ++ time->estimated = time->so_far * frames / frame; ++ if (samp_rate * time->estimated > 0) { ++ time->speed = frames * framesize / (1000.0 * samp_rate * time->estimated); ++ } else { ++ time->speed = 0; ++ } ++ time->eta = time->estimated - time->so_far; ++ } else { ++ time->estimated = 0; ++ time->speed = 0; ++ time->eta = 0; ++ } ++ } ++ ++ /*********************************************************/ ++ /* timestatus: display encoding process time information */ ++ /*********************************************************/ ++ void timestatus(layer *info,long frameNum,long totalframes) ++ { ++ const int samp_rate = (int)s_freq[info->version][info->sampling_frequency]; ++ ts_times real_time, process_time; ++ int percent; ++ int framesize = (info->version==0) ? 576 : 1152; ++ ++ real_time.so_far = ts_real_time(frameNum); ++ process_time.so_far = ts_process_time(frameNum); ++ ++ if (frameNum == 0) { ++ fprintf(stderr, " Frame | CPU/estimated | time/estimated | play/CPU | ETA\n"); ++ return; ++ } ++ ++ ts_calc_times(&real_time, samp_rate, frameNum, totalframes,framesize); ++ ts_calc_times(&process_time, samp_rate, frameNum, totalframes,framesize); ++ ++ if (totalframes > 1) { ++ percent = (int)(100.0 * frameNum / (totalframes - 1)); ++ } else { ++ percent = 100; ++ } ++ ++ #define TS_TIME_DECOMPOSE(time) \ ++ (int)((long)(time+.5) / 3600), \ ++ (int)((long)((time+.5) / 60) % 60), \ ++ (int)((long)(time+.5) % 60) ++ ++ fprintf(stderr, ++ "\r%6ld/%6ld(%3d%%)|%2d:%02d:%02d/%2d:%02d:%02d|%2d:%02d:%02d/%2d:%02d:%02d|%10.4f|%2d:%02d:%02d ", ++ frameNum, ++ totalframes - 1, ++ percent, ++ TS_TIME_DECOMPOSE(process_time.so_far), ++ TS_TIME_DECOMPOSE(process_time.estimated), ++ TS_TIME_DECOMPOSE(real_time.so_far), ++ TS_TIME_DECOMPOSE(real_time.estimated), ++ process_time.speed, ++ TS_TIME_DECOMPOSE(real_time.eta) ++ ); ++ ++ fflush(stderr); ++ } ++ +diff -r -c -N encoder/timestatus.h lame3.50/timestatus.h +*** encoder/timestatus.h Wed Dec 31 17:00:00 1969 +--- lame3.50/timestatus.h Wed Oct 20 22:58:21 1999 +*************** +*** 0 **** +--- 1,9 ---- ++ #ifndef TIMESTATUS_H_INCLUDED ++ #define TIMESTATUS_H_INCLUDED ++ ++ #include "util.h" ++ ++ void timestatus(layer *info,long frameNum,long totalframes); ++ ++ ++ #endif +diff -r -c -N encoder/tonal.c lame3.50/tonal.c +*** encoder/tonal.c Wed Jan 22 02:43:23 1997 +--- lame3.50/tonal.c Wed Dec 31 17:00:00 1969 +*************** +*** 1,1051 **** +- /********************************************************************** +- * ISO MPEG Audio Subgroup Software Simulation Group (1996) +- * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- * +- * $Id: tonal.c,v 1.1 1996/02/14 04:04:23 rowlands Exp $ +- * +- * $Log: tonal.c,v $ +- * Revision 1.1 1996/02/14 04:04:23 rowlands +- * Initial revision +- * +- * Received from Mike Coleman +- **********************************************************************/ +- /********************************************************************** +- * date programmers comment * +- * 2/25/91 Douglas Wong start of version 1.1 records * +- * 3/06/91 Douglas Wong rename: setup.h to endef.h * +- * updated I_psycho_one and II_psycho_one* +- * 3/11/91 W. J. Carter Added Douglas Wong's updates dated * +- * 3/9/91 for I_Psycho_One() and for * +- * II_Psycho_One(). * +- * 5/10/91 W. Joseph Carter Ported to Macintosh and Unix. * +- * Located and fixed numerous software * +- * bugs and table data errors. * +- * 6/11/91 Davis Pan corrected several bugs * +- * based on comments from H. Fuchs * +- * 01jul91 dpwe (Aware Inc.) Made pow() args float * +- * Removed logical bug in I_tonal_label: * +- * Sometimes *tone returned == STOP * +- * 7/10/91 Earle Jennings no change necessary in port to MsDos * +- * 11sep91 dpwe@aware.com Subtracted 90.3dB from II_f_f_t peaks * +- * 10/1/91 Peter W. Farrett Updated II_Psycho_One(),I_Psycho_One()* +- * to include comments. * +- *11/29/91 Masahiro Iwadare Bug fix regarding POWERNORM * +- * fixed several other miscellaneous bugs* +- * 2/11/92 W. Joseph Carter Ported new code to Macintosh. Most * +- * important fixes involved changing * +- * 16-bit ints to long or unsigned in * +- * bit alloc routines for quant of 65535 * +- * and passing proper function args. * +- * Removed "Other Joint Stereo" option * +- * and made bitrate be total channel * +- * bitrate, irrespective of the mode. * +- * Fixed many small bugs & reorganized. * +- * 2/12/92 Masahiro Iwadare Fixed some potential bugs in * +- * Davis Pan subsampling() * +- * 2/25/92 Masahiro Iwadare Fixed some more potential bugs * +- * 6/24/92 Tan Ah Peng Modified window for FFT * +- * (denominator N-1 to N) * +- * Updated all critical band rate & * +- * absolute threshold tables and critical* +- * boundaries for use with Layer I & II * +- * Corrected boundary limits for tonal * +- * component computation * +- * Placement of non-tonal component at * +- * geometric mean of critical band * +- * (previous placement method commented * +- * out - can be used if desired) * +- * 3/01/93 Mike Li Infinite looping fix in noise_label() * +- * 3/19/93 Jens Spille fixed integer overflow problem in * +- * psychoacoutic model 1 * +- * 3/19/93 Giorgio Dimino modifications to better account for * +- * tonal and non-tonal components * +- * 5/28/93 Sriram Jayasimha "London" mod. to psychoacoustic model1* +- * 8/05/93 Masahiro Iwadare noise_label modification "option" * +- * 1/21/94 Seymore Shlien fixed another infinite looping problem* +- * 7/12/95 Soeren H. Nielsen Changes for LSF, new tables * +- **********************************************************************/ +- +- #include "common.h" +- #include "encoder.h" +- #define LONDON /* enable "LONDON" modification */ +- #define MAKE_SENSE /* enable "MAKE_SENSE" modification */ +- #define MI_OPTION /* enable "MI_OPTION" modification */ +- /********************************************************************** +- * +- * This module implements the psychoacoustic model I for the +- * MPEG encoder layer II. It uses simplified tonal and noise masking +- * threshold analysis to generate SMR for the encoder bit allocation +- * routine. +- * +- **********************************************************************/ +- +- int crit_band; +- int FAR *cbound; +- int sub_size; +- +- void read_cbound(lay,freq) /* this function reads in critical */ +- int lay, freq; /* band boundaries */ +- { +- int i,j,k; +- FILE *fp; +- char r[16], t[80]; +- +- strcpy(r, "2cb1"); +- r[0] = (char) lay + '0'; +- r[3] = (char) freq + '0'; +- if( !(fp = OpenTableFile(r)) ){ /* check boundary values */ +- printf("Please check %s boundary table\n",r); +- exit(1); +- } +- fgets(t,80,fp); /* read input for critical bands */ +- sscanf(t,"%d\n",&crit_band); +- cbound = (int FAR *) mem_alloc(sizeof(int) * crit_band, "cbound"); +- for(i=0;i<crit_band;i++){ /* continue to read input for */ +- fgets(t,80,fp); /* critical band boundaries */ +- sscanf(t,"%d %d\n",&j, &k); +- if(i==j) cbound[j] = k; +- else { /* error */ +- printf("Please check index %d in cbound table %s\n",i,r); +- exit(1); +- } +- } +- fclose(fp); +- } +- +- void read_freq_band(ltg,lay,freq) /* this function reads in */ +- int lay, freq; /* frequency bands and bark */ +- g_ptr FAR *ltg; /* values */ +- { +- int i,j, k; +- double b,c; +- FILE *fp; +- char r[16], t[80]; +- +- strcpy(r, "2th1"); +- r[0] = (char) lay + '0'; +- r[3] = (char) freq + '0'; +- if( !(fp = OpenTableFile(r)) ){ /* check freq. values */ +- printf("Please check frequency and cband table %s\n",r); +- exit(1); +- } +- fgets(t,80,fp); /* read input for freq. subbands */ +- sscanf(t,"%d\n",&sub_size); +- *ltg = (g_ptr FAR ) mem_alloc(sizeof(g_thres) * sub_size, "ltg"); +- (*ltg)[0].line = 0; /* initialize global masking threshold */ +- (*ltg)[0].bark = 0; +- (*ltg)[0].hear = 0; +- for(i=1;i<sub_size;i++){ /* continue to read freq. subband */ +- fgets(t,80,fp); /* and assign */ +- sscanf(t,"%d %d %lf %lf\n",&j, &k, &b, &c); +- if(i == j){ +- (*ltg)[j].line = k; +- (*ltg)[j].bark = b; +- (*ltg)[j].hear = c; +- } +- else { /* error */ +- printf("Please check index %d in freq-cb table %s\n",i,r); +- exit(1); +- } +- } +- fclose(fp); +- } +- +- void make_map(power, ltg) /* this function calculates the */ +- mask FAR power[HAN_SIZE]; /* global masking threshold */ +- g_thres FAR *ltg; +- { +- int i,j; +- +- for(i=1;i<sub_size;i++) for(j=ltg[i-1].line;j<=ltg[i].line;j++) +- power[j].map = i; +- } +- +- double add_db(a,b) +- double a,b; +- { +- a = pow(10.0,a/10.0); +- b = pow(10.0,b/10.0); +- return 10 * log10(a+b); +- } +- +- /**************************************************************** +- * +- * Fast Fourier transform of the input samples. +- * +- ****************************************************************/ +- +- void II_f_f_t(sample, power) /* this function calculates an */ +- double FAR sample[FFT_SIZE]; /* FFT analysis for the freq. */ +- mask FAR power[HAN_SIZE]; /* domain */ +- { +- int i,j,k,L,l=0; +- int ip, le, le1; +- double t_r, t_i, u_r, u_i; +- static int M, MM1, init = 0, N; +- double *x_r, *x_i, *energy; +- static int *rev; +- static double *w_r, *w_i; +- +- x_r = (double *) mem_alloc(sizeof(DFFT), "x_r"); +- x_i = (double *) mem_alloc(sizeof(DFFT), "x_i"); +- energy = (double *) mem_alloc(sizeof(DFFT), "energy"); +- for(i=0;i<FFT_SIZE;i++) x_r[i] = x_i[i] = energy[i] = 0; +- if(!init){ +- rev = (int *) mem_alloc(sizeof(IFFT), "rev"); +- w_r = (double *) mem_alloc(sizeof(D10), "w_r"); +- w_i = (double *) mem_alloc(sizeof(D10), "w_i"); +- M = 10; +- MM1 = 9; +- N = FFT_SIZE; +- for(L=0;L<M;L++){ +- le = 1 << (M-L); +- le1 = le >> 1; +- w_r[L] = cos(PI/le1); +- w_i[L] = -sin(PI/le1); +- } +- for(i=0;i<FFT_SIZE;rev[i] = l,i++) for(j=0,l=0;j<10;j++){ +- k=(i>>j) & 1; +- l |= (k<<(9-j)); +- } +- init = 1; +- } +- memcpy( (char *) x_r, (char *) sample, sizeof(double) * FFT_SIZE); +- for(L=0;L<MM1;L++){ +- le = 1 << (M-L); +- le1 = le >> 1; +- u_r = 1; +- u_i = 0; +- for(j=0;j<le1;j++){ +- for(i=j;i<N;i+=le){ +- ip = i + le1; +- t_r = x_r[i] + x_r[ip]; +- t_i = x_i[i] + x_i[ip]; +- x_r[ip] = x_r[i] - x_r[ip]; +- x_i[ip] = x_i[i] - x_i[ip]; +- x_r[i] = t_r; +- x_i[i] = t_i; +- t_r = x_r[ip]; +- x_r[ip] = x_r[ip] * u_r - x_i[ip] * u_i; +- x_i[ip] = x_i[ip] * u_r + t_r * u_i; +- } +- t_r = u_r; +- u_r = u_r * w_r[L] - u_i * w_i[L]; +- u_i = u_i * w_r[L] + t_r * w_i[L]; +- } +- } +- for(i=0;i<N;i+=2){ +- ip = i + 1; +- t_r = x_r[i] + x_r[ip]; +- t_i = x_i[i] + x_i[ip]; +- x_r[ip] = x_r[i] - x_r[ip]; +- x_i[ip] = x_i[i] - x_i[ip]; +- x_r[i] = t_r; +- x_i[i] = t_i; +- energy[i] = x_r[i] * x_r[i] + x_i[i] * x_i[i]; +- } +- for(i=0;i<FFT_SIZE;i++) if(i<rev[i]){ +- t_r = energy[i]; +- energy[i] = energy[rev[i]]; +- energy[rev[i]] = t_r; +- } +- for(i=0;i<HAN_SIZE;i++){ /* calculate power density spectrum */ +- if (energy[i] < 1E-20) energy[i] = 1E-20; +- power[i].x = 10 * log10(energy[i]) + POWERNORM; +- power[i].next = STOP; +- power[i].type = FALSE; +- } +- mem_free((void **) &x_r); +- mem_free((void **) &x_i); +- mem_free((void **) &energy); +- } +- +- /**************************************************************** +- * +- * Window the incoming audio signal. +- * +- ****************************************************************/ +- +- void II_hann_win(sample) /* this function calculates a */ +- double FAR sample[FFT_SIZE]; /* Hann window for PCM (input) */ +- { /* samples for a 1024-pt. FFT */ +- register int i; +- register double sqrt_8_over_3; +- static int init = 0; +- static double FAR *window; +- +- if(!init){ /* calculate window function for the Fourier transform */ +- window = (double FAR *) mem_alloc(sizeof(DFFT), "window"); +- sqrt_8_over_3 = pow(8.0/3.0, 0.5); +- for(i=0;i<FFT_SIZE;i++){ +- /* Hann window formula */ +- window[i]=sqrt_8_over_3*0.5*(1-cos(2.0*PI*i/(FFT_SIZE)))/FFT_SIZE; +- } +- init = 1; +- } +- for(i=0;i<FFT_SIZE;i++) sample[i] *= window[i]; +- } +- +- /******************************************************************* +- * +- * This function finds the maximum spectral component in each +- * subband and return them to the encoder for time-domain threshold +- * determination. +- * +- *******************************************************************/ +- #ifndef LONDON +- void II_pick_max(power, spike) +- double FAR spike[SBLIMIT]; +- mask FAR power[HAN_SIZE]; +- { +- double max; +- int i,j; +- +- for(i=0;i<HAN_SIZE;spike[i>>4] = max, i+=16) /* calculate the */ +- for(j=0, max = DBMIN;j<16;j++) /* maximum spectral */ +- max = (max>power[i+j].x) ? max : power[i+j].x; /* component in each */ +- } /* subband from bound */ +- /* 4-16 */ +- #else +- void II_pick_max(power, spike) +- double FAR spike[SBLIMIT]; +- mask FAR power[HAN_SIZE]; +- { +- double sum; +- int i,j; +- +- for(i=0;i<HAN_SIZE;spike[i>>4] = 10.0*log10(sum), i+=16) +- /* calculate the */ +- for(j=0, sum = pow(10.0,0.1*DBMIN);j<16;j++) /* sum of spectral */ +- sum += pow(10.0,0.1*power[i+j].x); /* component in each */ +- } /* subband from bound */ +- /* 4-16 */ +- #endif +- +- /**************************************************************** +- * +- * This function labels the tonal component in the power +- * spectrum. +- * +- ****************************************************************/ +- +- void II_tonal_label(power, tone) /* this function extracts (tonal) */ +- mask FAR power[HAN_SIZE]; /* sinusoidals from the spectrum */ +- int *tone; +- { +- int i,j, last = LAST, first, run, last_but_one = LAST; /* dpwe */ +- double max; +- +- *tone = LAST; +- for(i=2;i<HAN_SIZE-12;i++){ +- if(power[i].x>power[i-1].x && power[i].x>=power[i+1].x){ +- power[i].type = TONE; +- power[i].next = LAST; +- if(last != LAST) power[last].next = i; +- else first = *tone = i; +- last = i; +- } +- } +- last = LAST; +- first = *tone; +- *tone = LAST; +- while(first != LAST){ /* the conditions for the tonal */ +- if(first<3 || first>500) run = 0;/* otherwise k+/-j will be out of bounds */ +- else if(first<63) run = 2; /* components in layer II, which */ +- else if(first<127) run = 3; /* are the boundaries for calc. */ +- else if(first<255) run = 6; /* the tonal components */ +- else run = 12; +- max = power[first].x - 7; /* after calculation of tonal */ +- for(j=2;j<=run;j++) /* components, set to local max */ +- if(max < power[first-j].x || max < power[first+j].x){ +- power[first].type = FALSE; +- break; +- } +- if(power[first].type == TONE){ /* extract tonal components */ +- int help=first; +- if(*tone==LAST) *tone = first; +- while((power[help].next!=LAST)&&(power[help].next-first)<=run) +- help=power[help].next; +- help=power[help].next; +- power[first].next=help; +- if((first-last)<=run){ +- if(last_but_one != LAST) power[last_but_one].next=first; +- } +- if(first>1 && first<500){ /* calculate the sum of the */ +- double tmp; /* powers of the components */ +- tmp = add_db(power[first-1].x, power[first+1].x); +- power[first].x = add_db(power[first].x, tmp); +- } +- for(j=1;j<=run;j++){ +- power[first-j].x = power[first+j].x = DBMIN; +- power[first-j].next = power[first+j].next = STOP; +- power[first-j].type = power[first+j].type = FALSE; +- } +- last_but_one=last; +- last = first; +- first = power[first].next; +- } +- else { +- int ll; +- if(last == LAST); /* *tone = power[first].next; dpwe */ +- else power[last].next = power[first].next; +- ll = first; +- first = power[first].next; +- power[ll].next = STOP; +- } +- } +- } +- +- /**************************************************************** +- * +- * This function groups all the remaining non-tonal +- * spectral lines into critical band where they are replaced by +- * one single line. +- * +- ****************************************************************/ +- +- void noise_label(power, noise, ltg) +- g_thres FAR *ltg; +- mask FAR *power; +- int *noise; +- { +- int i,j, centre, last = LAST; +- double index, weight, sum; +- /* calculate the remaining spectral */ +- for(i=0;i<crit_band-1;i++){ /* lines for non-tonal components */ +- for(j=cbound[i],weight = 0.0,sum = DBMIN;j<cbound[i+1];j++){ +- if(power[j].type != TONE){ +- if(power[j].x != DBMIN){ +- sum = add_db(power[j].x,sum); +- /* the line below and others under the "MAKE_SENSE" condition are an alternate +- interpretation of "geometric mean". This approach may make more sense but +- it has not been tested with hardware. */ +- #ifdef MAKE_SENSE +- /* weight += pow(10.0, power[j].x/10.0) * (ltg[power[j].map].bark-i); +- bad code [SS] 21-1-93 +- */ +- weight += pow(10.0,power[j].x/10.0) * (double) (j-cbound[i]) / +- (double) (cbound[i+1]-cbound[i]); /* correction */ +- #endif +- power[j].x = DBMIN; +- } +- } /* check to see if the spectral line is low dB, and if */ +- } /* so replace the center of the critical band, which is */ +- /* the center freq. of the noise component */ +- +- #ifdef MAKE_SENSE +- if(sum <= DBMIN) centre = (cbound[i+1]+cbound[i]) /2; +- else { +- index = weight/pow(10.0,sum/10.0); +- centre = cbound[i] + (int) (index * (double) (cbound[i+1]-cbound[i]) ); +- } +- #else +- index = (double)( ((double)cbound[i]) * ((double)(cbound[i+1]-1)) ); +- centre = (int)(pow(index,0.5)+0.5); +- #endif +- +- /* locate next non-tonal component until finished; */ +- /* add to list of non-tonal components */ +- #ifdef MI_OPTION +- /* Masahiro Iwadare's fix for infinite looping problem? */ +- if(power[centre].type == TONE) +- if (power[centre+1].type == TONE) centre++; else centre--; +- #else +- /* Mike Li's fix for infinite looping problem */ +- if(power[centre].type == FALSE) centre++; +- +- if(power[centre].type == NOISE){ +- if(power[centre].x >= ltg[power[i].map].hear){ +- if(sum >= ltg[power[i].map].hear) sum = add_db(power[j].x,sum); +- else +- sum = power[centre].x; +- } +- } +- #endif +- if(last == LAST) *noise = centre; +- else { +- power[centre].next = LAST; +- power[last].next = centre; +- } +- power[centre].x = sum; +- power[centre].type = NOISE; +- last = centre; +- } +- } +- +- /**************************************************************** +- * +- * This function reduces the number of noise and tonal +- * component for further threshold analysis. +- * +- ****************************************************************/ +- +- void subsampling(power, ltg, tone, noise) +- mask FAR power[HAN_SIZE]; +- g_thres FAR *ltg; +- int *tone, *noise; +- { +- int i, old; +- +- i = *tone; old = STOP; /* calculate tonal components for */ +- while(i!=LAST){ /* reduction of spectral lines */ +- if(power[i].x < ltg[power[i].map].hear){ +- power[i].type = FALSE; +- power[i].x = DBMIN; +- if(old == STOP) *tone = power[i].next; +- else power[old].next = power[i].next; +- } +- else old = i; +- i = power[i].next; +- } +- i = *noise; old = STOP; /* calculate non-tonal components for */ +- while(i!=LAST){ /* reduction of spectral lines */ +- if(power[i].x < ltg[power[i].map].hear){ +- power[i].type = FALSE; +- power[i].x = DBMIN; +- if(old == STOP) *noise = power[i].next; +- else power[old].next = power[i].next; +- } +- else old = i; +- i = power[i].next; +- } +- i = *tone; old = STOP; +- while(i != LAST){ /* if more than one */ +- if(power[i].next == LAST)break; /* tonal component */ +- if(ltg[power[power[i].next].map].bark - /* is less than .5 */ +- ltg[power[i].map].bark < 0.5) { /* bark, take the */ +- if(power[power[i].next].x > power[i].x ){/* maximum */ +- if(old == STOP) *tone = power[i].next; +- else power[old].next = power[i].next; +- power[i].type = FALSE; +- power[i].x = DBMIN; +- i = power[i].next; +- } +- else { +- power[power[i].next].type = FALSE; +- power[power[i].next].x = DBMIN; +- power[i].next = power[power[i].next].next; +- old = i; +- } +- } +- else { +- old = i; +- i = power[i].next; +- } +- } +- } +- +- /**************************************************************** +- * +- * This function calculates the individual threshold and +- * sum with the quiet threshold to find the global threshold. +- * +- ****************************************************************/ +- +- void threshold(power, ltg, tone, noise, bit_rate) +- mask FAR power[HAN_SIZE]; +- g_thres FAR *ltg; +- int *tone, *noise, bit_rate; +- { +- int k, t; +- double dz, tmps, vf; +- +- for(k=1;k<sub_size;k++){ +- ltg[k].x = DBMIN; +- t = *tone; /* calculate individual masking threshold for */ +- while(t != LAST){ /* components in order to find the global */ +- if(ltg[k].bark-ltg[power[t].map].bark >= -3.0 && /*threshold (LTG)*/ +- ltg[k].bark-ltg[power[t].map].bark <8.0){ +- dz = ltg[k].bark-ltg[power[t].map].bark; /* distance of bark value*/ +- tmps = -1.525-0.275*ltg[power[t].map].bark - 4.5 + power[t].x; +- /* masking function for lower & upper slopes */ +- if(-3<=dz && dz<-1) vf = 17*(dz+1)-(0.4*power[t].x +6); +- else if(-1<=dz && dz<0) vf = (0.4 *power[t].x + 6) * dz; +- else if(0<=dz && dz<1) vf = (-17*dz); +- else if(1<=dz && dz<8) vf = -(dz-1) * (17-0.15 *power[t].x) - 17; +- tmps += vf; +- ltg[k].x = add_db(ltg[k].x, tmps); +- } +- t = power[t].next; +- } +- +- t = *noise; /* calculate individual masking threshold */ +- while(t != LAST){ /* for non-tonal components to find LTG */ +- if(ltg[k].bark-ltg[power[t].map].bark >= -3.0 && +- ltg[k].bark-ltg[power[t].map].bark <8.0){ +- dz = ltg[k].bark-ltg[power[t].map].bark; /* distance of bark value */ +- tmps = -1.525-0.175*ltg[power[t].map].bark -0.5 + power[t].x; +- /* masking function for lower & upper slopes */ +- if(-3<=dz && dz<-1) vf = 17*(dz+1)-(0.4*power[t].x +6); +- else if(-1<=dz && dz<0) vf = (0.4 *power[t].x + 6) * dz; +- else if(0<=dz && dz<1) vf = (-17*dz); +- else if(1<=dz && dz<8) vf = -(dz-1) * (17-0.15 *power[t].x) - 17; +- tmps += vf; +- ltg[k].x = add_db(ltg[k].x, tmps); +- } +- t = power[t].next; +- } +- if(bit_rate<96)ltg[k].x = add_db(ltg[k].hear, ltg[k].x); +- else ltg[k].x = add_db(ltg[k].hear-12.0, ltg[k].x); +- } +- } +- +- /**************************************************************** +- * +- * This function finds the minimum masking threshold and +- * return the value to the encoder. +- * +- ****************************************************************/ +- +- void II_minimum_mask(ltg,ltmin,sblimit) +- g_thres FAR *ltg; +- double FAR ltmin[SBLIMIT]; +- int sblimit; +- { +- double min; +- int i,j; +- +- j=1; +- for(i=0;i<sblimit;i++) +- if(j>=sub_size-1) /* check subband limit, and */ +- ltmin[i] = ltg[sub_size-1].hear; /* calculate the minimum masking */ +- else { /* level of LTMIN for each subband*/ +- min = ltg[j].x; +- while(ltg[j].line>>4 == i && j < sub_size){ +- if(min>ltg[j].x) min = ltg[j].x; +- j++; +- } +- ltmin[i] = min; +- } +- } +- +- /***************************************************************** +- * +- * This procedure is called in musicin to pick out the +- * smaller of the scalefactor or threshold. +- * +- *****************************************************************/ +- +- void II_smr(ltmin, spike, scale, sblimit) +- double FAR spike[SBLIMIT], scale[SBLIMIT], ltmin[SBLIMIT]; +- int sblimit; +- { +- int i; +- double max; +- +- for(i=0;i<sblimit;i++){ /* determine the signal */ +- max = 20 * log10(scale[i] * 32768) - 10; /* level for each subband */ +- if(spike[i]>max) max = spike[i]; /* for the maximum scale */ +- max -= ltmin[i]; /* factors */ +- ltmin[i] = max; +- } +- } +- +- /**************************************************************** +- * +- * This procedure calls all the necessary functions to +- * complete the psychoacoustic analysis. +- * +- ****************************************************************/ +- +- void II_Psycho_One(buffer, scale, ltmin, fr_ps) +- short FAR buffer[2][1152]; +- double FAR scale[2][SBLIMIT], ltmin[2][SBLIMIT]; +- frame_params *fr_ps; +- { +- layer *info = fr_ps->header; +- int stereo = fr_ps->stereo; +- int sblimit = fr_ps->sblimit; +- int k,i, tone=0, noise=0; +- static char init = 0; +- static int off[2] = {256,256}; +- double *sample; +- DSBL *spike; +- static D1408 *fft_buf; +- static mask_ptr FAR power; +- static g_ptr FAR ltg; +- +- sample = (double *) mem_alloc(sizeof(DFFT), "sample"); +- spike = (DSBL *) mem_alloc(sizeof(D2SBL), "spike"); +- /* call functions for critical boundaries, freq. */ +- if(!init){ /* bands, bark values, and mapping */ +- fft_buf = (D1408 *) mem_alloc((long) sizeof(D1408) * 2, "fft_buf"); +- power = (mask_ptr FAR ) mem_alloc(sizeof(mask) * HAN_SIZE, "power"); +- if (info->version == MPEG_AUDIO_ID) { +- read_cbound(info->lay, info->sampling_frequency); +- read_freq_band(<g, info->lay, info->sampling_frequency); +- } else { +- read_cbound(info->lay, info->sampling_frequency + 4); +- read_freq_band(<g, info->lay, info->sampling_frequency + 4); +- } +- make_map(power,ltg); +- for (i=0;i<1408;i++) fft_buf[0][i] = fft_buf[1][i] = 0; +- init = 1; +- } +- for(k=0;k<stereo;k++){ /* check pcm input for 3 blocks of 384 samples */ +- for(i=0;i<1152;i++) fft_buf[k][(i+off[k])%1408]= (double)buffer[k][i]/SCALE; +- for(i=0;i<FFT_SIZE;i++) sample[i] = fft_buf[k][(i+1216+off[k])%1408]; +- off[k] += 1152; +- off[k] %= 1408; +- /* call functions for windowing PCM samples,*/ +- II_hann_win(sample); /* location of spectral components in each */ +- for(i=0;i<HAN_SIZE;i++) power[i].x = DBMIN; /*subband with labeling*/ +- II_f_f_t(sample, power); /*locate remaining non-*/ +- II_pick_max(power, &spike[k][0]); /*tonal sinusoidals, */ +- II_tonal_label(power, &tone); /*reduce noise & tonal */ +- noise_label(power, &noise, ltg); /*components, find */ +- subsampling(power, ltg, &tone, &noise); /*global & minimal */ +- threshold(power, ltg, &tone, &noise, /*threshold, and sgnl- */ +- bitrate[info->version][info->lay-1][info->bitrate_index]/stereo); /*to-mask ratio*/ +- II_minimum_mask(ltg, <min[k][0], sblimit); +- II_smr(<min[k][0], &spike[k][0], &scale[k][0], sblimit); +- } +- mem_free((void **) &sample); +- mem_free((void **) &spike); +- } +- +- /********************************************************************** +- * +- * This module implements the psychoacoustic model I for the +- * MPEG encoder layer I. It uses simplified tonal and noise masking +- * threshold analysis to generate SMR for the encoder bit allocation +- * routine. +- * +- **********************************************************************/ +- +- /**************************************************************** +- * +- * Fast Fourier transform of the input samples. +- * +- ****************************************************************/ +- +- void I_f_f_t(sample, power) /* this function calculates */ +- double FAR sample[FFT_SIZE/2]; /* an FFT analysis for the */ +- mask FAR power[HAN_SIZE/2]; /* freq. domain */ +- { +- int i,j,k,L,l=0; +- int ip, le, le1; +- double t_r, t_i, u_r, u_i; +- static int M, MM1, init = 0, N; +- double *x_r, *x_i, *energy; +- static int *rev; +- static double *w_r, *w_i; +- +- x_r = (double *) mem_alloc(sizeof(DFFT2), "x_r"); +- x_i = (double *) mem_alloc(sizeof(DFFT2), "x_i"); +- energy = (double *) mem_alloc(sizeof(DFFT2), "energy"); +- for(i=0;i<FFT_SIZE/2;i++) x_r[i] = x_i[i] = energy[i] = 0; +- if(!init){ +- rev = (int *) mem_alloc(sizeof(IFFT2), "rev"); +- w_r = (double *) mem_alloc(sizeof(D9), "w_r"); +- w_i = (double *) mem_alloc(sizeof(D9), "w_i"); +- M = 9; +- MM1 = 8; +- N = FFT_SIZE/2; +- for(L=0;L<M;L++){ +- le = 1 << (M-L); +- le1 = le >> 1; +- w_r[L] = cos(PI/le1); +- w_i[L] = -sin(PI/le1); +- } +- for(i=0;i<FFT_SIZE/2;rev[i] = l,i++) for(j=0,l=0;j<9;j++){ +- k=(i>>j) & 1; +- l |= (k<<(8-j)); +- } +- init = 1; +- } +- memcpy( (char *) x_r, (char *) sample, sizeof(double) * FFT_SIZE/2); +- for(L=0;L<MM1;L++){ +- le = 1 << (M-L); +- le1 = le >> 1; +- u_r = 1; +- u_i = 0; +- for(j=0;j<le1;j++){ +- for(i=j;i<N;i+=le){ +- ip = i + le1; +- t_r = x_r[i] + x_r[ip]; +- t_i = x_i[i] + x_i[ip]; +- x_r[ip] = x_r[i] - x_r[ip]; +- x_i[ip] = x_i[i] - x_i[ip]; +- x_r[i] = t_r; +- x_i[i] = t_i; +- t_r = x_r[ip]; +- x_r[ip] = x_r[ip] * u_r - x_i[ip] * u_i; +- x_i[ip] = x_i[ip] * u_r + t_r * u_i; +- } +- t_r = u_r; +- u_r = u_r * w_r[L] - u_i * w_i[L]; +- u_i = u_i * w_r[L] + t_r * w_i[L]; +- } +- } +- for(i=0;i<N;i+=2){ +- ip = i + 1; +- t_r = x_r[i] + x_r[ip]; +- t_i = x_i[i] + x_i[ip]; +- x_r[ip] = x_r[i] - x_r[ip]; +- x_i[ip] = x_i[i] - x_i[ip]; +- x_r[i] = t_r; +- x_i[i] = t_i; +- energy[i] = x_r[i] * x_r[i] + x_i[i] * x_i[i]; +- } +- for(i=0;i<FFT_SIZE/2;i++) if(i<rev[i]){ +- t_r = energy[i]; +- energy[i] = energy[rev[i]]; +- energy[rev[i]] = t_r; +- } +- for(i=0;i<HAN_SIZE/2;i++){ /* calculate power */ +- if(energy[i] < 1E-20) energy[i] = 1E-20; /* density spectrum */ +- power[i].x = 10 * log10(energy[i]) + POWERNORM; +- power[i].next = STOP; +- power[i].type = FALSE; +- } +- mem_free((void **) &x_r); +- mem_free((void **) &x_i); +- mem_free((void **) &energy); +- } +- +- /**************************************************************** +- * +- * Window the incoming audio signal. +- * +- ****************************************************************/ +- +- void I_hann_win(sample) /* this function calculates a */ +- double FAR sample[FFT_SIZE/2]; /* Hann window for PCM (input) */ +- { /* samples for a 512-pt. FFT */ +- register int i; +- register double sqrt_8_over_3; +- static int init = 0; +- static double FAR *window; +- +- if(!init){ /* calculate window function for the Fourier transform */ +- window = (double FAR *) mem_alloc(sizeof(DFFT2), "window"); +- sqrt_8_over_3 = pow(8.0/3.0, 0.5); +- for(i=0;i<FFT_SIZE/2;i++){ +- /* Hann window formula */ +- window[i]=sqrt_8_over_3*0.5*(1-cos(2.0*PI*i/(FFT_SIZE/2)))/(FFT_SIZE/2); +- } +- init = 1; +- } +- for(i=0;i<FFT_SIZE/2;i++) sample[i] *= window[i]; +- } +- +- /******************************************************************* +- * +- * This function finds the maximum spectral component in each +- * subband and return them to the encoder for time-domain threshold +- * determination. +- * +- *******************************************************************/ +- #ifndef LONDON +- void I_pick_max(power, spike) +- double FAR spike[SBLIMIT]; +- mask FAR power[HAN_SIZE/2]; +- { +- double max; +- int i,j; +- +- /* calculate the spectral component in each subband */ +- for(i=0;i<HAN_SIZE/2;spike[i>>3] = max, i+=8) +- for(j=0, max = DBMIN;j<8;j++) max = (max>power[i+j].x) ? max : power[i+j].x; +- } +- #else +- void I_pick_max(power, spike) +- double FAR spike[SBLIMIT]; +- mask FAR power[HAN_SIZE]; +- { +- double sum; +- int i,j; +- +- for(i=0;i<HAN_SIZE/2;spike[i>>3] = 10.0*log10(sum), i+=8) +- /* calculate the */ +- for(j=0, sum = pow(10.0,0.1*DBMIN);j<8;j++) /* sum of spectral */ +- sum += pow(10.0,0.1*power[i+j].x); /* component in each */ +- } /* subband from bound */ +- #endif +- /**************************************************************** +- * +- * This function labels the tonal component in the power +- * spectrum. +- * +- ****************************************************************/ +- +- void I_tonal_label(power, tone) /* this function extracts */ +- mask FAR power[HAN_SIZE/2]; /* (tonal) sinusoidals from */ +- int *tone; /* the spectrum */ +- { +- int i,j, last = LAST, first, run; +- double max; +- int last_but_one= LAST; +- +- *tone = LAST; +- for(i=2;i<HAN_SIZE/2-6;i++){ +- if(power[i].x>power[i-1].x && power[i].x>=power[i+1].x){ +- power[i].type = TONE; +- power[i].next = LAST; +- if(last != LAST) power[last].next = i; +- else first = *tone = i; +- last = i; +- } +- } +- last = LAST; +- first = *tone; +- *tone = LAST; +- while(first != LAST){ /* conditions for the tonal */ +- if(first<3 || first>250) run = 0; /* otherwise k+/-j will be out of bounds*/ +- else if(first<63) run = 2; /* components in layer I, which */ +- else if(first<127) run = 3; /* are the boundaries for calc. */ +- else run = 6; /* the tonal components */ +- max = power[first].x - 7; +- for(j=2;j<=run;j++) /* after calc. of tonal components, set to loc.*/ +- if(max < power[first-j].x || max < power[first+j].x){ /* max */ +- power[first].type = FALSE; +- break; +- } +- if(power[first].type == TONE){ /* extract tonal components */ +- int help=first; +- if(*tone == LAST) *tone = first; +- while((power[help].next!=LAST)&&(power[help].next-first)<=run) +- help=power[help].next; +- help=power[help].next; +- power[first].next=help; +- if((first-last)<=run){ +- if(last_but_one != LAST) power[last_but_one].next=first; +- } +- if(first>1 && first<255){ /* calculate the sum of the */ +- double tmp; /* powers of the components */ +- tmp = add_db(power[first-1].x, power[first+1].x); +- power[first].x = add_db(power[first].x, tmp); +- } +- for(j=1;j<=run;j++){ +- power[first-j].x = power[first+j].x = DBMIN; +- power[first-j].next = power[first+j].next = STOP; /*dpwe: 2nd was .x*/ +- power[first-j].type = power[first+j].type = FALSE; +- } +- last_but_one=last; +- last = first; +- first = power[first].next; +- } +- else { +- int ll; +- if(last == LAST) ; /* *tone = power[first].next; dpwe */ +- else power[last].next = power[first].next; +- ll = first; +- first = power[first].next; +- power[ll].next = STOP; +- } +- } +- } +- +- /**************************************************************** +- * +- * This function finds the minimum masking threshold and +- * return the value to the encoder. +- * +- ****************************************************************/ +- +- void I_minimum_mask(ltg,ltmin) +- g_thres FAR *ltg; +- double FAR ltmin[SBLIMIT]; +- { +- double min; +- int i,j; +- +- j=1; +- for(i=0;i<SBLIMIT;i++) +- if(j>=sub_size-1) /* check subband limit, and */ +- ltmin[i] = ltg[sub_size-1].hear; /* calculate the minimum masking */ +- else { /* level of LTMIN for each subband*/ +- min = ltg[j].x; +- while(ltg[j].line>>3 == i && j < sub_size){ +- if (min>ltg[j].x) min = ltg[j].x; +- j++; +- } +- ltmin[i] = min; +- } +- } +- +- /***************************************************************** +- * +- * This procedure is called in musicin to pick out the +- * smaller of the scalefactor or threshold. +- * +- *****************************************************************/ +- +- void I_smr(ltmin, spike, scale) +- double FAR spike[SBLIMIT], scale[SBLIMIT], ltmin[SBLIMIT]; +- { +- int i; +- double max; +- +- for(i=0;i<SBLIMIT;i++){ /* determine the signal */ +- max = 20 * log10(scale[i] * 32768) - 10; /* level for each subband */ +- if(spike[i]>max) max = spike[i]; /* for the scalefactor */ +- max -= ltmin[i]; +- ltmin[i] = max; +- } +- } +- +- /**************************************************************** +- * +- * This procedure calls all the necessary functions to +- * complete the psychoacoustic analysis. +- * +- ****************************************************************/ +- +- void I_Psycho_One(buffer, scale, ltmin, fr_ps) +- short FAR buffer[2][1152]; +- double FAR scale[2][SBLIMIT], ltmin[2][SBLIMIT]; +- frame_params *fr_ps; +- { +- int stereo = fr_ps->stereo; +- the_layer info = fr_ps->header; +- int k,i, tone=0, noise=0; +- static char init = 0; +- static int off[2] = {256,256}; +- double *sample; +- DSBL *spike; +- static D640 *fft_buf; +- static mask_ptr FAR power; +- static g_ptr FAR ltg; +- +- sample = (double *) mem_alloc(sizeof(DFFT2), "sample"); +- spike = (DSBL *) mem_alloc(sizeof(D2SBL), "spike"); +- /* call functions for critical boundaries, freq. */ +- if(!init){ /* bands, bark values, and mapping */ +- fft_buf = (D640 *) mem_alloc(sizeof(D640) * 2, "fft_buf"); +- power = (mask_ptr FAR ) mem_alloc(sizeof(mask) * HAN_SIZE/2, "power"); +- if (info->version == MPEG_AUDIO_ID) { +- read_cbound(info->lay, info->sampling_frequency); +- read_freq_band(<g, info->lay, info->sampling_frequency); +- } else { +- read_cbound(info->lay, info->sampling_frequency + 4); +- read_freq_band(<g, info->lay, info->sampling_frequency + 4); +- } +- make_map(power,ltg); +- for(i=0;i<640;i++) fft_buf[0][i] = fft_buf[1][i] = 0; +- init = 1; +- } +- for(k=0;k<stereo;k++){ /* check PCM input for a block of */ +- for(i=0;i<384;i++) /* 384 samples for a 512-pt. FFT */ +- fft_buf[k][(i+off[k])%640]= (double) buffer[k][i]/SCALE; +- for(i=0;i<FFT_SIZE/2;i++) +- sample[i] = fft_buf[k][(i+448+off[k])%640]; +- off[k] += 384; +- off[k] %= 640; +- /* call functions for windowing PCM samples, */ +- I_hann_win(sample); /* location of spectral components in each */ +- for(i=0;i<HAN_SIZE/2;i++) power[i].x = DBMIN; /* subband with */ +- I_f_f_t(sample, power); /* labeling, locate remaining */ +- I_pick_max(power, &spike[k][0]); /* non-tonal sinusoidals, */ +- I_tonal_label(power, &tone); /* reduce noise & tonal com., */ +- noise_label(power, &noise, ltg); /* find global & minimal */ +- subsampling(power, ltg, &tone, &noise); /* threshold, and sgnl- */ +- threshold(power, ltg, &tone, &noise, /* to-mask ratio */ +- bitrate[info->version][info->lay-1][info->bitrate_index]/stereo); +- I_minimum_mask(ltg, <min[k][0]); +- I_smr(<min[k][0], &spike[k][0], &scale[k][0]); +- } +- mem_free((void **) &sample); +- mem_free((void **) &spike); +- } +--- 0 ---- +diff -r -c -N encoder/util.c lame3.50/util.c +*** encoder/util.c Wed Dec 31 17:00:00 1969 +--- lame3.50/util.c Tue Oct 26 12:14:29 1999 +*************** +*** 0 **** +--- 1,413 ---- ++ #include "util.h" ++ #include "globalflags.h" ++ #include <assert.h> ++ ++ /*********************************************************************** ++ * ++ * Global Variable Definitions ++ * ++ ***********************************************************************/ ++ ++ ++ /* 1: MPEG-1, 0: MPEG-2 LSF, 1995-07-11 shn */ ++ double s_freq[2][4] = {{22.05, 24, 16, 0}, {44.1, 48, 32, 0}}; ++ ++ /* 1: MPEG-1, 0: MPEG-2 LSF, 1995-07-11 shn */ ++ int bitrate[2][3][15] = { ++ {{0,32,48,56,64,80,96,112,128,144,160,176,192,224,256}, ++ {0,8,16,24,32,40,48,56,64,80,96,112,128,144,160}, ++ {0,8,16,24,32,40,48,56,64,80,96,112,128,144,160}}, ++ {{0,32,64,96,128,160,192,224,256,288,320,352,384,416,448}, ++ {0,32,48,56,64,80,96,112,128,160,192,224,256,320,384}, ++ {0,32,40,48,56,64,80,96,112,128,160,192,224,256,320}} ++ }; ++ ++ ++ enum byte_order NativeByteOrder = order_unknown; ++ ++ /*********************************************************************** ++ * ++ * Global Function Definitions ++ * ++ ***********************************************************************/ ++ ++ int nint2( double in ) ++ { ++ int temp; ++ ++ if( in < 0 ) temp = (int)(in - 0.5); ++ else temp = (int)(in + 0.5); ++ ++ return(temp); ++ } ++ ++ ++ ++ /*********************************************************************** ++ * compute bitsperframe and mean_bits for a layer III frame ++ **********************************************************************/ ++ void getframebits(layer *info, int stereo, int *bitsPerFrame, int *mean_bits) { ++ int whole_SpF; ++ double bit_rate,samp; ++ int samplesPerFrame,bitsPerSlot; ++ int sideinfo_len; ++ int mode_gr; ++ ++ samp = s_freq[info->version][info->sampling_frequency]; ++ bit_rate = bitrate[info->version][info->lay-1][info->bitrate_index]; ++ samplesPerFrame = info->version == 1 ? 1152 : 576; ++ bitsPerSlot = 8; ++ mode_gr = (info->version == 1) ? 2 : 1; /* mode_gr = 2 */ ++ ++ /* determine the mean bitrate for main data */ ++ sideinfo_len = 32; ++ if ( info->version == 1 ) ++ { /* MPEG 1 */ ++ if ( stereo == 1 ) ++ sideinfo_len += 136; ++ else ++ sideinfo_len += 256; ++ } ++ else ++ { /* MPEG 2 */ ++ if ( stereo == 1 ) ++ sideinfo_len += 72; ++ else ++ sideinfo_len += 136; ++ } ++ ++ if (info->error_protection) sideinfo_len += 16; ++ ++ ++ whole_SpF = (samplesPerFrame /samp)*(bit_rate / (double)bitsPerSlot); ++ *bitsPerFrame = 8 * whole_SpF + (info->padding * 8); ++ *mean_bits = (*bitsPerFrame - sideinfo_len) / mode_gr; ++ } ++ ++ ++ ++ ++ void display_bitrates(int layr) ++ { ++ int index,version; ++ ++ version = MPEG_AUDIO_ID; ++ fprintf(stderr,"\n"); ++ fprintf(stderr,"MPEG1 samplerates(kHz): 32 44.1 48 \n"); ++ fprintf(stderr,"bitrates(kbs): "); ++ for (index=1;index<14;index++) { ++ fprintf(stderr,"%i ",bitrate[version][layr-1][index]); ++ } ++ fprintf(stderr,"\n"); ++ ++ version = MPEG_PHASE2_LSF; ++ fprintf(stderr,"\n"); ++ fprintf(stderr,"MPEG2 samplerates(kHz): 16 22.05 24 \n"); ++ fprintf(stderr,"bitrates(kbs): "); ++ for (index=1;index<15;index++) { ++ fprintf(stderr,"%i ",bitrate[version][layr-1][index]); ++ } ++ fprintf(stderr,"\n"); ++ } ++ ++ ++ int BitrateIndex( ++ int layr, /* 1 or 2 */ ++ int bRate, /* legal rates from 32 to 448 */ ++ int version, /* MPEG-1 or MPEG-2 LSF */ ++ int samplerate) /* convert bitrate in kbps to index */ ++ { ++ int index = 0; ++ int found = 0; ++ ++ while(!found && index<15) { ++ if(bitrate[version][layr-1][index] == bRate) ++ found = 1; ++ else ++ ++index; ++ } ++ if(found) ++ return(index); ++ else { ++ fprintf(stderr,"Bitrate %dkbs not legal for %iHz output sampling.\n", ++ bRate, samplerate); ++ return(-1); /* Error! */ ++ } ++ } ++ ++ int SmpFrqIndex( /* convert samp frq in Hz to index */ ++ long sRate, /* legal rates 16000, 22050, 24000, 32000, 44100, 48000 */ ++ int *version) ++ { ++ if (sRate == 44100L) { ++ *version = MPEG_AUDIO_ID; return(0); ++ } ++ else if (sRate == 48000L) { ++ *version = MPEG_AUDIO_ID; return(1); ++ } ++ else if (sRate == 32000L) { ++ *version = MPEG_AUDIO_ID; return(2); ++ } ++ else if (sRate == 24000L) { ++ *version = MPEG_PHASE2_LSF; return(1); ++ } ++ else if (sRate == 22050L) { ++ *version = MPEG_PHASE2_LSF; return(0); ++ } ++ else if (sRate == 16000L) { ++ *version = MPEG_PHASE2_LSF; return(2); ++ } ++ else { ++ fprintf(stderr, "SmpFrqIndex: %ldHz is not a legal sample rate\n", sRate); ++ return(-1); /* Error! */ ++ } ++ } ++ ++ /******************************************************************************* ++ * ++ * Allocate number of bytes of memory equal to "block". ++ * ++ *******************************************************************************/ ++ /* exit(0) changed to exit(1) on memory allocation ++ * error -- 1999/06 Alvaro Martinez Echevarria */ ++ ++ void *mem_alloc(unsigned long block, char *item) ++ { ++ ++ void *ptr; ++ ++ /* what kind of shit does ISO put out? */ ++ ptr = (void *) malloc((size_t) block/*<<1*/); /* allocate twice as much memory as needed. fixes dodgy ++ memory problem on most systems */ ++ ++ ++ if (ptr != NULL) { ++ memset(ptr, 0, (size_t) block); ++ } else { ++ fprintf(stderr,"Unable to allocate %s\n", item); ++ exit(1); ++ } ++ return(ptr); ++ } ++ ++ ++ ++ /***************************************************************************** ++ * ++ * Routines to determine byte order and swap bytes ++ * ++ *****************************************************************************/ ++ ++ enum byte_order DetermineByteOrder(void) ++ { ++ char s[ sizeof(long) + 1 ]; ++ union ++ { ++ long longval; ++ char charval[ sizeof(long) ]; ++ } probe; ++ probe.longval = 0x41424344L; /* ABCD in ASCII */ ++ strncpy( s, probe.charval, sizeof(long) ); ++ s[ sizeof(long) ] = '\0'; ++ /* fprintf( stderr, "byte order is %s\n", s ); */ ++ if ( strcmp(s, "ABCD") == 0 ) ++ return order_bigEndian; ++ else ++ if ( strcmp(s, "DCBA") == 0 ) ++ return order_littleEndian; ++ else ++ return order_unknown; ++ } ++ ++ void SwapBytesInWords( short *loc, int words ) ++ { ++ int i; ++ short thisval; ++ char *dst, *src; ++ src = (char *) &thisval; ++ for ( i = 0; i < words; i++ ) ++ { ++ thisval = *loc; ++ dst = (char *) loc++; ++ dst[0] = src[1]; ++ dst[1] = src[0]; ++ } ++ } ++ ++ ++ ++ ++ ++ ++ /***************************************************************************** ++ * ++ * bit_stream.c package ++ * Author: Jean-Georges Fritsch, C-Cube Microsystems ++ * ++ *****************************************************************************/ ++ ++ /******************************************************************** ++ This package provides functions to write (exclusive or read) ++ information from (exclusive or to) the bit stream. ++ ++ If the bit stream is opened in read mode only the get functions are ++ available. If the bit stream is opened in write mode only the put ++ functions are available. ++ ********************************************************************/ ++ ++ /*open_bit_stream_w(); open the device to write the bit stream into it */ ++ /*open_bit_stream_r(); open the device to read the bit stream from it */ ++ /*alloc_buffer(); open and initialize the buffer; */ ++ /*desalloc_buffer(); empty and close the buffer */ ++ /*back_track_buffer(); goes back N bits in the buffer */ ++ /*unsigned int get1bit(); read 1 bit from the bit stream */ ++ /*unsigned long look_ahead(); grep the next N bits in the bit stream without*/ ++ /* changing the buffer pointer */ ++ /*putbits(); write N bits from the bit stream */ ++ /*int seek_sync(); return 1 if a sync word was found in the bit stream */ ++ /* otherwise returns 0 */ ++ ++ ++ ++ /* empty the buffer to the output device when the buffer becomes full */ ++ /* mt 3/99: modified so we can call empty_buffer even if buffer is not full */ ++ /* add error checking -- 1999/06 Alvaro Martinez Echevarria */ ++ /* someone gimme a good reason for writing bytes one by ++ * one; changed -- 1999/06 Alvaro Martinez Echevarria */ ++ void write_buffer(bs) ++ Bit_stream_struc *bs; /* bit stream structure */ ++ { ++ register int i,j; ++ int minimum=1+bs->buf_byte_idx; /* end of the buffer to empty */ ++ unsigned char *tmpbuf; ++ if (bs->buf_size-minimum <= 0) return; ++ ++ tmpbuf=(unsigned char *)malloc(bs->buf_size-minimum); ++ if (tmpbuf==NULL) { ++ // fprintf(stderr,"error in malloc(): %s\n",sys_errlist[errno]); ++ perror("error in malloc()\n"); ++ exit(1); ++ } ++ for (i=bs->buf_size-1,j=0;i>=minimum;i--,j++) { ++ tmpbuf[j]=bs->buf[i]; ++ } ++ fwrite(tmpbuf,1,bs->buf_size-minimum,bs->pt); ++ if (ferror(bs->pt)) { ++ fprintf(stderr,"error en fwrite()\n"); ++ exit(1); ++ } ++ free(tmpbuf); ++ } ++ ++ int copy_buffer(char *buffer,Bit_stream_struc *bs) ++ { ++ int i,j=0; ++ for (i=bs->buf_size-1 ; i > bs->buf_byte_idx ; (i-- )) ++ buffer[j++]=bs->buf[i]; ++ return j; ++ } ++ ++ ++ void empty_buffer(Bit_stream_struc *bs) ++ { ++ int minimum=1+bs->buf_byte_idx; /* end of the buffer to empty */ ++ if (bs->buf_size-minimum <= 0) return; ++ bs->buf_byte_idx = bs->buf_size -1; ++ bs->buf_bit_idx = 8; ++ ++ bs->buf[bs->buf_byte_idx] = 0; /* what does this do? */ ++ ++ } ++ ++ ++ ++ /* open the device to write the bit stream into it */ ++ void open_bit_stream_w( ++ Bit_stream_struc* bs, /* bit stream structure */ ++ char* bs_filenam, /* name of the bit stream file */ ++ int size, /* size of the buffer */ ++ int nowrite /* =1 to disable output */) ++ { ++ if (!nowrite) { ++ if (!strcmp(bs_filenam, "-")) { ++ /* Write to standard output. */ ++ #ifdef __EMX__ ++ _fsetmode(stdout,"b"); ++ #elif (defined __BORLANDC__) ++ setmode(_fileno(stdout), O_BINARY); ++ #elif (defined __CYGWIN__) ++ setmode(fileno(stdout), _O_BINARY); ++ #elif (defined _WIN32) ++ _setmode(_fileno(stdout), _O_BINARY); ++ #endif ++ bs->pt = stdout; ++ } else { ++ if ((bs->pt = fopen(bs_filenam, "wb")) == NULL) { ++ fprintf(stderr,"Could not create \"%s\".\n", bs_filenam); ++ exit(1); ++ } ++ } ++ } ++ ++ alloc_buffer(bs, size); ++ bs->buf_byte_idx = size-1; ++ bs->buf_bit_idx=8; ++ bs->totbit=0; ++ } ++ ++ ++ /*open and initialize the buffer; */ ++ void alloc_buffer( ++ Bit_stream_struc *bs, /* bit stream structure */ ++ int size) ++ { ++ bs->buf = (unsigned char *) ++ mem_alloc((unsigned long) (size * sizeof(unsigned char)), "buffer"); ++ bs->buf_size = size; ++ } ++ ++ /*empty and close the buffer */ ++ void desalloc_buffer(Bit_stream_struc *bs) /* bit stream structure */ ++ { ++ free(bs->buf); ++ } ++ ++ int putmask[9]={0x0, 0x1, 0x3, 0x7, 0xf, 0x1f, 0x3f, 0x7f, 0xff}; ++ ++ ++ /*write N bits into the bit stream */ ++ void putbits( ++ Bit_stream_struc *bs, /* bit stream structure */ ++ unsigned int val, /* val to write into the buffer */ ++ int N) /* number of bits of val */ ++ { ++ register int j = N; ++ register int k, tmp; ++ ++ if (N > MAX_LENGTH) ++ fprintf(stderr,"Cannot read or write more than %d bits at a time.\n", MAX_LENGTH); ++ ++ bs->totbit += N; ++ while (j > 0) { ++ k = Min(j, bs->buf_bit_idx); ++ tmp = val >> (j-k); ++ bs->buf[bs->buf_byte_idx] |= (tmp&putmask[k]) << (bs->buf_bit_idx-k); ++ bs->buf_bit_idx -= k; ++ if (!bs->buf_bit_idx) { ++ bs->buf_bit_idx = 8; ++ bs->buf_byte_idx--; ++ assert(bs->buf_byte_idx >= 0); ++ bs->buf[bs->buf_byte_idx] = 0; ++ } ++ j -= k; ++ } ++ } ++ ++ ++ ++ /***************************************************************************** ++ * ++ * End of bit_stream.c package ++ * ++ *****************************************************************************/ ++ +diff -r -c -N encoder/util.h lame3.50/util.h +*** encoder/util.h Wed Dec 31 17:00:00 1969 +--- lame3.50/util.h Tue Oct 26 23:52:01 1999 +*************** +*** 0 **** +--- 1,181 ---- ++ #ifndef UTIL_DOT_H ++ #define UTIL_DOT_H ++ /*********************************************************************** ++ * ++ * Global Include Files ++ * ++ ***********************************************************************/ ++ #include "machine.h" ++ ++ ++ /*********************************************************************** ++ * ++ * Global Definitions ++ * ++ ***********************************************************************/ ++ ++ /* General Definitions */ ++ #ifndef FALSE ++ #define FALSE 0 ++ #endif ++ ++ #ifndef TRUE ++ #define TRUE 1 ++ #endif ++ ++ #define MAX_U_32_NUM 0xFFFFFFFF ++ #ifdef M_PI ++ #define PI M_PI ++ #endif ++ #ifndef PI ++ #define PI 3.14159265358979 ++ #endif ++ #ifdef M_PI_4 ++ #define PI4 M_PI_4 ++ #else ++ #define PI4 (PI/4) ++ #endif ++ #define PI64 (PI/64) ++ ++ #define SQRT2 1.4142135623730951454746218587388284504414 ++ ++ #define MPEG_AUDIO_ID 1 ++ #define MPEG_PHASE2_LSF 0 /* 1995-07-11 SHN */ ++ ++ #define BITS_IN_A_BYTE 8 ++ #define MAX_NAME_SIZE 300 ++ #define SBLIMIT 32 ++ #define SSLIMIT 18 ++ #define FFT_SIZE 1024 ++ #define HAN_SIZE 512 ++ #define CRC16_POLYNOMIAL 0x8005 ++ ++ /* MPEG Header Definitions - Mode Values */ ++ ++ #define MPG_MD_STEREO 0 ++ #define MPG_MD_JOINT_STEREO 1 ++ #define MPG_MD_DUAL_CHANNEL 2 ++ #define MPG_MD_MONO 3 ++ ++ /* Mode Extention */ ++ ++ #define MPG_MD_LR_LR 0 ++ #define MPG_MD_LR_I 1 ++ #define MPG_MD_MS_LR 2 ++ #define MPG_MD_MS_I 3 ++ ++ ++ /* "bit_stream.h" Definitions */ ++ ++ #define MINIMUM 4 /* Minimum size of the buffer in bytes */ ++ #define MAX_LENGTH 32 /* Maximum length of word written or ++ read from bit stream */ ++ #define BUFFER_SIZE 16384 ++ ++ #define Min(A, B) ((A) < (B) ? (A) : (B)) ++ #define Max(A, B) ((A) > (B) ? (A) : (B)) ++ ++ /*********************************************************************** ++ * ++ * Global Type Definitions ++ * ++ ***********************************************************************/ ++ ++ /* Structure for Reading Layer II Allocation Tables from File */ ++ ++ typedef struct { ++ unsigned int steps; ++ unsigned int bits; ++ unsigned int group; ++ unsigned int quant; ++ } sb_alloc, *alloc_ptr; ++ ++ typedef sb_alloc al_table[SBLIMIT][16]; ++ ++ /* Header Information Structure */ ++ ++ typedef struct { ++ int version; ++ int lay; ++ int error_protection; ++ int bitrate_index; ++ int sampling_frequency; ++ int padding; ++ int extension; ++ int mode; ++ int mode_ext; ++ int copyright; ++ int original; ++ int emphasis; ++ } layer, *the_layer; ++ ++ /* Parent Structure Interpreting some Frame Parameters in Header */ ++ ++ typedef struct { ++ layer *header; /* raw header information */ ++ int actual_mode; /* when writing IS, may forget if 0 chs */ ++ al_table *alloc; /* bit allocation table read in */ ++ int tab_num; /* number of table as loaded */ ++ int stereo; /* 1 for mono, 2 for stereo */ ++ int jsbound; /* first band of joint stereo coding */ ++ int sblimit; /* total number of sub bands */ ++ } frame_params; ++ ++ ++ enum byte_order { order_unknown, order_bigEndian, order_littleEndian }; ++ extern enum byte_order NativeByteOrder; ++ ++ /* "bit_stream.h" Type Definitions */ ++ ++ typedef struct bit_stream_struc { ++ unsigned char* pbtOutBuf; /* for .DLL code */ ++ int nOutBufPos; /* for .DLL code */ ++ FILE *pt; /* pointer to bit stream device */ ++ unsigned char *buf; /* bit stream buffer */ ++ int buf_size; /* size of buffer (in number of bytes) */ ++ unsigned long totbit; /* bit counter of bit stream */ ++ int buf_byte_idx; /* pointer to top byte in buffer */ ++ int buf_bit_idx; /* pointer to top bit of top byte in buffer */ ++ ++ /* format of file in rd mode (BINARY/ASCII) */ ++ } Bit_stream_struc; ++ ++ #include "l3side.h" ++ ++ /*********************************************************************** ++ * ++ * Global Variable External Declarations ++ * ++ ***********************************************************************/ ++ ++ extern double s_freq[2][4]; ++ extern int bitrate[2][3][15]; ++ ++ /*********************************************************************** ++ * ++ * Global Function Prototype Declarations ++ * ++ ***********************************************************************/ ++ ++ extern void display_bitrates(int layr); ++ extern int BitrateIndex(int, int, int,int); ++ extern int SmpFrqIndex(long, int*); ++ extern void *mem_alloc(unsigned long, char*); ++ extern void empty_buffer(Bit_stream_struc*); ++ extern int copy_buffer(char *buffer,Bit_stream_struc *bs); ++ extern void write_buffer(Bit_stream_struc*); ++ extern void open_bit_stream_w(Bit_stream_struc*, char*, int, int); ++ extern void close_bit_stream_w(Bit_stream_struc*); ++ extern void alloc_buffer(Bit_stream_struc*, int); ++ extern void desalloc_buffer(Bit_stream_struc*); ++ extern void putbits(Bit_stream_struc*, unsigned int, int); ++ ++ extern enum byte_order DetermineByteOrder(void); ++ extern void SwapBytesInWords( short *loc, int words ); ++ ++ extern void ++ getframebits(layer *info, int stereo, int *bitsPerFrame, int *mean_bits); ++ int nint2( double in ); ++ ++ void timestatus(layer *info,long frameNum,long totalframes); ++ #endif +diff -r -c -N encoder/version.c lame3.50/version.c +*** encoder/version.c Wed Dec 31 17:00:00 1969 +--- lame3.50/version.c Mon Sep 27 17:49:08 1999 +*************** +*** 0 **** +--- 1,57 ---- ++ /* ++ * Version numbering for LAME. ++ * ++ * Copyright (c) 1999 A.L. Faber ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License as published by ++ * the Free Software Foundation; either version 2, or (at your option) ++ * any later version. ++ * ++ * This program is distributed in the hope that it will be useful, ++ * but WITHOUT ANY WARRANTY; without even the implied warranty of ++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the ++ * GNU General Public License for more details. ++ * ++ * You should have received a copy of the GNU General Public License ++ * along with this program; see the file COPYING. If not, write to ++ * the Free Software Foundation, 675 Mass Ave, Cambridge, MA 02139, USA. ++ */ ++ ++ #include "version.h" ++ #include <stdio.h> ++ ++ static char lpszVersion[80]; ++ ++ char* get_lame_version(void) ++ { ++ if (LAME_ALPHAVERSION>0) ++ sprintf(lpszVersion,"%d.%02d (alpha %d)",LAME_MAJOR_VERSION,LAME_MINOR_VERSION,LAME_ALPHAVERSION); ++ else if (LAME_BETAVERSION>0) ++ sprintf(lpszVersion,"%d.%02d (beta %d)",LAME_MAJOR_VERSION,LAME_MINOR_VERSION,LAME_BETAVERSION); ++ else ++ sprintf(lpszVersion,"%d.%02d",LAME_MAJOR_VERSION,LAME_MINOR_VERSION); ++ return lpszVersion; ++ } ++ ++ char* get_psy_version(void) ++ { ++ if (PSY_ALPHAVERSION>0) ++ sprintf(lpszVersion,"%d.%02d (alpha %d)",PSY_MAJOR_VERSION,PSY_MINOR_VERSION,PSY_ALPHAVERSION); ++ else if (PSY_BETAVERSION>0) ++ sprintf(lpszVersion,"%d.%02d (beta %d)",PSY_MAJOR_VERSION,PSY_MINOR_VERSION,PSY_BETAVERSION); ++ else ++ sprintf(lpszVersion,"%d.%02d",PSY_MAJOR_VERSION,PSY_MINOR_VERSION); ++ return lpszVersion; ++ } ++ ++ char* get_mp3x_version(void) ++ { ++ if (MP3X_ALPHAVERSION>0) ++ sprintf(lpszVersion,"%d:%02d (alpha %d)",MP3X_MAJOR_VERSION,MP3X_MINOR_VERSION,MP3X_ALPHAVERSION); ++ else if (MP3X_BETAVERSION>0) ++ sprintf(lpszVersion,"%d:%02d (beta %d)",MP3X_MAJOR_VERSION,MP3X_MINOR_VERSION,MP3X_BETAVERSION); ++ else ++ sprintf(lpszVersion,"%d:%02d",MP3X_MAJOR_VERSION,MP3X_MINOR_VERSION); ++ return lpszVersion; ++ } +diff -r -c -N encoder/version.h lame3.50/version.h +*** encoder/version.h Wed Dec 31 17:00:00 1969 +--- lame3.50/version.h Mon Nov 1 18:22:45 1999 +*************** +*** 0 **** +--- 1,44 ---- ++ /* ++ * Version numbering for LAME. ++ * ++ * Copyright (c) 1999 A.L. Faber ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License as published by ++ * the Free Software Foundation; either version 2, or (at your option) ++ * any later version. ++ * ++ * This program is distributed in the hope that it will be useful, ++ * but WITHOUT ANY WARRANTY; without even the implied warranty of ++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the ++ * GNU General Public License for more details. ++ * ++ * You should have received a copy of the GNU General Public License ++ * along with this program; see the file COPYING. If not, write to ++ * the Free Software Foundation, 675 Mass Ave, Cambridge, MA 02139, USA. ++ */ ++ ++ #ifndef LAME_H_INCLUDED ++ #define LAME_H_INCLUDED ++ ++ #define LAME_MAJOR_VERSION 3 /* Major version number */ ++ #define LAME_MINOR_VERSION 50 /* Minor version number */ ++ #define LAME_ALPHAVERSION 0 /* Set number if this is an alpha version, otherwise zero */ ++ #define LAME_BETAVERSION 0 /* Set number if this is a beta version, otherwise zero */ ++ ++ #define PSY_MAJOR_VERSION 0 /* Major version number */ ++ #define PSY_MINOR_VERSION 74 /* Minor version number */ ++ #define PSY_ALPHAVERSION 0 /* Set number if this is an alpha version, otherwise zero */ ++ #define PSY_BETAVERSION 0 /* Set number if this is a beta version, otherwise zero */ ++ ++ #define MP3X_MAJOR_VERSION 0 /* Major version number */ ++ #define MP3X_MINOR_VERSION 73 /* Minor version number */ ++ #define MP3X_ALPHAVERSION 0 /* Set number if this is an alpha version, otherwise zero */ ++ #define MP3X_BETAVERSION 0 /* Set number if this is a beta version, otherwise zero */ ++ ++ ++ char* get_lame_version(void); /* returns lame version number string */ ++ char* get_psy_version(void); /* returns psy model version number string */ ++ char* get_mp3x_version(void); /* returns mp3x version number string */ ++ ++ #endif diff --git a/lame3.70.patch b/lame3.70.patch new file mode 100644 index 0000000..97e9d53 --- /dev/null +++ b/lame3.70.patch @@ -0,0 +1,44549 @@ +diff -r -c -N encoder/.cvsignore lame3.70/.cvsignore +*** encoder/.cvsignore Wed Dec 31 17:00:00 1969 +--- lame3.70/.cvsignore Wed Jan 12 15:38:44 2000 +*************** +*** 0 **** +--- 1,14 ---- ++ mp3x ++ lame ++ *.d ++ Debug ++ DebugGTK ++ Release ++ ReleaseGTK ++ WinGTK ++ *.ncb ++ *.plg ++ *.opt ++ *.dll ++ *.new.mp3 ++ *.wav.mp3 +\ No newline at end of file +diff -r -c -N encoder/API lame3.70/API +*** encoder/API Wed Dec 31 17:00:00 1969 +--- lame3.70/API Thu Mar 23 15:12:26 2000 +*************** +*** 0 **** +--- 1,68 ---- ++ The LAME API ++ ++ This is the simple interface to the lightweight encoding library ++ obtained by compiling libmp3lame.a without defining any of ++ the following extra features: ++ ++ #define HAVEMPGLIB to use mpglib's mp3 *decoding* capibility ++ #define AMIGA_MPEGA to use mpega.library (Amiga), don't use with HAVEMPGLIB ++ #define BRHIST to allow the display of the VBR historgram ++ #define LIBSNDFILE to use Erik de Castro Lopo's libsndfile ++ #define LAMESNDFILE to use LAME's minimial internal sndfile I/O ++ #define LAMEPARSE to use LAME's command line parsing/option setting routines ++ #define HAVEGTK to compile in support for the GTK mp3 frame analyzer ++ ++ To use any of the above features, see lame.h more for details. ++ For an example of a simple command line interface to lame, see ++ main.c ++ ++ ++ ========================================================================= ++ ++ 1. (optional) Get the version number of the encoder, if you are interested. ++ lame_version(char *); ++ ++ 2. Initialize the encoder. sets default for all encoder parameters, ++ returns pointer to encoder parameters listed above ++ ++ #include "lame.h" ++ lame_global_flags gf; ++ lame_init(%gf); ++ ++ Then override various default settings as necessary, for example: ++ ++ gf.num_channels=2; ++ gf.in_samplerate = 44100; ++ gf.brate = 128; ++ gf.mode = 0,1 or 3 /* stereo, jstereo, mono */ ++ gf.quality = 2,5 or 9 /* 2=high, 5=medium 9=low */ ++ ++ see lame.h for the complete list of options. ++ ++ ++ 3. sets more internal configuration based on data provided above: ++ lame_init_params(&gf); ++ ++ ++ ++ 4. Encode some data. input pcm data, output (maybe) mp3 frames. ++ This routine handles all buffering, resampling and filtering for you. ++ The required mp3buffer_size can be computed from num_samples, ++ samplerate and encoding rate, but here is a worst case estimate: ++ mp3buffer_size (in bytes) = 1.25*num_samples + 7200 ++ The return code = number of bytes output in mp3buffer. This can be 0. ++ If it is -1, an error occured. ++ ++ int lame_encode_buffer(lame_global_flags *gfp, ++ short int leftpcm[], short int rightpcm[], ++ int num_samples,char *mp3buffer,int mp3buffer_size); ++ ++ ++ 5. lame_encode_finish will flush the buffers and may return a ++ final few mp3 frames. mp3buffer should be at least 7200 bytes. ++ return code = number of bytes output to mp3buffer. This can be 0. ++ ++ int lame_encode_finish(lame_global_flags *gfp,char *mp3buffer); ++ ++ ++ +diff -r -c -N encoder/COPYING lame3.70/COPYING +*** encoder/COPYING Wed Dec 31 17:00:00 1969 +--- lame3.70/COPYING Wed Nov 24 01:40:26 1999 +*************** +*** 0 **** +--- 1,482 ---- ++ GNU LIBRARY GENERAL PUBLIC LICENSE ++ Version 2, June 1991 ++ ++ Copyright (C) 1991 Free Software Foundation, Inc. ++ 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA ++ Everyone is permitted to copy and distribute verbatim copies ++ of this license document, but changing it is not allowed. ++ ++ [This is the first released version of the library GPL. It is ++ numbered 2 because it goes with version 2 of the ordinary GPL.] ++ ++ Preamble ++ ++ The licenses for most software are designed to take away your ++ freedom to share and change it. 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Here is a sample; alter the names: ++ ++ Yoyodyne, Inc., hereby disclaims all copyright interest in the ++ library `Frob' (a library for tweaking knobs) written by James Random Hacker. ++ ++ <signature of Ty Coon>, 1 April 1990 ++ Ty Coon, President of Vice ++ ++ That's all there is to it! +diff -r -c -N encoder/CVS/Entries lame3.70/CVS/Entries +*** encoder/CVS/Entries Wed Dec 31 17:00:00 1969 +--- lame3.70/CVS/Entries Thu Apr 6 12:50:52 2000 +*************** +*** 0 **** +--- 1,80 ---- ++ D/Dll//// ++ D/doc//// ++ D/i386//// ++ D/libmp3lame//// ++ D/mpglib//// ++ /.cvsignore/1.5/Wed Jan 12 22:38:44 2000//Tlame3_70 ++ /API/1.5/Thu Mar 23 22:12:26 2000//Tlame3_70 ++ /COPYING/1.1.1.1/Wed Nov 24 08:40:26 1999//Tlame3_70 ++ /HACKING/1.4/Thu Apr 6 18:50:49 2000//Tlame3_70 ++ /INSTALL/1.3/Thu Mar 30 09:37:28 2000//Tlame3_70 ++ /LICENSE/1.1.1.1/Wed Nov 24 08:43:10 1999//Tlame3_70 ++ /Makefile/1.44.4.1/Thu Apr 6 18:50:49 2000//Tlame3_70 ++ /Makefile.MSVC/1.4/Thu Apr 6 18:50:49 2000//Tlame3_70 ++ /PRESETS.draft/1.2/Sat Jan 8 17:52:52 2000//Tlame3_70 ++ /README/1.2/Thu Mar 30 08:24:20 2000//Tlame3_70 ++ /README.WINGTK/1.3/Thu Mar 30 08:04:11 2000//Tlame3_70 ++ /TODO/1.6/Sun Jan 30 10:50:36 2000//Tlame3_70 ++ /USAGE/1.7/Wed Mar 22 02:03:03 2000//Tlame3_70 ++ /VbrTag.c/1.7.4.1/Thu Apr 6 18:50:49 2000//Tlame3_70 ++ /VbrTag.h/1.5/Thu Apr 6 18:50:49 2000//Tlame3_70 ++ /amiga_mpega.c/1.11.4.1/Thu Apr 6 18:50:49 2000//Tlame3_70 ++ /auenc/1.1.1.1/Wed Nov 24 08:41:13 1999//Tlame3_70 ++ /brhist.c/1.6.4.1/Thu Apr 6 18:50:49 2000//Tlame3_70 ++ /brhist.h/1.4/Tue Mar 21 23:02:17 2000//Tlame3_70 ++ /debugscalefac.c/1.1.1.1/Wed Nov 24 08:43:46 1999//Tlame3_70 ++ /encoder.h/1.6/Wed Mar 15 20:37:16 2000//Tlame3_70 ++ /fft.c/1.18.4.1/Thu Apr 6 18:50:49 2000//Tlame3_70 ++ /fft.h/1.7/Sun Feb 13 18:52:59 2000//Tlame3_70 ++ /formatBitstream.c/1.4/Sun Feb 20 14:09:34 2000//Tlame3_70 ++ /formatBitstream.h/1.2/Sat Feb 19 13:32:30 2000//Tlame3_70 ++ /get_audio.c/1.36.4.1/Thu Apr 6 18:50:49 2000//Tlame3_70 ++ /get_audio.h/1.7/Thu Mar 16 19:05:26 2000//Tlame3_70 ++ /gpkplotting.c/1.1.1.1/Wed Nov 24 08:42:07 1999//Tlame3_70 ++ /gpkplotting.h/1.1.1.1/Wed Nov 24 08:42:08 1999//Tlame3_70 ++ /gtkanal.c/1.26/Thu Apr 6 18:50:49 2000//Tlame3_70 ++ /gtkanal.h/1.10/Thu Mar 30 05:39:41 2000//Tlame3_70 ++ /id3tag.c/1.3/Sun Mar 5 18:37:28 2000//Tlame3_70 ++ /id3tag.h/1.4/Mon Mar 6 19:53:03 2000//Tlame3_70 ++ /ieeefloat.c/1.3/Mon Feb 21 23:05:05 2000//Tlame3_70 ++ /ieeefloat.h/1.1.1.1/Wed Nov 24 08:42:58 1999//Tlame3_70 ++ /l3bitstream-pvt.h/1.4/Thu Apr 6 18:50:49 2000//Tlame3_70 ++ /l3bitstream.c/1.20.2.1.2.2/Thu Apr 6 18:50:49 2000//Tlame3_70 ++ /l3bitstream.h/1.5/Thu Apr 6 18:50:49 2000//Tlame3_70 ++ /l3side.h/1.10/Thu Apr 6 18:50:49 2000//Tlame3_70 ++ /lame.c/1.99.2.3.4.4/Thu Apr 6 18:50:51 2000//Tlame3_70 ++ /lame.dsp/1.12/Thu Apr 6 18:50:51 2000//Tlame3_70 ++ /lame.dsw/1.1.1.1/Wed Nov 24 08:43:20 1999//Tlame3_70 ++ /lame.h/1.39/Thu Apr 6 18:50:51 2000//Tlame3_70 ++ /machine.h/1.14.4.1/Thu Apr 6 18:50:51 2000//Tlame3_70 ++ /main.c/1.25/Thu Apr 6 18:50:51 2000//Tlame3_70 ++ /mlame/1.3/Mon Jan 31 05:49:48 2000//Tlame3_70 ++ /mp3rtp.c/1.11/Thu Mar 16 19:05:26 2000//Tlame3_70 ++ /mp3x.c/1.7/Thu Mar 16 19:05:26 2000//Tlame3_70 ++ /newmdct.c/1.11.4.1/Thu Apr 6 18:50:51 2000//Tlame3_70 ++ /newmdct.h/1.4/Tue Mar 21 23:02:17 2000//Tlame3_70 ++ /parse.c/1.25.4.1.4.2/Thu Apr 6 18:50:51 2000//Tlame3_70 ++ /portableio.c/1.1.1.1/Wed Nov 24 08:43:35 1999//Tlame3_70 ++ /portableio.h/1.1.1.1/Wed Nov 24 08:43:37 1999//Tlame3_70 ++ /psymodel.c/1.52.4.2/Thu Apr 6 18:50:51 2000//Tlame3_70 ++ /psymodel.h/1.10/Fri Mar 24 18:07:30 2000//Tlame3_70 ++ /quantize-pvt.c/1.67.4.1/Thu Apr 6 18:50:51 2000//Tlame3_70 ++ /quantize-pvt.h/1.35/Thu Apr 6 18:50:51 2000//Tlame3_70 ++ /quantize.c/1.69.2.1.2.2/Thu Apr 6 18:50:51 2000//Tlame3_70 ++ /quantize.h/1.5/Thu Apr 6 18:50:51 2000//Tlame3_70 ++ /reservoir.c/1.11.4.1/Thu Apr 6 18:50:51 2000//Tlame3_70 ++ /reservoir.h/1.4/Tue Mar 21 23:02:17 2000//Tlame3_70 ++ /rtp.c/1.3/Thu Jan 27 04:35:48 2000//Tlame3_70 ++ /rtp.h/1.2/Tue Jan 25 17:48:23 2000//Tlame3_70 ++ /tables.c/1.4.2.1/Thu Apr 6 18:50:52 2000//Tlame3_70 ++ /tables.h/1.2/Thu Apr 6 18:50:52 2000//Tlame3_70 ++ /takehiro.c/1.33.2.1.2.2/Thu Apr 6 18:50:52 2000//Tlame3_70 ++ /testcase.mp3/1.11/Thu Apr 6 18:50:52 2000/-kb/Tlame3_70 ++ /testcase.wav/1.1/Sun Jan 9 20:55:56 2000/-kb/Tlame3_70 ++ /timestatus.c/1.4/Thu Mar 16 16:44:22 2000//Tlame3_70 ++ /timestatus.h/1.3/Tue Mar 14 20:45:04 2000//Tlame3_70 ++ /util.c/1.20.4.1/Thu Apr 6 18:50:52 2000//Tlame3_70 ++ /util.h/1.15/Thu Apr 6 18:50:52 2000//Tlame3_70 ++ /vbrquantize.c/1.20/Thu Apr 6 18:50:52 2000//Tlame3_70 ++ /version.c/1.3/Fri Mar 10 20:56:35 2000//Tlame3_70 ++ /version.h/1.12.2.1.2.1.4.4/Thu Apr 6 18:50:52 2000//Tlame3_70 +diff -r -c -N encoder/CVS/Repository lame3.70/CVS/Repository +*** encoder/CVS/Repository Wed Dec 31 17:00:00 1969 +--- lame3.70/CVS/Repository Thu Apr 6 12:50:20 2000 +*************** +*** 0 **** +--- 1 ---- ++ lame +diff -r -c -N encoder/CVS/Root lame3.70/CVS/Root +*** encoder/CVS/Root Wed Dec 31 17:00:00 1969 +--- lame3.70/CVS/Root Thu Apr 6 12:50:20 2000 +*************** +*** 0 **** +--- 1 ---- ++ markt@cvs.lame.sourceforge.net:/cvsroot/lame +diff -r -c -N encoder/CVS/Tag lame3.70/CVS/Tag +*** encoder/CVS/Tag Wed Dec 31 17:00:00 1969 +--- lame3.70/CVS/Tag Thu Apr 6 12:50:52 2000 +*************** +*** 0 **** +--- 1 ---- ++ Tlame3_70 +diff -r -c -N encoder/Dll/.cvsignore lame3.70/Dll/.cvsignore +*** encoder/Dll/.cvsignore Wed Dec 31 17:00:00 1969 +--- lame3.70/Dll/.cvsignore Tue Nov 30 12:30:46 1999 +*************** +*** 0 **** +--- 1,9 ---- ++ *.d ++ Debug ++ DebugGTK ++ Release ++ ReleaseGTK ++ *.ncb ++ *.plg ++ *.opt ++ *.dll +diff -r -c -N encoder/Dll/BladeMP3EncDLL.c lame3.70/Dll/BladeMP3EncDLL.c +*** encoder/Dll/BladeMP3EncDLL.c Wed Dec 31 17:00:00 1969 +--- lame3.70/Dll/BladeMP3EncDLL.c Thu Apr 6 12:50:52 2000 +*************** +*** 0 **** +--- 1,448 ---- ++ /* ++ * Blade DLL Interface for LAME. ++ * ++ * Copyright (c) 1999 A.L. Faber ++ * ++ * This library is free software; you can redistribute it and/or ++ * modify it under the terms of the GNU Library General Public ++ * License as published by the Free Software Foundation; either ++ * version 2 of the License, or (at your option) any later version. ++ * ++ * This library is distributed in the hope that it will be useful, ++ * but WITHOUT ANY WARRANTY; without even the implied warranty of ++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU ++ * Library General Public License for more details. ++ * ++ * You should have received a copy of the GNU Library General Public ++ * License along with this library; if not, write to the ++ * Free Software Foundation, Inc., 59 Temple Place - Suite 330, ++ * Boston, MA 02111-1307, USA. ++ */ ++ ++ #include "BladeMP3EncDLL.h" ++ #include <assert.h> ++ #include "version.h" ++ #include "VbrTag.h" ++ #include "lame.h" ++ /* ++ #include "get_audio.h" ++ #include "globalflags.h" ++ #include "machine.h" ++ #include "util.h" ++ */ ++ ++ #ifdef _DEBUG ++ #define _DEBUGDLL 1 ++ #endif ++ ++ ++ const int MAJORVERSION=1; ++ const int MINORVERSION=05; ++ const int CURRENT_STRUCT_VERSION=1; ++ const int CURRENT_STRUCT_SIZE=sizeof(BE_CONFIG); ++ ++ ++ ++ ++ ++ // Local variables ++ static int nPsychoModel=2; ++ static BOOL bFirstFrame=TRUE; ++ static DWORD dwSampleBufferSize=0; ++ ++ ++ #ifdef _DEBUGDLL ++ void dump_config( char *inPath, char *outPath); ++ #endif ++ ++ lame_global_flags gf; ++ ++ static void InitParams() ++ { ++ bFirstFrame=TRUE; ++ lame_init(&gf); ++ ++ } ++ ++ ++ ++ ++ __declspec(dllexport) BE_ERR beInitStream(PBE_CONFIG pbeConfig, PDWORD dwSamples, PDWORD dwBufferSize, PHBE_STREAM phbeStream) ++ { ++ int nDllArgC=0; ++ BE_CONFIG lameConfig; ++ ++ InitParams(); ++ ++ // clear out structure ++ memset(&lameConfig,0x00,CURRENT_STRUCT_SIZE); ++ ++ // Check if this is a regular BLADE_ENCODER header ++ if (pbeConfig->dwConfig!=BE_CONFIG_LAME) ++ { ++ int nCRC=pbeConfig->format.mp3.bCRC; ++ int nVBR=(nCRC>>12)&0x0F; ++ ++ // Copy parameter from old Blade structure ++ lameConfig.format.LHV1.dwSampleRate =pbeConfig->format.mp3.dwSampleRate; ++ //for low bitrates, LAME will automatically downsample for better ++ //sound quality. Forcing output samplerate = input samplerate is not a good idea ++ //unless the user specifically requests it: ++ //lameConfig.format.LHV1.dwReSampleRate=pbeConfig->format.mp3.dwSampleRate; ++ lameConfig.format.LHV1.nMode =(pbeConfig->format.mp3.byMode&0x0F); ++ lameConfig.format.LHV1.dwBitrate =pbeConfig->format.mp3.wBitrate; ++ lameConfig.format.LHV1.bPrivate =pbeConfig->format.mp3.bPrivate; ++ lameConfig.format.LHV1.bOriginal =pbeConfig->format.mp3.bOriginal; ++ lameConfig.format.LHV1.bCRC =nCRC&0x01; ++ lameConfig.format.LHV1.bCopyright =pbeConfig->format.mp3.bCopyright; ++ ++ // Fill out the unknowns ++ lameConfig.format.LHV1.dwStructSize=CURRENT_STRUCT_VERSION; ++ lameConfig.format.LHV1.dwStructVersion=CURRENT_STRUCT_SIZE; ++ ++ // Get VBR setting from fourth nibble ++ if (nVBR>0) ++ { ++ lameConfig.format.LHV1.bWriteVBRHeader=TRUE; ++ lameConfig.format.LHV1.bEnableVBR=TRUE; ++ lameConfig.format.LHV1.nVBRQuality=nVBR-1; ++ } ++ ++ // Get Quality from third nibble ++ lameConfig.format.LHV1.nQuality=(MPEG_QUALITY)((nCRC>>8)&0x0F); ++ ++ } ++ else ++ { ++ // Copy the parameters ++ memcpy(&lameConfig,pbeConfig,pbeConfig->format.LHV1.dwStructSize); ++ } ++ ++ ++ // Not used, always assign stream 1 ++ *phbeStream=1; ++ ++ ++ ++ // --------------- Set arguments to LAME encoder ------------------------- ++ ++ // Set zero argument, the filename ++ //strcpy(DllArgV[nDllArgC++],"LameDLLEncoder"); ++ ++ switch (lameConfig.format.LHV1.nMode) ++ { ++ case BE_MP3_MODE_STEREO: ++ gf.mode=0; ++ gf.mode_fixed=1; /* dont allow LAME to change the mode */ ++ gf.num_channels=2; ++ break; ++ case BE_MP3_MODE_JSTEREO: ++ gf.mode=1; ++ gf.mode_fixed=1; ++ gf.num_channels=2; ++ break; ++ case BE_MP3_MODE_MONO: ++ gf.mode=3; ++ gf.mode_fixed=1; ++ gf.num_channels=1; ++ break; ++ case BE_MP3_MODE_DUALCHANNEL: ++ gf.force_ms=1; ++ gf.mode=1; ++ gf.mode_fixed=1; ++ gf.num_channels=2; ++ break; ++ default: ++ { ++ char lpszError[255]; ++ sprintf(lpszError,"Invalid lameConfig.format.LHV1.nMode, value is %d\n",lameConfig.format.LHV1.nMode); ++ OutputDebugString(lpszError); ++ return BE_ERR_INVALID_FORMAT_PARAMETERS; ++ } ++ } ++ ++ switch (lameConfig.format.LHV1.nQuality) ++ { ++ case NORMAL_QUALITY: // Nothing special ++ break; ++ case LOW_QUALITY: // -f flag ++ gf.quality=9; ++ break; ++ case HIGH_QUALITY: // -h flag for high qualtiy ++ gf.quality=2; ++ break; ++ case VOICE_QUALITY: // --voice flag for experimental voice mode ++ gf.lowpassfreq=12000; ++ gf.VBR_max_bitrate_kbps=160; ++ gf.no_short_blocks=1; ++ break; ++ } ++ ++ if (lameConfig.format.LHV1.bEnableVBR) ++ { ++ // 0=no vbr 1..10 is VBR quality setting -1 ++ gf.VBR=1; ++ gf.VBR_q=lameConfig.format.LHV1.nVBRQuality; ++ } ++ ++ // Set frequency ++ gf.in_samplerate=lameConfig.format.LHV1.dwSampleRate; ++ ++ // Set frequency resampling rate, if specified ++ if (lameConfig.format.LHV1.dwReSampleRate>0) ++ gf.out_samplerate=lameConfig.format.LHV1.dwReSampleRate; ++ ++ ++ // Set bitrate. (CDex users always specify bitrate=Min bitrate when using VBR) ++ gf.brate=lameConfig.format.LHV1.dwBitrate; ++ gf.VBR_min_bitrate_kbps=gf.brate; ++ ++ // Set Maxbitrate, if specified ++ if (lameConfig.format.LHV1.dwMaxBitrate>0) ++ gf.VBR_max_bitrate_kbps=lameConfig.format.LHV1.dwMaxBitrate; ++ ++ // Set copyright flag? ++ if (lameConfig.format.LHV1.bCopyright) ++ gf.copyright=1; ++ ++ // Do we have to tag it as non original ++ if (!lameConfig.format.LHV1.bOriginal) ++ gf.original=0; ++ ++ // Add CRC? ++ if (lameConfig.format.LHV1.bCRC) ++ gf.error_protection=1; ++ ++ gf.silent=1; /* disable status ouput */ ++ ++ // Set private bit? ++ if (lameConfig.format.LHV1.bPrivate) ++ { ++ gf.extension = 0; ++ } ++ else ++ { ++ gf.extension = 1; ++ } ++ ++ lame_init_params(&gf); ++ ++ //LAME encoding call will accept any number of samples. ++ *dwSamples=1152*gf.num_channels; ++ ++ // Set the input sample buffer size, so we know what we can expect ++ dwSampleBufferSize=*dwSamples; ++ ++ // Set MP3 buffer size ++ // conservative estimate ++ *dwBufferSize=1.25*(*dwSamples/gf.num_channels) + 7200; ++ ++ ++ #ifdef _DEBUGDLL ++ dump_config(gf.inPath,gf.outPath); ++ #endif ++ ++ // Everything went OK, thus return SUCCESSFUL ++ return BE_ERR_SUCCESSFUL; ++ } ++ ++ ++ ++ __declspec(dllexport) BE_ERR beDeinitStream(HBE_STREAM hbeStream, PBYTE pOutput, PDWORD pdwOutput) ++ { ++ ++ *pdwOutput = lame_encode_finish(&gf,pOutput,0); ++ ++ if (*pdwOutput<0) { ++ *pdwOutput=0; ++ return BE_ERR_BUFFER_TOO_SMALL; ++ } ++ ++ return BE_ERR_SUCCESSFUL; ++ } ++ ++ ++ __declspec(dllexport) BE_ERR beCloseStream(HBE_STREAM hbeStream) ++ { ++ // DeInit encoder ++ // return DeInitEncoder(); ++ return BE_ERR_SUCCESSFUL; ++ } ++ ++ ++ ++ __declspec(dllexport) VOID beVersion(PBE_VERSION pbeVersion) ++ { ++ // DLL Release date ++ char lpszDate[20]; ++ char lpszTemp[5]; ++ ++ ++ // Set DLL interface version ++ pbeVersion->byDLLMajorVersion=MAJORVERSION; ++ pbeVersion->byDLLMinorVersion=MINORVERSION; ++ ++ // Set Engine version number (Same as Lame version) ++ pbeVersion->byMajorVersion=LAME_MAJOR_VERSION; ++ pbeVersion->byMinorVersion=LAME_MINOR_VERSION; ++ ++ // Get compilation date ++ strcpy(lpszDate,__DATE__); ++ ++ // Get the first three character, which is the month ++ strncpy(lpszTemp,lpszDate,3); ++ ++ // Set month ++ if (strcmp(lpszTemp,"Jan")==0) pbeVersion->byMonth=1; ++ if (strcmp(lpszTemp,"Feb")==0) pbeVersion->byMonth=2; ++ if (strcmp(lpszTemp,"Mar")==0) pbeVersion->byMonth=3; ++ if (strcmp(lpszTemp,"Apr")==0) pbeVersion->byMonth=4; ++ if (strcmp(lpszTemp,"May")==0) pbeVersion->byMonth=5; ++ if (strcmp(lpszTemp,"Jun")==0) pbeVersion->byMonth=6; ++ if (strcmp(lpszTemp,"Jul")==0) pbeVersion->byMonth=7; ++ if (strcmp(lpszTemp,"Aug")==0) pbeVersion->byMonth=8; ++ if (strcmp(lpszTemp,"Sep")==0) pbeVersion->byMonth=9; ++ if (strcmp(lpszTemp,"Oct")==0) pbeVersion->byMonth=10; ++ if (strcmp(lpszTemp,"Nov")==0) pbeVersion->byMonth=11; ++ if (strcmp(lpszTemp,"Dec")==0) pbeVersion->byMonth=12; ++ ++ // Get day of month string (char [4..5]) ++ pbeVersion->byDay=atoi(lpszDate+4); ++ ++ // Get year of compilation date (char [7..10]) ++ pbeVersion->wYear=atoi(lpszDate+7); ++ ++ memset(pbeVersion->zHomepage,0x00,BE_MAX_HOMEPAGE); ++ ++ strcpy(pbeVersion->zHomepage,"http://www.sulaco.org/mp3/"); ++ } ++ ++ __declspec(dllexport) BE_ERR beEncodeChunk(HBE_STREAM hbeStream, DWORD nSamples, ++ PSHORT pSamples, PBYTE pOutput, PDWORD pdwOutput) ++ { ++ ++ // Encode it ++ int dwSamples; ++ dwSamples=nSamples/gf.num_channels; ++ ++ // old versions of lame_enc.dll required exactly 1152 samples ++ // and worked even if nSamples accidently set to 2304 ++ // simulate this behavoir: ++ if (gf.num_channels==1 && nSamples == 2304) ++ dwSamples/=2; ++ ++ *pdwOutput=lame_encode_buffer_interleaved(&gf,pSamples,dwSamples,pOutput,0); ++ ++ ++ if (*pdwOutput<0) { ++ *pdwOutput=0; ++ return BE_ERR_BUFFER_TOO_SMALL; ++ } ++ ++ return BE_ERR_SUCCESSFUL; ++ } ++ ++ ++ __declspec(dllexport) BE_ERR beWriteVBRHeader(LPCSTR lpszFileName) ++ { ++ if (gf.bWriteVbrTag) ++ { ++ // Calculate relative quality of VBR stream ++ // 0=best, 100=worst ++ int nQuality=gf.VBR_q*100/9; ++ ++ // Write Xing header again ++ return PutVbrTag((LPSTR)lpszFileName,nQuality,1-gf.version); ++ } ++ return BE_ERR_INVALID_FORMAT_PARAMETERS; ++ } ++ ++ ++ BOOL APIENTRY DllMain(HANDLE hModule, ++ DWORD ul_reason_for_call, ++ LPVOID lpReserved) ++ { ++ switch( ul_reason_for_call ) ++ { ++ case DLL_PROCESS_ATTACH: ++ #ifdef _DEBUGDLL ++ OutputDebugString("Attach Process \n"); ++ #endif ++ break; ++ case DLL_THREAD_ATTACH: ++ #ifdef _DEBUGDLL ++ OutputDebugString("Attach Thread \n"); ++ #endif ++ break; ++ case DLL_THREAD_DETACH: ++ #ifdef _DEBUGDLL ++ OutputDebugString("Detach Thread \n"); ++ #endif ++ break; ++ case DLL_PROCESS_DETACH: ++ #ifdef _DEBUGDLL ++ OutputDebugString("Detach Process \n"); ++ #endif ++ break; ++ } ++ return TRUE; ++ } ++ ++ ++ #ifdef _DEBUGDLL ++ void dump_config( char *inPath, char *outPath) ++ { ++ char strTmp[255]; ++ ++ OutputDebugString("Encoding configuration:\n"); ++ ++ ++ sprintf(strTmp,"Write VBR Header=%s\n",(gf.bWriteVbrTag)?"Yes":"No"); ++ OutputDebugString(strTmp); ++ ++ sprintf(strTmp,"version=%d\n",gf.version); ++ OutputDebugString(strTmp); ++ ++ ++ sprintf(strTmp,"Layer=3 mode=%d \n",gf.mode); ++ OutputDebugString(strTmp); ++ ++ ++ sprintf(strTmp,"samp frq=%.1f kHz total bitrate=%d kbps\n",gf.in_samplerate/1000.0); ++ OutputDebugString(strTmp); ++ ++ sprintf(strTmp,"de-emph=%d c/right=%d orig=%d errprot=%s\n",gf.emphasis, gf.copyright, gf.original,((gf.error_protection) ? "on" : "off")); ++ OutputDebugString(strTmp); ++ ++ // sprintf(strTmp,"16 Khz cut off is %s\n",(0)?"enabled":"disabled"); ++ // OutputDebugString(strTmp); ++ ++ sprintf(strTmp,"Fast mode is %s\n",(gf.quality==9)?"enabled":"disabled"); ++ OutputDebugString(strTmp); ++ ++ sprintf(strTmp,"Force ms %s\n",(gf.force_ms)?"enabled":"disabled"); ++ OutputDebugString(strTmp); ++ ++ // sprintf(strTmp,"GPsycho acoustic model is %s\n",(gpsycho)?"enabled":"disabled"); ++ // OutputDebugString(strTmp); ++ ++ sprintf(strTmp,"VRB is %s, VBR_q value is %d\n",(gf.VBR)?"enabled":"disabled",gf.VBR_q); ++ OutputDebugString(strTmp); ++ ++ sprintf(strTmp,"input file: '%s' output file: '%s'\n", inPath, outPath); ++ OutputDebugString(strTmp); ++ ++ // sprintf(strTmp,"Voice mode %s\n",(voice_mode)?"enabled":"disabled"); ++ // OutputDebugString(strTmp); ++ ++ sprintf(strTmp,"Encoding as %.1f kHz %d kbps %d MPEG-%d LayerIII file\n",gf.out_samplerate/1000.0,gf.brate,gf.mode,3 - gf.mode_gr); ++ OutputDebugString(strTmp); ++ } ++ ++ ++ void DispErr(LPSTR strErr) ++ { ++ MessageBox(NULL,strErr,"",MB_OK); ++ } ++ ++ #endif +diff -r -c -N encoder/Dll/BladeMP3EncDLL.def lame3.70/Dll/BladeMP3EncDLL.def +*** encoder/Dll/BladeMP3EncDLL.def Wed Dec 31 17:00:00 1969 +--- lame3.70/Dll/BladeMP3EncDLL.def Sun Nov 28 15:21:47 1999 +*************** +*** 0 **** +--- 1,9 ---- ++ LIBRARY lame_enc.DLL ++ EXPORTS ++ ++ beInitStream @1 ++ beEncodeChunk @2 ++ beDeinitStream @3 ++ beCloseStream @4 ++ beVersion @5 ++ beWriteVBRHeader @6 +diff -r -c -N encoder/Dll/BladeMP3EncDLL.h lame3.70/Dll/BladeMP3EncDLL.h +*** encoder/Dll/BladeMP3EncDLL.h Wed Dec 31 17:00:00 1969 +--- lame3.70/Dll/BladeMP3EncDLL.h Thu Apr 6 12:50:52 2000 +*************** +*** 0 **** +--- 1,196 ---- ++ /* ++ ++ bladedll.h ++ ++ +++++++++++++++++++++++++++ ++ + Blade's Encoder DLL + ++ +++++++++++++++++++++++++++ ++ ++ ------------------------------------------------------ ++ - Version 1.00 (7 November 1998) - Jukka Poikolainen - ++ ------------------------------------------------------ ++ ++ Initial version ++ ++ ------------------------------------------------------ ++ - Version x.xx (x xxxxxxxx xxxx) - xxxxx xxxxxxxxxxx - ++ ------------------------------------------------------ ++ ++ */ ++ ++ #ifndef ___BLADEDLL_H_INCLUDED___ ++ #define ___BLADEDLL_H_INCLUDED___ ++ ++ #pragma pack(push) ++ #pragma pack(1) ++ ++ /* encoding formats */ ++ ++ #define BE_CONFIG_MP3 0 ++ #define BE_CONFIG_LAME 256 ++ ++ /* type definitions */ ++ ++ typedef unsigned long HBE_STREAM; ++ typedef HBE_STREAM *PHBE_STREAM; ++ typedef unsigned long BE_ERR; ++ ++ /* error codes */ ++ ++ #define BE_ERR_SUCCESSFUL 0x00000000 ++ #define BE_ERR_INVALID_FORMAT 0x00000001 ++ #define BE_ERR_INVALID_FORMAT_PARAMETERS 0x00000002 ++ #define BE_ERR_NO_MORE_HANDLES 0x00000003 ++ #define BE_ERR_INVALID_HANDLE 0x00000004 ++ #define BE_ERR_BUFFER_TOO_SMALL 0x00000005 ++ ++ /* other constants */ ++ ++ #define BE_MAX_HOMEPAGE 256 ++ ++ /* format specific variables */ ++ ++ #define BE_MP3_MODE_STEREO 0 ++ #define BE_MP3_MODE_JSTEREO 1 ++ #define BE_MP3_MODE_DUALCHANNEL 2 ++ #define BE_MP3_MODE_MONO 3 ++ ++ #define MPEG1 1 ++ #define MPEG2 0 ++ ++ #ifdef _BLADEDLL ++ #undef FLOAT ++ #include <Windows.h> ++ #endif ++ ++ ++ typedef enum ++ { ++ NORMAL_QUALITY=0, ++ LOW_QUALITY, ++ HIGH_QUALITY, ++ VOICE_QUALITY ++ } MPEG_QUALITY; ++ ++ typedef struct { ++ DWORD dwConfig; // BE_CONFIG_XXXXX ++ // Currently only BE_CONFIG_MP3 is supported ++ union { ++ ++ struct { ++ ++ DWORD dwSampleRate; // 48000, 44100 and 32000 allowed ++ BYTE byMode; // BE_MP3_MODE_STEREO, BE_MP3_MODE_DUALCHANNEL, BE_MP3_MODE_MONO ++ WORD wBitrate; // 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256 and 320 allowed ++ BOOL bPrivate; ++ BOOL bCRC; ++ BOOL bCopyright; ++ BOOL bOriginal; ++ ++ } mp3; // BE_CONFIG_MP3 ++ ++ struct ++ { ++ // STRUCTURE INFORMATION ++ DWORD dwStructVersion; ++ DWORD dwStructSize; ++ ++ // BASIC ENCODER SETTINGS ++ DWORD dwSampleRate; // ALLOWED SAMPLERATE VALUES DEPENDS ON dwMPEGVersion ++ DWORD dwReSampleRate; // DOWNSAMPLERATE, 0=ENCODER DECIDES ++ INT nMode; // BE_MP3_MODE_STEREO, BE_MP3_MODE_DUALCHANNEL, BE_MP3_MODE_MONO ++ DWORD dwBitrate; // CBR bitrate, VBR min bitrate ++ DWORD dwMaxBitrate; // CBR ignored, VBR Max bitrate ++ MPEG_QUALITY nQuality; // Quality setting (NORMAL,HIGH,LOW,VOICE) ++ DWORD dwMpegVersion; // MPEG-1 OR MPEG-2 ++ DWORD dwPsyModel; // FUTURE USE, SET TO 0 ++ DWORD dwEmphasis; // FUTURE USE, SET TO 0 ++ ++ // BIT STREAM SETTINGS ++ BOOL bPrivate; // Set Private Bit (TRUE/FALSE) ++ BOOL bCRC; // Insert CRC (TRUE/FALSE) ++ BOOL bCopyright; // Set Copyright Bit (TRUE/FALSE) ++ BOOL bOriginal; // Set Original Bit (TRUE/FALSE) ++ ++ // VBR STUFF ++ BOOL bWriteVBRHeader; // WRITE XING VBR HEADER (TRUE/FALSE) ++ BOOL bEnableVBR; // USE VBR ENCODING (TRUE/FALSE) ++ INT nVBRQuality; // VBR QUALITY 0..9 ++ ++ BYTE btReserved[255]; // FUTURE USE, SET TO 0 ++ ++ } LHV1; // LAME header version 1 ++ ++ struct { ++ ++ DWORD dwSampleRate; ++ BYTE byMode; ++ WORD wBitrate; ++ BYTE byEncodingMethod; ++ ++ } aac; ++ ++ } format; ++ ++ } BE_CONFIG, *PBE_CONFIG; ++ ++ ++ typedef struct { ++ ++ // BladeEnc DLL Version number ++ ++ BYTE byDLLMajorVersion; ++ BYTE byDLLMinorVersion; ++ ++ // BladeEnc Engine Version Number ++ ++ BYTE byMajorVersion; ++ BYTE byMinorVersion; ++ ++ // DLL Release date ++ ++ BYTE byDay; ++ BYTE byMonth; ++ WORD wYear; ++ ++ // BladeEnc Homepage URL ++ ++ CHAR zHomepage[BE_MAX_HOMEPAGE + 1]; ++ ++ } BE_VERSION, *PBE_VERSION; ++ ++ #ifndef _BLADEDLL ++ ++ typedef BE_ERR (*BEINITSTREAM) (PBE_CONFIG, PDWORD, PDWORD, PHBE_STREAM); ++ typedef BE_ERR (*BEENCODECHUNK) (HBE_STREAM, DWORD, PSHORT, PBYTE, PDWORD); ++ typedef BE_ERR (*BEDEINITSTREAM) (HBE_STREAM, PBYTE, PDWORD); ++ typedef BE_ERR (*BECLOSESTREAM) (HBE_STREAM); ++ typedef VOID (*BEVERSION) (PBE_VERSION); ++ ++ #define TEXT_BEINITSTREAM "beInitStream" ++ #define TEXT_BEENCODECHUNK "beEncodeChunk" ++ #define TEXT_BEDEINITSTREAM "beDeinitStream" ++ #define TEXT_BECLOSESTREAM "beCloseStream" ++ #define TEXT_BEVERSION "beVersion" ++ ++ /* ++ BE_ERR beInitStream(PBE_CONFIG pbeConfig, PDWORD dwSamples, PDWORD dwBufferSize, PHBE_STREAM phbeStream); ++ BE_ERR beEncodeChunk(HBE_STREAM hbeStream, DWORD nSamples, PSHORT pSamples, PBYTE pOutput, PDWORD pdwOutput); ++ BE_ERR beDeinitStream(HBE_STREAM hbeStream, PBYTE pOutput, PDWORD pdwOutput); ++ BE_ERR beCloseStream(HBE_STREAM hbeStream); ++ VOID beVersion(PBE_VERSION pbeVersion); ++ */ ++ ++ #else ++ ++ __declspec(dllexport) BE_ERR beInitStream(PBE_CONFIG pbeConfig, PDWORD dwSamples, PDWORD dwBufferSize, PHBE_STREAM phbeStream); ++ __declspec(dllexport) BE_ERR beEncodeChunk(HBE_STREAM hbeStream, DWORD nSamples, PSHORT pSamples, PBYTE pOutput, PDWORD pdwOutput); ++ __declspec(dllexport) BE_ERR beDeinitStream(HBE_STREAM hbeStream, PBYTE pOutput, PDWORD pdwOutput); ++ __declspec(dllexport) BE_ERR beCloseStream(HBE_STREAM hbeStream); ++ __declspec(dllexport) VOID beVersion(PBE_VERSION pbeVersion); ++ ++ #endif ++ ++ #pragma pack(pop) ++ ++ #endif +diff -r -c -N encoder/Dll/CVS/Entries lame3.70/Dll/CVS/Entries +*** encoder/Dll/CVS/Entries Wed Dec 31 17:00:00 1969 +--- lame3.70/Dll/CVS/Entries Thu Apr 6 12:50:52 2000 +*************** +*** 0 **** +--- 1,7 ---- ++ /.cvsignore/1.1/Tue Nov 30 19:30:46 1999//Tlame3_70 ++ /BladeMP3EncDLL.c/1.16.4.8/Thu Apr 6 18:50:52 2000//Tlame3_70 ++ /BladeMP3EncDLL.def/1.2/Sun Nov 28 22:21:47 1999//Tlame3_70 ++ /BladeMP3EncDLL.h/1.3.8.1/Thu Apr 6 18:50:52 2000//Tlame3_70 ++ /MP3EncDll.dsp/1.13/Thu Apr 6 18:50:52 2000//Tlame3_70 ++ /MP3EncDll.dsw/1.4/Wed Mar 15 01:20:31 2000//Tlame3_70 ++ D +diff -r -c -N encoder/Dll/CVS/Repository lame3.70/Dll/CVS/Repository +*** encoder/Dll/CVS/Repository Wed Dec 31 17:00:00 1969 +--- lame3.70/Dll/CVS/Repository Thu Apr 6 12:50:35 2000 +*************** +*** 0 **** +--- 1 ---- ++ lame/Dll +diff -r -c -N encoder/Dll/CVS/Root lame3.70/Dll/CVS/Root +*** encoder/Dll/CVS/Root Wed Dec 31 17:00:00 1969 +--- lame3.70/Dll/CVS/Root Thu Apr 6 12:50:35 2000 +*************** +*** 0 **** +--- 1 ---- ++ markt@cvs.lame.sourceforge.net:/cvsroot/lame +diff -r -c -N encoder/Dll/CVS/Tag lame3.70/Dll/CVS/Tag +*** encoder/Dll/CVS/Tag Wed Dec 31 17:00:00 1969 +--- lame3.70/Dll/CVS/Tag Thu Apr 6 12:50:52 2000 +*************** +*** 0 **** +--- 1 ---- ++ Tlame3_70 +diff -r -c -N encoder/Dll/MP3EncDll.dsp lame3.70/Dll/MP3EncDll.dsp +*** encoder/Dll/MP3EncDll.dsp Wed Dec 31 17:00:00 1969 +--- lame3.70/Dll/MP3EncDll.dsp Thu Apr 6 12:50:52 2000 +*************** +*** 0 **** +--- 1,298 ---- ++ # Microsoft Developer Studio Project File - Name="MP3EncDll" - Package Owner=<4> ++ # Microsoft Developer Studio Generated Build File, Format Version 5.00 ++ # ** DO NOT EDIT ** ++ ++ # TARGTYPE "Win32 (x86) Dynamic-Link Library" 0x0102 ++ ++ CFG=MP3EncDll - Win32 Debug ++ !MESSAGE This is not a valid makefile. To build this project using NMAKE, ++ !MESSAGE use the Export Makefile command and run ++ !MESSAGE ++ !MESSAGE NMAKE /f "MP3EncDll.mak". ++ !MESSAGE ++ !MESSAGE You can specify a configuration when running NMAKE ++ !MESSAGE by defining the macro CFG on the command line. For example: ++ !MESSAGE ++ !MESSAGE NMAKE /f "MP3EncDll.mak" CFG="MP3EncDll - Win32 Debug" ++ !MESSAGE ++ !MESSAGE Possible choices for configuration are: ++ !MESSAGE ++ !MESSAGE "MP3EncDll - Win32 Release" (based on\ ++ "Win32 (x86) Dynamic-Link Library") ++ !MESSAGE "MP3EncDll - Win32 Debug" (based on\ ++ "Win32 (x86) Dynamic-Link Library") ++ !MESSAGE ++ ++ # Begin Project ++ # PROP Scc_ProjName "" ++ # PROP Scc_LocalPath "" ++ CPP=xicl.exe ++ MTL=midl.exe ++ RSC=rc.exe ++ ++ !IF "$(CFG)" == "MP3EncDll - Win32 Release" ++ ++ # PROP BASE Use_MFC 0 ++ # PROP BASE Use_Debug_Libraries 0 ++ # PROP BASE Output_Dir "Release" ++ # PROP BASE Intermediate_Dir "Release" ++ # PROP BASE Target_Dir "" ++ # PROP Use_MFC 0 ++ # PROP Use_Debug_Libraries 0 ++ # PROP Output_Dir "Release" ++ # PROP Intermediate_Dir "Release" ++ # PROP Ignore_Export_Lib 0 ++ # PROP Target_Dir "" ++ # ADD BASE CPP /nologo /MT /W3 /GX /O2 /D "WIN32" /D "NDEBUG" /D "_WINDOWS" /YX /FD /c ++ # ADD CPP /nologo /Zp2 /MT /W3 /GX /Ox /Ot /Og /Oy /Ob2 /I "..\\" /D "NDEBUG" /D "WIN32" /D "_WINDOWS" /D "_BLADEDLL" /YX /FD -Qmem -Qip /c ++ # SUBTRACT CPP /Oa /Ow /Oi /Os ++ # ADD BASE MTL /nologo /D "NDEBUG" /mktyplib203 /o NUL /win32 ++ # ADD MTL /nologo /D "NDEBUG" /mktyplib203 /o NUL /win32 ++ # ADD BASE RSC /l 0x409 /d "NDEBUG" ++ # ADD RSC /l 0x409 /d "NDEBUG" ++ BSC32=bscmake.exe ++ # ADD BASE BSC32 /nologo ++ # ADD BSC32 /nologo ++ LINK32=xilink.exe ++ # ADD BASE LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:windows /dll /machine:I386 ++ # ADD LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:windows /dll /map /machine:I386 /out:"..\..\lame_enc.dll" ++ ++ !ELSEIF "$(CFG)" == "MP3EncDll - Win32 Debug" ++ ++ # PROP BASE Use_MFC 0 ++ # PROP BASE Use_Debug_Libraries 1 ++ # PROP BASE Output_Dir "Debug" ++ # PROP BASE Intermediate_Dir "Debug" ++ # PROP BASE Target_Dir "" ++ # PROP Use_MFC 0 ++ # PROP Use_Debug_Libraries 1 ++ # PROP Output_Dir "Debug" ++ # PROP Intermediate_Dir "Debug" ++ # PROP Ignore_Export_Lib 0 ++ # PROP Target_Dir "" ++ # ADD BASE CPP /nologo /MTd /W3 /Gm /GX /Zi /Od /D "WIN32" /D "_DEBUG" /D "_WINDOWS" /YX /FD /c ++ # ADD CPP /nologo /Zp2 /MTd /W3 /Gm /GX /Zi /Od /I "..\\" /D "_DEBUG" /D "WIN32" /D "_WINDOWS" /D "_BLADEDLL" /YX /FD /c ++ # ADD BASE MTL /nologo /D "_DEBUG" /mktyplib203 /o NUL /win32 ++ # ADD MTL /nologo /D "_DEBUG" /mktyplib203 /o NUL /win32 ++ # ADD BASE RSC /l 0x409 /d "_DEBUG" ++ # ADD RSC /l 0x409 /d "_DEBUG" ++ BSC32=bscmake.exe ++ # ADD BASE BSC32 /nologo ++ # ADD BSC32 /nologo ++ LINK32=xilink.exe ++ # ADD BASE LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:windows /dll /debug /machine:I386 /pdbtype:sept ++ # ADD LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:windows /dll /debug /machine:I386 /out:"..\..\lame_enc.dll" /pdbtype:sept ++ # SUBTRACT LINK32 /map ++ ++ !ENDIF ++ ++ # Begin Target ++ ++ # Name "MP3EncDll - Win32 Release" ++ # Name "MP3EncDll - Win32 Debug" ++ # Begin Source File ++ ++ SOURCE=.\BladeMP3EncDLL.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\BladeMP3EncDLL.def ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\BladeMP3EncDLL.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\brhist.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\brhist.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\encoder.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\fft.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\fft.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\formatBitstream.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\formatBitstream.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\gpkplotting.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\gpkplotting.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\gtkanal.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\gtkanal.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\huffman.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\id3tag.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\id3tag.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\ieeefloat.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\ieeefloat.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE="..\l3bitstream-pvt.h" ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\l3bitstream.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\l3bitstream.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\l3side.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\lame.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\lame.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\newmdct.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\newmdct.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\parse.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\portableio.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\portableio.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\psymodel.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\psymodel.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE="..\quantize-pvt.c" ++ # End Source File ++ # Begin Source File ++ ++ SOURCE="..\quantize-pvt.h" ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\quantize.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\quantize.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\reservoir.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\reservoir.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\sqrttab.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\tables.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\tables.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\timestatus.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\timestatus.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\takehiro.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\util.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\util.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\VbrTag.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\VbrTag.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\version.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\version.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=..\wavheader.h ++ # End Source File ++ # End Target ++ # End Project +diff -r -c -N encoder/Dll/MP3EncDll.dsw lame3.70/Dll/MP3EncDll.dsw +*** encoder/Dll/MP3EncDll.dsw Wed Dec 31 17:00:00 1969 +--- lame3.70/Dll/MP3EncDll.dsw Tue Mar 14 18:20:31 2000 +*************** +*** 0 **** +--- 1,29 ---- ++ Microsoft Developer Studio Workspace File, Format Version 5.00 ++ # WARNING: DO NOT EDIT OR DELETE THIS WORKSPACE FILE! ++ ++ ############################################################################### ++ ++ Project: "MP3EncDll"=.\MP3EncDll.dsp - Package Owner=<4> ++ ++ Package=<5> ++ {{{ ++ }}} ++ ++ Package=<4> ++ {{{ ++ }}} ++ ++ ############################################################################### ++ ++ Global: ++ ++ Package=<5> ++ {{{ ++ }}} ++ ++ Package=<3> ++ {{{ ++ }}} ++ ++ ############################################################################### ++ +diff -r -c -N encoder/HACKING lame3.70/HACKING +*** encoder/HACKING Wed Dec 31 17:00:00 1969 +--- lame3.70/HACKING Thu Apr 6 12:50:49 2000 +*************** +*** 0 **** +--- 1,106 ---- ++ ************************************************************************ ++ LAME API ++ ++ For a general outline of the code, see the file API. ++ Also, main.c is a simple front end to libmp3lame.a ++ ++ The guts of the code are called from lame_encode_buffer(). ++ ++ lame_encode_buffer() handles buffering, resampling, filtering, and ++ then calls lame_encode_frame() for each frame: ++ ++ lame_encode_frame(): ++ l3psycho_anal() compute masking thresholds ++ mdct_sub() compute MDCT coefficients ++ iteration_loop() choose scalefactors (via iteration) ++ which determine noise shapping, and ++ choose best huffman tables for lossless compression ++ ++ III_formatBitstream format the bitstream and store in bit_buffer ++ copy_buffer() make a copy of bit_buffer, to return to calling program ++ write_buffer() optionally output bit_buffer to a file ++ empty_buffer() mark bit_buffer as empty ++ ++ ++ ++ ++ ++ ************************************************************************ ++ We are in the process of switching the whole code to FLOAT or FLOAT8, ++ which ever is faster (defined in machine.h). ++ Avoid using float or double, and instead use: ++ ++ FLOAT 4 byte floating point. ++ FLOAT8 8 byte floating point (only when necessary) ++ ++ ++ ************************************************************************ ++ Initialization: LAME can be used as a library, and there are two ++ kinds of static data: data that has to be initialized only once ++ when the library is called, and data that has to be initialized ++ every time you start encoding a new sample. ++ ++ These routines should be coded something like: ++ ++ #include "globalflags" ++ void subroutine(void) ++ { ++ static int firstcall=1; ++ if (firstcall) { ++ firstcall=0; ++ /* code that needs to be done only once when routine is loaded */ ++ } ++ if (gf.frameNum==0) { ++ /* code that needs to be done before encoding each file/stream */ ++ } ++ } ++ ++ The frameNum==0 code only works for routines which are only called ++ once per frame. More general code would be: ++ ++ ++ #include "globalflags" ++ void subroutine(void) ++ { ++ static int init=0; ++ if (gf.frameNum==0 && !init) { ++ /* code that needs to be done before encoding each file/stream */ ++ init=1; ++ } ++ if (gf.frameNum>0) init=0; ++ } ++ ++ ++ ++ ++ ++ ************************************************************************ ++ Global flags ++ ++ For better or worse, there are many global flags in lame, ++ in the 'gf' structure. ++ ++ The rules for adding a global flag are: ++ ++ 1. declair in struct gf in lame.h ++ 2. default values must be set in lame_init() ++ 3. include "globalflags.h" in routines that need access to these flags. ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ +diff -r -c -N encoder/INSTALL lame3.70/INSTALL +*** encoder/INSTALL Wed Jan 22 02:43:12 1997 +--- lame3.70/INSTALL Thu Mar 30 02:37:28 2000 +*************** +*** 1,36 **** +! Building the MPEG audio encoding software: + +! If you have gnu make and are on a Unix-type system, you should only have to +! type: + +! ./configure +! make + +- This will create a Makefile customized to your environment, overwriting the +- previous Makefile, using Makefile.in as a template. + +- For other systems you may have to modify the Makefile. Some flags of +- interest are: + +- BS_FORMAT +- Set to BINARY to work with most MPEG audio decoders (and save space). You +- can set it to ASCII, but your bitstream will only work with musicout(). + +- HAVE_NINT +- Define this if your math library already has nint(). + +! NDEBUG +! Define this to turn off all the assertions. Don't define this if you +! are developing code. + +- DEBUG +- Define this to get debugging messages printed to stdout. + +- For non-Unix systems you may have to define other things. Be sure to look +- at common.h. If you build this on a Macintosh, be sure to use a fairly +- recent set of Universal Headers. I have built this code on SunOS 4.1 and +- Linux 1.3 (using gcc 2.6.3), and on the Power Macintosh (using Metrowerks +- CodeWarrior 6). + + +- <mc@fivebats.com> 1995/10/16 +--- 1,69 ---- +! LAME 3.xx 3/00 Mark Taylor (http://www.sulaco.org/mp3) + +! ======================================================================= +! Building the MPEG audio encoding software on Windows: +! ======================================================================= +! There are MSVC project files, and a Makefile.MSVC included +! with the project. For production use, be sure to compile +! a "release" version and compile with the "maximum speed" compile +! option, and #define NDEBUG. + +! Note that MSVC "professional" claims to produce faster executables +! then MSVC "standard". It is possible to compile the GTK frame +! analyzer under windows, see README.WINGTK + + + + + +! ======================================================================= +! Building the MPEG audio encoding software on *NIX platforms: +! ======================================================================= +! If you are reading this, you have already applied the patch to the +! dist10 ISO demonstration source (or downloaded a full source +! distrubtion). If you do not have GTK 1.1.16 or higher, or do not want +! the graphical frame analyzer (it uses a lot of memory), edit the +! Makefile and dont define GTK and GTKLIBS. If you have trouble with +! libncurses/libtermcap, you can also disable that in the Makefile. +! +! Now just type: +! +! % make +! +! If it doesn't work, figure out why and send me the fix! +! If you are using a newer version of EGCS, check the Makefile. +! There are some flags which may improve performance. +! +! LAME has built in support for 16bit raw pcm, aiff and simple wav +! formats. Some complex .wav headers will confuse LAME. If you have +! trouble with this, or want to encode from other sound formats, you can +! now compile LAME to use Erik de Castro Lopo's libsndfile. To use +! libsndfile in LAME: +! +! 1. install libsndfile (see http://www.zip.com.au/~erikd/libsndfile/). +! 2. set SNDLIB and LIBSNDFILE as shown in the LAME Makefile. +! 3. make clean ; make +! +! Thanks to Albert Faber for adding the libsndfile support! +! +! LAME can also be compiled into a BladeEnc compatiable MS Windows .dll +! file. See the Dll/ directory for details. +! +! +! +! ======================================================================= +! To encode: +! ======================================================================= +! see the file "USAGE" for some usage guidlines. +! +! +! +! ======================================================================= +! Portability +! ======================================================================= +! LAME 3.x has been run on every modern OS. see the Makefile. +! + + + + +diff -r -c -N encoder/LICENSE lame3.70/LICENSE +*** encoder/LICENSE Wed Dec 31 17:00:00 1969 +--- lame3.70/LICENSE Wed Nov 24 01:43:10 1999 +*************** +*** 0 **** +--- 1,44 ---- ++ Can I use LAME in my commercial program? ++ ++ Yes, you can, under the restrictions of the LGPL. In particular, you ++ can include a compiled version of the LAME library (for example, ++ lame.dll) with a commercial program. Some notable requirements of ++ the LGPL: ++ ++ 1. In your program, you cannot include any source code from LAME, with ++ the exception of files whose only purpose is to describe the library ++ interface (such as lame.h). ++ ++ 2. Any modifications of LAME must be released under the LGPL. ++ The LAME project (www.sulaco.org/mp3) would appreciate being ++ notified of any modifications. ++ ++ 3. You must give prominent notice that your program is: ++ A. using LAME (including version number) ++ B. LAME is under the LGPL ++ C. Provide a copy of the LGPL. (the file COPYING contains the LGPL) ++ D. Provide a copy of LAME source, or a pointer where the LAME ++ source can be obtained (such as www.sulaco.org/mp3) ++ An example of prominent notice would be an "About the LAME encoding engine" ++ button in some pull down menu within the executable of your program. ++ ++ 4. If you determine that distribution of LAME requires a patent license, ++ you must obtain such license. ++ ++ ++ ***IMPORTANT NOTE*** ++ ++ The decoding functions provided in LAME use the mpglib decoding ++ engine which is under the GPL. They may not be used by any ++ program not released under the GPL unless you obtain such ++ permission from the MPG123 project. (www.mpg123.de). ++ ++ LAME has built in support to read raw pcm and some wav and aiff files. ++ More robust file I/O can be handled by compiling in LIBSNDFILE, ++ but LIBSNDFILE is also under the GPL and my not be used by other ++ programs not under the GPL. ++ ++ ++ ++ ++ +diff -r -c -N encoder/Makefile lame3.70/Makefile +*** encoder/Makefile Wed Jan 22 02:43:23 1997 +--- lame3.70/Makefile Thu Apr 6 12:50:49 2000 +*************** +*** 1,72 **** +! # Generated automatically from Makefile.in by configure. +! ############################################################################ +! ## ISO MPEG Audio Subgroup Software Simulation Group (1996) +! ## ISO 13818-3 MPEG-2 Audio Decoder - Lower Sampling Frequency Extension +! ## +! ## $Id: Makefile,v 1.2 1996/08/14 23:18:34 rowlands Exp $ +! ## +! ## Makefile for encoder. Requies GNU make. +! ## +! ## $Log: Makefile,v $ +! ## Revision 1.2 1996/08/14 23:18:34 rowlands +! ## Cleanups. +! ## +! ## Revision 1.2 1996/02/14 05:18:05 rowlands +! ## Cleanups. +! ## +! ## Revision 1.1 1996/02/14 04:04:23 rowlands +! ## Initial revision +! ## +! ## Received from Mike Coleman +! ############################################################################ + + CC = gcc + + c_sources = \ +! common.c \ +! encode.c \ + formatBitstream.c \ +! huffman.c \ +! ieeefloat.c \ + l3bitstream.c \ +! l3psy.c \ +! loop.c \ +! mdct.c \ +! musicin.c \ + portableio.c \ +! psy.c \ + reservoir.c \ +! subs.c \ +! tonal.c + + OBJ = $(c_sources:.c=.o) + DEP = $(c_sources:.c=.d) + +- NINT_SWITCH = + +- CC_SWITCHES = -g -O -DUNIX -DBS_FORMAT=BINARY $(NINT_SWITCH) -DNDEBUG + +! PGM = encode + +! LIBS = -lm + + +! %.o: %.c +! $(CC) $(CC_SWITCHES) -c $< -o $@ + +! %.d: %.c +! $(SHELL) -ec '$(CC) -M $(CC_SWITCHES) $< | sed '\''s/$*.o/& $@/g'\'' > $@' + + +! $(PGM): $(OBJ) Makefile +! $(CC) -o $(PGM) $(OBJ) $(LIBS) + + clean: +! -rm $(OBJ) $(DEP) + + tags: TAGS + + TAGS: ${c_sources} + etags -T ${c_sources} + +! -include $(DEP) +--- 1,326 ---- +! # Makefile for LAME 3.xx +! # +! # LAME is reported to work under: +! # Linux (i86), NetBSD 1.3.2 (StrongARM), FreeBSD (i86) +! # Compaq Alpha(OSF, Linux, Tru64 Unix), Sun Solaris, SGI IRIX, +! # OS2 Warp, Macintosh PPC, BeOS, Amiga and even VC++ +! # +! UNAME = $(shell uname) +! ARCH = $(shell uname -m) + ++ ++ # defaults: ++ PGM = lame + CC = gcc ++ CC_OPTS = -O ++ GTK = ++ GTKLIBS = ++ SNDLIB = -DLAMESNDFILE ++ LIBSNDFILE = ++ LIBS = -lm ++ MAKEDEP = -M ++ BRHIST_SWITCH = ++ LIBTERMCAP = ++ RM = rm -f ++ CPP_OPTS = -DHAVEMPGLIB -DLAMEPARSE ++ ++ ++ ++ ########################################################################## ++ # To remove support for mp3 *decoding*, ++ # remove -DHAVEMPGLIB to CPP_OPTS ++ ########################################################################## ++ ++ ++ ########################################################################## ++ # Define these to compile in code for the optional VBR bitrate histogram. ++ # Requires ncurses, but libtermcap also works. ++ # If you have any trouble, just dont define these ++ ########################################################################## ++ #BRHIST_SWITCH = -DBRHIST ++ #LIBTERMCAP = -lncurses ++ #LIBTERMCAP = -ltermcap ++ ++ ++ ########################################################################## ++ # SNDLIB = no file i/o ++ # SNDLIB = -DLAMESNDFILE to use internal LAME soundfile routines ++ # SNDLIB = -DLIBSNDFILE to use Erik de Castro Lopo's libsndfile ++ # http://www.zip.com.au/~erikd/libsndfile/ ++ # otherwise LAME can only read 16bit wav, aiff and pcm inputfiles. ++ # Note: at present, libsndfile does not support input from stdin. ++ ########################################################################## ++ #SNDLIB = -DLIBSNDFILE ++ #LIBSNDFILE=-lsndfile ++ # if libsndfile is in a custom location, try: ++ #LIBSNDFILE=-L $(LIBSNDHOME) -lsndfile -I $(LIBSNDHOME) ++ ++ ++ ########################################################################## ++ # define these to use compile in support for the GTK mp3 frame analyzer ++ # Requires -DHAVEMPGLIB ++ # and SNDLIB = -DLAME or -DLIBSNDFILE ++ ########################################################################## ++ #GTK = -DHAVEGTK `gtk-config --cflags` ++ #GTKLIBS = `gtk-config --libs` ++ ++ ++ ########################################################################## ++ # LINUX ++ ########################################################################## ++ ifeq ($(UNAME),Linux) ++ # remove these lines if you dont have GTK, or dont want the GTK frame analyzer ++ GTK = -DHAVEGTK `gtk-config --cflags` ++ GTKLIBS = `gtk-config --libs` ++ # Comment out next 2 lines if you want to remove VBR histogram capability ++ BRHIST_SWITCH = -DBRHIST ++ LIBTERMCAP = -lncurses ++ ++ ++ # suggested for gcc-2.7.x ++ CC_OPTS = -O3 -fomit-frame-pointer -funroll-loops -ffast-math -finline-functions -Wall ++ # CC_OPTS = -O9 -fomit-frame-pointer -fno-strength-reduce -mpentiumpro -ffast-math -finline-functions -funroll-loops -Wall -malign-double -g -march=pentiumpro -mfancy-math-387 -pipe ++ ++ # for debugging: ++ # CC_OPTS = -UNDEBUG -O -Wall -g -DABORTFP ++ ++ # for lots of debugging: ++ # CC_OPTS = -DDEBUG -UNDEBUG -O -Wall -g -DABORTFP ++ ++ ++ ++ # The following features are experimental code, ++ # so don't use them if you don't know exactly what you do! (RH 2000-01-30) ++ # FEATURES = -DRH_QUALITY_CONTROL # tuned VBR quality control ++ # FEATURES += -DRH_SIDE_VBR # turns side channel reduction off (VBR) ++ # FEATURES += -DRH_SIDE_CBR # turns side channel reduction off (CBR) ++ # FEATURES += -DRH_ATH # max noise instead of average noise ++ # these options for gcc-2.95.2 to produce fast code ++ # CC_OPTS = $(FEATURES)\ ++ # -Wall -O9 -fomit-frame-pointer -march=pentium \ ++ # -finline-functions -fexpensive-optimizations \ ++ # -ffast-math -malign-double -mfancy-math-387 \ ++ # -funroll-loops -funroll-all-loops -pipe -fschedule-insns2 \ ++ # -fno-strength-reduce ++ ++ ++ ########################################################################## ++ # LINUX on Digital/Compaq Alpha CPUs ++ ########################################################################## ++ ifeq ($(ARCH),alpha) ++ # double is faster than float on Alpha ++ CC_OPTS = -O4 -Wall -fomit-frame-pointer -ffast-math -funroll-loops \ ++ -mfp-regs -fschedule-insns -fschedule-insns2 \ ++ -finline-functions \ ++ # -DFLOAT=double ++ # add "-mcpu=21164a -Wa,-m21164a" to optimize for 21164a (ev56) CPU ++ ++ # Compaq's C Compiler ++ #CC = ccc ++ # Options for Compaq's C Compiler ++ #CC_OPTS = -fast -Wall ++ ++ # standard Linux libm ++ #LIBS = -lm ++ # optimized libffm (free fast math library) ++ #LIBS = -lffm ++ # Compaq's fast math library ++ LIBS = -lcpml ++ endif ++ endif ++ ++ ++ ++ ########################################################################## ++ # FreeBSD ++ ########################################################################## ++ ifeq ($(UNAME),FreeBSD) ++ # remove if you do not have GTK or do not want the GTK frame analyzer ++ GTK = -DHAVEGTK `gtk12-config --cflags` ++ GTKLIBS = `gtk12-config --libs` ++ # Comment out next 2 lines if you want to remove VBR histogram capability ++ BRHIST_SWITCH = -DBRHIST ++ LIBTERMCAP = -lncurses ++ ++ endif ++ ++ ++ ########################################################################## ++ # SunOS ++ ########################################################################## ++ ifeq ($(UNAME),SunOS) ++ CC = cc ++ CC_OPTS = -O -xCC ++ MAKEDEP = -xM ++ endif ++ ++ ++ ########################################################################## ++ # SGI ++ ########################################################################## ++ ifeq ($(UNAME),IRIX64) ++ CC = cc ++ endif ++ ++ ++ ########################################################################## ++ # Compaq Alpha running Dec Unix (OSF) ++ ########################################################################## ++ ifeq ($(UNAME),OSF1) ++ CC = cc ++ CC_OPTS = -fast -O3 -std -g3 -non_shared ++ endif ++ ++ ########################################################################## ++ # BeOS ++ ########################################################################## ++ ifeq ($(UNAME),BeOS) ++ CC = $(BE_C_COMPILER) ++ LIBS = ++ ifeq ($(ARCH),BePC) ++ CC_OPTS = -O9 -fomit-frame-pointer -march=pentium \ ++ -mcpu=pentium -ffast-math -funroll-loops \ ++ -fprofile-arcs -fbranch-probabilities ++ else ++ CC_OPTS = -opt all ++ MAKEDEP = -make ++ endif ++ endif ++ ++ ########################################################################### ++ # MOSXS (Rhapsody PPC) ++ ########################################################################### ++ ifeq ($(UNAME),Rhapsody) ++ CC = cc ++ LIBS = ++ CC_OPTS = -O9 -ffast-math -funroll-loops -fomit-frame-pointer ++ MAKEDEP = -make ++ ++ endif ++ ########################################################################## ++ # OS/2 ++ ########################################################################## ++ # Properly installed EMX runtime & development package is a prerequisite. ++ # tools I used: make 3.76.1, uname 1.12, sed 2.05, PD-ksh 5.2.13 ++ # ++ ########################################################################## ++ ifeq ($(UNAME),OS/2) ++ SHELL=sh ++ CC = gcc ++ CC_OPTS = -O3 ++ PGM = lame.exe ++ LIBS = ++ ++ # I use the following for slightly better performance on my Pentium-II ++ # using pgcc-2.91.66: ++ # CC_OPTS = -O6 -ffast-math -funroll-loops -mpentiumpro -march=pentiumpro ++ ++ # Comment out next 2 lines if you want to remove VBR histogram capability ++ BRHIST_SWITCH = -DBRHIST ++ LIBTERMCAP = -ltermcap ++ ++ # Uncomment & inspect the 2 GTK lines to use MP3x GTK frame analyzer. ++ # Properly installed XFree86/devlibs & GTK+ is a prerequisite. ++ # The following works for me using Xfree86/OS2 3.3.5 and GTK+ 1.2.3: ++ # GTK = -DHAVEGTK -IC:/XFree86/include/gtk12 -Zmt -D__ST_MT_ERRNO__ -IC:/XFree86/include/glib12 -IC:/XFree86/include ++ # GTKLIBS = -LC:/XFree86/lib -Zmtd -Zsysv-signals -Zbin-files -lgtk12 -lgdk12 -lgmodule -lglib12 -lXext -lX11 -lshm -lbsd -lsocket -lm ++ endif ++ ++ ++ + ++ # 10/99 added -D__NO_MATH_INLINES to fix a bug in *all* versions of ++ # gcc 2.8+ as of 10/99. ++ ++ CC_SWITCHES = -DNDEBUG -D__NO_MATH_INLINES $(CC_OPTS) $(SNDLIB) $(GTK) $(BRHIST_SWITCH) + c_sources = \ +! brhist.c \ + formatBitstream.c \ +! fft.c \ +! get_audio.c \ + l3bitstream.c \ +! id3tag.c \ +! ieeefloat.c \ +! lame.c \ +! newmdct.c \ +! parse.c \ + portableio.c \ +! psymodel.c \ +! quantize.c \ +! quantize-pvt.c \ +! vbrquantize.c \ + reservoir.c \ +! tables.c \ +! takehiro.c \ +! timestatus.c \ +! util.c \ +! VbrTag.c \ +! version.c \ +! gtkanal.c \ +! gpkplotting.c \ +! mpglib/common.c \ +! mpglib/dct64_i386.c \ +! mpglib/decode_i386.c \ +! mpglib/layer3.c \ +! mpglib/tabinit.c \ +! mpglib/interface.c \ +! mpglib/main.c + + OBJ = $(c_sources:.c=.o) + DEP = $(c_sources:.c=.d) + + + +! %.o: %.c +! $(CC) $(CPP_OPTS) $(CC_SWITCHES) -c $< -o $@ + +! %.d: %.c +! $(SHELL) -ec '$(CC) $(MAKEDEP) $(CPP_OPTS) $(CC_SWITCHES) $< | sed '\''s;$*.o;& $@;g'\'' > $@' + ++ #$(PGM): main.o $(OBJ) Makefile ++ # $(CC) -o $(PGM) main.o $(OBJ) $(LIBS) $(CPP_OPTS) $(LIBSNDFILE) $(GTKLIBS) $(LIBTERMCAP) + +! $(PGM): main.o libmp3lame.a +! $(CC) -o $(PGM) main.o -L. -lmp3lame $(LIBS) $(LIBSNDFILE) $(GTKLIBS) $(LIBTERMCAP) + +! mp3x: mp3x.o libmp3lame.a +! $(CC) -o mp3x mp3x.o $(OBJ) $(LIBS) $(LIBSNDFILE) $(GTKLIBS) $(LIBTERMCAP) + ++ mp3rtp: rtp.o mp3rtp.o libmp3lame.a ++ $(CC) -o mp3rtp mp3rtp.o rtp.o $(OBJ) $(LIBS) $(LIBSNDFILE) $(GTKLIBS) $(LIBTERMCAP) + +! libmp3lame.a: $(OBJ) Makefile +! # cd libmp3lame +! # make libmp3lame +! ar cr libmp3lame.a $(OBJ) + + clean: +! -$(RM) $(OBJ) $(DEP) $(PGM) main.o rtp.o mp3rtp mp3rtp.o \ +! mp3x.o mp3x libmp3lame.a +! + + tags: TAGS + + TAGS: ${c_sources} + etags -T ${c_sources} + +! ifneq ($(MAKECMDGOALS),clean) +! -include $(DEP) +! endif +! +! +! # +! # testcase.mp3 is a 2926 byte file. The first number output by +! # wc is the number of bytes which differ between new output +! # and 'official' results. +! # +! # Because of compilier options and effects of roundoff, the +! # number of bytes which are different may not be zero, but +! # should be at most 30. +! # +! test: $(PGM) +! ./lame --nores -h testcase.wav testcase.new.mp3 +! cmp -l testcase.new.mp3 testcase.mp3 | wc +! +! testg: $(PGM) +! ./lame -g -h ../test/castanets.wav +diff -r -c -N encoder/Makefile.MSVC lame3.70/Makefile.MSVC +*** encoder/Makefile.MSVC Wed Dec 31 17:00:00 1969 +--- lame3.70/Makefile.MSVC Thu Apr 6 12:50:49 2000 +*************** +*** 0 **** +--- 1,86 ---- ++ # Makefile.MSVC: MSVC Makefile for LAME 3.57beta ++ ++ PGM = lame ++ CC = cl ++ LN = link ++ ++ # debugging options ++ # CC_OPTS = /nologo /Zi /Ge /GZ ++ # LN_OPTS = /nologo /debug:full /debugtype:cv /fixed:no ++ ++ # profiling options ++ # CC_OPTS = /nologo /Zi /O2ab2gitwy /G6AFs /DNDEBUG ++ # LN_OPTS = /nologo /debug:full /debugtype:cv /fixed:no /profile ++ ++ # release options ++ CC_OPTS = /nologo /O2ab2gitwy /G6AFs /DNDEBUG ++ LN_OPTS = /nologo ++ ++ GTK = ++ GTKLIBS = ++ SNDLIB = ++ LIBSNDFILE = ++ LIBS = ++ MAKEDEP = ++ TIMER_SWITCH = ++ BRHIST_SWITCH = ++ LIBTERMCAP = ++ ++ CC_SWITCHES = $(CC_OPTS) $(DISTRIB) $(SNDLIB) $(GTK) \ ++ /DBS_FORMAT=BINARY $(TIMER_SWITCH) $(BRHIST_SWITCH) ++ LN_SWITCHES = $(LN_OPTS) ++ ++ c_sources = \ ++ main.c \ ++ brhist.c \ ++ formatBitstream.c \ ++ fft.c \ ++ get_audio.c \ ++ l3bitstream.c \ ++ id3tag.c \ ++ ieeefloat.c \ ++ lame.c \ ++ newmdct.c \ ++ parse.c \ ++ portableio.c \ ++ psymodel.c \ ++ quantize.c \ ++ quantize-pvt.c \ ++ vbrquantize.c \ ++ loopold.c \ ++ reservoir.c \ ++ tables.c \ ++ takehiro.c \ ++ timestatus.c \ ++ util.c \ ++ VbrTag.c \ ++ version.c \ ++ gtkanal.c \ ++ gpkplotting.c \ ++ mpglib/common.c \ ++ mpglib/dct64_i386.c \ ++ mpglib/decode_i386.c \ ++ mpglib/layer3.c \ ++ mpglib/tabinit.c \ ++ mpglib/interface.c \ ++ mpglib/main.c ++ ++ OBJ = $(c_sources:.c=.obj) ++ ++ .c.obj: ++ @$(CC) $(CC_SWITCHES) /c $< /Fo$@ ++ ++ $(PGM).exe: $(OBJ) Makefile.MSVC ++ @echo $(PGM).exe ++ @$(LN) $(LN_SWITCHES) $(OBJ) $(LIBS) $(LIBSNDFILE) $(GTKLIBS) \ ++ $(LIBTERMCAP) /out:$(PGM).exe /map:$(PGM).map ++ ++ clean: ++ @-del *.obj ++ @-del dll\*.obj ++ @-del mpglib\*.obj ++ ++ rebuild: clean $(PGM).exe ++ ++ ++ +diff -r -c -N encoder/Makefile.in lame3.70/Makefile.in +*** encoder/Makefile.in Wed Jan 22 02:43:12 1997 +--- lame3.70/Makefile.in Wed Dec 31 17:00:00 1969 +*************** +*** 1,68 **** +- ############################################################################ +- ## ISO MPEG Audio Subgroup Software Simulation Group (1996) +- ## ISO 13818-3 MPEG-2 Audio Decoder - Lower Sampling Frequency Extension +- ## +- ## $Id: Makefile.in,v 1.2 1996/02/14 05:18:05 rowlands Exp $ +- ## +- ## Makefile for encoder. Requies GNU make. +- ## +- ## $Log: Makefile.in,v $ +- ## Revision 1.2 1996/02/14 05:18:05 rowlands +- ## Cleanups. +- ## +- ## Revision 1.1 1996/02/14 04:04:23 rowlands +- ## Initial revision +- ## +- ## Received from Mike Coleman +- ############################################################################ +- +- CC = gcc +- +- c_sources = \ +- common.c \ +- encode.c \ +- formatBitstream.c \ +- huffman.c \ +- ieeefloat.c \ +- l3bitstream.c \ +- l3psy.c \ +- loop.c \ +- mdct.c \ +- musicin.c \ +- portableio.c \ +- psy.c \ +- reservoir.c \ +- subs.c \ +- tonal.c +- +- OBJ = $(c_sources:.c=.o) +- DEP = $(c_sources:.c=.d) +- +- NINT_SWITCH = @NINTSW@ +- +- CC_SWITCHES = -g -O -DUNIX -DBS_FORMAT=BINARY $(NINT_SWITCH) -DNDEBUG +- +- PGM = encode +- +- LIBS = @LIBS@ @MATHLIBS@ +- +- +- %.o: %.c +- $(CC) $(CC_SWITCHES) -c $< -o $@ +- +- %.d: %.c +- $(SHELL) -ec '$(CC) -M $(CC_SWITCHES) $< | sed '\''s/$*.o/& $@/g'\'' > $@' +- +- +- $(PGM): $(OBJ) Makefile +- $(CC) -o $(PGM) $(OBJ) $(LIBS) +- +- clean: +- -rm $(OBJ) $(DEP) +- +- tags: TAGS +- +- TAGS: ${c_sources} +- etags -T ${c_sources} +- +- -include $(DEP) +--- 0 ---- +diff -r -c -N encoder/PRESETS.draft lame3.70/PRESETS.draft +*** encoder/PRESETS.draft Wed Dec 31 17:00:00 1969 +--- lame3.70/PRESETS.draft Sat Jan 8 10:52:52 2000 +*************** +*** 0 **** +--- 1,45 ---- ++ -- ++ ++ Phone 300 - 3400 hz, mono: ++ --highpass 0.300 ++ --lowpass 3.400 ++ --noshort (preechoes are a minor issue) ++ --resample 16 (8 would be better, needs MPEG2.5) ++ -m m -b 16 (8 is a litle bit to low :( ++ ++ Tape 20 - 18000 hz, stereo: ++ --highpass 0.020 ++ --lowpass 18.000 ++ --resample 44.1 ++ -m j -b 128 ++ ++ HIFI 20 - 20000 hz, stereo: ++ --highpass 0.020 ++ --lowpass 20.000 ++ --resample 44.1 ++ -m j -b 160 ++ ++ CD 0 - 22050 hz, stereo: (full 44.1kHz frequency range) ++ -m s -b 192 ++ ++ Studio 0 - 24000 hz, stereo: (full 48kHz frequency range) ++ -m s -b 256 ++ ++ ++ ++ -- thanks to Ross Levis: ++ ++ FM 30 - 15000 hz, stereo: ++ --lowpass 15.000 ++ --lowpass-width 0 (sharp cut-off) ++ --resample 32 ++ -m j -b 128 ++ ++ HQvoice (interviews) 100 - 12000 hz, mono: ++ --lowpass 12.000 ++ --highpass 0.100 ++ --resample 24 ++ -m m -b 64 ++ ++ ++ -- feel free to extend this list further -- +diff -r -c -N encoder/README lame3.70/README +*** encoder/README Wed Dec 31 17:00:00 1969 +--- lame3.70/README Thu Mar 30 01:24:20 2000 +*************** +*** 0 **** +--- 1,52 ---- ++ LAME 3.xx ++ LAME Ain't an MP3 Encoder ++ http://www.sulaco.org/mp3 ++ October 1999 ++ ++ Originally developed by Mike Cheng (www.uq.net.au/~zzmcheng). Now ++ maintained by Mark Taylor (www.sulaco.org/mp3). ++ ++ This is a patch against the ISO MPEG2 dist10 demonstration source. The ++ modifications are distributed under the GNU LESSER PUBLIC LICENSE ++ (LGPL, see www.gnu.org) with the following modification: ++ ++ 1. If you determine that distribution of LAME requires a patent license, ++ and you obtain a patent license, you may distribute LAME even though ++ redistribution of LAME may also require a patent license. ++ ++ 2. You agree not to enforce any patent claims for any aspect of ++ MPEG audio compression, or any other techniques contained in ++ the LAME source code. ++ ++ ++ ============================================================================ ++ ++ see the file "INSTALL" for installation (compiling) instructions. ++ see the file "USAGE" for the most up-to-date guide to the command line options. ++ see the file "LICENSE" for details on how to use LAME in non-GPL programs. ++ see the file "HACKING" for some general codeing guidelines ++ ++ There is HTML documentation and a man page in the doc directory. ++ The HTML docs were contributed by Gabriel Bouvigne (www.mp3tech.org). ++ See www.mp3tech.org for more audio compression information and links. ++ ++ ============================================================================ ++ ++ The graphical frame analyzer uses the MPGLIB decoding engine, from the ++ MPG123 package, written by: Michael Hipp (www.mpg123.de) MPGLIB is ++ released under the GPL, while MPG123 is released under a more ++ restrictive agreement. ++ ++ Various components of the patch are: ++ ++ Copyrights (c) 1999 by Mark Taylor: new psy-model, quantization and bit allocation ++ Copyrights (c) 1998 by Michael Cheng: new transforms and many other improvements ++ Copyrights (c) 1995,1996,1997 by Michael Hipp: mpglib ++ ++ ++ The ISO demonstration source (distributed separately) has its own ++ patent, copyright and distribution issues. The code is freely ++ available, but any commercial use (including distributing free ++ encoders) may require a patent license from Thomson Multimedia. (see ++ http://www.iis.fhg.de/amm/legal/) ++ +diff -r -c -N encoder/README.WINGTK lame3.70/README.WINGTK +*** encoder/README.WINGTK Wed Dec 31 17:00:00 1969 +--- lame3.70/README.WINGTK Thu Mar 30 01:04:11 2000 +*************** +*** 0 **** +--- 1,42 ---- ++ March 30, 2000 ++ ++ Release notes MP3X (the LAME frame analyzer) for WIN32 ++ ++ =========================================================================== ++ How to compile Lame with MP3x support: ++ ++ You first need the proper GTK and GLIB include files and lib files. ++ You can get them from the WINGTK project ++ http://user.sgic.fi/~tml/gimp/win32/ ++ (or see http://www.gtk.org, and click on the WINGTK link). ++ ++ Download: glib-dev-VERSION.zip, gtk+-dev-VERSION.zip, extralibs-dev-VERSION.zip ++ where VERSION = something like 20000201 ++ ++ unzip the files. this will make a "src" directory. (for example, c:\gtk\src) ++ Then, in MSVC, under Tools->options->directories, add ++ to the "include files" and "library files" settings: ++ c:\gtk\src\glib ++ c:\gtk\src\gtk+\glib ++ c:\gtk\src\gtk+ ++ ++ ++ =========================================================================== ++ To run MP3x.exe, or lame.exe with MP3x support, you need ++ the GTK DLL files: Either instal them on your system, ++ or put them in the same directory as lame.exe. ++ ++ glib-1.3.dll ++ gdk-1.3.dll ++ gtk-1.3.dll ++ gmodule-1.3.dll ++ gnu-intl.dll ++ ++ lame -g myfile ++ ++ myfile can be a mp3 file, or a wav file. ++ ++ ++ ++ ++ +diff -r -c -N encoder/TODO lame3.70/TODO +*** encoder/TODO Wed Dec 31 17:00:00 1969 +--- lame3.70/TODO Sun Jan 30 03:50:36 2000 +*************** +*** 0 **** +--- 1,44 ---- ++ 0. ./configure script ++ ++ things to check for: ++ ++ -llibsndfile (allow user to enable) ++ -lncurses ++ -lGTK (allow user to disable) ++ make depend option for CC ++ ++ ++ ++ ++ 1. Modularization: ++ ++ BladeEnc.dll was written before lame was 'modularized' ++ this should be cleaned up to look like main.c: It should be possible ++ to write the BladeEnc/lame dll so that it just uses ++ #include "lame.h" and then calls to various lame_*() routines ++ ++ ++ 2. ++ Does stdin work when LAME is compiled to use libsndfile? ++ (new version of libsndfile will support this - try this out) ++ ++ 3. ++ LAME uses a linear downsampling with simple lowpass filter ++ on the MDCT/polyphase coefficients. Is this good enough, or ++ would a more expensive filter give better results? ++ ++ 4. ++ make use of subblock_gain (crude algorithm enabled with -Z) ++ One method: loop over all possible subblock_gains and ++ see which combination gives best distortion? But this puts a lot ++ of faith in the short block psycho acoustics ++ ++ 5. ++ make use of scalefactor_select (crude algorithm enabled with -Y) ++ ++ 6. ++ 10Hz high-pass filter. Probably wont help much. ++ ++ ++ ++ +diff -r -c -N encoder/USAGE lame3.70/USAGE +*** encoder/USAGE Wed Dec 31 17:00:00 1969 +--- lame3.70/USAGE Tue Mar 21 19:03:03 2000 +*************** +*** 0 **** +--- 1,539 ---- ++ ++ % lame [options] inputfile [outputfile] ++ ++ ======================================================================= ++ Examples: ++ ======================================================================= ++ fixed bit rate jstereo 128kbs encoding: ++ % lame sample.wav sample.mp3 ++ ++ fixed bit rate jstereo 128kbs encoding, highest quality: (recommended) ++ % lame -h sample.wav sample.mp3 ++ ++ To disable joint stereo encoding (slightly faster, but less quality at bitrates<=128kbs) ++ % lame -m s sample.wav sample.mp3 ++ ++ Fast encode, low quality (no psycho-acoustics) ++ % lame -f sample.wav sample.mp3 ++ ++ Variable Bitrate (VBR): (use -V n to adjust quality/filesize) ++ % lame -h -v sample.wav sample.mp3 ++ ++ Note: VBR is currently under heavy development. Right now it can ++ often result in too much compression. I would recommend using VBR ++ with a minimum bitrate of 112kbs. This will let LAME increase ++ the bitrate for difficult-to-encode frames, but prevent LAME from ++ being too aggressive for simple frames: ++ ++ % lame -h -v -b 112 sample.wav sample.mp3 ++ ++ ++ ======================================================================= ++ LOW BITRATES ++ ======================================================================= ++ At lower bitrates, (like 24kbs per channel), it is recommended that ++ you use a 16kHz sampling rate combined with lowpass filtering. LAME, ++ as well as commercial encoders (FhG, Xing) will do this automatically. ++ However, if you feel there is too much (or not enough) lowpass ++ filtering, you may need to try different values of the lowpass cutoff ++ and passband width (--lowpass and --lowpass-width options). ++ ++ ++ ++ ++ ======================================================================= ++ STREAMING EXAMPLES ++ ======================================================================= ++ Streaming mono 22.05kHz raw pcm, 24kbs output: ++ % cat inputfile | lame -r -m m -b 24 -s 22.05 - - > output ++ ++ Streaming mono 22.05kHz raw pcm, with downsampling to 16kHz: ++ % cat inputfile | sox -t raw -x -w -s -c 1 -r 22050 - -t raw -x -w -s -c 1 -r 16000 - resample 0.66 | lame -r -m m -b 24 -s 16 - - > output ++ ++ You may or may not need one or both of the "-x" (swap bytes) option in Sox. ++ ++ ++ ++ ++ ======================================================================= ++ ++ For more options, just type: ++ % lame --help ++ ++ Scripts are included to run lame on multiple files: ++ ++ bach script: mlame Run "mlame -h" for instructions. ++ sh script: auenc Run auenc for instructions ++ ++ ++ ++ ======================================================================= ++ options guide: ++ ======================================================================= ++ These options are explained in detail below. ++ ++ ++ Quality related: ++ ++ -m m/s/j/f mode selection ++ -k disable all filtering ++ -d allow block types to differ between channels ++ --athonly ignore psy-model output, only use masking from the ATH ++ --voice experimental voice encoding mode ++ --noshort disable short blocks ++ ++ ++ Constant Bit Rate (CBR) ++ -b n set bitrate (8,16,24,...,320) ++ -h higher quality but slower ++ -f disable psycho-acoustics. Encoding much faster but lower quality ++ ++ ++ Variable Bit Rate (VBR) ++ -v VBR ++ -V n VBR quality setting (0=highest quality, 9=lowest) ++ -b n specify a minimum allowed bitrate (8,16,24,...,320) ++ -B n specify a maximum allowed bitrate (8,16,24,...,320) ++ -t disable Xing VBR informational tag ++ --nohist disable display of VBR bitrate histogram ++ ++ ++ Experimental (undocumented): may work better or worse: ++ ++ -X n try different quality measures (when comparing quantizations) ++ -Y try to use scalefac_select ++ -Z try to use subblock_gain ++ ++ ++ Operational: ++ ++ -r assume input file is raw PCM ++ -s n input sampling frequency in kHz (for raw PCM input files) ++ --resample n output sampling frequency ++ --mp3input input file is an MP3 file. decode using mpglib/mpg123 ++ -x swap bytes of input file ++ -a downmix stereo input file to mono .mp3 ++ -e n/5/c de-emphasis ++ -p add CRC error protection ++ -c mark the encoded file as copyrighted ++ -o mark the encoded file as a copy ++ -S don't print progress report, VBR histogram ++ -g run MP3x, the graphical frame analyzer ++ ++ ++ id3 tagging: ++ ++ --tt "title" title of song (max 30 chars) ++ --ta "artist" artist who did the song (max 30 chars) ++ --tl "album" album where it came from (max 30 chars) ++ --ty "year" year in which the song/album was made (max 4 chars) ++ --tc "comment" additional info (max 30 chars) ++ --tg "genre" genre of song (name or number) ++ ++ ++ ++ options not yet described: ++ --nores disable bit reservoir ++ --noath disable ATH ++ --cwlimit <freq> specify range of tonality calculation ++ ++ --lowpass ++ --lowpass-width ++ --highpass ++ --highpass-width ++ ++ ++ ++ ++ ++ ======================================================================= ++ Detailed description of all options in alphabetical order ++ ======================================================================= ++ ++ ++ ======================================================================= ++ downmix ++ ======================================================================= ++ -a ++ ++ mix the stereo input file to mono and encode as mono. ++ ++ This option is only needed in the case of raw PCM stereo input ++ (because LAME cannot determine the number of channels in the input file). ++ To encode a stereo PCM input file as mono, use "lame -m s -a" ++ ++ For WAV and AIFF input files, using "-m m" will always produce a ++ mono .mp3 file from both mono and stereo input. ++ ++ ======================================================================= ++ ATH only ++ ======================================================================= ++ --athonly ++ ++ This option causes LAME to ignore the output of the psy-model and ++ only use masking from the ATH. Might be useful at very high bitrates ++ or for testing the ATH. ++ ++ ++ ++ ++ ======================================================================= ++ bitrate ++ ======================================================================= ++ -b n ++ ++ For MPEG1 (sampling frequencies of 32, 44.1 and 48kHz) ++ n = 32,40,48,56,64,80,96,112,128,160,192,224,256,320 ++ ++ For MPEG2 (sampling frequencies of 16, 22.05 and 24kHz) ++ n = 8,16,24,32,40,48,56,64,80,96,112,128,144,160 ++ ++ ++ The bitrate to be used. Default is 128kbs MPEG1, 80kbs MPEG2. ++ ++ When used with variable bitrate encodings (VBR), -b specifies the ++ minimum bitrate to use. This is useful to prevent LAME VBR from ++ using some very aggressive compression which can cause some distortion ++ due to small flaws in the psycho-acoustic model. ++ ++ ======================================================================= ++ max bitrate ++ ======================================================================= ++ -B n ++ ++ For MPEG1 (sampling frequencies of 32, 44.1 and 48kHz) ++ n = 32,40,48,56,64,80,96,112,128,160,192,224,256,320 ++ ++ For MPEG2 (sampling frequencies of 16, 22.05 and 24kHz) ++ n = 8,16,24,32,40,48,56,64,80,96,112,128,144,160 ++ ++ Maximum allowed bitrate when using VBR. ++ ++ ++ ======================================================================= ++ copyright ++ ======================================================================= ++ -c ++ ++ mark the encoded file as copyrighted ++ ++ ++ ++ ======================================================================= ++ block type control ++ ======================================================================= ++ -d ++ ++ Allows the left and right channels to use different block types. ++ Normally this is not allowed, only because the FhG encoder does ++ not seem to allow it either. If anyone finds a sample where -d ++ produces better results, let me know. (mt@sulaco.org) ++ ++ ++ ++ ======================================================================= ++ de-emphasis ++ ======================================================================= ++ -e n/5/c ++ ++ n = (none, default) ++ 5 = 0/15 microseconds ++ c = citt j.17 ++ ++ All this does is set a flag in the bitstream. If you have a PCM ++ input file where one of the above types of (obsolete) emphasis has ++ been applied, you can set this flag in LAME. Then the mp3 decoder ++ should de-emphasize the output during playback, although most ++ decoders ignore this flag. ++ ++ A better solution would be to apply the de-emphasis with a standalone ++ utility before encoding, and then encode without -e. ++ ++ ++ ++ ======================================================================= ++ fast mode ++ ======================================================================= ++ -f ++ ++ disable psycho-acoustics. Encoding much faster but lower quality ++ ++ ++ ++ ======================================================================= ++ graphical frame analyzer ++ ======================================================================= ++ -g ++ ++ run MP3x, the graphical frame analyzer analysis on the inputfile. The ++ inputfile can be either an .mp3 file or uncompressed audio file. MP3x ++ support must be compiled into LAME, and requires GTK 1.2. ++ Documentation is under the About pull down menu. ++ ++ ++ ++ ======================================================================= ++ high quality ++ ======================================================================= ++ -h ++ ++ use (maybe) some quality improvements ++ ++ LAME 3.21 and up: -h enables specialized mid/side masking thresholds to ++ be used in jstereo mode. Will sound better in jstereo mode ++ but is 20% slower. No effect for mono files. ++ ++ LAME 3.58beta and up: -h also enables a more accurate but slightly ++ slower quantization formula. ++ ++ ++ ++ ======================================================================= ++ sfb=21 cutoff ++ ======================================================================= ++ -k ++ ++ keep all frequencies. (Disable all filters) ++ ++ Without -k, LAME will automatically apply various types of lowpass ++ filters. This is because the high frequency coefficients can take up ++ a lot of bits that would be better used for lower, more important ++ frequencies. ++ ++ ++ ++ ++ ======================================================================= ++ Modes: ++ ======================================================================= ++ ++ -m m mono. ++ -m s stereo ++ -m j jstereo ++ -m f forced mid/side stereo ++ ++ mono is the default mode for mono input files. If "-m m" is specified ++ for a stereo input file, the two channels will be averaged into a mono ++ signal. ++ ++ jstereo is the default mode for stereo files with fixed bitrates of ++ 128kbs or less. At higher fixed bitrates, the default is stereo. ++ For VBR encoding, jstereo is the default for VBR_q >4, and stereo ++ is the default for VBR_q <=4. You can override all of these defaults ++ by specifing the mode on the command line. ++ ++ jstereo means the encoder can use (on a frame by frame bases) either ++ regular stereo (just encode left and right channels independently) ++ or mid/side stereo. In mid/side stereo, the mid (L+R) and side (L-R) ++ channels are encoded, and more bits are allocated to the mid channel ++ than the side channel. This will effectively increase the bandwidth ++ if the signal does not have too much stereo separation. ++ ++ Mid/side stereo is basically a trick to increase bandwidth. At 128kbs, ++ it is clearly worth while. At higher bitrates it is less usefull. ++ ++ Using mid/side stereo inappropriately can result in audible ++ compression artifacts. To much switching between mid/side and regular ++ stereo can also sound bad. To determine when to switch to mid/side ++ stereo, LAME uses a much more sophisticated algorithm than that ++ described in the ISO documentation. ++ ++ -m f forces all frames to be encoded mid/side stereo. It ++ should only be used if you are sure every frame of the input file ++ has very little stereo seperation. ++ ++ ++ ++ ======================================================================= ++ MP3 input file ++ ======================================================================= ++ --mp3input ++ ++ Assume the input file is a MP3 file. Usefull for downsampling from ++ one mp3 to another. If the filename ends in ".mp3" LAME will assume ++ it is an MP3. For stdin or MP3 files which dont end in .mp3 you need ++ to use this switch. ++ ++ ++ ======================================================================= ++ disable historgram display ++ ======================================================================= ++ --nohist ++ ++ By default, LAME will display a bitrate histogram while producing ++ VBR mp3 files. This will disable that feature. ++ ++ ++ ======================================================================= ++ disable short blocks ++ ======================================================================= ++ --noshort ++ ++ Encode all frames using long blocks. ++ ++ ++ ++ ======================================================================= ++ non-original ++ ======================================================================= ++ -o ++ ++ mark the encoded file as a copy ++ ++ ++ ======================================================================= ++ CRC error protection ++ ======================================================================= ++ -p ++ ++ turn on CRC error protection. ++ Yes this really does work correctly in LAME. However, it takes ++ 16 bits per frame that would otherwise be used for encoding. ++ ++ ++ ++ ======================================================================= ++ input file is raw pcm ++ ======================================================================= ++ -r ++ ++ Assume the input file is raw pcm. Sampling rate and mono/stereo/jstereo ++ must be specified on the command line. Without -r, LAME will perform ++ several fseek()'s on the input file looking for WAV and AIFF headers. ++ ++ Not supported if LAME is compiled to use LIBSNDFILE. ++ ++ ++ ++ ======================================================================= ++ output sampling frequency in kHZ ++ ======================================================================= ++ --resample n ++ ++ where n = 16, 22.05, 24, 32, 44.1, 48 ++ ++ Output sampling frequency. Resample the input if necessary. ++ ++ If not specified, LAME may sometimes resample automatically ++ when faced with extreme compression conditions (like encoding ++ a 44.1khz input file at 16kbs). ++ ++ ++ ++ ======================================================================= ++ sampling frequency in kHZ ++ ======================================================================= ++ -s n ++ ++ where n = sampling rate in kHz. ++ ++ Required for raw PCM input files. Otherwise it will be determined ++ from the header information in the input file. ++ ++ LAME will automatically resample the input file to one of the ++ supported MP3 samplerates if necessary. ++ ++ ++ ======================================================================= ++ silent operation ++ ======================================================================= ++ -S ++ ++ don't print progress report ++ ++ ++ ++ ======================================================================= ++ disable Xing VBR tag ++ ======================================================================= ++ -t ++ ++ Disable writing of the Xing VBR Tag (only valid if -v flag is ++ specified) This tag in embedded in frame 0 of the MP3 file. It lets ++ VBR aware players correctly seek and compute playing times of VBR ++ files. ++ ++ ++ ++ ======================================================================= ++ variable bit rate (VBR) ++ ======================================================================= ++ -v ++ ++ Turn on VBR. There are several ways you can use VBR. I personally ++ like using VBR to get files slightly bigger than 128kbs files, where ++ the extra bits are used for the occasional difficult-to-encode frame. ++ For this, try specifying a minimum bitrate to use with VBR: ++ ++ lame -v -b 112 input.wav output.mp3 ++ ++ If the file is too big, use -V n, where n=0..9 ++ ++ lame -v -V n -b 112 input.wav output.mp3 ++ ++ ++ If you wan to use VBR to get the maximum compression possible, ++ and for this, you can try: ++ ++ lame -v input.wav output.mp3 ++ lame -v -V n input.wav output.mp3 (to very quality/filesize) ++ ++ ++ ++ ++ ======================================================================= ++ VBR quality setting ++ ======================================================================= ++ -V n ++ ++ n=0..9. Specifies the value of VBR_q. default=4. 0=highest quality. ++ ++ ++ ++ How is VBR_q used? ++ ++ OVER = number of scalefactor bands with distortion that exceeds the ++ allowed distortion given by the masking thresholds. OVER is computed ++ by outer_loop, and the masking thresholds are computed by the ++ psycho-acoustic model. ++ ++ VBR_q = the minimum value of OVER which is to be allowed. ++ LAME will choose the smallest bitrate for which OVER <= VBR_q. ++ (a minimum allowed bitrate can be set with -b. default=64kbs) ++ ++ If the frame contains short blocks, then the minimum bitrate is made ++ much larger since the OVER does not adequately measure distortion ++ caused by pre-echo. LAME uses bitrates of at least 160kbs for short ++ blocks to make sure they sound good. ++ ++ *NOTE* No psy-model is perfect, so there can often be distortion which ++ is audible even though the psy-model claims it is not! Thus using a ++ small minimum bitrate can result in some aggressive compression and ++ audible distortion even with -V 0. Thus using -V 0 does not sound ++ better than a fixed 256kbs encoding. For example: suppose in the 1kHz ++ frequency band the psy-model claims 20db of distortion will not be ++ detectable by the human ear, so LAME VBR-0 will compress that ++ frequency band as much as possible and introduce at most 20db of ++ distortion. Using a fixed 256kbit framesize, LAME could end up ++ introducing only 2db of distortion. If the psy-model was correct, ++ they will both sound the same. If the psy-model was wrong, the VBR-0 ++ result can sound worse. ++ ++ ++ ======================================================================= ++ voice encoding mode ++ ======================================================================= ++ --voice ++ ++ An experimental voice encoding mode. Tuned for 44.1kHz input files. ++ ++ ++ ======================================================================= ++ swapbytes ++ ======================================================================= ++ -x ++ ++ swap bytes in the input file. for sorting out little endian/big endian ++ type problems. If your encodings sound like static, try this first. ++ ++ +diff -r -c -N encoder/VbrTag.c lame3.70/VbrTag.c +*** encoder/VbrTag.c Wed Dec 31 17:00:00 1969 +--- lame3.70/VbrTag.c Thu Apr 6 12:50:49 2000 +*************** +*** 0 **** +--- 1,494 ---- ++ /* ++ * Xing VBR tagging for LAME. ++ * ++ * Copyright (c) 1999 A.L. Faber ++ * ++ * This library is free software; you can redistribute it and/or ++ * modify it under the terms of the GNU Library General Public ++ * License as published by the Free Software Foundation; either ++ * version 2 of the License, or (at your option) any later version. ++ * ++ * This library is distributed in the hope that it will be useful, ++ * but WITHOUT ANY WARRANTY; without even the implied warranty of ++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU ++ * Library General Public License for more details. ++ * ++ * You should have received a copy of the GNU Library General Public ++ * License along with this library; if not, write to the ++ * Free Software Foundation, Inc., 59 Temple Place - Suite 330, ++ * Boston, MA 02111-1307, USA. ++ */ ++ ++ #include "machine.h" ++ #include <math.h> ++ #include "VbrTag.h" ++ #include "version.h" ++ ++ #ifdef _DEBUG ++ /* #define DEBUG_VBRTAG */ ++ #endif ++ ++ ++ int SizeOfEmptyFrame[2][2]= ++ { ++ {32,17}, ++ {17,9}, ++ }; ++ ++ static u_char pbtStreamBuffer[216]; ++ static long g_Position[NUMTOCENTRIES]; ++ static int nZeroStreamSize=0; ++ static int TotalFrameSize=0; ++ static char VBRTag[]={"Xing"}; ++ ++ ++ int* pVbrFrames=NULL; ++ int nVbrNumFrames=0; ++ int nVbrFrameBufferSize=0; ++ ++ /**************************************************************************** ++ * AddVbrFrame: Add VBR entry, used to fill the VBR the TOC entries ++ * Paramters: ++ * nStreamPos: how many bytes did we write to the bitstream so far ++ * (in Bytes NOT Bits) ++ **************************************************************************** ++ */ ++ void AddVbrFrame(int nStreamPos) ++ { ++ /* Simple exponential growing buffer */ ++ if (pVbrFrames==NULL || nVbrFrameBufferSize==0) ++ { ++ /* Start with 100 frames */ ++ nVbrFrameBufferSize=100; ++ ++ /* Allocate them */ ++ pVbrFrames=(int*)malloc((size_t)(nVbrFrameBufferSize*sizeof(int))); ++ } ++ ++ /* Is buffer big enough to store this new frame */ ++ if (nVbrNumFrames==nVbrFrameBufferSize) ++ { ++ /* Guess not, double th e buffer size */ ++ nVbrFrameBufferSize*=2; ++ ++ /* Allocate new buffer */ ++ pVbrFrames=(int*)realloc(pVbrFrames,(size_t)(nVbrFrameBufferSize*sizeof(int))); ++ } ++ ++ /* Store values */ ++ pVbrFrames[nVbrNumFrames++]=nStreamPos; ++ } ++ ++ ++ /*-------------------------------------------------------------*/ ++ static int ExtractI4(unsigned char *buf) ++ { ++ int x; ++ /* big endian extract */ ++ x = buf[0]; ++ x <<= 8; ++ x |= buf[1]; ++ x <<= 8; ++ x |= buf[2]; ++ x <<= 8; ++ x |= buf[3]; ++ return x; ++ } ++ ++ void CreateI4(unsigned char *buf, int nValue) ++ { ++ /* big endian create */ ++ buf[0]=(nValue>>24)&0xff; ++ buf[1]=(nValue>>16)&0xff; ++ buf[2]=(nValue>> 8)&0xff; ++ buf[3]=(nValue )&0xff; ++ } ++ ++ ++ /*-------------------------------------------------------------*/ ++ /* Same as GetVbrTag below, but only checks for the Xing tag. ++ requires buf to contain only 40 bytes */ ++ /*-------------------------------------------------------------*/ ++ int CheckVbrTag(unsigned char *buf) ++ { ++ int h_id, h_mode, h_sr_index; ++ ++ /* get selected MPEG header data */ ++ h_id = (buf[1] >> 3) & 1; ++ h_sr_index = (buf[2] >> 2) & 3; ++ h_mode = (buf[3] >> 6) & 3; ++ ++ /* determine offset of header */ ++ if( h_id ) ++ { ++ /* mpeg1 */ ++ if( h_mode != 3 ) buf+=(32+4); ++ else buf+=(17+4); ++ } ++ else ++ { ++ /* mpeg2 */ ++ if( h_mode != 3 ) buf+=(17+4); ++ else buf+=(9+4); ++ } ++ ++ if( buf[0] != VBRTag[0] ) return 0; /* fail */ ++ if( buf[1] != VBRTag[1] ) return 0; /* header not found*/ ++ if( buf[2] != VBRTag[2] ) return 0; ++ if( buf[3] != VBRTag[3] ) return 0; ++ return 1; ++ } ++ ++ int GetVbrTag(VBRTAGDATA *pTagData, unsigned char *buf) ++ { ++ int i, head_flags; ++ int h_id, h_mode, h_sr_index; ++ static int sr_table[4] = { 44100, 48000, 32000, 99999 }; ++ ++ /* get Vbr header data */ ++ pTagData->flags = 0; ++ ++ /* get selected MPEG header data */ ++ h_id = (buf[1] >> 3) & 1; ++ h_sr_index = (buf[2] >> 2) & 3; ++ h_mode = (buf[3] >> 6) & 3; ++ ++ /* determine offset of header */ ++ if( h_id ) ++ { ++ /* mpeg1 */ ++ if( h_mode != 3 ) buf+=(32+4); ++ else buf+=(17+4); ++ } ++ else ++ { ++ /* mpeg2 */ ++ if( h_mode != 3 ) buf+=(17+4); ++ else buf+=(9+4); ++ } ++ ++ if( buf[0] != VBRTag[0] ) return 0; /* fail */ ++ if( buf[1] != VBRTag[1] ) return 0; /* header not found*/ ++ if( buf[2] != VBRTag[2] ) return 0; ++ if( buf[3] != VBRTag[3] ) return 0; ++ ++ buf+=4; ++ ++ pTagData->h_id = h_id; ++ ++ pTagData->samprate = sr_table[h_sr_index]; ++ ++ if( h_id == 0 ) ++ pTagData->samprate >>= 1; ++ ++ head_flags = pTagData->flags = ExtractI4(buf); buf+=4; /* get flags */ ++ ++ if( head_flags & FRAMES_FLAG ) ++ { ++ pTagData->frames = ExtractI4(buf); buf+=4; ++ } ++ ++ if( head_flags & BYTES_FLAG ) ++ { ++ pTagData->bytes = ExtractI4(buf); buf+=4; ++ } ++ ++ if( head_flags & TOC_FLAG ) ++ { ++ if( pTagData->toc != NULL ) ++ { ++ for(i=0;i<NUMTOCENTRIES;i++) ++ pTagData->toc[i] = buf[i]; ++ } ++ buf+=NUMTOCENTRIES; ++ } ++ ++ pTagData->vbr_scale = -1; ++ ++ if( head_flags & VBR_SCALE_FLAG ) ++ { ++ pTagData->vbr_scale = ExtractI4(buf); buf+=4; ++ } ++ ++ #ifdef DEBUG_VBRTAG ++ printf("\n\n********************* VBR TAG INFO *****************\n"); ++ printf("tag :%s\n",VBRTag); ++ printf("head_flags :%d\n",head_flags); ++ printf("bytes :%d\n",pTagData->bytes); ++ printf("frames :%d\n",pTagData->frames); ++ printf("VBR Scale :%d\n",pTagData->vbr_scale); ++ printf("toc:\n"); ++ if( pTagData->toc != NULL ) ++ { ++ for(i=0;i<NUMTOCENTRIES;i++) ++ { ++ if( (i%10) == 0 ) printf("\n"); ++ printf(" %3d", (int)(pTagData->toc[i])); ++ } ++ } ++ printf("\n***************** END OF VBR TAG INFO ***************\n"); ++ #endif ++ return 1; /* success */ ++ } ++ ++ ++ /**************************************************************************** ++ * InitVbrTag: Initializes the header, and write empty frame to stream ++ * Paramters: ++ * fpStream: pointer to output file stream ++ * nVersion: 0= MPEG1 1=MPEG2 ++ * nMode : Channel Mode: 0=STEREO 1=JS 2=DS 3=MONO ++ **************************************************************************** ++ */ ++ int InitVbrTag(Bit_stream_struc* pBs,int nVersion, int nMode, int SampIndex) ++ { ++ int i; ++ ++ /* Clear Frame position array variables */ ++ pVbrFrames=NULL; ++ nVbrNumFrames=0; ++ nVbrFrameBufferSize=0; ++ ++ /* Clear struct */ ++ memset(g_Position,0x00,sizeof(g_Position)); ++ ++ /* Clear stream buffer */ ++ memset(pbtStreamBuffer,0x00,sizeof(pbtStreamBuffer)); ++ ++ /* Set TOC values to 255 */ ++ for (i=0;i<NUMTOCENTRIES;i++) ++ { ++ g_Position[i]=-1; ++ } ++ ++ ++ ++ /* Reserve the proper amount of bytes */ ++ if (nMode==3) ++ { ++ nZeroStreamSize=SizeOfEmptyFrame[nVersion][1]+4; ++ } ++ else ++ { ++ nZeroStreamSize=SizeOfEmptyFrame[nVersion][0]+4; ++ } ++ ++ /* ++ // Xing VBR pretends to be a 48kbs layer III frame. (at 44.1kHz). ++ // (at 48kHz they use 56kbs since 48kbs frame not big enough for ++ // table of contents) ++ // let's always embed Xing header inside a 64kbs layer III frame. ++ // this gives us enough room for a LAME version string too. ++ // size determined by sampling frequency (MPEG1) ++ // 32kHz: 216 bytes@48kbs 288bytes@ 64kbs ++ // 44.1kHz: 156 bytes 208bytes@64kbs (+1 if padding = 1) ++ // 48kHz: 144 bytes 192 ++ // ++ // MPEG 2 values are the since the framesize and samplerate ++ // are each reduced by a factor of 2. ++ */ ++ { ++ int tot; ++ static const int framesize[3]={208,192,288}; /* 64kbs MPEG1 or MPEG2 framesize */ ++ /* static int framesize[3]={156,144,216}; */ /* 48kbs framesize */ ++ ++ if (SampIndex>2) { ++ fprintf(stderr,"illegal sampling frequency index\n"); ++ exit(-1); ++ } ++ TotalFrameSize= framesize[SampIndex]; ++ tot = (nZeroStreamSize+VBRHEADERSIZE); ++ tot += 20; /* extra 20 bytes for LAME & version string */ ++ ++ if (TotalFrameSize < tot ) { ++ fprintf(stderr,"Xing VBR header problem...use -t\n"); ++ exit(-1); ++ } ++ } ++ ++ ++ /* Put empty bytes into the bitstream */ ++ for (i=0;i<TotalFrameSize;i++) ++ { ++ /* Write a byte to the bitstream */ ++ putbits(pBs,0,8); ++ } ++ ++ /* Success */ ++ return 0; ++ } ++ ++ ++ ++ /**************************************************************************** ++ * PutVbrTag: Write final VBR tag to the file ++ * Paramters: ++ * lpszFileName: filename of MP3 bit stream ++ * nVersion: 0= MPEG1 1=MPEG2 ++ * nVbrScale : encoder quality indicator (0..100) ++ **************************************************************************** ++ */ ++ int PutVbrTag(char* lpszFileName,int nVbrScale,int nVersion) ++ { ++ int i; ++ long lFileSize; ++ int nStreamIndex; ++ char abyte; ++ u_char btToc[NUMTOCENTRIES]; ++ FILE *fpStream; ++ char str1[80]; ++ ++ ++ if (nVbrNumFrames==0 || pVbrFrames==NULL) ++ return -1; ++ ++ /* Open the bitstream again */ ++ fpStream=fopen(lpszFileName,"rb+"); ++ ++ /* Assert stream is valid */ ++ if (fpStream==NULL) ++ return -1; ++ ++ /* Clear stream buffer */ ++ memset(pbtStreamBuffer,0x00,sizeof(pbtStreamBuffer)); ++ ++ /* Seek to end of file*/ ++ fseek(fpStream,0,SEEK_END); ++ ++ /* Get file size */ ++ lFileSize=ftell(fpStream); ++ ++ /* Abort if file has zero length. Yes, it can happen :) */ ++ if (lFileSize==0) ++ return -1; ++ ++ /* Seek to first real frame */ ++ fseek(fpStream,(long)TotalFrameSize,SEEK_SET); ++ ++ /* Read the header (first valid frame) */ ++ fread(pbtStreamBuffer,4,1,fpStream); ++ ++ /* the default VBR header. 48kbs layer III, no padding, no crc */ ++ /* but sampling freq, mode andy copyright/copy protection taken */ ++ /* from first valid frame */ ++ pbtStreamBuffer[0]=(u_char) 0xff; ++ if (nVersion==0) { ++ pbtStreamBuffer[1]=(u_char) 0xfb; ++ abyte = pbtStreamBuffer[2] & (char) 0x0c; ++ pbtStreamBuffer[2]=(char) 0x50 | abyte; /* 64kbs MPEG1 frame */ ++ }else{ ++ pbtStreamBuffer[1]=(u_char) 0xf3; ++ abyte = pbtStreamBuffer[2] & (char) 0x0c; ++ pbtStreamBuffer[2]=(char) 0x80 | abyte; /* 64kbs MPEG2 frame */ ++ } ++ ++ ++ /*Seek to the beginning of the stream */ ++ fseek(fpStream,0,SEEK_SET); ++ ++ /* Clear all TOC entries */ ++ memset(btToc,0,sizeof(btToc)); ++ ++ for (i=1;i<NUMTOCENTRIES;i++) /* Don't touch zero point... */ ++ { ++ /* Calculate frame from given percentage */ ++ int frameNum=(int)(floor(0.01*i*nVbrNumFrames)); ++ ++ /* Calculate relative file postion, normalized to 0..256!(?) */ ++ float fRelStreamPos=(float)256.0*(float)pVbrFrames[frameNum]/(float)lFileSize; ++ ++ /* Just to be safe */ ++ if (fRelStreamPos>255) fRelStreamPos=255; ++ ++ /* Assign toc entry value */ ++ btToc[i]=(u_char) fRelStreamPos; ++ } ++ ++ ++ ++ /* Start writing the tag after the zero frame */ ++ nStreamIndex=nZeroStreamSize; ++ ++ /* Put Vbr tag */ ++ pbtStreamBuffer[nStreamIndex++]=VBRTag[0]; ++ pbtStreamBuffer[nStreamIndex++]=VBRTag[1]; ++ pbtStreamBuffer[nStreamIndex++]=VBRTag[2]; ++ pbtStreamBuffer[nStreamIndex++]=VBRTag[3]; ++ ++ /* Put header flags */ ++ CreateI4(&pbtStreamBuffer[nStreamIndex],FRAMES_FLAG+BYTES_FLAG+TOC_FLAG+VBR_SCALE_FLAG); ++ nStreamIndex+=4; ++ ++ /* Put Total Number of frames */ ++ CreateI4(&pbtStreamBuffer[nStreamIndex],nVbrNumFrames); ++ nStreamIndex+=4; ++ ++ /* Put Total file size */ ++ CreateI4(&pbtStreamBuffer[nStreamIndex],(int)lFileSize); ++ nStreamIndex+=4; ++ ++ /* Put TOC */ ++ memcpy(&pbtStreamBuffer[nStreamIndex],btToc,sizeof(btToc)); ++ nStreamIndex+=sizeof(btToc); ++ ++ /* Put VBR SCALE */ ++ CreateI4(&pbtStreamBuffer[nStreamIndex],nVbrScale); ++ nStreamIndex+=4; ++ ++ /* Put LAME id */ ++ sprintf(str1,"LAME%s",get_lame_version()); ++ strncpy((char *)&pbtStreamBuffer[nStreamIndex],str1,(size_t) 20); ++ nStreamIndex+=20; ++ ++ ++ #ifdef DEBUG_VBRTAG ++ { ++ VBRTAGDATA TestHeader; ++ GetVbrTag(&TestHeader,pbtStreamBuffer); ++ } ++ #endif ++ ++ /* Put it all to disk again */ ++ if (fwrite(pbtStreamBuffer,TotalFrameSize,1,fpStream)!=1) ++ { ++ return -1; ++ } ++ fclose(fpStream); ++ ++ /* Save to delete the frame buffer */ ++ free(pVbrFrames); ++ pVbrFrames=NULL; ++ ++ return 0; /* success */ ++ } ++ ++ /*-------------------------------------------------------------*/ ++ int SeekPoint(unsigned char TOC[NUMTOCENTRIES], int file_bytes, float percent) ++ { ++ /* interpolate in TOC to get file seek point in bytes */ ++ int a, seekpoint; ++ float fa, fb, fx; ++ ++ ++ if( percent < (float)0.0 ) percent = (float)0.0; ++ if( percent > (float)100.0 ) percent = (float)100.0; ++ ++ a = (int)percent; ++ if( a > 99 ) a = 99; ++ fa = TOC[a]; ++ if( a < 99 ) { ++ fb = TOC[a+1]; ++ } ++ else { ++ fb = (float)256.0; ++ } ++ ++ ++ fx = fa + (fb-fa)*(percent-a); ++ ++ seekpoint = (int)(((float)(1.0/256.0))*fx*file_bytes); ++ ++ ++ return seekpoint; ++ } ++ /*-------------------------------------------------------------*/ +diff -r -c -N encoder/VbrTag.h lame3.70/VbrTag.h +*** encoder/VbrTag.h Wed Dec 31 17:00:00 1969 +--- lame3.70/VbrTag.h Thu Apr 6 12:50:49 2000 +*************** +*** 0 **** +--- 1,86 ---- ++ /* ++ * Xing VBR tagging for LAME. ++ * ++ * Copyright (c) 1999 A.L. Faber ++ * ++ * This library is free software; you can redistribute it and/or ++ * modify it under the terms of the GNU Library General Public ++ * License as published by the Free Software Foundation; either ++ * version 2 of the License, or (at your option) any later version. ++ * ++ * This library is distributed in the hope that it will be useful, ++ * but WITHOUT ANY WARRANTY; without even the implied warranty of ++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU ++ * Library General Public License for more details. ++ * ++ * You should have received a copy of the GNU Library General Public ++ * License along with this library; if not, write to the ++ * Free Software Foundation, Inc., 59 Temple Place - Suite 330, ++ * Boston, MA 02111-1307, USA. ++ */ ++ ++ #ifndef VRBTAG_H_INCLUDED ++ #define VRBTAG_H_INCLUDED ++ ++ /* ----------------------------------------------------------- ++ * A Vbr header may be present in the ancillary ++ * data field of the first frame of an mp3 bitstream ++ * The Vbr header (optionally) contains ++ * frames total number of audio frames in the bitstream ++ * bytes total number of bytes in the bitstream ++ * toc table of contents ++ ++ * toc (table of contents) gives seek points ++ * for random access ++ * the ith entry determines the seek point for ++ * i-percent duration ++ * seek point in bytes = (toc[i]/256.0) * total_bitstream_bytes ++ * e.g. half duration seek point = (toc[50]/256.0) * total_bitstream_bytes ++ */ ++ ++ #include "l3bitstream.h" ++ ++ #define FRAMES_FLAG 0x0001 ++ #define BYTES_FLAG 0x0002 ++ #define TOC_FLAG 0x0004 ++ #define VBR_SCALE_FLAG 0x0008 ++ ++ #define NUMTOCENTRIES 100 ++ ++ #define FRAMES_AND_BYTES (FRAMES_FLAG | BYTES_FLAG) ++ ++ /*structure to receive extracted header */ ++ /* toc may be NULL*/ ++ typedef struct ++ { ++ int h_id; /* from MPEG header, 0=MPEG2, 1=MPEG1 */ ++ int samprate; /* determined from MPEG header */ ++ int flags; /* from Vbr header data */ ++ int frames; /* total bit stream frames from Vbr header data */ ++ int bytes; /* total bit stream bytes from Vbr header data*/ ++ int vbr_scale; /* encoded vbr scale from Vbr header data*/ ++ u_char toc[NUMTOCENTRIES]; /* may be NULL if toc not desired*/ ++ } VBRTAGDATA; ++ /* ++ // 4 bytes for Header Tag ++ // 4 bytes for Header Flags ++ // 100 bytes for entry (NUMTOCENTRIES) ++ // 4 bytes for FRAME SIZE ++ // 4 bytes for STREAM_SIZE ++ // 4 bytes for VBR SCALE. a VBR quality indicator: 0=best 100=worst ++ // 20 bytes for LAME tag. for example, "LAME3.12 (beta 6)" ++ // ___________ ++ // 140 bytes ++ */ ++ ++ #define VBRHEADERSIZE (NUMTOCENTRIES+4+4+4+4+4) ++ ++ int CheckVbrTag(unsigned char *buf); ++ int GetVbrTag(VBRTAGDATA *pTagData, unsigned char *buf); ++ int SeekPoint(unsigned char TOC[NUMTOCENTRIES], int file_bytes, float percent); ++ int InitVbrTag(Bit_stream_struc* pBs,int nVersion,int nMode, int SampIndex); ++ int PutVbrTag(char* lpszFileName,int nVbrScale,int nVersion); ++ void AddVbrFrame(int nStreamPos); ++ ++ ++ #endif +diff -r -c -N encoder/amiga_mpega.c lame3.70/amiga_mpega.c +*** encoder/amiga_mpega.c Wed Dec 31 17:00:00 1969 +--- lame3.70/amiga_mpega.c Thu Apr 6 12:50:49 2000 +*************** +*** 0 **** +--- 1,96 ---- ++ /* MPGLIB replacement using mpega.library (AmigaOS) ++ * Written by Thomas Wenzel and Sigbj�rn (CISC) Skj�ret. ++ * ++ * Big thanks to St�phane Tavernard for mpega.library. ++ * ++ */ ++ ++ #ifdef AMIGA_MPEGA ++ ++ #include "lame.h" ++ #include "util.h" ++ #include <stdio.h> ++ #include <stdlib.h> ++ #include <proto/exec.h> ++ #include <dos.h> ++ #include <proto/mpega.h> ++ ++ struct Library *MPEGABase; ++ MPEGA_STREAM *mstream=NULL; ++ MPEGA_CTRL mctrl; ++ ++ ++ static int break_cleanup(void) ++ { ++ /* Dummy break function to make atexit() work. :P */ ++ return 1; ++ } ++ ++ static void exit_cleanup(void) ++ { ++ if(mstream) { ++ MPEGA_close(mstream); ++ mstream = NULL; ++ } ++ if(MPEGABase) { ++ CloseLibrary(MPEGABase); ++ MPEGABase = NULL; ++ } ++ } ++ ++ ++ int lame_decode_initfile(const char *fullname, int *stereo, int *samp, int *bitrate, unsigned long *nsamp) ++ { ++ mctrl.bs_access = NULL; ++ ++ mctrl.layer_1_2.mono.quality = 2; ++ mctrl.layer_1_2.stereo.quality = 2; ++ mctrl.layer_1_2.mono.freq_div = 1; ++ mctrl.layer_1_2.stereo.freq_div = 1; ++ mctrl.layer_1_2.mono.freq_max = 48000; ++ mctrl.layer_1_2.stereo.freq_max = 48000; ++ mctrl.layer_3.mono.quality = 2; ++ mctrl.layer_3.stereo.quality = 2; ++ mctrl.layer_3.mono.freq_div = 1; ++ mctrl.layer_3.stereo.freq_div = 1; ++ mctrl.layer_3.mono.freq_max = 48000; ++ mctrl.layer_3.stereo.freq_max = 48000; ++ mctrl.layer_1_2.force_mono = 0; ++ mctrl.layer_3.force_mono = 0; ++ ++ MPEGABase = OpenLibrary("mpega.library", 2); ++ if(!MPEGABase) { ++ fprintf(stderr, "Unable to open mpega.library v2\n"); ++ exit(1); ++ } ++ onbreak(break_cleanup); ++ atexit(exit_cleanup); ++ ++ mstream=MPEGA_open(fullname, &mctrl); ++ if(!mstream) { return (-1); } ++ ++ *stereo = mstream->dec_channels; ++ *samp = mstream->dec_frequency; ++ *bitrate = mstream->bitrate; ++ /* *nsamp = MAX_U_32_NUM; */ ++ *nsamp = (FLOAT)mstream->ms_duration/1000 * mstream->dec_frequency; ++ ++ return 0; ++ } ++ ++ int lame_decode_fromfile(FILE *fd, short pcm_l[],short pcm_r[]) ++ { ++ int outsize=0; ++ WORD *b[MPEGA_MAX_CHANNELS]; ++ ++ b[0]=pcm_l; ++ b[1]=pcm_r; ++ ++ while (outsize == 0) ++ outsize = MPEGA_decode_frame(mstream, b); ++ ++ if (outsize < 0) { return (-1); } ++ else { return outsize; } ++ } ++ ++ #endif /* AMIGA_MPEGA */ +diff -r -c -N encoder/auenc lame3.70/auenc +*** encoder/auenc Wed Dec 31 17:00:00 1969 +--- lame3.70/auenc Wed Nov 24 01:41:13 1999 +*************** +*** 0 **** +--- 1,39 ---- ++ #!/bin/sh ++ # ++ # auenc -- version 0.1 ++ # ++ # A wrapper for lame to encode multiple files. By default, a .wav ++ # extension is removed and replaced by .mp3 . ++ # ++ # (C) 1999 Gerhard Wesp <gwesp@cosy.sbg.ac.at> under the GPL. ++ ++ # set the variables below according to your taste ++ LAME=/usr/bin/lame ++ LAME_OPTS="-S -h -v -V 0 -b 256" # high quality, silent operation ++ ++ if [ $# -lt 1 ] ; then ++ exec 1>&2 ++ cat << _EOF_ ++ usage: $0 [options] file... ++ options: ++ -d --delete: delete original file after successful encoding ++ _EOF_ ++ exit 1 ++ fi ++ ++ unset DELETE ++ case "$1" in ( -d | --delete ) ++ DELETE=1 ++ shift ;; ++ esac ++ ++ for f in $* ; do ++ if ! $LAME $LAME_OPTS "$f" `basename "$f" .wav`.mp3 ; then ++ exec 1>&2 ++ echo "encoding of $f failed, aborting..." ++ exit 1 ++ fi ++ if [ -n "$DELETE" ] ; then ++ rm -f "$f" ++ fi ++ done +diff -r -c -N encoder/brhist.c lame3.70/brhist.c +*** encoder/brhist.c Wed Dec 31 17:00:00 1969 +--- lame3.70/brhist.c Thu Apr 6 12:50:49 2000 +*************** +*** 0 **** +--- 1,135 ---- ++ #ifdef BRHIST ++ #include <string.h> ++ #include "brhist.h" ++ #include "util.h" ++ #include <termcap.h> ++ ++ ++ #define BRHIST_BARMAX 50 ++ int disp_brhist = 1; ++ long brhist_count[15]; ++ long brhist_temp[15]; ++ int brhist_vbrmin; ++ int brhist_vbrmax; ++ long brhist_max; ++ char brhist_bps[15][5]; ++ char brhist_backcur[200]; ++ char brhist_bar[BRHIST_BARMAX+10]; ++ char brhist_spc[BRHIST_BARMAX+1]; ++ ++ char stderr_buff[BUFSIZ]; ++ ++ ++ void brhist_init(lame_global_flags *gfp,int br_min, int br_max) ++ { ++ int i; ++ char term_buff[1024]; ++ char *termname; ++ char *tp; ++ char tc[10]; ++ ++ for(i = 0; i < 15; i++) ++ { ++ sprintf(brhist_bps[i], "%3d:", bitrate_table[gfp->version][i]); ++ brhist_count[i] = 0; ++ brhist_temp[i] = 0; ++ } ++ ++ brhist_vbrmin = br_min; ++ brhist_vbrmax = br_max; ++ ++ brhist_max = 0; ++ ++ memset(&brhist_bar[0], '*', BRHIST_BARMAX); ++ brhist_bar[BRHIST_BARMAX] = '\0'; ++ memset(&brhist_spc[0], ' ', BRHIST_BARMAX); ++ brhist_spc[BRHIST_BARMAX] = '\0'; ++ brhist_backcur[0] = '\0'; ++ ++ if ((termname = getenv("TERM")) == NULL) ++ { ++ fprintf(stderr, "can't get TERM environment string.\n"); ++ disp_brhist = 0; ++ return; ++ } ++ ++ if (tgetent(term_buff, termname) != 1) ++ { ++ fprintf(stderr, "can't find termcap entry: %s\n", termname); ++ disp_brhist = 0; ++ return; ++ } ++ ++ tc[0] = '\0'; ++ tp = &tc[0]; ++ tp=tgetstr("up", &tp); ++ brhist_backcur[0] = '\0'; ++ for(i = br_min-1; i <= br_max; i++) ++ strcat(brhist_backcur, tp); ++ setbuf(stderr, stderr_buff); ++ } ++ ++ void brhist_add_count(void) ++ { ++ int i; ++ ++ for(i = brhist_vbrmin; i <= brhist_vbrmax; i++) ++ { ++ brhist_count[i] += brhist_temp[i]; ++ if (brhist_count[i] > brhist_max) ++ brhist_max = brhist_count[i]; ++ brhist_temp[i] = 0; ++ } ++ } ++ ++ void brhist_disp(void) ++ { ++ int i; ++ long full; ++ int barlen; ++ ++ full = (brhist_max < BRHIST_BARMAX) ? BRHIST_BARMAX : brhist_max; ++ fputc('\n', stderr); ++ for(i = brhist_vbrmin; i <= brhist_vbrmax; i++) ++ { ++ barlen = (brhist_count[i]*BRHIST_BARMAX+full-1) / full; ++ fputs(brhist_bps[i], stderr); ++ fputs(&brhist_bar[BRHIST_BARMAX - barlen], stderr); ++ fputs(&brhist_spc[barlen], stderr); ++ fputc('\n', stderr); ++ } ++ fputs(brhist_backcur, stderr); ++ fflush(stderr); ++ } ++ ++ void brhist_disp_total(lame_global_flags *gfp) ++ { ++ int i; ++ FLOAT ave; ++ ++ for(i = brhist_vbrmin; i <= brhist_vbrmax; i++) ++ fputc('\n', stderr); ++ ++ ave=0; ++ for(i = brhist_vbrmin; i <= brhist_vbrmax; i++) ++ ave += bitrate_table[gfp->version][i]* ++ (FLOAT)brhist_count[i] / gfp->totalframes; ++ fprintf(stderr, "\naverage: %2.0f kbs\n",ave); ++ ++ #if 0 ++ fprintf(stderr, "----- bitrate statistics -----\n"); ++ fprintf(stderr, " [kbps] frames\n"); ++ for(i = brhist_vbrmin; i <= brhist_vbrmax; i++) ++ { ++ fprintf(stderr, " %3d %8ld (%.1f%%)\n", ++ bitrate_table[gfp->version][i], ++ brhist_count[i], ++ (FLOAT)brhist_count[i] / gfp->totalframes * 100.0); ++ } ++ #endif ++ fflush(stderr); ++ } ++ ++ #endif /* BRHIST */ ++ ++ +diff -r -c -N encoder/brhist.h lame3.70/brhist.h +*** encoder/brhist.h Wed Dec 31 17:00:00 1969 +--- lame3.70/brhist.h Tue Mar 21 16:02:17 2000 +*************** +*** 0 **** +--- 1,15 ---- ++ #ifdef BRHIST ++ #ifndef BRHIST_H_INCLUDED ++ #define BRHIST_H_INCLUDED ++ ++ extern int disp_brhist; ++ ++ #include "lame.h" ++ void brhist_init(lame_global_flags *gfp,int br_min, int br_max); ++ void brhist_add_count(void); ++ void brhist_disp(void); ++ void brhist_disp_total(lame_global_flags *gfp); ++ extern long brhist_temp[15]; ++ #endif ++ ++ #endif +diff -r -c -N encoder/common.c lame3.70/common.c +*** encoder/common.c Wed Jan 22 02:43:13 1997 +--- lame3.70/common.c Wed Dec 31 17:00:00 1969 +*************** +*** 1,1439 **** +- /********************************************************************** +- * ISO MPEG Audio Subgroup Software Simulation Group (1996) +- * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- * +- * $Id: common.c,v 1.2 1997/01/19 22:28:29 rowlands Exp $ +- * +- * $Log: common.c,v $ +- * Revision 1.2 1997/01/19 22:28:29 rowlands +- * Layer 3 bug fixes from Seymour Shlien +- * +- * Revision 1.1 1996/02/14 04:04:23 rowlands +- * Initial revision +- * +- * Received from Mike Coleman +- **********************************************************************/ +- /********************************************************************** +- * date programmers comment * +- * 2/25/91 Doulas Wong, start of version 1.0 records * +- * Davis Pan * +- * 5/10/91 W. Joseph Carter Created this file for all common * +- * functions and global variables. * +- * Ported to Macintosh and Unix. * +- * Added Jean-Georges Fritsch's * +- * "bitstream.c" package. * +- * Added routines to handle AIFF PCM * +- * sound files. * +- * Added "mem_alloc()" and "mem_free()" * +- * routines for memory allocation * +- * portability. * +- * Added routines to convert between * +- * Apple SANE extended floating point * +- * format and IEEE double precision * +- * floating point format. For AIFF. * +- * 02jul91 dpwe (Aware Inc) Moved allocation table input here; * +- * Tables read from subdir TABLES_PATH. * +- * Added some debug printout fns (Write*)* +- * 7/10/91 Earle Jennings replacement of the one float by FLOAT * +- * port to MsDos from MacIntosh version * +- * 8/ 5/91 Jean-Georges Fritsch fixed bug in open_bit_stream_r() * +- *10/ 1/91 S.I. Sudharsanan, Ported to IBM AIX platform. * +- * Don H. Lee, * +- * Peter W. Farrett * +- *10/3/91 Don H. Lee implemented CRC-16 error protection * +- * newly introduced functions are * +- * I_CRC_calc, II_CRC_calc and * +- * update_CRC. Additions and revisions * +- * are marked with dhl for clarity * +- *10/18/91 Jean-Georges Fritsch fixed bug in update_CRC(), * +- * II_CRC_calc() and I_CRC_calc() * +- * 2/11/92 W. Joseph Carter Ported new code to Macintosh. Most * +- * important fixes involved changing * +- * 16-bit ints to long or unsigned in * +- * bit alloc routines for quant of 65535 * +- * and passing proper function args. * +- * Removed "Other Joint Stereo" option * +- * and made bitrate be total channel * +- * bitrate, irrespective of the mode. * +- * Fixed many small bugs & reorganized. * +- * 3/20/92 Jean-Georges Fritsch fixed bug in start-of-frame search * +- * 6/15/92 Juan Pineda added refill_buffer(bs) "n" * +- * initialization * +- * 7/08/92 Susanne Ritscher MS-DOS, MSC6.0 port fixes * +- * 7/27/92 Mike Li (re-)Port to MS-DOS * +- * 8/19/92 Soren H. Nielsen Fixed bug in I_CRC_calc and in * +- * II_CRC_calc. Added function: new_ext * +- * for better MS-DOS compatability * +- * 3/10/93 Kevin Peterson changed aiff_read_headers to handle * +- * chunks in any order. now returns * +- * position of sound data in file. * +- * 3/31/93 Jens Spille changed IFF_* string compares to use * +- * strcmp() * +- * 5/30/93 Masahiro Iwadare removed the previous modification * +- * for UNIX. * +- * 8/27/93 Seymour Shlien, Fixes in Unix and MSDOS ports, * +- * Daniel Lauzon, and * +- * Bill Truerniet * +- *--------------------------------------------------------------------* +- * 8/24/93 Masahiro Iwadare Included IS modification in Layer III.* +- * Changed for 1 pass decoding. * +- * 9/07/93 Toshiyuki Ishino Integrated Layer III with Ver 3.9. * +- *--------------------------------------------------------------------* +- * 11/20/93 Masahiro Iwadare Integrated Layer III with Ver 4.0. * +- *--------------------------------------------------------------------* +- * 7/14/94 Juergen Koller rewind of bitbuffer added * +- * 6/12/95 Soeren H. Nielsen Bug fix in new_ext(). * +- * 7/11/95 Soeren H. Nielsen Changes for MPEG-2 LSF Layer I and II * +- *--------------------------------------------------------------------* +- * 8/02/95 mc@fivebats.com Added code to determine byte-order, * +- * fixes to AIFF routines. Modified Mac * +- * code to work with new Apple headers * +- **********************************************************************/ +- +- /*********************************************************************** +- * +- * Global Include Files +- * +- ***********************************************************************/ +- +- #include "common.h" +- #include <string.h> /* 1995-07-11 shn */ +- #include <ctype.h> +- #include <stdlib.h> +- +- /*********************************************************************** +- * +- * Global Variable Definitions +- * +- ***********************************************************************/ +- +- char *mode_names[4] = { "stereo", "j-stereo", "dual-ch", "single-ch" }; +- char *layer_names[3] = { "I", "II", "III" }; +- char *version_names[2] = { "MPEG-2 LSF", "MPEG-1" }; +- +- /* 1: MPEG-1, 0: MPEG-2 LSF, 1995-07-11 shn */ +- double s_freq[2][4] = {{22.05, 24, 16, 0}, {44.1, 48, 32, 0}}; +- +- /* 1: MPEG-1, 0: MPEG-2 LSF, 1995-07-11 shn */ +- int bitrate[2][3][15] = { +- {{0,32,48,56,64,80,96,112,128,144,160,176,192,224,256}, +- {0,8,16,24,32,40,48,56,64,80,96,112,128,144,160}, +- {0,8,16,24,32,40,48,56,64,80,96,112,128,144,160}}, +- {{0,32,64,96,128,160,192,224,256,288,320,352,384,416,448}, +- {0,32,48,56,64,80,96,112,128,160,192,224,256,320,384}, +- {0,32,40,48,56,64,80,96,112,128,160,192,224,256,320}} +- }; +- +- double FAR multiple[64] = { +- 2.00000000000000, 1.58740105196820, 1.25992104989487, +- 1.00000000000000, 0.79370052598410, 0.62996052494744, 0.50000000000000, +- 0.39685026299205, 0.31498026247372, 0.25000000000000, 0.19842513149602, +- 0.15749013123686, 0.12500000000000, 0.09921256574801, 0.07874506561843, +- 0.06250000000000, 0.04960628287401, 0.03937253280921, 0.03125000000000, +- 0.02480314143700, 0.01968626640461, 0.01562500000000, 0.01240157071850, +- 0.00984313320230, 0.00781250000000, 0.00620078535925, 0.00492156660115, +- 0.00390625000000, 0.00310039267963, 0.00246078330058, 0.00195312500000, +- 0.00155019633981, 0.00123039165029, 0.00097656250000, 0.00077509816991, +- 0.00061519582514, 0.00048828125000, 0.00038754908495, 0.00030759791257, +- 0.00024414062500, 0.00019377454248, 0.00015379895629, 0.00012207031250, +- 0.00009688727124, 0.00007689947814, 0.00006103515625, 0.00004844363562, +- 0.00003844973907, 0.00003051757813, 0.00002422181781, 0.00001922486954, +- 0.00001525878906, 0.00001211090890, 0.00000961243477, 0.00000762939453, +- 0.00000605545445, 0.00000480621738, 0.00000381469727, 0.00000302772723, +- 0.00000240310869, 0.00000190734863, 0.00000151386361, 0.00000120155435, +- 1E-20 +- }; +- +- enum byte_order NativeByteOrder = order_unknown; +- +- /*********************************************************************** +- * +- * Global Function Definitions +- * +- ***********************************************************************/ +- +- /* The system uses a variety of data files. By opening them via this +- function, we can accommodate various locations. */ +- +- FILE *OpenTableFile(name) +- char *name; +- { +- char fulname[80]; +- char *envdir; +- FILE *f; +- +- fulname[0] = '\0'; +- +- #ifdef TABLES_PATH +- strcpy(fulname, TABLES_PATH); /* default relative path for tables */ +- #endif /* TABLES_PATH */ /* (includes terminal path seperator */ +- +- #ifdef UNIX /* envir. variables for UNIX only */ +- { +- char *getenv(); +- +- envdir = getenv(MPEGTABENV); /* check for environment */ +- if(envdir != NULL) +- strcpy(fulname, envdir); +- strcat(fulname, PATH_SEPARATOR); /* add a "/" on the end */ +- } +- #endif /* UNIX */ +- +- strcat(fulname, name); +- if( (f=fopen(fulname,"r"))==NULL ) { +- fprintf(stderr,"OpenTable: could not find %s\n", fulname); +- +- #ifdef UNIX +- if(envdir != NULL) +- fprintf(stderr,"Check %s directory '%s'\n",MPEGTABENV, envdir); +- else +- fprintf(stderr,"Check local directory './%s' or setenv %s\n", +- TABLES_PATH, MPEGTABENV); +- #else /* not unix : no environment variables */ +- +- #ifdef TABLES_PATH +- fprintf(stderr,"Check local directory './%s'\n",TABLES_PATH); +- #endif /* TABLES_PATH */ +- +- #endif /* UNIX */ +- +- } +- return f; +- } +- +- /*********************************************************************** +- * +- * Read one of the data files ("alloc_*") specifying the bit allocation/ +- * quatization parameters for each subband in layer II encoding +- * +- **********************************************************************/ +- +- int read_bit_alloc(table, alloc) /* read in table, return # subbands */ +- int table; +- al_table *alloc; +- { +- unsigned int a, b, c, d, i, j; +- FILE *fp; +- char name[16], t[80]; +- int sblim; +- +- strcpy(name, "alloc_0"); +- +- switch (table) { +- case 0 : name[6] = '0'; break; +- case 1 : name[6] = '1'; break; +- case 2 : name[6] = '2'; break; +- case 3 : name[6] = '3'; break; +- case 4 : name[6] = '4'; break; /* LSF, added 1995-07-11 shn */ +- default : name[6] = '0'; +- } +- +- if (!(fp = OpenTableFile(name))) { +- printf("Please check bit allocation table %s\n", name); +- exit(1); +- } +- +- printf("using bit allocation table %s\n", name); +- +- fgets(t, 80, fp); +- sscanf(t, "%d\n", &sblim); +- while (!feof(fp)) { +- fgets(t, 80, fp); +- sscanf(t, "%d %d %d %d %d %d\n", &i, &j, &a, &b, &c, &d); +- (*alloc)[i][j].steps = a; +- (*alloc)[i][j].bits = b; +- (*alloc)[i][j].group = c; +- (*alloc)[i][j].quant = d; +- } +- fclose(fp); +- return sblim; +- } +- +- /*********************************************************************** +- * +- * Using the decoded info the appropriate possible quantization per +- * subband table is loaded +- * +- **********************************************************************/ +- +- int pick_table(fr_ps) /* choose table, load if necess, return # sb's */ +- frame_params *fr_ps; +- { +- int table, lay, ws, bsp, br_per_ch, sfrq; +- int sblim = fr_ps->sblimit; /* return current value if no load */ +- +- lay = fr_ps->header->lay - 1; +- bsp = fr_ps->header->bitrate_index; +- br_per_ch = bitrate[fr_ps->header->version][lay][bsp] / fr_ps->stereo; +- ws = fr_ps->header->sampling_frequency; +- sfrq = s_freq[fr_ps->header->version][ws]; +- /* decision rules refer to per-channel bitrates (kbits/sec/chan) */ +- if (fr_ps->header->version == MPEG_AUDIO_ID) { /* MPEG-1 */ +- if ((sfrq == 48 && br_per_ch >= 56) || +- (br_per_ch >= 56 && br_per_ch <= 80)) table = 0; +- else if (sfrq != 48 && br_per_ch >= 96) table = 1; +- else if (sfrq != 32 && br_per_ch <= 48) table = 2; +- else table = 3; +- } +- else { /* MPEG-2 LSF */ +- table = 4; +- } +- if (fr_ps->tab_num != table) { +- if (fr_ps->tab_num >= 0) +- mem_free((void **)&(fr_ps->alloc)); +- fr_ps->alloc = (al_table FAR *) mem_alloc(sizeof(al_table), +- "alloc"); +- sblim = read_bit_alloc(fr_ps->tab_num = table, fr_ps->alloc); +- } +- return sblim; +- } +- +- int js_bound(lay, m_ext) +- int lay, m_ext; +- { +- static int jsb_table[3][4] = { { 4, 8, 12, 16 }, { 4, 8, 12, 16}, +- { 0, 4, 8, 16} }; /* lay+m_e -> jsbound */ +- +- if(lay<1 || lay >3 || m_ext<0 || m_ext>3) { +- fprintf(stderr, "js_bound bad layer/modext (%d/%d)\n", lay, m_ext); +- exit(1); +- } +- return(jsb_table[lay-1][m_ext]); +- } +- +- void hdr_to_frps(fr_ps) /* interpret data in hdr str to fields in fr_ps */ +- frame_params *fr_ps; +- { +- layer *hdr = fr_ps->header; /* (or pass in as arg?) */ +- +- fr_ps->actual_mode = hdr->mode; +- fr_ps->stereo = (hdr->mode == MPG_MD_MONO) ? 1 : 2; +- if (hdr->lay == 2) fr_ps->sblimit = pick_table(fr_ps); +- else fr_ps->sblimit = SBLIMIT; +- if(hdr->mode == MPG_MD_JOINT_STEREO) +- fr_ps->jsbound = js_bound(hdr->lay, hdr->mode_ext); +- else +- fr_ps->jsbound = fr_ps->sblimit; +- /* alloc, tab_num set in pick_table */ +- } +- +- void WriteHdr(fr_ps, s) +- frame_params *fr_ps; +- FILE *s; +- { +- layer *info = fr_ps->header; +- +- fprintf(s, "HDR: s=FFF, id=%X, l=%X, ep=%s, br=%X, sf=%X, pd=%X, ", +- info->version, info->lay, ((info->error_protection) ? "on" : "off"), +- info->bitrate_index, info->sampling_frequency, info->padding); +- fprintf(s, "pr=%X, m=%X, js=%X, c=%X, o=%X, e=%X\n", +- info->extension, info->mode, info->mode_ext, +- info->copyright, info->original, info->emphasis); +- fprintf(s, "alg.=%s, layer=%s, tot bitrate=%d, sfrq=%.1f\n", +- version_names[info->version], layer_names[info->lay-1], +- bitrate[info->version][info->lay-1][info->bitrate_index], +- s_freq[info->version][info->sampling_frequency]); +- fprintf(s, "mode=%s, sblim=%d, jsbd=%d, ch=%d\n", +- mode_names[info->mode], fr_ps->sblimit, fr_ps->jsbound, fr_ps->stereo); +- fflush(s); +- } +- +- void WriteBitAlloc(bit_alloc, f_p, s) +- unsigned int bit_alloc[2][SBLIMIT]; +- frame_params *f_p; +- FILE *s; +- { +- int i,j; +- int st = f_p->stereo; +- int sbl = f_p->sblimit; +- int jsb = f_p->jsbound; +- +- fprintf(s, "BITA "); +- for(i=0; i<sbl; ++i) { +- if(i == jsb) fprintf(s,"-"); +- for(j=0; j<st; ++j) +- fprintf(s, "%1x", bit_alloc[j][i]); +- } +- fprintf(s, "\n"); fflush(s); +- } +- +- void WriteScale(bit_alloc, scfsi, scalar, fr_ps, s) +- unsigned int bit_alloc[2][SBLIMIT], scfsi[2][SBLIMIT], scalar[2][3][SBLIMIT]; +- frame_params *fr_ps; +- FILE *s; +- { +- int stereo = fr_ps->stereo; +- int sblimit = fr_ps->sblimit; +- int lay = fr_ps->header->lay; +- int i,j,k; +- +- if(lay == 2) { +- fprintf(s, "SFSI "); +- for (i=0;i<sblimit;i++) for (k=0;k<stereo;k++) +- if (bit_alloc[k][i]) fprintf(s,"%d",scfsi[k][i]); +- fprintf(s, "\nSCFs "); +- for (k=0;k<stereo;k++) { +- for (i=0;i<sblimit;i++) +- if (bit_alloc[k][i]) +- switch (scfsi[k][i]) { +- case 0: for (j=0;j<3;j++) +- fprintf(s,"%2d%c",scalar[k][j][i], +- (j==2)?';':'-'); +- break; +- case 1: +- case 3: fprintf(s,"%2d-",scalar[k][0][i]); +- fprintf(s,"%2d;",scalar[k][2][i]); +- break; +- case 2: fprintf(s,"%2d;",scalar[k][0][i]); +- } +- fprintf(s, "\n"); +- } +- } +- else{ /* lay == 1 */ +- fprintf(s, "SCFs "); +- for (i=0;i<sblimit;i++) for (k=0;k<stereo;k++) +- if (bit_alloc[k][i]) fprintf(s,"%2d;",scalar[k][0][i]); +- fprintf(s, "\n"); +- } +- } +- +- void WriteSamples(ch, sample, bit_alloc, fr_ps, s) +- int ch; +- unsigned int FAR sample[SBLIMIT]; +- unsigned int bit_alloc[SBLIMIT]; +- frame_params *fr_ps; +- FILE *s; +- { +- int i; +- int stereo = fr_ps->stereo; +- int sblimit = fr_ps->sblimit; +- +- fprintf(s, "SMPL "); +- for (i=0;i<sblimit;i++) +- if ( bit_alloc[i] != 0) +- fprintf(s, "%d:", sample[i]); +- if(ch==(stereo-1) ) fprintf(s, "\n"); +- else fprintf(s, "\t"); +- } +- +- int NumericQ(s) /* see if a string lookd like a numeric argument */ +- char *s; +- { +- char c; +- +- while( (c = *s++)!='\0' && isspace((int)c)) /* strip leading ws */ +- ; +- if( c == '+' || c == '-' ) +- c = *s++; /* perhaps skip leading + or - */ +- return isdigit((int)c); +- } +- +- int BitrateIndex(layr, bRate, version) /* convert bitrate in kbps to index */ +- int layr; /* 1 or 2 */ +- int bRate; /* legal rates from 32 to 448 */ +- int version; /* MPEG-1 or MPEG-2 LSF */ +- { +- int index = 0; +- int found = 0; +- +- while(!found && index<15) { +- if(bitrate[version][layr-1][index] == bRate) +- found = 1; +- else +- ++index; +- } +- if(found) +- return(index); +- else { +- fprintf(stderr, "BitrateIndex: %d (layer %d) is not a legal bitrate\n", +- bRate, layr); +- return(-1); /* Error! */ +- } +- } +- +- int SmpFrqIndex(sRate, version) /* convert samp frq in Hz to index */ +- long sRate; /* legal rates 16000, 22050, 24000, 32000, 44100, 48000 */ +- int *version; +- { +- if (sRate == 44100L) { +- *version = MPEG_AUDIO_ID; return(0); +- } +- else if (sRate == 48000L) { +- *version = MPEG_AUDIO_ID; return(1); +- } +- else if (sRate == 32000L) { +- *version = MPEG_AUDIO_ID; return(2); +- } +- else if (sRate == 24000L) { +- *version = MPEG_PHASE2_LSF; return(1); +- } +- else if (sRate == 22050L) { +- *version = MPEG_PHASE2_LSF; return(0); +- } +- else if (sRate == 16000L) { +- *version = MPEG_PHASE2_LSF; return(2); +- } +- else { +- fprintf(stderr, "SmpFrqIndex: %ld is not a legal sample rate\n", sRate); +- return(-1); /* Error! */ +- } +- } +- +- /******************************************************************************* +- * +- * Allocate number of bytes of memory equal to "block". +- * +- *******************************************************************************/ +- +- void FAR *mem_alloc(block, item) +- unsigned long block; +- char *item; +- { +- +- void *ptr; +- +- #ifdef MACINTOSH +- ptr = NewPtr(block); +- #endif +- +- #ifdef MSC60 +- /*ptr = (void FAR *) _fmalloc((unsigned int)block);*/ /* far memory, 92-07-08 sr */ +- ptr = (void FAR *) malloc((unsigned int)block); /* far memory, 93-08-24 ss */ +- #endif +- +- #if ! defined (MACINTOSH) && ! defined (MSC60) +- ptr = (void FAR *) malloc(block); +- #endif +- +- if (ptr != NULL){ +- #ifdef MSC60 +- _fmemset(ptr, 0, (unsigned int)block); /* far memory, 92-07-08 sr */ +- #else +- memset(ptr, 0, block); +- #endif +- } +- else{ +- printf("Unable to allocate %s\n", item); +- exit(0); +- } +- return(ptr); +- } +- +- +- /**************************************************************************** +- * +- * Free memory pointed to by "*ptr_addr". +- * +- *****************************************************************************/ +- +- void mem_free(ptr_addr) +- void **ptr_addr; +- { +- +- if (*ptr_addr != NULL){ +- #ifdef MACINTOSH +- DisposPtr(*ptr_addr); +- #else +- free(*ptr_addr); +- #endif +- *ptr_addr = NULL; +- } +- +- } +- +- /******************************************************************************* +- * +- * Check block of memory all equal to a single byte, else return FALSE +- * +- *******************************************************************************/ +- +- int memcheck(array, test, num) +- char *array; +- int test; /* but only tested as a char (bottom 8 bits) */ +- int num; +- { +- int i=0; +- +- while (array[i] == test && i<num) i++; +- if (i==num) return TRUE; +- else return FALSE; +- } +- +- /***************************************************************************** +- * +- * Routines to determine byte order and swap bytes +- * +- *****************************************************************************/ +- +- enum byte_order DetermineByteOrder() +- { +- char s[ sizeof(long) + 1 ]; +- union +- { +- long longval; +- char charval[ sizeof(long) ]; +- } probe; +- probe.longval = 0x41424344L; /* ABCD in ASCII */ +- strncpy( s, probe.charval, sizeof(long) ); +- s[ sizeof(long) ] = '\0'; +- /* fprintf( stderr, "byte order is %s\n", s ); */ +- if ( strcmp(s, "ABCD") == 0 ) +- return order_bigEndian; +- else +- if ( strcmp(s, "DCBA") == 0 ) +- return order_littleEndian; +- else +- return order_unknown; +- } +- +- void SwapBytesInWords( short *loc, int words ) +- { +- int i; +- short thisval; +- char *dst, *src; +- src = (char *) &thisval; +- for ( i = 0; i < words; i++ ) +- { +- thisval = *loc; +- dst = (char *) loc++; +- dst[0] = src[1]; +- dst[1] = src[0]; +- } +- } +- +- /***************************************************************************** +- * +- * Read Audio Interchange File Format (AIFF) headers. +- * +- *****************************************************************************/ +- +- int aiff_read_headers( FILE *file_ptr, IFF_AIFF *aiff_ptr ) +- { +- int chunkSize, subSize, sound_position; +- +- if ( fseek(file_ptr, 0, SEEK_SET) != 0 ) +- return -1; +- +- if ( Read32BitsHighLow(file_ptr) != IFF_ID_FORM ) +- return -1; +- +- chunkSize = Read32BitsHighLow( file_ptr ); +- +- if ( Read32BitsHighLow(file_ptr) != IFF_ID_AIFF ) +- return -1; +- +- sound_position = 0; +- while ( chunkSize > 0 ) +- { +- chunkSize -= 4; +- switch ( Read32BitsHighLow(file_ptr) ) +- { +- +- case IFF_ID_COMM: +- chunkSize -= subSize = Read32BitsHighLow( file_ptr ); +- aiff_ptr->numChannels = Read16BitsHighLow( file_ptr ); +- subSize -= 2; +- aiff_ptr->numSampleFrames = Read32BitsHighLow( file_ptr ); +- subSize -= 4; +- aiff_ptr->sampleSize = Read16BitsHighLow( file_ptr ); +- subSize -= 2; +- aiff_ptr->sampleRate = ReadIeeeExtendedHighLow( file_ptr ); +- subSize -= 10; +- while ( subSize > 0 ) +- { +- getc( file_ptr ); +- subSize -= 1; +- } +- break; +- +- case IFF_ID_SSND: +- chunkSize -= subSize = Read32BitsHighLow( file_ptr ); +- aiff_ptr->blkAlgn.offset = Read32BitsHighLow( file_ptr ); +- subSize -= 4; +- aiff_ptr->blkAlgn.blockSize = Read32BitsHighLow( file_ptr ); +- subSize -= 4; +- sound_position = ftell( file_ptr ) + aiff_ptr->blkAlgn.offset; +- if ( fseek(file_ptr, (long) subSize, SEEK_CUR) != 0 ) +- return -1; +- aiff_ptr->sampleType = IFF_ID_SSND; +- break; +- +- default: +- chunkSize -= subSize = Read32BitsHighLow( file_ptr ); +- while ( subSize > 0 ) +- { +- getc( file_ptr ); +- subSize -= 1; +- } +- break; +- } +- } +- return sound_position; +- } +- +- /***************************************************************************** +- * +- * Seek past some Audio Interchange File Format (AIFF) headers to sound data. +- * +- *****************************************************************************/ +- +- int aiff_seek_to_sound_data( FILE *file_ptr ) +- { +- if ( fseek(file_ptr, AIFF_FORM_HEADER_SIZE + AIFF_SSND_HEADER_SIZE, SEEK_SET) != 0 ) +- return(-1); +- return(0); +- } +- +- /******************************************************************************* +- * +- * Write Audio Interchange File Format (AIFF) headers. +- * +- *******************************************************************************/ +- +- int aiff_write_headers( FILE *file_ptr, IFF_AIFF *aiff_ptr ) +- { +- int chunkSize; +- int sampleBytes = (aiff_ptr->sampleSize / 8) + (aiff_ptr->sampleSize % 8 ? 1 : 0); +- +- if ( fseek(file_ptr, 0L, SEEK_SET) != 0 ) +- return -1; +- +- /* write FORM chunk */ +- chunkSize = 8 + 18 + 8 + aiff_ptr->numChannels * aiff_ptr->numSampleFrames * sampleBytes; +- Write32BitsHighLow( file_ptr, IFF_ID_FORM ); +- Write32BitsHighLow( file_ptr, chunkSize ); +- Write32BitsHighLow( file_ptr, IFF_ID_AIFF ); +- /* write COMM chunk */ +- Write32BitsHighLow( file_ptr, IFF_ID_COMM ); +- Write32BitsHighLow( file_ptr, 18 ); /* chunk size */ +- Write16BitsHighLow( file_ptr, aiff_ptr->numChannels ); +- Write32BitsHighLow( file_ptr, aiff_ptr->numSampleFrames ); +- Write16BitsHighLow( file_ptr, aiff_ptr->sampleSize ); +- WriteIeeeExtendedHighLow( file_ptr, aiff_ptr->sampleRate ); +- /* write SSND chunk header */ +- chunkSize = 8 + aiff_ptr->numChannels * aiff_ptr->numSampleFrames * sampleBytes; +- Write32BitsHighLow( file_ptr, IFF_ID_SSND ); +- Write32BitsHighLow( file_ptr, chunkSize ); +- Write32BitsHighLow( file_ptr, 0 ); /* offset */ +- Write32BitsHighLow( file_ptr, 0 ); /* block size */ +- return 0; +- } +- +- /***************************************************************************** +- * +- * bit_stream.c package +- * Author: Jean-Georges Fritsch, C-Cube Microsystems +- * +- *****************************************************************************/ +- +- /******************************************************************** +- This package provides functions to write (exclusive or read) +- information from (exclusive or to) the bit stream. +- +- If the bit stream is opened in read mode only the get functions are +- available. If the bit stream is opened in write mode only the put +- functions are available. +- ********************************************************************/ +- +- /*open_bit_stream_w(); open the device to write the bit stream into it */ +- /*open_bit_stream_r(); open the device to read the bit stream from it */ +- /*close_bit_stream(); close the device containing the bit stream */ +- /*alloc_buffer(); open and initialize the buffer; */ +- /*desalloc_buffer(); empty and close the buffer */ +- /*back_track_buffer(); goes back N bits in the buffer */ +- /*unsigned int get1bit(); read 1 bit from the bit stream */ +- /*unsigned long getbits(); read N bits from the bit stream */ +- /*unsigned long byte_ali_getbits(); read the next byte aligned N bits from*/ +- /* the bit stream */ +- /*unsigned long look_ahead(); grep the next N bits in the bit stream without*/ +- /* changing the buffer pointer */ +- /*put1bit(); write 1 bit from the bit stream */ +- /*put1bit(); write 1 bit from the bit stream */ +- /*putbits(); write N bits from the bit stream */ +- /*byte_ali_putbits(); write byte aligned the next N bits into the bit stream*/ +- /*unsigned long sstell(); return the current bit stream length (in bits) */ +- /*int end_bs(); return 1 if the end of bit stream reached otherwise 0 */ +- /*int seek_sync(); return 1 if a sync word was found in the bit stream */ +- /* otherwise returns 0 */ +- +- /* refill the buffer from the input device when the buffer becomes empty */ +- int refill_buffer(bs) +- Bit_stream_struc *bs; /* bit stream structure */ +- { +- register int i=bs->buf_size-2-bs->buf_byte_idx; +- register unsigned long n=1; +- register int index=0; +- char val[2]; +- +- while ((i>=0) && (!bs->eob)) { +- +- if (bs->format == BINARY) +- n = fread(&bs->buf[i--], sizeof(unsigned char), 1, bs->pt); +- +- else { +- while((index < 2) && n) { +- n = fread(&val[index], sizeof(char), 1, bs->pt); +- switch (val[index]) { +- case 0x30: +- case 0x31: +- case 0x32: +- case 0x33: +- case 0x34: +- case 0x35: +- case 0x36: +- case 0x37: +- case 0x38: +- case 0x39: +- case 0x41: +- case 0x42: +- case 0x43: +- case 0x44: +- case 0x45: +- case 0x46: +- index++; +- break; +- default: break; +- } +- } +- +- if (val[0] <= 0x39) bs->buf[i] = (val[0] - 0x30) << 4; +- else bs->buf[i] = (val[0] - 0x37) << 4; +- if (val[1] <= 0x39) bs->buf[i--] |= (val[1] - 0x30); +- else bs->buf[i--] |= (val[1] - 0x37); +- index = 0; +- } +- +- if (!n) { +- bs->eob= i+1; +- } +- +- } +- return 0; +- } +- +- static char *he = "0123456789ABCDEF"; +- +- /* empty the buffer to the output device when the buffer becomes full */ +- void empty_buffer(bs, minimum) +- Bit_stream_struc *bs; /* bit stream structure */ +- int minimum; /* end of the buffer to empty */ +- { +- register int i; +- +- #if BS_FORMAT == BINARY +- for (i=bs->buf_size-1;i>=minimum;i--) +- fwrite(&bs->buf[i], sizeof(unsigned char), 1, bs->pt); +- #else +- for (i=bs->buf_size-1;i>=minimum;i--) { +- char val[2]; +- val[0] = he[((bs->buf[i] >> 4) & 0x0F)]; +- val[1] = he[(bs->buf[i] & 0x0F)]; +- fwrite(val, sizeof(char), 2, bs->pt); +- } +- #endif +- fflush(bs->pt); /* NEW SS to assist in debugging*/ +- +- for (i=minimum-1; i>=0; i--) +- bs->buf[bs->buf_size - minimum + i] = bs->buf[i]; +- +- bs->buf_byte_idx = bs->buf_size -1 - minimum; +- bs->buf_bit_idx = 8; +- } +- +- /* open the device to write the bit stream into it */ +- void open_bit_stream_w(bs, bs_filenam, size) +- Bit_stream_struc *bs; /* bit stream structure */ +- char *bs_filenam; /* name of the bit stream file */ +- int size; /* size of the buffer */ +- { +- if ((bs->pt = fopen(bs_filenam, "wb")) == NULL) { +- printf("Could not create \"%s\".\n", bs_filenam); +- exit(1); +- } +- alloc_buffer(bs, size); +- bs->buf_byte_idx = size-1; +- bs->buf_bit_idx=8; +- bs->totbit=0; +- bs->mode = WRITE_MODE; +- bs->eob = FALSE; +- bs->eobs = FALSE; +- } +- +- /* open the device to read the bit stream from it */ +- void open_bit_stream_r(bs, bs_filenam, size) +- Bit_stream_struc *bs; /* bit stream structure */ +- char *bs_filenam; /* name of the bit stream file */ +- int size; /* size of the buffer */ +- { +- register unsigned long n; +- register unsigned char flag = 1; +- unsigned char val; +- +- if ((bs->pt = fopen(bs_filenam, "rb")) == NULL) { +- printf("Could not find \"%s\".\n", bs_filenam); +- exit(1); +- } +- +- do { +- n = fread(&val, sizeof(unsigned char), 1, bs->pt); +- switch (val) { +- case 0x30: +- case 0x31: +- case 0x32: +- case 0x33: +- case 0x34: +- case 0x35: +- case 0x36: +- case 0x37: +- case 0x38: +- case 0x39: +- case 0x41: +- case 0x42: +- case 0x43: +- case 0x44: +- case 0x45: +- case 0x46: +- case 0xa: /* \n */ +- case 0xd: /* cr */ +- case 0x1a: /* sub */ +- break; +- +- default: /* detection of an binary character */ +- flag--; +- break; +- } +- +- } while (flag & n); +- +- if (flag) { +- printf ("the bit stream file %s is an ASCII file\n", bs_filenam); +- bs->format = ASCII; +- } +- else { +- bs->format = BINARY; +- printf ("the bit stream file %s is a BINARY file\n", bs_filenam); +- } +- +- fclose(bs->pt); +- +- if ((bs->pt = fopen(bs_filenam, "rb")) == NULL) { +- printf("Could not find \"%s\".\n", bs_filenam); +- exit(1); +- } +- +- alloc_buffer(bs, size); +- bs->buf_byte_idx=0; +- bs->buf_bit_idx=0; +- bs->totbit=0; +- bs->mode = READ_MODE; +- bs->eob = FALSE; +- bs->eobs = FALSE; +- } +- +- /*close the device containing the bit stream after a read process*/ +- void close_bit_stream_r(bs) +- Bit_stream_struc *bs; /* bit stream structure */ +- { +- fclose(bs->pt); +- desalloc_buffer(bs); +- } +- +- /*close the device containing the bit stream after a write process*/ +- void close_bit_stream_w(bs) +- Bit_stream_struc *bs; /* bit stream structure */ +- { +- empty_buffer(bs, bs->buf_byte_idx); +- fclose(bs->pt); +- desalloc_buffer(bs); +- } +- +- /*open and initialize the buffer; */ +- void alloc_buffer(bs, size) +- Bit_stream_struc *bs; /* bit stream structure */ +- int size; +- { +- bs->buf = (unsigned char FAR *) mem_alloc(size*sizeof(unsigned +- char), "buffer"); +- bs->buf_size = size; +- } +- +- /*empty and close the buffer */ +- void desalloc_buffer(bs) +- Bit_stream_struc *bs; /* bit stream structure */ +- { +- free(bs->buf); +- } +- +- int putmask[9]={0x0, 0x1, 0x3, 0x7, 0xf, 0x1f, 0x3f, 0x7f, 0xff}; +- int clearmask[9]={0xff, 0xfe, 0xfc, 0xf8, 0xf0, 0xe0, 0xc0, 0x80, 0x0}; +- +- void back_track_buffer(bs, N) /* goes back N bits in the buffer */ +- Bit_stream_struc *bs; /* bit stream structure */ +- int N; +- { +- int tmp = N - (N/8)*8; +- register int i; +- +- bs->totbit -= N; +- for (i=bs->buf_byte_idx;i< bs->buf_byte_idx+N/8-1;i++) bs->buf[i] = 0; +- bs->buf_byte_idx += N/8; +- if ( (tmp + bs->buf_bit_idx) <= 8) { +- bs->buf_bit_idx += tmp; +- } +- else { +- bs->buf_byte_idx ++; +- bs->buf_bit_idx += (tmp - 8); +- } +- bs->buf[bs->buf_byte_idx] &= clearmask[bs->buf_bit_idx]; +- } +- +- int mask[8]={0x1, 0x2, 0x4, 0x8, 0x10, 0x20, 0x40, 0x80}; +- +- /*read 1 bit from the bit stream */ +- unsigned int get1bit(bs) +- Bit_stream_struc *bs; /* bit stream structure */ +- { +- unsigned int bit; +- register int i; +- +- bs->totbit++; +- +- if (!bs->buf_bit_idx) { +- bs->buf_bit_idx = 8; +- bs->buf_byte_idx--; +- if ((bs->buf_byte_idx < MINIMUM) || (bs->buf_byte_idx < bs->eob)) { +- if (bs->eob) +- bs->eobs = TRUE; +- else { +- for (i=bs->buf_byte_idx; i>=0;i--) +- bs->buf[bs->buf_size-1-bs->buf_byte_idx+i] = bs->buf[i]; +- refill_buffer(bs); +- bs->buf_byte_idx = bs->buf_size-1; +- } +- } +- } +- bit = bs->buf[bs->buf_byte_idx]&mask[bs->buf_bit_idx-1]; +- bit = bit >> (bs->buf_bit_idx-1); +- bs->buf_bit_idx--; +- return(bit); +- } +- +- /*write 1 bit from the bit stream */ +- void put1bit(bs, bit) +- Bit_stream_struc *bs; /* bit stream structure */ +- int bit; /* bit to write into the buffer */ +- { +- bs->totbit++; +- +- bs->buf[bs->buf_byte_idx] |= (bit&0x1) << (bs->buf_bit_idx-1); +- bs->buf_bit_idx--; +- if (!bs->buf_bit_idx) { +- bs->buf_bit_idx = 8; +- bs->buf_byte_idx--; +- if (bs->buf_byte_idx < 0) +- empty_buffer(bs, MINIMUM); +- bs->buf[bs->buf_byte_idx] = 0; +- } +- } +- +- /*look ahead for the next N bits from the bit stream */ +- unsigned long look_ahead(bs, N) +- Bit_stream_struc *bs; /* bit stream structure */ +- int N; /* number of bits to read from the bit stream */ +- { +- unsigned long val=0; +- register int j = N; +- register int k, tmp; +- register int bit_idx = bs->buf_bit_idx; +- register int byte_idx = bs->buf_byte_idx; +- +- if (N > MAX_LENGTH) +- printf("Cannot read or write more than %d bits at a time.\n", MAX_LENGTH); +- +- while (j > 0) { +- if (!bit_idx) { +- bit_idx = 8; +- byte_idx--; +- } +- k = MIN (j, bit_idx); +- tmp = bs->buf[byte_idx]&putmask[bit_idx]; +- tmp = tmp >> (bit_idx-k); +- val |= tmp << (j-k); +- bit_idx -= k; +- j -= k; +- } +- return(val); +- } +- +- /*read N bit from the bit stream */ +- unsigned long getbits(bs, N) +- Bit_stream_struc *bs; /* bit stream structure */ +- int N; /* number of bits to read from the bit stream */ +- { +- unsigned long val=0; +- register int i; +- register int j = N; +- register int k, tmp; +- +- if (N > MAX_LENGTH) +- printf("Cannot read or write more than %d bits at a time.\n", MAX_LENGTH); +- +- bs->totbit += N; +- while (j > 0) { +- if (!bs->buf_bit_idx) { +- bs->buf_bit_idx = 8; +- bs->buf_byte_idx--; +- if ((bs->buf_byte_idx < MINIMUM) || (bs->buf_byte_idx < bs->eob)) { +- if (bs->eob) +- bs->eobs = TRUE; +- else { +- for (i=bs->buf_byte_idx; i>=0;i--) +- bs->buf[bs->buf_size-1-bs->buf_byte_idx+i] = bs->buf[i]; +- refill_buffer(bs); +- bs->buf_byte_idx = bs->buf_size-1; +- } +- } +- } +- k = MIN (j, bs->buf_bit_idx); +- tmp = bs->buf[bs->buf_byte_idx]&putmask[bs->buf_bit_idx]; +- tmp = tmp >> (bs->buf_bit_idx-k); +- val |= tmp << (j-k); +- bs->buf_bit_idx -= k; +- j -= k; +- } +- return(val); +- } +- +- /*write N bits into the bit stream */ +- void putbits(bs, val, N) +- Bit_stream_struc *bs; /* bit stream structure */ +- unsigned int val; /* val to write into the buffer */ +- int N; /* number of bits of val */ +- { +- register int j = N; +- register int k, tmp; +- +- if (N > MAX_LENGTH) +- printf("Cannot read or write more than %d bits at a time.\n", MAX_LENGTH); +- +- bs->totbit += N; +- while (j > 0) { +- k = MIN(j, bs->buf_bit_idx); +- tmp = val >> (j-k); +- bs->buf[bs->buf_byte_idx] |= (tmp&putmask[k]) << (bs->buf_bit_idx-k); +- bs->buf_bit_idx -= k; +- if (!bs->buf_bit_idx) { +- bs->buf_bit_idx = 8; +- bs->buf_byte_idx--; +- if (bs->buf_byte_idx < 0) +- empty_buffer(bs, MINIMUM); +- bs->buf[bs->buf_byte_idx] = 0; +- } +- j -= k; +- } +- } +- +- /*write N bits byte aligned into the bit stream */ +- void byte_ali_putbits(bs, val, N) +- Bit_stream_struc *bs; /* bit stream structure */ +- unsigned int val; /* val to write into the buffer */ +- int N; /* number of bits of val */ +- { +- unsigned long aligning, sstell(); +- +- if (N > MAX_LENGTH) +- printf("Cannot read or write more than %d bits at a time.\n", MAX_LENGTH); +- aligning = sstell(bs)%8; +- if (aligning) +- putbits(bs, (unsigned int)0, (int)(8-aligning)); +- +- putbits(bs, val, N); +- } +- +- /*read the next bute aligned N bits from the bit stream */ +- unsigned long byte_ali_getbits(bs, N) +- Bit_stream_struc *bs; /* bit stream structure */ +- int N; /* number of bits of val */ +- { +- unsigned long aligning, sstell(); +- +- if (N > MAX_LENGTH) +- printf("Cannot read or write more than %d bits at a time.\n", MAX_LENGTH); +- aligning = sstell(bs)%8; +- if (aligning) +- getbits(bs, (int)(8-aligning)); +- +- return(getbits(bs, N)); +- } +- +- /*return the current bit stream length (in bits)*/ +- unsigned long sstell(bs) +- Bit_stream_struc *bs; /* bit stream structure */ +- { +- return(bs->totbit); +- } +- +- /*return the status of the bit stream*/ +- /* returns 1 if end of bit stream was reached */ +- /* returns 0 if end of bit stream was not reached */ +- int end_bs(bs) +- Bit_stream_struc *bs; /* bit stream structure */ +- { +- return(bs->eobs); +- } +- +- /*this function seeks for a byte aligned sync word in the bit stream and +- places the bit stream pointer right after the sync. +- This function returns 1 if the sync was found otherwise it returns 0 */ +- int seek_sync(bs, sync, N) +- Bit_stream_struc *bs; /* bit stream structure */ +- long sync; /* sync word maximum 32 bits */ +- int N; /* sync word length */ +- { +- #if defined(MACINTOSH) && !defined(__powerc) +- double pow(); +- #endif +- unsigned long aligning, stell(); +- unsigned long val; +- long maxi = (int)pow(2.0, (FLOAT)N) - 1; +- +- aligning = sstell(bs)%ALIGNING; +- if (aligning) +- getbits(bs, (int)(ALIGNING-aligning)); +- +- val = getbits(bs, N); +- while (((val&maxi) != sync) && (!end_bs(bs))) { +- val <<= ALIGNING; +- val |= getbits(bs, ALIGNING); +- } +- +- if (end_bs(bs)) return(0); +- else return(1); +- } +- /***************************************************************************** +- * +- * End of bit_stream.c package +- * +- *****************************************************************************/ +- +- /***************************************************************************** +- * +- * CRC error protection package +- * +- *****************************************************************************/ +- +- void I_CRC_calc(fr_ps, bit_alloc, crc) +- frame_params *fr_ps; +- unsigned int bit_alloc[2][SBLIMIT]; +- unsigned int *crc; +- { +- int i, k; +- layer *info = fr_ps->header; +- int stereo = fr_ps->stereo; +- int jsbound = fr_ps->jsbound; +- +- *crc = 0xffff; /* changed from '0' 92-08-11 shn */ +- update_CRC(info->bitrate_index, 4, crc); +- update_CRC(info->sampling_frequency, 2, crc); +- update_CRC(info->padding, 1, crc); +- update_CRC(info->extension, 1, crc); +- update_CRC(info->mode, 2, crc); +- update_CRC(info->mode_ext, 2, crc); +- update_CRC(info->copyright, 1, crc); +- update_CRC(info->original, 1, crc); +- update_CRC(info->emphasis, 2, crc); +- +- for (i=0;i<SBLIMIT;i++) +- for (k=0;k<((i<jsbound)?stereo:1);k++) +- update_CRC(bit_alloc[k][i], 4, crc); +- } +- +- void II_CRC_calc(fr_ps, bit_alloc, scfsi, crc) +- frame_params *fr_ps; +- unsigned int bit_alloc[2][SBLIMIT], scfsi[2][SBLIMIT]; +- unsigned int *crc; +- { +- int i, k; +- layer *info = fr_ps->header; +- int stereo = fr_ps->stereo; +- int sblimit = fr_ps->sblimit; +- int jsbound = fr_ps->jsbound; +- al_table *alloc = fr_ps->alloc; +- +- *crc = 0xffff; /* changed from '0' 92-08-11 shn */ +- update_CRC(info->bitrate_index, 4, crc); +- update_CRC(info->sampling_frequency, 2, crc); +- update_CRC(info->padding, 1, crc); +- update_CRC(info->extension, 1, crc); +- update_CRC(info->mode, 2, crc); +- update_CRC(info->mode_ext, 2, crc); +- update_CRC(info->copyright, 1, crc); +- update_CRC(info->original, 1, crc); +- update_CRC(info->emphasis, 2, crc); +- +- for (i=0;i<sblimit;i++) +- for (k=0;k<((i<jsbound)?stereo:1);k++) +- update_CRC(bit_alloc[k][i], (*alloc)[i][0].bits, crc); +- +- for (i=0;i<sblimit;i++) +- for (k=0;k<stereo;k++) +- if (bit_alloc[k][i]) +- update_CRC(scfsi[k][i], 2, crc); +- } +- +- void update_CRC(data, length, crc) +- unsigned int data, length, *crc; +- { +- unsigned int masking, carry; +- +- masking = 1 << length; +- +- while((masking >>= 1)){ +- carry = *crc & 0x8000; +- *crc <<= 1; +- if (!carry ^ !(data & masking)) +- *crc ^= CRC16_POLYNOMIAL; +- } +- *crc &= 0xffff; +- } +- +- /***************************************************************************** +- * +- * End of CRC error protection package +- * +- *****************************************************************************/ +- +- #ifdef MACINTOSH +- /***************************************************************************** +- * +- * Set Macintosh file attributes. +- * +- *****************************************************************************/ +- void set_mac_file_attr( char fileName[MAX_NAME_SIZE], short vRefNum, OSType creator, OSType fileType) +- { +- char pascal_fileName[MAX_NAME_SIZE]; +- FInfo fndrInfo; +- OSErr anErr; +- long dirID = 0L; +- +- CtoPstr(strcpy(pascal_fileName, fileName)); +- anErr = HGetFInfo(vRefNum, dirID, (ConstStr255Param) pascal_fileName, &fndrInfo); +- if ( anErr != noErr ) +- return; +- fndrInfo.fdCreator = creator; +- fndrInfo.fdType = fileType; +- anErr = HSetFInfo(vRefNum, dirID, (ConstStr255Param) pascal_fileName, &fndrInfo); +- } +- +- #endif +- +- +- #ifdef MS_DOS +- /* ------------------------------------------------------------------------ +- new_ext() +- Puts a new extension name on a file name <filename>. +- Removes the last extension name, if any. +- 1992-08-19, 1995-06-12 shn +- ------------------------------------------------------------------------ */ +- void +- new_ext(char *filename, char *extname, char *newname) +- { +- int found, dotpos; +- +- /* First, strip the extension */ +- dotpos=strlen(filename); found=0; +- do +- { +- switch (filename[dotpos]) +- { +- case '.' : found=1; break; +- case '\\': +- case '/' : +- case ':' : found=-1; break; +- default : dotpos--; if (dotpos<0) found=-1; break; +- } +- } while (found==0); +- if (found==-1) strcpy(newname,filename); +- if (found== 1) { strncpy(newname,filename,dotpos); newname[dotpos]='\0'; } +- strcat(newname,extname); +- } +- #endif +- +- +- +- #define BUFSIZE 4096 +- static unsigned long offset,totbit=0, buf_byte_idx=0; +- static unsigned int buf[BUFSIZE]; +- static unsigned int buf_bit_idx=8; +- +- /*return the current bit stream length (in bits)*/ +- unsigned long hsstell() +- { +- return(totbit); +- } +- +- /* int putmask[9]={0x0, 0x1, 0x3, 0x7, 0xf, 0x1f, 0x3f, 0x7f, 0xff}; */ +- extern int putmask[9]; +- +- /*read N bit from the bit stream */ +- unsigned long hgetbits(N) +- int N; /* number of bits to read from the bit stream */ +- { +- unsigned long val=0; +- register int j = N; +- register int k, tmp; +- +- /* +- if (N > MAX_LENGTH) +- printf("Cannot read or write more than %d bits at a time.\n", MAX_LENGTH); +- */ +- totbit += N; +- while (j > 0) { +- if (!buf_bit_idx) { +- buf_bit_idx = 8; +- buf_byte_idx++; +- if (buf_byte_idx > offset) +- { printf("Buffer overflow !!\n");exit(3); } +- } +- k = MIN (j, buf_bit_idx); +- tmp = buf[buf_byte_idx%BUFSIZE]&putmask[buf_bit_idx]; +- tmp = tmp >> (buf_bit_idx-k); +- val |= tmp << (j-k); +- buf_bit_idx -= k; +- j -= k; +- } +- return(val); +- } +- +- unsigned int hget1bit() +- { +- return(hgetbits(1)); +- } +- +- /*write N bits into the bit stream */ +- void hputbuf(val, N) +- unsigned int val; /* val to write into the buffer */ +- int N; /* number of bits of val */ +- { +- if (N != 8) { printf("Not Supported yet!!\n"); exit(-3); } +- buf[offset % BUFSIZE] = val; +- offset++; +- } +- +- void rewindNbits( N ) +- int N; +- { +- totbit -= N; +- buf_bit_idx += N; +- while( buf_bit_idx >= 8 ) +- { buf_bit_idx -= 8; +- buf_byte_idx--; +- } +- } +- +- void rewindNbytes( N ) +- int N; +- { +- totbit -= N*8; +- buf_byte_idx -= N; +- } +--- 0 ---- +diff -r -c -N encoder/common.h lame3.70/common.h +*** encoder/common.h Wed Jan 22 02:43:13 1997 +--- lame3.70/common.h Wed Dec 31 17:00:00 1969 +*************** +*** 1,491 **** +- /********************************************************************** +- * ISO MPEG Audio Subgroup Software Simulation Group (1996) +- * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- * +- * $Id: common.h,v 1.1 1996/02/14 04:04:23 rowlands Exp $ +- * +- * $Log: common.h,v $ +- * Revision 1.1 1996/02/14 04:04:23 rowlands +- * Initial revision +- * +- * Received from Mike Coleman +- **********************************************************************/ +- /********************************************************************** +- * date programmers comment * +- * 2/25/91 Doulas Wong, start of version 1.0 records * +- * Davis Pan * +- * 5/10/91 W. Joseph Carter Reorganized & renamed all ".h" files * +- * into "common.h" and "encoder.h". * +- * Ported to Macintosh and Unix. * +- * Added additional type definitions for * +- * AIFF, double/SANE and "bitstream.c". * +- * Added function prototypes for more * +- * rigorous type checking. * +- * 27jun91 dpwe (Aware) Added "alloc_*" defs & prototypes * +- * Defined new struct 'frame_params'. * +- * Changed info.stereo to info.mode_ext * +- * #define constants for mode types * +- * Prototype arguments if PROTO_ARGS * +- * 5/28/91 Earle Jennings added MS_DOS definition * +- * MsDos function prototype declarations * +- * 7/10/91 Earle Jennings added FLOAT definition as double * +- *10/ 3/91 Don H. Lee implemented CRC-16 error protection * +- * 2/11/92 W. Joseph Carter Ported new code to Macintosh. Most * +- * important fixes involved changing * +- * 16-bit ints to long or unsigned in * +- * bit alloc routines for quant of 65535 * +- * and passing proper function args. * +- * Removed "Other Joint Stereo" option * +- * and made bitrate be total channel * +- * bitrate, irrespective of the mode. * +- * Fixed many small bugs & reorganized. * +- * Modified some function prototypes. * +- * Changed BUFFER_SIZE back to 4096. * +- * 7/27/92 Michael Li (re-)Ported to MS-DOS * +- * 7/27/92 Masahiro Iwadare Ported to Convex * +- * 8/07/92 mc@tv.tek.com * +- * 8/10/92 Amit Gulati Ported to the AIX Platform (RS6000) * +- * AIFF string constants redefined * +- * 8/27/93 Seymour Shlien, Fixes in Unix and MSDOS ports, * +- * Daniel Lauzon, and * +- * Bill Truerniet * +- *--------------------------------------------------------------------* +- * 4/23/92 J. Pineda Added code for Layer III. * +- * 11/9/92 Amit Gulati Added defines for layerIII stereo * +- * modes. * +- * 8/24/93 Masahiro Iwadare Included IS modification in Layer III.* +- * Changed for 1 pass decoding. * +- * 9/07/93 Toshiyuki Ishino Integrated Layer III with Ver 3.9. * +- *--------------------------------------------------------------------* +- * 11/20/93 Masahiro Iwadare Integrated Layer III with Ver 4.0. * +- *--------------------------------------------------------------------* +- * 7/14/94 Juergen Koller Fix for HPUX an IRIX in AIFF-Strings * +- * 6/12/95 Soeren H. Nielsen Bug fix in new_ext(). * +- * 7/11/95 Soeren H. Nielsen Added defs. for MPEG-2 LSF * +- *--------------------------------------------------------------------* +- * 8/02/95 mc@fivebats.com Changes to AIFF stuff for portability * +- * to little-endian machines * +- **********************************************************************/ +- #ifndef COMMON_DOT_H +- #define COMMON_DOT_H +- /*********************************************************************** +- * +- * Global Conditional Compile Switches +- * +- ***********************************************************************/ +- +- #if 0 +- #define UNIX /* Unix conditional compile switch */ +- #define MACINTOSH /* Macintosh conditional compile switch */ +- #define MS_DOS /* IBM PC conditional compile switch */ +- #define MSC60 /* Compiled for MS_DOS with MSC v6.0 */ +- #define AIX /* AIX conditional compile switch */ +- #define CONVEX /* CONVEX conditional compile switch */ +- #endif +- +- #if defined(MSC60) +- #ifndef MS_DOS +- #define MS_DOS +- #endif +- #ifndef PROTO_ARGS +- #define PROTO_ARGS +- #endif +- #endif +- +- #ifdef UNIX +- #define TABLES_PATH "tables" /* to find data files */ +- /* name of environment variable holding path of table files */ +- #define MPEGTABENV "MPEGTABLES" +- #define PATH_SEPARATOR "/" /* how to build paths */ +- #endif /* UNIX */ +- +- #ifdef MACINTOSH +- #define TABLES_PATH ":tables:" /* where to find data files */ +- #endif /* MACINTOSH */ +- +- /* +- * Don't define FAR to far unless you're willing to clean up the +- * prototypes +- */ +- #define FAR /*far*/ +- +- #ifdef __STDC__ +- #ifndef PROTO_ARGS +- #define PROTO_ARGS +- #endif +- #endif +- +- #ifdef CONVEX +- #define SEEK_SET 0 +- #define SEEK_CUR 1 +- #define SEEK_END 2 +- #endif +- +- /* MS_DOS and VMS do not define TABLES_PATH, so OpenTableFile will default +- to finding the data files in the default directory */ +- +- /*********************************************************************** +- * +- * Global Include Files +- * +- ***********************************************************************/ +- +- #include <stdio.h> +- #include <string.h> +- #include <math.h> +- #include "portableio.h" +- #include "ieeefloat.h" +- +- #ifdef UNIX +- #include <unistd.h> +- #endif /* UNIX */ +- +- #ifdef __sgi +- #include <stdlib.h> +- #endif +- +- #ifdef MACINTOSH +- #include <stdlib.h> +- #include <console.h> +- #endif /* MACINTOSH */ +- +- #ifdef MS_DOS +- #include <stdlib.h> +- #ifdef MSC60 +- #include <memory.h> +- #else +- #include <alloc.h> +- #include <mem.h> +- #endif /* MSC60 */ +- #endif /* MS_DOS */ +- +- #ifndef EXIT_SUCCESS +- #define EXIT_SUCCESS 0 +- #endif +- +- #ifndef EXIT_FAILURE +- #define EXIT_FAILURE 1 +- #endif +- +- /*********************************************************************** +- * +- * Global Definitions +- * +- ***********************************************************************/ +- +- /* General Definitions */ +- +- #ifdef MS_DOS +- #define FLOAT double +- #else +- #define FLOAT float +- #endif +- #ifndef FALSE +- #define FALSE 0 +- #endif +- +- #ifndef TRUE +- #define TRUE 1 +- #endif +- +- #define NULL_CHAR '\0' +- +- #define MAX_U_32_NUM 0xFFFFFFFF +- #ifndef PI +- #define PI 3.14159265358979 +- #endif +- #define PI4 PI/4 +- #define PI64 PI/64 +- #define LN_TO_LOG10 0.2302585093 +- +- #define VOL_REF_NUM 0 +- #define MPEG_AUDIO_ID 1 +- #define MPEG_PHASE2_LSF 0 /* 1995-07-11 SHN */ +- #define MAC_WINDOW_SIZE 24 +- +- #define MONO 1 +- #define STEREO 2 +- #define BITS_IN_A_BYTE 8 +- #define WORD 16 +- #define MAX_NAME_SIZE 81 +- #define SBLIMIT 32 +- #define SSLIMIT 18 +- #define FFT_SIZE 1024 +- #define HAN_SIZE 512 +- #define SCALE_BLOCK 12 +- #define SCALE_RANGE 64 +- #define SCALE 32768 +- #define CRC16_POLYNOMIAL 0x8005 +- +- /* MPEG Header Definitions - Mode Values */ +- +- #define MPG_MD_STEREO 0 +- #define MPG_MD_JOINT_STEREO 1 +- #define MPG_MD_DUAL_CHANNEL 2 +- #define MPG_MD_MONO 3 +- +- /* Mode Extention */ +- +- #define MPG_MD_LR_LR 0 +- #define MPG_MD_LR_I 1 +- #define MPG_MD_MS_LR 2 +- #define MPG_MD_MS_I 3 +- +- +- /* "bit_stream.h" Definitions */ +- +- #define MINIMUM 4 /* Minimum size of the buffer in bytes */ +- #define MAX_LENGTH 32 /* Maximum length of word written or +- read from bit stream */ +- #define READ_MODE 0 +- #define WRITE_MODE 1 +- #define ALIGNING 8 +- #define BINARY 0 +- #define ASCII 1 +- +- #ifndef BS_FORMAT +- #define BS_FORMAT ASCII /* BINARY or ASCII = 2x bytes */ +- #endif +- +- #define BUFFER_SIZE 4096 +- +- #define MIN(A, B) ((A) < (B) ? (A) : (B)) +- #define MAX(A, B) ((A) > (B) ? (A) : (B)) +- +- /*********************************************************************** +- * +- * Global Type Definitions +- * +- ***********************************************************************/ +- +- /* Structure for Reading Layer II Allocation Tables from File */ +- +- typedef struct { +- unsigned int steps; +- unsigned int bits; +- unsigned int group; +- unsigned int quant; +- } sb_alloc, *alloc_ptr; +- +- typedef sb_alloc al_table[SBLIMIT][16]; +- +- /* Header Information Structure */ +- +- typedef struct { +- int version; +- int lay; +- int error_protection; +- int bitrate_index; +- int sampling_frequency; +- int padding; +- int extension; +- int mode; +- int mode_ext; +- int copyright; +- int original; +- int emphasis; +- } layer, *the_layer; +- +- /* Parent Structure Interpreting some Frame Parameters in Header */ +- +- typedef struct { +- layer *header; /* raw header information */ +- int actual_mode; /* when writing IS, may forget if 0 chs */ +- al_table *alloc; /* bit allocation table read in */ +- int tab_num; /* number of table as loaded */ +- int stereo; /* 1 for mono, 2 for stereo */ +- int jsbound; /* first band of joint stereo coding */ +- int sblimit; /* total number of sub bands */ +- } frame_params; +- +- +- /* AIFF Definitions */ +- +- #define IFF_ID_FORM 0x464f524d /* "FORM" */ +- #define IFF_ID_AIFF 0x41494646 /* "AIFF" */ +- #define IFF_ID_COMM 0x434f4d4d /* "COMM" */ +- #define IFF_ID_SSND 0x53534e44 /* "SSND" */ +- #define IFF_ID_MPEG 0x4d504547 /* "MPEG" */ +- +- #define AIFF_FORM_HEADER_SIZE 12 +- #define AIFF_SSND_HEADER_SIZE 16 +- +- +- typedef struct blockAlign_struct { +- unsigned long offset; +- unsigned long blockSize; +- } blockAlign; +- +- typedef struct IFF_AIFF_struct { +- short numChannels; +- unsigned long numSampleFrames; +- short sampleSize; +- double sampleRate; +- unsigned long sampleType; +- blockAlign blkAlgn; +- } IFF_AIFF; +- +- enum byte_order { order_unknown, order_bigEndian, order_littleEndian }; +- extern enum byte_order NativeByteOrder; +- +- /* "bit_stream.h" Type Definitions */ +- +- typedef struct bit_stream_struc { +- FILE *pt; /* pointer to bit stream device */ +- unsigned char *buf; /* bit stream buffer */ +- int buf_size; /* size of buffer (in number of bytes) */ +- long totbit; /* bit counter of bit stream */ +- int buf_byte_idx; /* pointer to top byte in buffer */ +- int buf_bit_idx; /* pointer to top bit of top byte in buffer */ +- int mode; /* bit stream open in read or write mode */ +- int eob; /* end of buffer index */ +- int eobs; /* end of bit stream flag */ +- char format; +- +- /* format of file in rd mode (BINARY/ASCII) */ +- } Bit_stream_struc; +- +- #include "l3side.h" +- +- /*********************************************************************** +- * +- * Global Variable External Declarations +- * +- ***********************************************************************/ +- +- extern char *mode_names[4]; +- extern char *layer_names[3]; +- extern char *version_names[2]; +- extern double s_freq[2][4]; +- extern int bitrate[2][3][15]; +- extern double FAR multiple[64]; +- +- /*********************************************************************** +- * +- * Global Function Prototype Declarations +- * +- ***********************************************************************/ +- +- /* The following functions are in the file "common.c" */ +- +- #ifdef PROTO_ARGS +- extern FILE *OpenTableFile(char*); +- extern int read_bit_alloc(int, al_table*); +- extern int pick_table(frame_params*); +- extern int js_bound(int, int); +- extern void hdr_to_frps(frame_params*); +- extern void WriteHdr(frame_params*, FILE*); +- extern void WriteBitAlloc(unsigned int[2][SBLIMIT], frame_params*, +- FILE*); +- extern void WriteScale(unsigned int[2][SBLIMIT], +- unsigned int[2][SBLIMIT], unsigned int[2][3][SBLIMIT], +- frame_params*, FILE*); +- extern void WriteSamples(int, unsigned int FAR [SBLIMIT], +- unsigned int[SBLIMIT], frame_params*, FILE*); +- extern int NumericQ(char*); +- extern int BitrateIndex(int, int, int); +- extern int SmpFrqIndex(long, int*); +- extern int memcheck(char*, int, int); +- extern void FAR *mem_alloc(unsigned long, char*); +- extern void mem_free(void**); +- extern void double_to_extended(double*, char[10]); +- extern void extended_to_double(char[10], double*); +- extern int aiff_read_headers(FILE*, IFF_AIFF*); +- extern int aiff_seek_to_sound_data(FILE*); +- extern int aiff_write_headers(FILE*, IFF_AIFF*); +- extern int refill_buffer(Bit_stream_struc*); +- extern void empty_buffer(Bit_stream_struc*, int); +- extern void open_bit_stream_w(Bit_stream_struc*, char*, int); +- extern void open_bit_stream_r(Bit_stream_struc*, char*, int); +- extern void close_bit_stream_r(Bit_stream_struc*); +- extern void close_bit_stream_w(Bit_stream_struc*); +- extern void alloc_buffer(Bit_stream_struc*, int); +- extern void desalloc_buffer(Bit_stream_struc*); +- extern void back_track_buffer(Bit_stream_struc*, int); +- extern unsigned int get1bit(Bit_stream_struc*); +- extern void put1bit(Bit_stream_struc*, int); +- extern unsigned long look_ahead(Bit_stream_struc*, int); +- extern unsigned long getbits(Bit_stream_struc*, int); +- extern void putbits(Bit_stream_struc*, unsigned int, int); +- extern void byte_ali_putbits(Bit_stream_struc*, unsigned int, int); +- extern unsigned long byte_ali_getbits(Bit_stream_struc*, int); +- extern unsigned long sstell(Bit_stream_struc*); +- extern int end_bs(Bit_stream_struc*); +- extern int seek_sync(Bit_stream_struc*, long, int); +- extern void I_CRC_calc(frame_params*, unsigned int[2][SBLIMIT], +- unsigned int*); +- extern void II_CRC_calc(frame_params*, unsigned int[2][SBLIMIT], +- unsigned int[2][SBLIMIT], unsigned int*); +- extern void update_CRC(unsigned int, unsigned int, unsigned int*); +- extern void read_absthr(FLOAT*, int); +- extern unsigned int hget1bit(); /* MI */ +- extern unsigned long hgetbits(int); +- extern unsigned long hsstell(); +- extern void hputbuf(unsigned int,int); +- extern enum byte_order DetermineByteOrder(); +- extern void SwapBytesInWords( short *loc, int words ); +- +- +- #ifdef MACINTOSH +- extern void set_mac_file_attr(char[MAX_NAME_SIZE], short, OSType, +- OSType); +- #endif +- #ifdef MS_DOS +- extern void new_ext(char *filename, char *extname, char *newname); +- #endif +- +- #else +- extern FILE *OpenTableFile(); +- extern int read_bit_alloc(); +- extern int pick_table(); +- extern int js_bound(); +- extern void hdr_to_frps(); +- extern void WriteHdr(); +- extern void WriteBitAlloc(); +- extern void WriteScale(); +- extern void WriteSamples(); +- extern int NumericQ(); +- extern int BitrateIndex(); +- extern int SmpFrqIndex(); +- extern int memcheck(); +- extern void FAR *mem_alloc(); +- extern void mem_free(); +- extern void double_to_extended(); +- extern void extended_to_double(); +- extern int aiff_read_headers(); +- extern int aiff_seek_to_sound_data(); +- extern int aiff_write_headers(); +- extern int refill_buffer(); +- extern void empty_buffer(); +- extern void open_bit_stream_w(); +- extern void open_bit_stream_r(); +- extern void close_bit_stream_r(); +- extern void close_bit_stream_w(); +- extern void alloc_buffer(); +- extern void desalloc_buffer(); +- extern void back_track_buffer(); +- extern unsigned int get1bit(); +- extern void put1bit(); +- extern unsigned long look_ahead(); +- extern unsigned long getbits(); +- extern void putbits(); +- extern void byte_ali_putbits(); +- extern unsigned long byte_ali_getbits(); +- extern unsigned long sstell(); +- extern int end_bs(); +- extern int seek_sync(); +- extern void I_CRC_calc(); +- extern void II_CRC_calc(); +- extern void update_CRC(); +- extern void read_absthr(); +- +- extern unsigned int hget1bit(); +- extern unsigned long hgetbits(); +- extern unsigned long hsstell(); +- extern void hputbuf(); +- +- #ifdef MS_DOS +- extern void new_ext(); +- #endif +- #endif +- #endif +--- 0 ---- +diff -r -c -N encoder/configure lame3.70/configure +*** encoder/configure Wed Jan 22 02:43:13 1997 +--- lame3.70/configure Wed Dec 31 17:00:00 1969 +*************** +*** 1,742 **** +- #!/bin/sh +- ############################################################################ +- ## ISO MPEG Audio Subgroup Software Simulation Group (1996) +- ## ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- ## +- ## $Id: configure,v 1.1 1996/02/14 04:06:24 rowlands Exp $ +- ## +- ## $Log: configure,v $ +- ## Revision 1.1 1996/02/14 04:06:24 rowlands +- ## Initial revision +- ## +- ## Received from Mike Coleman +- ## +- ############################################################################ +- +- # Guess values for system-dependent variables and create Makefiles. +- # Generated automatically using autoconf version 2.3 +- # Copyright (C) 1992, 1993, 1994 Free Software Foundation, Inc. +- # +- # This configure script is free software; the Free Software Foundation +- # gives unlimited permission to copy, distribute and modify it. +- +- # Defaults: +- ac_help= +- ac_default_prefix=/usr/local +- # Any additions from configure.in: +- +- # Initialize some variables set by options. +- # The variables have the same names as the options, with +- # dashes changed to underlines. +- build=NONE +- cache_file=./config.cache +- exec_prefix=NONE +- host=NONE +- no_create= +- nonopt=NONE +- no_recursion= +- prefix=NONE +- program_prefix=NONE +- program_suffix=NONE +- program_transform_name=s,x,x, +- silent= +- site= +- srcdir= +- target=NONE +- verbose= +- x_includes=NONE +- x_libraries=NONE +- +- # Initialize some other variables. +- subdirs= +- +- ac_prev= +- for ac_option +- do +- +- # If the previous option needs an argument, assign it. +- if test -n "$ac_prev"; then +- eval "$ac_prev=\$ac_option" +- ac_prev= +- continue +- fi +- +- case "$ac_option" in +- -*=*) ac_optarg=`echo "$ac_option" | sed 's/[-_a-zA-Z0-9]*=//'` ;; +- *) ac_optarg= ;; 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It is not useful on other systems. +- # If it contains results you don't want to keep, you may remove or edit it. +- # +- # By default, configure uses ./config.cache as the cache file, +- # creating it if it does not exist already. You can give configure +- # the --cache-file=FILE option to use a different cache file; that is +- # what configure does when it calls configure scripts in +- # subdirectories, so they share the cache. +- # Giving --cache-file=/dev/null disables caching, for debugging configure. +- # config.status only pays attention to the cache file if you give it the +- # --recheck option to rerun configure. +- # +- EOF +- # Ultrix sh set writes to stderr and can't be redirected directly, +- # and sets the high bit in the cache file unless we assign to the vars. +- (set) 2>&1 | +- sed -n "s/^\([a-zA-Z0-9_]*_cv_[a-zA-Z0-9_]*\)=\(.*\)/\1=\${\1='\2'}/p" \ +- >> confcache +- if cmp -s $cache_file confcache; then +- : +- else +- if test -w $cache_file; then +- echo "updating cache $cache_file" +- cat confcache > $cache_file +- else +- echo "not updating unwritable cache $cache_file" +- fi +- fi +- rm -f confcache +- +- trap 'rm -fr conftest* confdefs* core core.* *.core $ac_clean_files; exit 1' 1 2 15 +- +- test "x$prefix" = xNONE && prefix=$ac_default_prefix +- # Let make expand exec_prefix. +- test "x$exec_prefix" = xNONE && exec_prefix='${prefix}' +- +- # Any assignment to VPATH causes Sun make to only execute +- # the first set of double-colon rules, so remove it if not needed. +- # If there is a colon in the path, we need to keep it. +- if test "x$srcdir" = x.; then +- ac_vpsub='/^[ ]*VPATH[ ]*=[^:]*$/d' +- fi +- +- trap 'rm -f $CONFIG_STATUS conftest*; exit 1' 1 2 15 +- +- # Transform confdefs.h into DEFS. +- # Protect against shell expansion while executing Makefile rules. +- # Protect against Makefile macro expansion. +- cat > conftest.defs <<\EOF +- s%#define \([A-Za-z_][A-Za-z0-9_]*\) \(.*\)%-D\1=\2%g +- s%[ `~#$^&*(){}\\|;'"<>?]%\\&%g +- s%\[%\\&%g +- s%\]%\\&%g +- s%\$%$$%g +- EOF +- DEFS=`sed -f conftest.defs confdefs.h | tr '\012' ' '` +- rm -f conftest.defs +- +- +- # Without the "./", some shells look in PATH for config.status. +- : ${CONFIG_STATUS=./config.status} +- +- echo creating $CONFIG_STATUS +- rm -f $CONFIG_STATUS +- cat > $CONFIG_STATUS <<EOF +- #! /bin/sh +- # Generated automatically by configure. +- # Run this file to recreate the current configuration. +- # This directory was configured as follows, +- # on host `(hostname || uname -n) 2>/dev/null | sed 1q`: +- # +- # $0 $ac_configure_args +- # +- # Compiler output produced by configure, useful for debugging +- # configure, is in ./config.log if it exists. +- +- ac_cs_usage="Usage: $CONFIG_STATUS [--recheck] [--version] [--help]" +- for ac_option +- do +- case "\$ac_option" in +- -recheck | --recheck | --rechec | --reche | --rech | --rec | --re | --r) +- echo "running \${CONFIG_SHELL-/bin/sh} $0 $ac_configure_args --no-create --no-recursion" +- exec \${CONFIG_SHELL-/bin/sh} $0 $ac_configure_args --no-create --no-recursion ;; +- -version | --version | --versio | --versi | --vers | --ver | --ve | --v) +- echo "$CONFIG_STATUS generated by autoconf version 2.3" +- exit 0 ;; +- -help | --help | --hel | --he | --h) +- echo "\$ac_cs_usage"; exit 0 ;; +- *) echo "\$ac_cs_usage"; exit 1 ;; +- esac +- done +- +- ac_given_srcdir=$srcdir +- +- trap 'rm -fr `echo "Makefile" | sed "s/:[^ ]*//g"` conftest*; exit 1' 1 2 15 +- +- # Protect against being on the right side of a sed subst in config.status. +- sed 's/%@/@@/; s/@%/@@/; s/%g$/@g/; /@g$/s/[\\\\&%]/\\\\&/g; +- s/@@/%@/; s/@@/@%/; s/@g$/%g/' > conftest.subs <<\CEOF +- $ac_vpsub +- $extrasub +- s%@CFLAGS@%$CFLAGS%g +- s%@CPPFLAGS@%$CPPFLAGS%g +- s%@CXXFLAGS@%$CXXFLAGS%g +- s%@DEFS@%$DEFS%g +- s%@LDFLAGS@%$LDFLAGS%g +- s%@LIBS@%$LIBS%g +- s%@exec_prefix@%$exec_prefix%g +- s%@prefix@%$prefix%g +- s%@program_transform_name@%$program_transform_name%g +- s%@CC@%$CC%g +- s%@MATHLIBS@%$MATHLIBS%g +- s%@NINTSW@%$NINTSW%g +- +- CEOF +- EOF +- cat >> $CONFIG_STATUS <<EOF +- +- CONFIG_FILES=\${CONFIG_FILES-"Makefile"} +- EOF +- cat >> $CONFIG_STATUS <<\EOF +- for ac_file in .. $CONFIG_FILES; do if test "x$ac_file" != x..; then +- # Support "outfile[:infile]", defaulting infile="outfile.in". +- case "$ac_file" in +- *:*) ac_file_in=`echo "$ac_file"|sed 's%.*:%%'` +- ac_file=`echo "$ac_file"|sed 's%:.*%%'` ;; +- *) ac_file_in="${ac_file}.in" ;; +- esac +- +- # Adjust relative srcdir, etc. for subdirectories. +- +- # Remove last slash and all that follows it. 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+- esac +- sed -e "$ac_comsub +- s%@configure_input@%$configure_input%g +- s%@srcdir@%$srcdir%g +- s%@top_srcdir@%$top_srcdir%g +- " -f conftest.subs $ac_given_srcdir/$ac_file_in > $ac_file +- fi; done +- rm -f conftest.subs +- +- +- +- exit 0 +- EOF +- chmod +x $CONFIG_STATUS +- rm -fr confdefs* $ac_clean_files +- test "$no_create" = yes || ${CONFIG_SHELL-/bin/sh} $CONFIG_STATUS || exit 1 +- +--- 0 ---- +diff -r -c -N encoder/debugscalefac.c lame3.70/debugscalefac.c +*** encoder/debugscalefac.c Wed Dec 31 17:00:00 1969 +--- lame3.70/debugscalefac.c Wed Nov 24 01:43:46 1999 +*************** +*** 0 **** +--- 1,139 ---- ++ { ++ /******************************************************************* ++ ++ DEBUG INFORMATION ++ criticalbands (scalefactor bands) ++ partition bands ++ FFT bands ++ MDCT bands ++ ++ *******************************************************************/ ++ ++ #include "tables.h" ++ int *scalefac_band_long; ++ int *scalefac_band_short; ++ FLOAT fstart,fstop; ++ FLOAT mstart,mstop; ++ ++ ++ scalefac_band_long = &sfBandIndex[info->sampling_frequency + (info->version * 3)].l[0]; ++ scalefac_band_short = &sfBandIndex[info->sampling_frequency + (info->version * 3)].s[0]; ++ ++ printf("\n\n long block partiton bands \n"); ++ for (b = 0; b<CBANDS; b++) { ++ /* look for FFT partition band bu_l[b]*/ ++ for (i=0;i<HBLKSIZE;i++, fstart=i) { ++ if (partition_l[i]==b) break; ++ } ++ fstop=0; ++ for (i=HBLKSIZE-1;i>=0;i--, fstart=i) { ++ if (partition_l[i]==b) break; ++ } ++ ++ if(fstart>0) fstart -= .5; ++ fstop += .5; ++ printf("pb=%2i FFT=(%4.1f,%4.1f) numlines=%i norm=%f \n", ++ b,fstart,fstop,numlines_l[b],norm_l[b]); ++ ++ } ++ ++ ++ ++ printf("\n\n long block critial bands \n"); ++ printf("crit band,part_band, part_band range FFT range MDCT range \n"); ++ for ( b = 0;b < SBPSY_l; b++ ) { ++ ++ /* look for FFT partition band bu_l[b]*/ ++ for (i=0;i<HBLKSIZE;i++, fstart=i) { ++ if (partition_l[i]==bu_l[b]) break; ++ } ++ for (i=HBLKSIZE-1;i>=0;i--, fstop=i) { ++ if (partition_l[i]==bo_l[b]) break; ++ } ++ ++ ++ /* numlines_l[pb] = number of FFT lines in partition band pb */ ++ /* w1 = 1, then all FFT lines belong to this partition band */ ++ fstart += ( (1-w1_l[b])*numlines_l[ bu_l[b] ] ) ; ++ ++ /* w2 = 1, then all FFT lines belong to this partition band */ ++ fstop -= ( (1-w2_l[b])*numlines_l[ bo_l[b] ] ) ; ++ ++ ++ if (fstart>0) fstart -= .5; /* contribution extends to center of freq band */ ++ fstop += .5; ++ ++ mstart = scalefac_band_long[b]; ++ mstop = scalefac_band_long[b+1]-1; ++ if (mstart>0) mstart -= .5; ++ mstop += .5; ++ ++ printf("cb=%2i(%2i)(%2i,%2i)(%3.2f,%3.2f) FFT:(%4.1f,%4.1f)(%5.0f,%5.0f)Hz MDCT:(%5.1f,%5.1f)(%5.0f,%5.0f)Hz \n", ++ b,npart_l_orig,bu_l[b],bo_l[b],w1_l[b],w2_l[b], ++ fstart,fstop,sfreq*fstart/BLKSIZE,sfreq*fstop/BLKSIZE, ++ mstart,mstop,mstart*.5*sfreq/576,mstop*.5*sfreq/576 ++ ); ++ } ++ ++ ++ ++ ++ ++ printf("\n\n short block partition bands \n"); ++ for (b = 0; b<CBANDS; b++) { ++ /* look for FFT partition band bu_s[b]*/ ++ for (i=0;i<HBLKSIZE_s;i++,fstart=i) { ++ if (partition_s[i]==b) break; ++ } ++ for (i=HBLKSIZE_s-1; i>=0; i--,fstop=i) { ++ if (partition_s[i]==b) break; ++ } ++ ++ if(fstart>0) fstart -= .5; ++ fstop += .5; ++ printf("pb=%2i FFT=(%4.1f,%4.1f) numlines=%i norm=%f \n", ++ b,fstart,fstop,numlines_s[b],norm_s[b]); ++ ++ } ++ ++ ++ ++ printf("\n\n short block critial bands \n"); ++ printf("crit band,part_band, part_band range FFT range MDCT range \n"); ++ for ( b = 0;b < SBPSY_s; b++ ) { ++ ++ /* look for FFT partition band bu_s[b]*/ ++ for (i=0;i<HBLKSIZE_s;i++) ++ if (partition_s[i]==bu_s[b]) {fstart=i;break;} ++ for (i=HBLKSIZE_s-1;i>=0;i--) ++ if (partition_s[i]==bo_s[b]) {fstop=i; break;} ++ ++ ++ /* numlines_s[pb] = number of FFT lines in partition band pb */ ++ /* w1 = 1, then all FFT lines belong to this partition band */ ++ fstart += ( (1-w1_s[b])*numlines_s[ bu_s[b] ] ) ; ++ ++ /* w2 = 1, then all FFT lines belong to this partition band */ ++ fstop -= ( (1-w2_s[b])*numlines_s[ bo_s[b] ] ) ; ++ ++ ++ if (fstart>0) fstart -= .5; /* contribution extends to center of freq band */ ++ fstop += .5; ++ ++ mstart = scalefac_band_short[b]; ++ mstop = scalefac_band_short[b+1]-1; ++ if (mstart>0) mstart -= .5; ++ mstop += .5; ++ ++ printf("cb=%2i(%2i)(%2i,%2i)(%3.2f,%3.2f) FFT:(%4.1f,%4.1f)(%5.0f,%5.0f)Hz MDCT:(%5.1f,%5.1f)(%5.0f,%5.0f)Hz \n", ++ b,npart_s_orig,bu_s[b],bo_s[b],w1_s[b],w2_s[b], ++ fstart,fstop,sfreq*fstart/BLKSIZE_s,sfreq*fstop/BLKSIZE_s, ++ mstart,mstop,mstart*.5*sfreq/192,mstop*.5*sfreq/192 ++ ); ++ } ++ ++ ++ /******************************************************************* ++ END DEBUG INFORMATION ++ *******************************************************************/ ++ } +diff -r -c -N encoder/doc/CVS/Entries lame3.70/doc/CVS/Entries +*** encoder/doc/CVS/Entries Wed Dec 31 17:00:00 1969 +--- lame3.70/doc/CVS/Entries Thu Apr 6 12:50:48 2000 +*************** +*** 0 **** +--- 1,2 ---- ++ D/html//// ++ D/man//// +diff -r -c -N encoder/doc/CVS/Repository lame3.70/doc/CVS/Repository +*** encoder/doc/CVS/Repository Wed Dec 31 17:00:00 1969 +--- lame3.70/doc/CVS/Repository Thu Apr 6 12:50:35 2000 +*************** +*** 0 **** +--- 1 ---- ++ lame/doc +diff -r -c -N encoder/doc/CVS/Root lame3.70/doc/CVS/Root +*** encoder/doc/CVS/Root Wed Dec 31 17:00:00 1969 +--- lame3.70/doc/CVS/Root Thu Apr 6 12:50:35 2000 +*************** +*** 0 **** +--- 1 ---- ++ markt@cvs.lame.sourceforge.net:/cvsroot/lame +diff -r -c -N encoder/doc/CVS/Tag lame3.70/doc/CVS/Tag +*** encoder/doc/CVS/Tag Wed Dec 31 17:00:00 1969 +--- lame3.70/doc/CVS/Tag Thu Apr 6 12:50:52 2000 +*************** +*** 0 **** +--- 1 ---- ++ Tlame3_70 +diff -r -c -N encoder/doc/html/CVS/Entries lame3.70/doc/html/CVS/Entries +*** encoder/doc/html/CVS/Entries Wed Dec 31 17:00:00 1969 +--- lame3.70/doc/html/CVS/Entries Thu Apr 6 12:50:53 2000 +*************** +*** 0 **** +--- 1,11 ---- ++ /contributors.html/1.5/Wed Mar 29 20:49:02 2000//Tlame3_70 ++ /examples.html/1.3/Wed Mar 29 20:49:02 2000//Tlame3_70 ++ /history.html/1.11.4.2/Thu Apr 6 18:50:53 2000//Tlame3_70 ++ /id3.html/1.4/Wed Mar 29 20:49:02 2000//Tlame3_70 ++ /index.html/1.6/Wed Mar 29 20:49:02 2000//Tlame3_70 ++ /lame.css/1.3/Wed Mar 29 20:49:02 2000//Tlame3_70 ++ /node5.html/1.1.1.1/Wed Nov 24 08:44:08 1999//Tlame3_70 ++ /node6.html/1.4/Wed Mar 29 20:49:02 2000//Tlame3_70 ++ /node7.html/1.3/Mon Jan 3 05:15:41 2000//Tlame3_70 ++ /switchs.html/1.6/Wed Mar 29 20:49:02 2000//Tlame3_70 ++ D +diff -r -c -N encoder/doc/html/CVS/Repository lame3.70/doc/html/CVS/Repository +*** encoder/doc/html/CVS/Repository Wed Dec 31 17:00:00 1969 +--- lame3.70/doc/html/CVS/Repository Thu Apr 6 12:50:35 2000 +*************** +*** 0 **** +--- 1 ---- ++ lame/doc/html +diff -r -c -N encoder/doc/html/CVS/Root lame3.70/doc/html/CVS/Root +*** encoder/doc/html/CVS/Root Wed Dec 31 17:00:00 1969 +--- lame3.70/doc/html/CVS/Root Thu Apr 6 12:50:35 2000 +*************** +*** 0 **** +--- 1 ---- ++ markt@cvs.lame.sourceforge.net:/cvsroot/lame +diff -r -c -N encoder/doc/html/CVS/Tag lame3.70/doc/html/CVS/Tag +*** encoder/doc/html/CVS/Tag Wed Dec 31 17:00:00 1969 +--- lame3.70/doc/html/CVS/Tag Thu Apr 6 12:50:53 2000 +*************** +*** 0 **** +--- 1 ---- ++ Tlame3_70 +diff -r -c -N encoder/doc/html/contributors.html lame3.70/doc/html/contributors.html +*** encoder/doc/html/contributors.html Wed Dec 31 17:00:00 1969 +--- lame3.70/doc/html/contributors.html Wed Mar 29 13:49:02 2000 +*************** +*** 0 **** +--- 1,132 ---- ++ <!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN"> ++ <!-- saved from url=(0051)http://internet.roadrunner.com/~mt/mp3/history.html --> ++ <HTML><HEAD><TITLE>Contributors ++ ++ ++ ++
    ++

    Contributors

    ++
    ++
    ++

    The following people contributed to the LAME development:

    ++

    Lauri Ahonen
    ++ Sakari Ailus
    ++ Tero Auvinen
    ++ Alex Ballegooy
    ++ Wilfried Behne
    ++ Bob Bell
    ++ Segher Boessenkool
    ++ Lionel Bonnet
    ++ Gabriel Bouvigne
    ++ Erik de Castro Lopo
    ++ Michael Cheng
    ++ Goran Dokic
    ++ James Droppo
    ++ Sergey Dubov
    ++ Alvaro Martinez Echevarria
    ++ Joakim Enerstam
    ++ Albert Faber
    ++ Nils Faerber
    ++ Sami Farin
    ++ Mikhail M. Fedotov
    ++ Joseph Fourier
    ++ Jani Frilander
    ++ Richard Gorton
    ++ Steffan Haeuser
    ++ Jeremy Hall
    ++ Paul Hartman
    ++ John Hayward-Warburton
    ++ Robert Hegemann
    ++ Mathew Hendry
    ++ Ben "Jacobs"
    ++ Mo Katz
    ++ Iwasa Kazmi
    ++ Joachim Kuebart
    ++ Leonid Kulakov
    ++ An van Lam
    ++ Juha Laukala
    ++ Felix von Leitner
    ++ Rafael Luebbert
    ++ Macik
    ++ Lars Magne Ingebrigtsen
    ++ Scott Manley
    ++ Vladimir Marek
    ++ Sergey A. Maslyakov
    ++ Chris Matrakidis
    ++ Greg Maxwell
    ++ Scott Miller
    ++ Kimmo Mustonen
    ++ Dan Nelson
    ++ Anton Oleynikov
    ++ Mike Oliphant
    ++ André Osterhues
    ++ Johannes Overmann
    ++ Jan Peman
    ++ Jan Rafaj
    ++ Gertjan van Ratingen
    ++ Miguel Revilla Rodriguez
    ++ Tim Ruddick
    ++ Ingo Saitz
    ++ Conrad Sanderson
    ++ Sergey Sapelin
    ++ William Schelter
    ++ Sigbj�rn Skj�ret
    ++ Nathan Slingerland
    ++ Patrick De Smet
    ++ Acy Stapp
    ++ Mark Stephens
    ++ Stephane Tavenard
    ++ Mark Taylor
    ++ Brad Threatt
    ++ Takehiro Tominaga
    ++ Warren Toomey
    ++ Martin Weghofer
    ++ Gerhard Wesp
    ++ Alfred Weyers
    ++ Ethan Yeo
    ++ Chuck Zenkus

    ++

     

    ++

    Original ISO contributors:

    ++

    Bill Aspromonte
    ++ Shaun Astarabadi
    ++ R. Bittner
    ++ Karlheinz Brandenburg
    ++ W. Joseph Carter
    ++ Jack Chang
    ++ Mike Coleman
    ++ Johnathan Devine
    ++ Ernst Eberlein
    ++ Dan Ellis
    ++ Peter Farrett
    ++ Jean-Georges Fritsch
    ++ Vlad Fruchter
    ++ Hendrik Fuchs
    ++ Bernhard Grill
    ++ Amit Gulati
    ++ Munsi Haque
    ++ Chuck Hsiao
    ++ Toshiyuki Ishino
    ++ Masahiro Iwadare
    ++ Earl Jennings
    ++ James Johnston
    ++ Leon v.d. Kerkhof
    ++ Don Lee
    ++ Mike Li
    ++ Yu-Tang Lin
    ++ Soren Neilsen
    ++ Simao F. Campos Neto
    ++ Mark Paley
    ++ Davis Pan
    ++ Tan Ah Peng
    ++ Kevin Peterson
    ++ Juan Pineda
    ++ Ernst F. Schroeder
    ++ Peter Siebert
    ++ Jens Spille
    ++ Sam Stewart
    ++ Al Tabayoyon
    ++ Kathy Wang
    ++ Franz-Otto Witte
    ++ Douglas Wong

    ++ ++ +diff -r -c -N encoder/doc/html/examples.html lame3.70/doc/html/examples.html +*** encoder/doc/html/examples.html Wed Dec 31 17:00:00 1969 +--- lame3.70/doc/html/examples.html Wed Mar 29 13:49:02 2000 +*************** +*** 0 **** +--- 1,52 ---- ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ Some command line examples ++ ++ ++ ++ ++ ++ ++ ++

    Some command line examples

    ++
      ++
    • Fixed bit rate jstereo 128kbs encoding:
      ++ lame sample.wav sample.mp3
      ++
      ++
      • ++
    • Fixed bit rate jstereo 128kbs encoding, highest quality (recommended):
      ++ lame -h sample.wav sample.mp3
      ++
      ++
      • ++
    • Fixed bit rate jstereo 112kbs encoding:
      ++ lame -b 112 sample.wav sample.mp3
      ++
      ++
      • ++
    • To disable joint stereo encoding (slightly faster, but less quality at bitrates<=128kbs):
      ++ lame -m s sample.wav sample.mp3
      ++
      ++
      • ++
    • Fast encode, low quality (no psycho-acoustics):
      ++ lame -f sample.wav sample.mp3
      ++
      ++
      • ++
    • Variable bitrate (use -V n to adjust quality/filesize):
      ++ lame -h -V 6 sample.wav sample.mp3
      ++
      ++
      • ++
    • Streaming mono 22.05kHz raw pcm, 24kbs output:
      ++ cat inputfile | lame -r -m m -b 24 -s 22.05 -- > output
      ++
      ++
      • ++
    • Streaming mono 44.1kHz raw pcm, with downsampling to 22.05 kHz:
      ++ cat inputfile | lame -r -m m -b 24 --resample 22.05 -- > output
      • ++
    ++ ++ +diff -r -c -N encoder/doc/html/history.html lame3.70/doc/html/history.html +*** encoder/doc/html/history.html Wed Dec 31 17:00:00 1969 +--- lame3.70/doc/html/history.html Thu Apr 6 12:50:53 2000 +*************** +*** 0 **** +--- 1,1632 ---- ++ ++ ++ ++ ++ ++ LAME Changelog ++ ++ ++ ++
    ++

    ++ History

    ++ Starting with LAME 3.0: ++
    red = features and bug fixes which effect quality ++
    blue = features and bug fixes which effect speed ++
    black = usability, portability, other ++

    ++ ++

    ++ ++

    ++ LAME 3.80beta  (CVS)

    ++ ++
      ++
    • ++ Takehiro Tominaga:  more efficient and faster ++ huffman encoding!
      • ++ ++
    • ++ Tomasz Motylewski and Mark Taylor:  MPEG2.5 ++ now supported!
      • ++ ++
    • ++ Mark Taylor: incorporated Takehiro's bitstream.c!  ++ This replaces the last of the ISO code.
      • ++ ++
    • ++ Scott  Manley and Mark Taylor:  good resampling ++ routine finaly in LAME.  uses a 19 point FIR filter with Blackman ++ window.  Very slow for non integer resampling ratios.
      • ++
    ++ ++

    ++ LAME 3.70   April 6 2000

    ++ ++
      ++
    • ++ "spahm":  default mode selection bug fixed.  In some cases, lame ++ was defaulting to regular stereo instead of jstereo when the user did not ++ specify a mode.
      • ++
    ++ ++

    ++ LAME 3.68beta  April 4 2000

    ++ ++
      ++
    • ++ Mark Taylor: mono encoding bug in DLL fixed.
      • ++ ++
    • ++ Ingo Saitz: bug in --cwlimit argument parsing fixed.
      • ++ ++
    • ++ Scott Manly: bug in 4-point resample code fixed.
      • ++
    ++ ++

    ++ LAME 3.67beta  March 27 2000

    ++ ++
      ++
    • ++ Robert Hegemann:  jstereo now enabled for MPEG2 ++ encodings
      • ++ ++
    • ++ Mark Taylor: old M/S stereo mode which used L/R maskings has been removed.
      • ++ ++
    • ++ Mark Taylor: Xing MPEG2 VBR headers now working.
      • ++ ++
    • ++ Mark Taylor:  When quantized coefficients are ++ all 0 in a band, set scalefactors to 0 also to save a few bits.
      • ++ ++
    • ++ Ingo Saitz:  Problems with framesize calculation ++ when using -f fast-math option fixed.
      • ++
    ++ ++

    ++ LAME 3.66beta March 21 2000

    ++ ++
      ++
    • ++ Bug fixes in BladeEnc DLL, possible click in last mp3 frame, VBR historgram ++ display, byteswapping option, ASM quantize routines work for both float ++ and double.
      • ++
    ++ ++

    ++ LAME 3.65beta   March 17 2000

    ++ ++
      ++
    • ++ Enabled ASM version of quantize_xrpow() - accidently disabled in lame3.64.
      • ++
    ++ ++

    ++ LAME 3.64beta  March 16 2000

    ++ ++
      ++
    • ++ Don Melton:  id3v1.1 tags & id3 bugfixes
      • ++ ++
    • ++ Gabriel Bouvigne:  L/R matching block type fix
      • ++ ++
    • ++ Bug fixed which was allowing quantized values to ++ exceed the maximum when not using -h
      • ++ ++
    • ++ Mark Taylor: Fitlers based on polyphase filterbank.  ++ should be slightly better since the responce is independent of the blocktype, ++ and they are slightly faster.
      • ++ ++
    • ++ Mark Taylor: API:  the API changed slightly - and this should be the ++ final version. There is a new routine: lame_encode_buffer() which takes ++ an arbritray sized input buffer, resamples & filters if necessary, ++ encodes, and returns the mp3buffer.  There are also several new #defines, ++ so it is possible to compile a simple encoding library with no decoding ++ or file I/O or command line parsing.  see the file API for details.
      • ++ ++
    • ++ Mark Taylor: MSVC stuff:  lame.exe (with and without the frame analyzer) ++ and the CDex lame_enc.dll
      • ++ ++
      should compile under MSVC.  The MSVC5 project files may need some ++ tweaking.  In particular, ++
      you need to make sure LAMEPARSE, LAMESNDFILE and HAVEMPGLIB ++
      are defined.  (and HAVEGTK for the GTK stuff).
    ++ ++

    ++ LAME 3.63beta  February 20  2000

    ++ ++
      ++
    • ++ Robert Hegemann:  FPE with -h fixed?
      • ++ ++
    • ++ Mathey Hendry:  FPE error catching for Cygwin, FPE fix for vbr mode ++ and output to /dev/null
      • ++ ++
    • ++ Jeremy Hall:  Fixed problems with input files where the number of ++ samples is not known.
      • ++ ++
    • ++ Mathew Hendry:  ASM quantize_xrpow() for GNU ++ i386
      • ++ ++
    • ++ Wilfried Behne  quantize_xrpow ()for PowerPC ++ and non-ASM
      • ++ ++
    • ++ Takehiro Tominaga:  GOGO FFTs  (not yet ++ used?)
      • ++ ++
       
    ++ ++

    ++ LAME 3.62beta   February 9 2000

    ++ ++
      ++
    • ++ Iwasa Kazmi:  frame analyzer short block display ++ of single subblocks (press 1,2  or 3)
      • ++ ++
    • ++ Ingo Saitz:  --help option added, with output ++ to stdout
      • ++ ++
    • ++ Alfred Weyers: short block AAC spreading function ++ bug fixed
      • ++ ++
    • ++ Takehiro Tominaga:  new scalefac data structure ++ - improves performance!
      • ++ ++
    • ++ Lionel Bonnet:  Bug fixed in MPEG2 scalefactor ++ routine: scalefactors were being severly limited.
      • ++ ++
    • ++ Takehiro Tominaga:  faster FFT routines from.  ++ These routines are also compatible with the GOGO routines, in case someone ++ is interested in porting them back to LAME.
      • ++ ++
    • ++ Sigbjørn Skjæret, Takehiro Tominaga:  ++ faster pow() code.
      • ++ ++
    • ++ Joachim Kuebart:  Found some unitialized variables ++ that were effecting quality for encodings which did not use the -h option ++ (now fixed).
      • ++ ++
    • ++ Mark Taylor: More modularization work.   It is now possible to ++ use LAME as a library where you can set the encoding parameters directly ++ and do your own file i/o.   The calling program is now it's own ++ mp3 output.  For an example of the LAME API, see main.c, or mp3rtp.c ++ or mp3x.c.  These can all be compiled as stand alone programs which ++ link with libmp3lame.a.
      • ++ ++
    • ++ Felix vos Leitner:  mp3rtp fixes.  mp3rtp is a standalone program ++ which will encode and stream with RTP.
      • ++ ++
    • ++ Robert Hegemann:  Information written to stderr displaying exactly ++ which type of lowpass filter (if any) is being used.
      • ++ ++
    • ++ Iwasa Kazmi:  mpglib (the mpg123 decoder) scsfi decoding fixes.
      • ++ ++
    • ++ Takehiro Tominaga:  More mpglib scsfi decoding fixes.
      • ++
    ++ ++

    ++ LAME 3.61beta  January 14 2000

    ++ ++
      ++
    • ++ Mark Taylor: Fixed bug with lowpass filters when ++ using VBR with a 64kbs or lower min bitrate setting.
      • ++ ++
    • ++ Takehiro Tominaga:  more efficient huffman encoding ++ splitting.
      • ++
    ++ ++

    ++ LAME 3.60beta   January 9 2000

    ++ ++
      ++
    • ++ Mark Taylor: Distribution now comes with self test.  Needs work to ++ be automated, see 'make test' in Makefile.
      • ++ ++
    • ++ Mark Taylor: AAC spreading function now the default
      • ++ ++
    • ++ Gabriel Bouvigne: updated HTML docs
      • ++ ++
    • ++ Felix von Leitner: compute correct file length from Xing header (if present) ++ when input file is a mp3 file
      • ++ ++
    • ++ Felix von Leitner: mp3rtp (standalone) program now included.  Not ++ yet tested.  mp3rtp ip:port:ttl  <infile>   /dev/null ++ will stream directly to ip:port using  RTP.
      • ++ ++
       
    ++ ++

    ++ LAME 3.59beta  January 4 2000

    ++ ++
      ++
    • ++ Takehiro Tominaga:  --noath option.  Disables ATH maskings.
      • ++ ++
    • ++ Gabriel Bouvigne:  updated HTML docs.
      • ++ ++
    • ++ Iwasa Kazmi:  makefile fixes
      • ++ ++
    • ++ Mark Taylor:  Fixed bug where first frame of data was always overwritten ++ with 0's.  Thanks to 'gol'
      • ++ ++
    • ++ Mark Taylor:  bug fixes in mid/side masking ++ ratios (thanks to Menno Bakker)
      • ++ ++
    • ++ Mark Taylor:  replaced norm_l, norm_s table data with formulas.
      • ++
    ++ ++

    ++ LAME 3.58beta  December 13 1999

    ++ ++
      ++
    • ++ Segher Boessenkool:  More accurate quantization ++ procedure!  Enabled with -h.
      • ++ ++
    • ++ Mathew Hendry, Acy Stapp and Takehiro Tominaga: ASM ++ optimizations for quantize_xrpow and quantize_xrpow_ISO.
      • ++ ++
    • ++ Chuck Zenkus:  "encoder inside" logo on web page
      • ++ ++
    • ++ Mark Taylor:   a couple people have asked for this.   ++ Allow LAME to overide VBR_min_bitrate if analog_silence detected.   ++ Analog_silence defined a la Robert:   energy  < ATH.
      • ++ ++
    • ++ An Van Lam: Valid bitrates were being printed for layer 2, not layer 3!
      • ++ ++
    • ++ Ethan Yeo:  Makefile.MSVC updated
      • ++ ++
    • ++ Mark Stephens:  updated all MSVC project files
      • ++ ++
    • ++ Robert Hegemann:  lowpass and highpass filters can be enabled with ++ --lowpass, --highpass
      • ++ ++
    • ++ Mark Taylor:  MS switching is now smoother: ++ ms_ratio average over 4 granules
      • ++ ++
    • ++ Takehiro Tominaga:  Scalefactor pre-emphasis ++ fixed (and now turned back on)
      • ++ ++
    • ++ Takehiro Tominaga:  Bug in M/S maskings:  ++ switch to turn on stereo demasking code was buggy.
      • ++
    ++ ++

    ++ LAME 3.57beta  November 22 1999

    ++ ++
      ++
    • ++ Sigbjørn Skjæret, patch to allow encoding from 8bit input ++ files when using LIBSNDFILE
      • ++ ++
    • ++ Mark Taylor: Automatic downsampling to nearest valid samplerate.
      • ++ ++
    • ++ Mark Taylor: Scalefactor bands demarked on MDCT plot in frameanalyzer
      • ++ ++
    • ++ Mark Taylor: Scalefactor preemphasis disabled for now.   The ++ algorithm was often doing more harm than good.
      • ++
    ++ ++

    ++ LAME 3.56beta  November 19 1999

    ++ ++
      ++
    • ++ Kimmo Mustonen: portabilty code cleanup.
      • ++ ++
    • ++ Vladimir Marek: id3 genre patch.
      • ++ ++
    • ++ Conrad Sanderson: new applypatch script.
      • ++ ++
    • ++ Mark Taylor: Initial window type now "STOP_TYPE" to reduce initial attenuation.  ++ This is needed because the new encoder delay is so short.  With a ++ NORM_TYPE, the first 240 samples would be attenuated.
      • ++ ++
    • ++ Mark Taylor: Padding at end of file now adjusted (hopefully!) to produce ++ as little padding as possible while still guarantee all input samples are ++ encoded.
      • ++ ++
    • ++ Takehiro Tominaga:  Reduced shortblock extra ++ bit allocation formulas by 10% since new huffman coding is at least 10% ++ more efficient.
      • ++
    ++ ++

    ++ LAME 3.55beta  November 11 1999

    ++ ++
      ++
    • ++ Albert Faber:  updated BladeEnc.dll
      • ++ ++
    • ++ Mark Taylor: Simple lowpass filter added to linear downsampling routine.
      • ++ ++
    • ++ Nils Faerber: updated man page.
      • ++ ++
    • ++ Mark Taylor: All floating point variables are delcared FLOAT or  FLOAT8.  ++ Change the definition of FLOAT8 in machine.h to run at 32bit preceision.
      • ++ ++
    • ++ Mark Taylor: Bug (introduced in 3.54beta) in stereo->mono downsampling ++ fixed.
      • ++
    ++ ++

    ++ LAME 3.54beta  November 8 1999

    ++ ++
      ++
    • ++ Mark Taylor: Encoder delay is now 48 samples.  Can be adjusted to ++ 1160 to sync with FhG (see ENCDELAY in encoder.h)  This is kind of ++ amazing, since if Takehiro put his MDCT/filterbank routine in a decoder, ++ we could have a total  delay of only 96 samples.
      • ++ ++
    • ++ Mark Taylor: More inconstancies found and fixed in ++ MPEG2 tables.
      • ++ ++
    • ++ Mark Taylor: Resampling from an MP3 input file now works.  But we ++ still dont have a lowpass filter so dont expect good results.
      • ++
    ++ ++

    ++ LAME 3.53beta  November 8 1999

    ++ ++
      ++
    • ++ Takehiro Tominaga:  Fixed MPEG2 problem in new ++ MDCT routines.  Takehiro's combined filterbank/MDCT routine is now ++ the default.  Removes all buffering from psymodel.c and the filterbanks/MDCT ++ routines.
      • ++
    ++ ++

    ++ LAME 3.52beta  November 8 1999

    ++ ++
      ++
    • ++ By permission of copyright holders of all GPL code in LAME,  all GPL ++ code is now released under a modified version of the LGPL (see the README ++ file)
      • ++ ++
    • ++ By popular demand, all C++ comments changed to C style comments
      • ++ ++
    • ++ Mark Taylor: Linear resampling now works.  Use --resample to set an ++ output samplerate different from the input samplerate.  (doesn't seem ++ to work with mp3 input files, and there is no lowpass filter, so dont expect ++ good results just yet)
      • ++ ++
    • ++ Takehiro Tominaga:  Faster Huffman encoding ++ routines
      • ++
    ++ The following changes are disabled because of MPEG2 ++ problems.  But to try them, set MDCTDELAY=48 in encoder.h, instead ++ of MDCTDELAY=528.: ++
      ++
    • ++ Takehiro Tominaga:  New MDCT routines with shorter ++ delay (48 samples instead of 528) and even faster than the old routines.
      • ++ ++
    • ++ Takehiro Tominaga:  Removed extra buffering ++ in psymodel.c
      • ++
    ++ ++

    ++ LAME 3.51  November 7 1999

    ++ ++
      ++
    • ++ Takehiro Tominaga: Bug in quantize.c absolute threshold of hearing calculation ++ for non-44.1kHz input files.
      • ++
    ++ ++

    ++ LAME 3.50  November 1 1999

    ++ ++
      ++
    • ++ LAME 3.37beta becomes official LAME 3.50 release
      • ++
    ++ ++

    ++ LAME 3.37beta  November 1 1999

    ++ ++
      ++
    • ++ Lionel Bonnet:  Found severe bug in MPEG2 Short ++ block SNR.
      • ++ ++
    • ++ Sergey Sapelin:  VBR Toc improvement.
      • ++ ++
    • ++ Sergey Dubov: fskip() routine
      • ++ ++
    • ++ Conrad Sanderson: replacement for filterbank.c.   Not much faster ++ but amazingly simpler.
      • ++
    ++ ++

    ++ LAME 3.36beta  October 25 1999

    ++ ++
      ++
    • ++ Albert Faber:  more MSVC and BladeDLL updates
      • ++ ++
    • ++ Kimmo Mustonen:  Much code cleanup and Amiga updates
      • ++ ++
    • ++ Anton Oleynikov: Borland C updates
      • ++ ++
    • ++ Mark Taylor: More stdin fixes:  For some reason, forward fseek()'s ++ would fail when used on pipes even though it is okay with redirection from ++ "<". So I changed all the forward fseek()'s to use fread().  This ++ should improve stdin support for wav/aiff files.  If you know the ++ input file is raw pcm, you can still use the '-r' option to avoid *all* ++ seeking of any kind.
      • ++ ++
       
    ++ ++

    ++ LAME 3.35beta  October 21 1999

    ++ ++
      ++
    • ++ Leonid Kulakov:  Serious bug in MPEG2 scalefactor ++ band tables fixed.
      • ++ ++
    • ++ Portability patches from:  Anton Oleynikov, Sigbjørn Skjæret, ++ Mathew Hendry, Richard Gorton
      • ++ ++
    • ++ Alfred Weyers: compiler options, updated timestatus.
      • ++ ++
    • ++ Albert Faber:  BladeDll and other updates (new machine.h).
      • ++ ++
    • ++ Monty:  updated Makefile to fix gcc inline math bug.
      • ++ ++
       
    ++ ++

    ++ LAME 3.34beta  October 12 1999

    ++ ++
      ++
    • ++ Mark Taylor: Bug fixed:  minimum bitrate in ++ VBR mode could be ignored for a few frames.
      • ++ ++
    • ++ Mark Taylor: New (minor) VBR tunings.
      • ++ ++
    • ++ Tim Ruddick: New wav/aiff header parsing routines.  Better parsing ++ and fewer fseek()'s.
      • ++ ++
    • ++ Anton Oleynikov:  patches to work with Borland C
      • ++ ++
    • ++ Gabriel Bouvigne:  Experimental voice option ++ enabled with --voice
      • ++
    ++ ++

    ++ LAME 3.33beta  October 11 1999

    ++ ++
      ++
    • ++ Robert Hegemann: RH VBR mode now the default and ++ only VBR mode.  The new code will always quantize to 0 distortion ++ and the quality is increased by reducing the masking from the psy-model.  ++ -X0 is still the default for now.
      • ++ ++
    • ++ Robert Hegemann: new -X5 mode
      • ++ ++
    • ++ Mathew Hendry: New timing code, removes the need for HAVETIMES
      • ++ ++
    • ++ Mathew Hendry:  assembler quantize_xrpow for ++ Windows
      • ++ ++
    • ++ Iwasa Kazmi:  stdin/stdout patch for Windows
      • ++ ++
    • ++ Mark Taylor: New option: "--athonly" will ignore the psy-model output and ++ use only the absolute threshold of hearing for the masking.
      • ++ ++
       
    ++ ++

    ++ LAME 3.32beta  October 8 1999

    ++ ++
      ++
    • ++ Takehiro Tominaga:  faster long block spreading ++ function convolution for non 44.1kHz sampling frequencies, and faster short ++ block spreading function convolution for all sampling frequencies.
      • ++ ++
    • ++ Takehiro Tominaga:  Completly rewritten huffman ++ table selection and count_bits().  More efficient table selection ++ results in many more bits per frame.
      • ++ ++
    • ++ Takehiro Tominaga:  More efficient scalefac ++ compress setting.
      • ++ ++
    • ++ Mike Cheng: new calc_noise2()
      • ++ ++
    • ++ Alfred Weyers: patch for timestatus() seconds rollover
      • ++ ++
       
    ++ ++

    ++ LAME 3.31beta  September 28 1999

    ++ ++
      ++
    • ++ Albert Faber:  updated his BladeDLL code.  This allows LAME to ++ be compiled into a BladeEnc compatiable .dll.
      • ++ ++
    • ++ Mike Cheng: faster l3psycho_ener() routine.
      • ++ ++
    • ++ Sigbjørn Skjæret: more code cleanup.
      • ++
    ++ ++

    ++ LAME  3.30beta  September 27 1999

    ++ ++
      ++
    • ++ Conrad Sanderson:  ID3 tag code added (type 'lame' for instructions)
      • ++ ++
    • ++ new mdct.c from Mike Cheng (no faster, but much cleaner code)
      • ++ ++
    • ++ Mathew Hendry: Microsoft nmake makefile and a couple other changes for ++ MSVC
      • ++ ++
    • ++ More modulization work:  One input sound file interface handles mp3's, ++ uncompressed audio, with or without LIBSNDFILE.  Fixes (hopefully) ++ a bunch of file I/O bugs introduced in 3.29 (Mark Taylor)
      • ++ ++
    • ++ LAME will now print valid samplerate/bitrate combinations (Mark Taylor)
      • ++ ++
    • ++ stdin/stdout fix for OS/2 (Paul Hartman)
      • ++ ++
    • ++ For mp3 input files, totalframes estimated based on filesize and first ++ frame bitrate. (Mark Taylor)
      • ++ ++
    • ++ Updated all functions with new style prototypes.  (Sigbjørn ++ Skjæret)
      • ++ ++
       
    ++ ++

    ++ LAME 3.29beta  September 21 1999

    ++ ++
      ++
    • ++ Bug in bigv_bitcount fixed.  Loop.c was overestimating ++ the number of bits needed, resulting in wasted bits every frame.  ++ (Leonid A. Kulakov)
      • ++ ++
    • ++ Bug in *_choose_table() fixed   These routines ++ would not sellect the optimal Huffman table in some cases.   ++ (Leonid A. Kulakov)
      • ++ ++
    • ++ Tuning of ATH normalization (macik)
      • ++ ++
    • ++ Removed unused variables and fixed function prototypes (Sigbjørn ++ Skjæret)
      • ++ ++
    • ++ Sami Farin sent a  .wav  file that  LAME built in support ++ choked on.   I added a slightly more sophisticated wav header ++ parsing to handle this, but if you have trouble, use libsndfile.
      • ++ ++
    • ++ Resampling hooks and options added.  Buffering and resampling routines ++ need to be written.
      • ++ ++
    • ++ LAME will now take an mp3 file as input.  When resampling code is ++ working, LAME will be able to (for example) convert a high bitrate stereo ++ mp3 to a low bitrate mono mp3 for streaming.
      • ++
    ++ ++

    ++ LAME 3.28beta  September 15 1999

    ++ ++
      ++
    • ++ Serious bug fixed in high frequency MDCT coefficients.  ++ Huffman coding was reversing the order of the count1 block quadruples.   ++ (Leonid A. Kulakov)
      • ++ ++
    • ++ nint() problems under Tru64 unix fixed and preprocessor variable HAVE_NINT ++ removed.  (Bob Bell)
      • ++ ++
    • ++ Compiler warning fixes and code cleanup   (Sigbjørn Skjæret, ++ Lionel Bonnet)
      • ++ ++
    • ++ USAGE file now includes suggestions for downsampling.  For low bitrate ++ encodings, proper downsampling can give dramatically better results.  ++ (John Hayward-Warburton)
      • ++
    ++ ++

    ++ LAME 3.27beta  September 12 1999

    ++ ++
      ++
    • ++ Several bugs in encode.c and l3bitstream.c fixed by Lionel Bonnet.
      • ++ ++
    • ++ Bugs in new VBR (#define RH) formula for mono input file and mid/side encoding ++ fixed.
      • ++
    ++ ++

    ++ LAME 3.26beta  September 10 1999

    ++ ++
      ++
    • ++ The "-m m" option (mono .mp3 file) will automatically mix left and right ++ channels if the input file is stereo.  (Alfred Weyers)
      • ++ ++
    • ++ New quant_compare algorithm (method for deciding ++ which of two quantizations is better) enabled with -X4 (Greg Maxwell)
      • ++ ++
    • ++ New mid/side VBR bit allocation formula.  Mid ++ channel bits are set by the quality requirements, and then the side channel ++ uses a reduced number of bits (in a proportion coming from the fixed bitrate ++ code).  This might not be optimal, but it should be pretty good and ++ no one knows what the optimal solution should be.  (Greg Maxwell)
      • ++ ++
    • ++ New VBR (#define RH) tunings based on detailed listening ++ tests by Macik and Greg Maxwell.
      • ++ ++
    • ++ Sigbjørn Skjæret fixed several compiler warnings (which turned ++ out to be potential bugs)
      • ++ ++
    • ++ Takehiro Tominaga fixed a low bitrate bug in reduce_side()
      • ++ ++
    • ++ Alfred Weyers fixed some buffer overflows.
      • ++ ++
    • ++ New ATH (absolute threshold of hearing) formula replaces ++ buggy ISO code, and adds  analog silence treatment  (removal ++ of coefficients below below ATH).   These are turned on by default ++ but have not been fully tested.  (Robert Hegemann)
      • ++ ++
    • ++ Bug in short block spreading function fixed.  ++ (Robert Hegemann)
      • ++
    ++ ++

    ++ LAME 3.25beta  August 22 1999

    ++ ++
      ++
    • ++ Sigbjørn Skjæret fixed a zero byte malloc call.  This ++ bug  was introduced in 3.24 and causes problems on non Linux systems.
      • ++ ++
    • ++ Bit allocation routines would sometimes allocate more than 4095 bits to ++ one channel of one granule.  A couple of people reported problems ++ that might be caused by this, especially at higher bitrates.
      • ++ ++
    • ++ Nils Faerber updated the man page and fixed many of the compiler warnings.
      • ++ ++
       
    ++ ++

    ++ LAME 3.24beta   August 15 1999

    ++ ++
      ++
    • ++ This release contains the following new code (for developers) which is ++ disabled by default:
      • ++ ++
    • ++ Robert Hegemann:  Completely overhauled VBR code.  Now computes ++ exact number of bits required for the given qualty and then quantized with ++ the appropriate bitrate.
      • ++ ++
    • ++ Several new quantization quality measures.
      • ++
    ++ ++

    ++ LAME 3.23beta  August 8 1999

    ++ ++
      ++
    • ++ Very nice continuously updated VBR histogram display from Iwasa Kazmi.  ++ (disabled with --nohist).
      • ++ ++
    • ++ More modulerization work.  The encoding engine can now be compiled ++ into libmp3lame, but the interface is awkward.
      • ++ ++
    • ++ Bug fixed in FFT Hann window formula (Leonid A. Kulakov).
      • ++ ++
    • ++ New LAME logo on the download page.  Created by Chris Michalisles.
      • ++ ++
    • ++ Several VBR algorithm improvements from Robert Hegemann.  ++ New quantization noise metrics and VBR quality measure takes into account ++ mid/side encoding.  Should produce smaller files with the same quality, ++ especially when using jstereo.
      • ++
    ++ ++

    ++ LAME 3.22beta  July 27 1999

    ++ ++
      ++
    • ++ Downsampling (stereo to mono) bug with MPEG2 fixed.  (Mike Oliphant)
      • ++ ++
    • ++ Downsampling now merges L & R channels - before it only took the L ++ channel.
      • ++ ++
    • ++ More modularization and code cleanup from Albert Faber and myself.
      • ++ ++
    • ++ Input filesize limit removed for raw pcm input files.  For other file ++ types, LAME will still only read the first 2^32 samples, (27 hours of playing ++ time at 44.1kHz).
      • ++ ++
       
    ++ ++

    ++ LAME 3.21beta  July 26 1999

    ++ ++
      ++
    • ++ Correct Mid/Side masking thresholds for JSTEREO mode!  ++ This is enabled with -h.  It  makes LAME about 20% slower since ++ it computes psycho-acoustics for L,R Mid and Side channels.
      • ++ ++
    • ++ "Analog silence" threshold added.  Keeps VBR ++ from upping the bitrate during very quite passages.  (Robert.Hegemann)
      • ++ ++
    • ++ New VBR quality setting from Robert Hegemann.  ++ It is based on the idea that distortion at lower bit rates sounds worse ++ than at higher bitrates, and so the allowed distortion (VBR quality setting) ++ is proportional to the bitrate.  Because of this, default minimum ++ bitrate is now 32kbs.
      • ++ ++
    • ++ Expermental subblock gain code enabled with -Z.
      • ++ ++
    • ++ New "-r" option for raw pcm input files.  With -r, LAME will not do ++ any fseek()'s or look for wav and aiff headers on the input file.
      • ++ ++
    • ++ Bug fixes in mp3x (frame analyzer) for viewing frames near end of the file.
      • ++ ++
    • ++ Bug fixed to allow setting the sampling rate of raw pcm input files.
      • ++
    ++ ++

    ++ LAME 3.20beta  July 19 1999

    ++ ++
      ++
    • ++ Bug in get_audio.c fixed.  Libsndfile wrappers would not compile (Miguel ++ Revilla  Rodriguez)
      • ++ ++
    • ++ Nils Faerber found some unitialized variables and some wierd extranous ++ computations in filter_subband, now fixed.  This was causing seg faults ++ on some machines.
      • ++ ++
       
    ++ ++

    ++ LAME 3.19beta  July 18 1999

    ++ ++
      ++
    • ++ Oops!  Robert Hegemann immediatly found a bug ++ in the new  (old -Z option) quantization code.  calc_noise1 was ++ not returning tot_noise, so non ms-stereo frames were buggy.
      • ++ ++
       
    ++ ++

    ++ LAME 3.18beta  July 17 1999

    ++ ++
      ++
    • ++ Many psycho-acoustic bug fixes.  Dan Nelson ++ discovered a bug in MPEG2: For short blocks, the code assumes 42 partition ++ bands.  MPEG1 sometimes has less, MPEG2 can have more.  In MPEG1, ++ this bug would not have effected the output if your compiler initializes ++ static variables to 0 on creation.  In MPEG2 it leads to array out-of-bounds ++ access errors. Finally, there was a related bug in MPEG1/MPEG2, short & ++ long blocks where the energy above 16kHz was all added to partition band ++ 0. (the lowest frequeny partition band!)
      • ++ ++
    • ++ The -Z option (Gabriel Bouvigne's idea of using total ++ quantization noise to choose between two quantizations with the same value ++ of "over") is now the default.  I believe this helps remove the trilling ++ sound in Jan's testsignal4.wav.  The quality of testsignal2.wav and ++ testsignal4.wav are now better than Xing and getting closer to FhG.
      • ++ ++
    • ++ Bug fixes in frame & sample count for downsampling mode. (ben "jacobs")
      • ++ ++
    • ++ Patches to improve modulization.  (ben "jacobs")
      • ++
    ++ ++

    ++ LAME 3.17beta  July 11 1999

    ++ ++
      ++
    • ++ substantial code cleanup towards goal of making LAME more modular.
      • ++
    ++ ++

    ++ LAME 3.16beta  July 11 1999

    ++ ++
      ++
    • ++ New tunings of window switching, and better bit allocation ++ based on pe.  (Jan Rafaj.  improves both testsignal2.wav and ++ testsignal4.wav).
      • ++ ++
    • ++ Bug in mid/side quantization when side channel was ++ zero fixed.  (Albert Faber)
      • ++ ++
    • ++ Removed some extranous computations in l3psy.c (Robert Hegemann)
      • ++ ++
    • ++ More detailed timing status info, including hours display. (Sakari Ailus) ++ and percentage indicator (Conrad Sanderson).
      • ++ ++
    • ++ Window_subband and calc_noise1,calc_noise2 speedups.  ++ Quantize_xrpow speedup should be significant on non GNU/intel systems. ++ (Mike Cheng)
      • ++ ++
    • ++ Better initial guess for VBR bitrate.  Should ++ speed up VBR encoding.  (Gabriel Bouvigne)
      • ++ ++
    • ++ More advanced .wav header parsing.  fixes bugs involving click in ++ first frame. (Robert.Hegemann)
      • ++ ++
    • ++ Correct filesize and total frame computation when using LIBSNDFILE (ben ++ "jacobs")
      • ++ ++
    • ++ Click in last frame (buffering problem) when using libsndfile fixed.
      • ++ ++
    • ++ Audio I/O code overhauled.  There is now a uniform audio i/o interface ++ to libsndfile or the LAME built in wav/aiff routines.  All audio i/o ++ code localized to get_audio.c.
      • ++ ++
       
    ++ ++

    ++ LAME 3.15beta

    ++ ++
      ++
    • ++ times()/clock() problem fixed for non-unix OS.  (Ben "Jacobs")
      • ++ ++
    • ++ Fixed uninitialized pe[] when using fast mode.  (Ben "Jacobs")
      • ++
    ++ ++

    ++ LAME 3.13  June 24 1999

    ++ ++
      ++
    • ++ Patches for BeOS from Gertjan van Ratingen.
      • ++ ++
    • ++ Makefile info for OS/2 Warp 4.0  (from dink.org).
      • ++ ++
    • ++ Status display now based on wall clock time, not cpu time.
      • ++ ++
    • ++ mem_alloc no longer allocates twice as much memory as needed (Jan Peman).
      • ++
    ++ ++

    ++ 3.12pre9

    ++ ++
      ++
    • ++ Updated BLADEDLL code to handle recent changes (Albert Faber).
      • ++ ++
    • ++ Bug fixed in parsing options when not using GTK (Albert Faber).
      • ++ ++
    • ++ MPEG2 Layer III psycho acoustics now working.
      • ++ ++
    • ++ Improved huffman encoding Chris Matrakidis. (10% ++ faster).  I dont know how he finds these improvements!  LAME ++ with full quality now encodes faster than real time on my PII 266.
      • ++ ++
    • ++ Fixed time display when encoding takes more than 60 minutes.
      • ++
    ++ ++

    ++ 3.12pre8

    ++ ++
      ++
    • ++ New mid/side stereo ++ criterion.  LAME will use mid/side stereo only when the difference ++ between L & R masking thresholds (averaged over all scalefactors) is ++ less then 5db.  In several test samples it does a very good job mimicking ++ the FhG encoder.
      • ++ ++
    • ++ Bug in mid/side stereo fixed:  independent variation ++ of mid & side channel scalefactors disabled.  Because of the way ++ outer_loop is currently coded, when encoding mid/side coefficietns using ++ left/right thresholds, you have to vary the scalefactors simultaneously.
      • ++ ++
    • ++ Bug in side/mid energy ratio calculation fixed. (Thanks ++ to Robert Hegemann)
      • ++ ++
    • ++ Default mode is stereo (not jstereo) if bitrate is chosen as 192kbs or ++ higher.  Tero Auvinen first pointed out that FhG seems to think at ++ 160kbs, their encoder is so good it doesn't need jstereo tricks. Since ++ LAME is not as good as FhG, I am going to claim that 192kbs LAME is so ++ good it doens't need jstereo tricks, and thus it is disabled by default.
      • ++ ++
    • ++ WAV header parsing for big-endian machines, and automatic detection of ++ big-endian machines.  (Thanks to  Sigbjørn Skjæret).
      • ++ ++
    • ++ added 56 sample delay to sync LAME with FhG.
      • ++ ++
    • ++ MP3x (frame analyzer) can now handle MPEG2 streams.
      • ++
    ++ ++

    ++ 3.12pre7

    ++ ++
      ++
    • ++ MPEG2 layer III now working!  lower bit rates (down to 8kbs) and 3 ++ more sampling frequencies:  16000, 22050, 24000Hz. Quality is poor ++ - the psy-model does not yet work with these sampling frequencies.
      • ++ ++
    • ++ Fixed "ERROR: outer_loop(): huff_bits < 0." bug when using VBR.
      • ++ ++
    • ++ bash and sh scripts to run LAME on multiple files now included.  (from ++ Robert Hegemann and Gerhard Wesp respectively)
      • ++ ++
    • ++ bug fix in encoding times for longer files from  (Alvaro Martinez ++ Echevarria)
      • ++ ++
    • ++ yet another segfault in the frame analyzer fixed.
      • ++ ++
    • ++ ISO psy-model/bit allocation routines removed.  This allowed makeframe() ++ to be made much simpler, and most of the complicated buffering is now gone. ++ Eventually I would like the encoding engine to be a stand alone library.
      • ++
    ++ ++

    ++ 3.12pre6

    ++ ++
      ++
    • ++ Better VBR tuning.  Find minimum bitrate with ++ distortion less than the allows maximum.  A minimum bit rate is imposed ++ on frames with short blocks (where the measured distortion can not be trusted).   ++ A minimum frame bitrate can be specified with -b, default=64kbs.
      • ++ ++
    • ++ LIBSNDFILE support.  ++ With libsndfile, LAME can encode almost all sound formats.  Albert ++ Faber did the work for this, including getting libsndfile running under ++ win32.
      • ++ ++
    • ++ CRC checksum now working!   (Thanks to Johannes Overmann )
      • ++ ++
    • ++ frame analyzer will now work with mono .mp3 files
      • ++ ++
    • ++ more code tweeks from Jan Peman.
      • ++ ++
    • ++ Compaq-Alpha(Linux) fixes and speedups from Nils ++ Faerber.
      • ++ ++
    • ++ Faster bin_search_StepSize from Juha Laukala.
      • ++ ++
    • ++ Faster quantize() from Mike Cheng
      • ++ ++
    • ++ Faster quantize_xrpow() from Chris Matrakidis.  ++ xrpow_flag removed since this option is now on by default.
      • ++ ++
    • ++ Fixed .wav header parsing from Nils Faerber.
      • ++ ++
    • ++ Xing VBR frame info header code from Albert Faber.   "Xing" and ++ "LAME 3.12" embedded in first frame.
      • ++ ++
    • ++ Bug in VBR bit allocation based on "over" value fixed.
      • ++
    ++ ++

    ++ LAME 3.11  June 3 1999

    ++ ++
    ++
  • ++ Almost all warnings (-Wall) now fixed!  (Thanks to Jan Peman)
    • ++ ++
  • ++ More coding improvements from Gabriel Bouvigne and Warren Toomey.
    • ++ ++
  • ++ VBR  (variable bit rate).   Increases ++ bit rate for short blocks and for frames where the number of bands containing ++ audible distortion is greater than a given value.  Much tuning needs ++ to be done.
    • ++ ++
  • ++ Patch to remove all atan() calls from James Droppo.
    • ++
    ++ ++

    ++ LAME 3.10 May 30 1999

    ++ ++
      ++
    • ++ Fast mode (-f)  disables psycho-acoustic model ++ for real time encoding on older machines.  Thanks to Lauri Ahonen ++ who first sent a patch for this.
      • ++ ++
    • ++ New bit reservoir usage scheme to accommodate the ++ new pre-echo detection formulas.
      • ++ ++
    • ++ Tuning of AWS and ENER_AWS pre-echo formulas by Gabriel ++ Bouvigne and myself.  They work great!  now on by default.
      • ++ ++
    • ++ In jstereo, force blocktypes for left & right channels to be identical.  ++ FhG seems to do this.  It can be disabled with "-d".
      • ++ ++
    • ++ Patches to compile MP3x under win32 (Thanks to Albert Faber).
      • ++ ++
    • ++ bin_serach_stepsize limited to a quantizationStepSize ++ of  -210 through 45.
      • ++ ++
    • ++ outer_loop()  will now vary Mid & Side scalefactors ++ independently.  Can lead to better quantizations, but it is slower ++ (twice as many quantizations to look at).  Running with "-m f" does ++ not need this and will run at the old speed
      • ++ ++
    • ++ Bug in inner_loop would allow quantizations larger ++ than allowed.  (introduced in lame3.04, now fixed.)
      • ++ ++
    • ++ Updated HTML documentation from Gabriel Bouvigne.
      • ++ ++
    • ++ Unix  man page from William Schelter.
      • ++ ++
    • ++ numlines[] bug fixed.  (Thanks to Rafael Luebbert, ++ MPecker author).
      • ++ ++
    • ++ Quantization speed improvements from Chirs Matrakidis.
      • ++ ++
    • ++ When comparing quantizations with the same number ++ of bands with audible distortion, use the one with the largest scalefactors, ++ not the first one outer_loop happened to find.
      • ++ ++
    • ++ Improved defination of best quantization when using -f (fast mode).
      • ++ ++
    • ++ subblock code now working.  But no algorithm to choose subblock gains ++ yet.
      • ++ ++
    • ++ Linux now segfaults on floating point exceptions.  Should prevent ++ me from releasing binaries that crash on other operating systems.
      • ++
    ++ ++

    ++ May 22 1999

    ++ ++
      ++
    • ++ Version 3.04 released.
      • ++ ++
    • ++ Preliminary documentation from Gabriel Bouvigne.
      • ++ ++
    • ++ I wouldn't have thought it was possible, but now ++ there are even more speed improvements from Chris Matrakidis!  Removed ++ one FFT when using joint stereo, and many improvements in loop.c.
      • ++ ++
    • ++ "Fake" ms_stereo mode renamed "Force" ms_stereo since it forces mid/side ++ stereo on all frames.  For some music this is said to be a problem, ++ but for most music mode is probably better than the default jstereo because ++ it uses specialized mid/side channel masking thresholds.
      • ++ ++
    • ++ Small bugs in Force ms_stereo mode fixed.
      • ++ ++
    • ++ Compaq Alpha fixes from Nathan Slingerland.
      • ++ ++
    • ++ Some new experimental pre-echo detection formulas ++ in l3psy.c (#ifdef AWS and #ifdef ENER_AWS, both off by default.  ++ Thanks to Gabriel Bouvigne and Andre Osterhues)
      • ++ ++
    • ++ Several bugs in the syncing of data displayed by mp3x (the frame analyzer) ++ were fixed.
      • ++ ++
    • ++ highq (-h) option added.  This turns on things (just one so far) that ++ should sound better but slow down LAME.
      • ++
    ++ May 18 1999 ++
      ++
    • ++ Version 3.03 released.
      • ++ ++
    • ++ Faster (20%) & cleaner FFT (Thanks to Chris Matrakidis  ++ http://www.geocities.com/ResearchTriangle/8869/fft_summary.html)
      • ++ ++
    • ++ mods so it works with VC++ (Thanks to Gabriel Bouvigne, www.mp3tech.org)
      • ++ ++
    • ++ MP3s marked "original" by default  (Thanks to Gabriel Bouvigne, www.mp3tech.org)
      • ++ ++
    • ++ Can now be compiled into a BladeEnc compatible .DLL   (Thanks ++ to  Albert Faber, CDex author)
      • ++ ++
    • ++ Patches for "silent mode" and stdin/stdout  (Thanks to Lars Magne ++ Ingebrigtsen)
      • ++ ++
    • ++ Fixed rare bug: if a long_block is sandwiched between ++ two short_blocks, it must be changed to a short_block, but the short_block ++ ratios have not been computed in l3psy.c.  Now always compute short_block ++ ratios just in case.
      • ++ ++
    • ++ Fixed bug with initial quantize step size when many ++ coefficients are zero.  (Thanks to Martin Weghofer).
      • ++ ++
    • ++ Bug fixed in MP3x display of audible distortion.
      • ++ ++
    • ++ improved status display (Thanks to Lauri Ahonen).
      • ++
    ++ ++

    ++ May 12 1999

    ++ ++
      ++
    • ++ Version 3.02 released.
      • ++ ++
    • ++ encoder could use ms_stereo even if channel 0 and ++ 1 block types were different.  (Thanks to Jan Rafaj)
      • ++ ++
    • ++ added -k option to disable the 16kHz cutoff at 128kbs.  ++ This cutoff is never used at higher bitrates. (Thanks to Jan Rafaj)
      • ++ ++
    • ++ modified pe bit allocation formula to make sense ++ at bit rates other than 128kbs.
      • ++ ++
    • ++ fixed l3_xmin initialization problem which showed up under FreeBSD.  ++ (Thanks to Warren Toomey)
      • ++
    ++ May 11 1999 ++
      ++
    • ++ Version 3.01 released
      • ++ ++
    • ++ max_name_size increased to 300  (Thanks to Mike Oliphant)
      • ++ ++
    • ++ patch to allow seeks on input file (Thanks to Scott Manley)
      • ++ ++
    • ++ fixes for mono modes (Thanks to everyone who pointed this out)
      • ++ ++
    • ++ overflow in calc_noise2 fixed
      • ++ ++
    • ++ bit reservoir overflow when encoding lots of frames with all zeros  ++ (Thanks to Jani Frilander)
      • ++
    ++ ++


    May 10 1999 ++

      ++
    • ++ Version 3.0 released
      • ++ ++
    • ++ added GPSYCHO (developed by Mark Taylor)
      • ++ ++
    • ++ added MP3x (developed by Mark Taylor)
      • ++ ++
    • ++ LAME now maintained by Mark Taylor
      • ++
    ++ November 8 1998 ++
      ++
    • ++ Version 2.1f released
      • ++ ++
    • ++ 50% faster filter_subband() routine in encode.c contributed by James Droppo
      • ++
    ++ November 2 1998 ++
      ++
    • ++ Version 2.1e released.
      • ++ ++
    • ++ New command line switch -a auto-resamples a stereo input file to ++ mono.
      • ++ ++
    • ++ New command line switch -r resamples from 44.1khz to 32khz [this ++ switch doesn't work really well. Very tinny sounding output files. Has ++ to do with the way I do the resampling probably]
      • ++ ++
    • ++ Both of these were put into the ISO code in the encode.c file, and are ++ simply different ways of filling the input buffers from a file.
      • ++
    ++ October 31 1998 ++
      ++
    • ++ Version 2.1d released
      • ++ ++
    • ++ Fixed memory alloc in musicin.c (for l3_sb_sample)
      • ++ ++
    • ++ Added new command line switch (-x) to force swapping of byte order
      • ++ ++
    • ++ Cleaned up memory routines in l3psy.c. All the mem_alloc() and free() routines ++ where changed so that it was only done once and not every single ++ time the routine was called.
      • ++ ++
    • ++ Added a compile time switch -DTIMER that includes all timing info. It's ++ a switch for the time being until some other people have tested on their ++ system. Timing code has a tendency to do different things on different ++ platforms.
      • ++
    ++ October 18 1998 ++
      ++
    • ++ Version 2.1b released.
      • ++ ++
    • ++ Fixed up bug: all PCM files were being read as WAV.
      • ++ ++
    • ++ Played with the mem_alloc routine to fix crash under amigaos (just allocating ++ twice as much memory as needed). Might see if we can totally do without ++ this routine. Individual malloc()s where they are needed instead
      • ++ ++
    • ++ Put Jan Peman's quality switch back in. This reduces quality via the '-q ' ++ switch. Fun speedup which is mostly harmless if you're not concerned with ++ quality.
      • ++ ++
    • ++ Compiling with amiga-gcc works fine
      • ++
    ++ October 16 1998 ++
      ++
    • ++ Version 2.1a released. User input/output has been cleaned up a bit. WAV ++ file reading is there in a very rudimentary sense ie the program will recognize ++ the header and skip it, but not read it. The WAV file is assumed to be ++ 16bit stereo 44.1khz.
      • ++
    ++ October 6 1998 ++
      ++
    • ++ Version 2.1 released with all tables now incorporated into the exe. Thanks ++ to Lars Magne Ingebrigtseni(larsi@ifi.uio.no)
      • ++
    ++ October 4 1998In response to some concerns about the quality of ++ the encoder, I have rebuilt the encoder from scratch and carefully compared ++ output at all stages with the output of the unmodified ISO encoder. Version2.0 ++ of LAME is built from the ISO source code (dist10), and incorporates modifications ++ from myself and the 8hz effort. The output file from LAME v2.0 is identical ++ to the output of the ISO encoder for my test file.Since I do not have heaps ++ of time, I left the ISO AIFF file reader in the code, and did not incorporate ++ a WAV file reader.Added section on qualityOctober ++ 1 1998 ++
      ++
    • ++ Updated web page and released LAME v1.0
      • ++
    ++ Up to September 1998 ++
      Working on the 8hz source code... ++
        ++
      • ++ Patched the 8hz source code
        • ++ ++
      • ++ 45% faster than original source (on my freebsd p166).
        • ++ ++
          ++
        • ++ m1 - sped up the mdct.c and quantize() functions [MDCTD, MDCTD2, LOOPD]
          • ++ ++
        • ++ m2 - sped up the filter_subband routine using Stephane Tavenard's ++ work from musicin [FILTST]
          • ++ ++
        • ++ m2 - minor cleanup of window_subband [WINDST2]
          • ++ ++
        • ++ m2 - Cleaned up a few bits in l3psy.c. Replaced a sparse matrix multiply ++ with a hand configured unrolling [PSYD]
          • ++ ++
        • ++ m3 - (amiga only) Added in the asm FFT for m68k (based on sources from ++ Henryk ++ Richter and Stephane Tavenard)
          • ++ ++
        • ++ m4 - raw pcm support back in
          • ++ ++
        • ++ m5 - put in a byte-ordering switch for raw PCM reading (just in case)
          • ++ ++
        • ++ m6 - reworked the whole fft.c file. fft now 10-15% faster.
          • ++ ++
        • ++ m7 - totally new fft routine. exploits fact that this is a real->complex ++ fft. About twice as fast as previous fastest fft (in m6). (C fft routine ++ is faster than the asm one on an m68k!)
          • ++ ++
        • ++ m8
          • ++ ++
            ++
          • ++ - Now encodes from stdin. Use '-' as the input filename. Thanks to Brad ++ Threatt
            • ++ ++
          • ++ - Worked out that the 1024point FFT only ever uses the first 6 phi values, ++ and the first 465 energy values. Saves a bunch of calculations.
            • ++ ++
          • ++ - Added a speed-up/quality switch. Speed is increased but quality is decreased ++ slightly. ++ My ears are bad enough not to be able to notice the difference in quality ++ at low settings :). Setting '-q 1' improves speed by about 10%. '-q 100' ++ improves speed by about 26%. Enoding of my test track goes from 111s (at ++ default '-q 0') to 82s (at -q 100). Thanks to Jan Peman for this ++ tip.
            • ++
          ++ ++
        • ++ m9 - fixed an error in l3psy.c. numlines[] is overwritten with incorrect ++ data. Added a new variable numlines_s[] to fix this. Thanks again to Jan ++ Peman.
          • ++ ++
        • ++ m10 - Down to 106 seconds by selecting a few more compiler options. Also ++ added a pow20() function in l3loop.c to speed up (ever so slightly) calls ++ to pow(2.0, x)
          • ++ ++
        • ++ m11
          • ++ ++
            ++
          • ++ No speedups. Just cleaned up some bits of the code.
            • ++ ++
          • ++ Changed K&R prototyping to 'normal' format. Thanks to Steffan Haeuser ++ for his help here.
            • ++ ++
          • ++ Changed some C++ style comments to normal C comments in huffman.c
            • ++ ++
          • ++ Removed the #warning from psy_data.h (it was getting annoying!)
            • ++ ++
          • ++ Removed reference in bitstream.c to malloc.h. Is there a system left where ++ malloc.h hasn't been superceded by stdlib.h?
            • ++
          ++ ++
        • ++ In Progess:
          • ++ ++
            ++
          • ++ my PSYD hack for the spreading functions is only valid for 44.1khz - Should ++ really put in a "if freq = 44.1khz" switch for it. Someone might want to ++ extend the speedup for 48 and 32khz.
            • ++ ++
          • ++ Putting in Jan Peman's quantanf_init speedup.
            • ++
          ++
        ++
      ++
    ++ ++ ++ +diff -r -c -N encoder/doc/html/id3.html lame3.70/doc/html/id3.html +*** encoder/doc/html/id3.html Wed Dec 31 17:00:00 1969 +--- lame3.70/doc/html/id3.html Wed Mar 29 13:49:02 2000 +*************** +*** 0 **** +--- 1,221 ---- ++ ++ ++ ++ ID3 tags ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++

    ID3 tags

    ++

     

    ++

    LAME is able to embed ID3 v1 or v1.1 tags inside the encoded MP3 file. This ++ allows to have some usefull information about the music track included inside ++ the file. Those data can be read by most MP3 players.

    ++

    ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++
    switchparameter
    --tt "title"title of song (max 30 chars)
    --ta "artist"artist who did the song (max 30 chars)
    --tl "album"album where it came from (max 30 chars)
    --ty "year"year in which the song/album was made (max 4 chars)
    --tc "comment"additional info (max 30 chars or max 28 chars if using ++ the "track" option)
    --tn "track"track number of the song on the CD (1 to 99)
    ++ using this option will add an ID3v1.1 tag
    --tg "genre" genre of song (name or number)
    ++


    ++ The following genres are supported:
    ++
    ++

    ++ ++ ++ ++ ++ ++
    00 - Blues
    ++ 01 - Classic Rock
    ++ 02 - Country
    ++ 03 - Dance
    ++ 04 - Disco
    ++ 05 - Funk
    ++ 06 - Grunge
    ++ 07 - Hip-Hop
    ++ 08 - Jazz
    ++ 09 - Metal
    ++ 10 - New Age
    ++ 11 - Oldies
    ++ 12 - Other
    ++ 13 - Pop
    ++ 14 - R&B
    ++ 15 - Rap
    ++ 16 - Reggae
    ++ 17 - Rock
    ++ 18 - Techno
    ++ 19 - Industrial
    ++ 20 - Alternative
    ++ 21 - Ska
    ++ 22 - Death Metal
    ++ 23 - Pranks
    ++ 24 - Soundtrack
    ++ 25 - Euro-Techno
    ++ 26 - Ambient
    ++ 27 - Trip-Hop
    ++ 28 - Vocal
    ++ 29 - Jazz+Funk
    ++ 30 - Fusion
    ++ 31 - Trance
    ++ 32 - Classical
    ++ 33 - Instrumental
    ++ 34 - Acid
    ++ 35 - House
    ++ 36 - Game
    ++ 37 - Sound Clip
    ++ 38 - Gospel
    ++ 39 - Noise
    ++ 40 - Alternative Rock
    ++ 41 - Bass
    ++ 43 - Punk
    ++ 44 - Space
    ++ 45 - Meditative
    ++ 46 - Instrumental Pop
    ++ 47 - Instrumental Rock
    ++ 48 - Ethnic
    ++ 49 - Gothic
    ++ 50 - Darkwave
    ++ 51 - Techno-Industrial
    ++ 52 - Electronic
    ++ 53 - Pop-Folk
    ++ 54 - Eurodance
    ++ 55 - Dream
    ++ 56 - Southern Rock
    ++ 57 - Comedy
    ++ 58 - Cult
    ++ 59 - Gangsta
    ++ 60 - Top 40
    ++ 61 - Christian Rap
    ++ 62 - Pop/Funk
    ++ 63 - Jungle
    ++ 64 - Native US
    ++ 65 - Cabaret
    ++ 66 - New Wave
    ++ 67 - Psychadelic
    ++ 68 - Rave
    ++ 69 - Showtunes
    ++ 70 - Trailer
    ++ 71 - Lo-Fi
    ++ 72 - Tribal
    ++ 73 - Acid Punk
    ++ 74 - Acid Jazz
    ++ 75 - Polka
    ++ 76 - Retro
    ++ 77 - Musical
    ++ 78 - Rock & Roll
    ++ 79 - Hard Rock
    ++ 80 - Folk
    ++ 81 - Folk-Rock
    ++ 82 - National Folk
    ++ 83 - Swing
    ++ 84 - Fast Fusion
    ++ 85 - Bebob
    ++ 86 - Latin
    ++ 87 - Revival
    ++ 88 - Celtic
    ++ 89 - Bluegrass
    ++ 90 - Avantgarde
    ++ 91 - Gothic Rock
    ++ 92 - Progressive Rock
    ++ 93 - Psychedelic Rock
    ++ 94 - Symphonic Rock
    ++ 95 - Slow Rock
    ++ 96 - Big Band
    ++ 97 - Chorus
    ++ 98 - Easy Listening
    ++ 99 - Acoustic     		100 - Humour
    ++ 101 - Speech
    ++ 102 - Chanson
    ++ 103 - Opera
    ++ 104 - Chamber Music
    ++ 105 - Sonata
    ++ 106 - Symphony
    ++ 107 - Booty Bass
    ++ 108 - Primus
    ++ 109 - Porn Groove
    ++ 110 - Satire
    ++ 111 - Slow Jam
    ++ 112 - Club
    ++ 113 - Tango
    ++ 114 - Samba
    ++ 115 - Folklore
    ++ 116 - Ballad
    ++ 117 - Power Ballad
    ++ 118 - Rhytmic Soul
    ++ 119 - Freestyle
    ++ 120 - Duet
    ++ 121 - Punk Rock
    ++ 122 - Drum Solo
    ++ 123 - Acapella
    ++ 124 - Euro-House
    ++ 125 - Dance Hall
    ++ 126 - Goa
    ++ 127 - Drum & Bass
    ++ 128 - Club-House
    ++ 129 - Hardcore
    ++ 130 - Terror
    ++ 131 - Indie
    ++ 132 - BritPop
    ++ 133 - Negerpunk
    ++ 134 - Polsk Punk
    ++ 135 - Beat
    ++ 136 - Christian Gangsta
    ++ 137 - Heavy Metal
    ++ 138 - Black Metal
    ++ 139 - Crossover
    ++ 140 - Contemporary C
    ++ 141 - Christian Rock
    ++ 142 - Merengue
    ++ 143 - Salsa
    ++ 144 - Thrash Metal
    ++ 145 - Anime
    ++ 146 - JPop
    ++ 147 - SynthPop
    ++


    ++
    ++
    ++

    ++ ++ +diff -r -c -N encoder/doc/html/index.html lame3.70/doc/html/index.html +*** encoder/doc/html/index.html Wed Dec 31 17:00:00 1969 +--- lame3.70/doc/html/index.html Wed Mar 29 13:49:02 2000 +*************** +*** 0 **** +--- 1,53 ---- ++ ++ ++ ++ LAME V3 Next Generation High-End MPEG Layer III Encoding ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++

    ++

    LAME ++ V3.70
    ++     ++


    ++     Lame ++ Ain't an MP3 Encoder ++


    ++     ++

    ++

    Open Source Project
    ++ http://www.sulaco.org/mp3

    ++

    ++

    ++

    ++

    ++

    ++
    ++
    ++
    ++
  • Introduction ++ ++
  • Some command line examples ++
  • Full command line switch reference ++
  • ID3 tags ++
  • History ++
  • Contributors
    ++     ++ ++ +diff -r -c -N encoder/doc/html/lame.css lame3.70/doc/html/lame.css +*** encoder/doc/html/lame.css Wed Dec 31 17:00:00 1969 +--- lame3.70/doc/html/lame.css Wed Mar 29 13:49:02 2000 +*************** +*** 0 **** +--- 1,24 ---- ++ BODY { ++ font-family: Arial, Helvetica, sans-serif; ++ position: absolute; ++ padding-left: 1cm; ++ margin-left: 1cm; ++ } ++ ++ h1, h2, h3 { ++ margin-left: -1cm; text-align: left ++ } ++ ++ TABLE { text-align: left; font-family: Arial, Helvetica, sans-serif;} ++ TH {background-color: #C0C0C0} ++ TD {font-family: Arial, Helvetica, sans-serif;} ++ ++ SMALL.TINY { font-size : xx-small } ++ SMALL.SCRIPTSIZE { font-size : xx-small } ++ SMALL.FOOTNOTESIZE { font-size : x-small } ++ SMALL.SMALL { font-size : small } ++ BIG.LARGE { font-size : large } ++ BIG.XLARGE { font-size : x-large } ++ BIG.XXLARGE { font-size : xx-large } ++ BIG.HUGE { font-size : xx-large } ++ BIG.XHUGE { font-size : xx-large } +diff -r -c -N encoder/doc/html/node5.html lame3.70/doc/html/node5.html +*** encoder/doc/html/node5.html Wed Dec 31 17:00:00 1969 +--- lame3.70/doc/html/node5.html Wed Nov 24 01:44:08 1999 +*************** +*** 0 **** +--- 1,51 ---- ++ ++ ++ ++ For the impatient ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++

    For the impatient

    ++ If you are new to audio compression, you should read section 1.1 ++ for an introduction about audio compression and MPEG Layer-3. ++

    If, however, you want to jump right into the business of sound compression, ++ then Section 1.2 will show you some prefabricated ++ command lines that will give you compressed audio streams right away. ++

    If you are an expert in audio coding already, the command line switch reference ++ page might come in handy. ++


    ++

    ++   ++ ++
    ++     ++ ++ +diff -r -c -N encoder/doc/html/node6.html lame3.70/doc/html/node6.html +*** encoder/doc/html/node6.html Wed Dec 31 17:00:00 1969 +--- lame3.70/doc/html/node6.html Wed Mar 29 13:49:02 2000 +*************** +*** 0 **** +--- 1,133 ---- ++ ++ ++ ++ Introduction ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ Subsections ++ ++ ++
    ++

    Introduction

    ++   There is a lot of confusion surrounding the terms ++  audio compression,  audio ++ encoding, and  audio decoding. This section ++ will give you an overview what audio coding (another one of these terms...) is ++ all about. ++

    ++

    The purpose of audio compression ++

    ++

    Up to the advent of audio compression, high-quality digital audio data took ++ a lot of hard disk space to store. Let us go through a short example. ++

    You want to, say, sample your favorite 1-minute song and store it on your ++ harddisk. Because you want CD quality, you sample at 44.1 kHz, stereo, ++ with 16 bits per sample. ++

    44100 Hz means that you have 44100 values per second coming in from your sound ++ card (or input file). Multiply that by two because you have two channels. Multiply ++ by another factor of two because you have two bytes per value (that's what 16 ++ bit means). The song will take up 44100 samples/s · 2 channels ++ · 2 bytes/sample · 60 s/min ~ 10 MBytes of storage space ++ on your harddisk. ++

    If you wanted to download that over the internet, given an average 56k modem ++ connected at 44k (which is a typical case), it would take you (at least) 10000000 ++ bytes · 8 bits/byte / (44000 bits/s) · / (60 s/min) ~ 30 minutes ++

    Just to download one minute of music!
    ++

    Digital audio coding, which - in this context - is synonymously called digital ++ audio compression as well, is the art of minimizing storage space (or channel ++ bandwidth) requirements for audio data. Modern perceptual audio coding techniques ++ (like MPEG Layer III) exploit the properties of the human ear (the perception ++ of sound) to achieve a size reduction by a factor of 11 with little or no perceptible ++ loss of quality. ++

    Therefore, such schemes are the key technology for high quality low bit-rate ++ applications, like soundtracks for CD-ROM games, solid-state sound memories, ++ Internet audio, digital audio broadcasting systems, and the like. ++

    ++

    The two parts of audio compression ++

    ++

    Audio compression really consists of two parts. The first part, called  encoding, ++ transforms the digital audio data that resides, say, in a WAVE file, into a ++ highly compressed form called  bitstream. To play ++ the bitstream on your soundcard, you need the second part, called  decoding. ++ Decoding takes the bitstream and re-expands it to a WAVE file. ++

    The program that effects the first part is called an audio encoder. ++ LAME is such an encoder . The program that does the second part is called ++ an audio decoder. One well-known MPEG Layer III decoder is Xmms, ++ another mpg123. Both can be found on www.mp3tech.org . ++

    Compression ratios, bitrate and quality ++

    ++

    It has not been explicitly mentioned up to now: What you end up with after ++ encoding and decoding is not the same sound file anymore: All superflous information ++ has been squeezed out, so to say. It is not the same file, but it will ++ sound the same - more or less, depending on how much compression had ++ been performed on it. ++

    Generally speaking, the lower the compression ratio achieved, the better the ++ sound quality will be in the end - and vice versa. Table 1.1 ++ gives you an overview about quality achievable. ++

    Because compression ratio is a somewhat unwieldy measure, experts use the ++ term  bitrate when speaking of the strength of ++ compression. Bitrate denotes the average number of bits that one second of audio ++ data will take up in your compressed bitstream. Usually the units used will ++ be kbps, which is kbits/s, or 1000 bits/s. ++ To calculate the number of bytes per second of audio data, simply divide the ++ number of bits per second by eight. ++


    ++

        ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++
    Table 1.1: Bitrate versus sound quality
    BitrateBandwidthQuality comparable to or better than
    8 kbps2.5 kHzPOTS (telephone sound)
    16 kbps4.5 kHzshortwave radio
    32 kbps7.5 kHzAM radio
    64 kbps11 kHzFM radio
    128 kbps16 kHzCD
    ++
    ++
    ++     ++ ++ +diff -r -c -N encoder/doc/html/node7.html lame3.70/doc/html/node7.html +*** encoder/doc/html/node7.html Wed Dec 31 17:00:00 1969 +--- lame3.70/doc/html/node7.html Sun Jan 2 22:15:41 2000 +*************** +*** 0 **** +--- 1,33 ---- ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ Some examples ++ ++ ++ ++ ++ ++ ++ ++

    Some examples

    ++ ++
      ++
    • Recommended usage: Encode a WAVE-file myfile.wav ++ to a bitrate of 128000 bits/s, writing to a plain bitstream myfile.mp3 ++ and using high quality mode:
      • ++
      ++ lame -h myfile.wav myfile.mp3
      ++
      ++
    • Encode a plain PCM stream (2-channel, 44.1 kHz) ++ to a plain 56 kBit/s Layer-3 stream, using the encoder as filter
      • ++
      ++ lame -r -s 44.1 -b 56 - - ++
    ++ ++ +diff -r -c -N encoder/doc/html/switchs.html lame3.70/doc/html/switchs.html +*** encoder/doc/html/switchs.html Wed Dec 31 17:00:00 1969 +--- lame3.70/doc/html/switchs.html Wed Mar 29 13:49:02 2000 +*************** +*** 0 **** +--- 1,521 ---- ++ ++ ++ ++ Full command line switch reference ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++

    Full command line switch reference

    ++

    ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++
    switchparameter
    -adownmix stereo file to mono
    --athonlyATH only
    -bbitrate (8...320)
    -Bmax VBR bitrate (8...320)
    -ccopyright
    -dblock type control
    -ede-emphasis (n, 5, c)
    -f fast mode
    -gturn on MP3x graphical frame analyser (may not be ++ available on your release)
    -hhigh quality
    --help help
    --highpass highpass filtering frequency in kHz
    --highpass-width width of highpass filtering in kHz
    -k full bandwidth
    --lowpass lowpass filtering frequency in kHz
    --lowpass-width width of lowpass filtering in kHz
    -mstereo mode (s, j, f, m)
    --mp3inputmp3 input file
    --noathno ATH
    --nohistdisable histogram display
    --noresdisable bit reservoir
    --noshortdisable short blocks frames
    -onon-original
    -perror protection
    --presetuse built-in preset (phone, voice, fm, tape, hifi, ++ cd, studio)
    -rinput file is raw pcm
    --resampleoutput sampling frequency in kHz
    -ssampling frequency in kHz
    -Ssilent operation
    -tdisable VBR tag
    -VVBR quality setting (0...9)
    --voicevoice encoding mode
    -xswapbytes
    ++
    ++

    ++
    * -a    downmix  ++
    Mix the stereo input file to mono and encode as mono. This option is only ++ needed in the case of raw PCM stereo input (because LAME cannot determine ++ the number of channels in the input file).
    ++ To encode a stereo PCM input file as mono, use "lame -m s -a".
    ++ For WAV and AIFF input files, using "-m m" will always produce a mono .mp3 ++ file from both mono and stereo input.
    ++ The downmix is calculated as the sum of the left and right channel, attenuated ++ by 6 dB. ++

    ++
    ++
    ++
    * --athonly    ATH ++ only
    ++
    ++
    ++
    This option causes LAME to ignore the output of the psy-model and only use ++ masking from the ATH. Might be useful at very high bitrates or for testing ++ the ATH. ++

    ++
    ++
    ++
    ++
    * -b    bitrate ++
    ++
    ++
    ++
    For MPEG1 (sampling frequencies of 32, 44.1 and 48kHz)
    ++ n = 32,40,48,56,64,80,96,112,128,160,192,224,256,320
    ++
    ++ For MPEG2 (sampling frequencies of 16, 22.05 and 24kHz)
    ++ n = 8,16,24,32,40,48,56,64,80,96,112,128,144,160
    ++
    ++ Default is 128kbs for MPEG1 and 80kbs for MPEG2.
    ++
    ++ When used with variable bitrate encoding (VBR), -b specifies the minimum bitrate ++ to be used. ++

    ++
    ++
    ++
    ++
    * -B    maximum VBR ++ bitrate 
    ++
    ++
    ++
    For MPEG1 (sampling frequencies of 32, 44.1 and 48kHz)
    ++ n = 32,40,48,56,64,80,96,112,128,160,192,224,256,320
    ++
    ++ For MPEG2 (sampling frequencies of 16, 22.05 and 24kHz)
    ++ n = 8,16,24,32,40,48,56,64,80,96,112,128,144,160
    ++
    ++ Specifies the maximum allowed bitrate when using VBR.
    ++
    ++ note: If you own an mp3 hardware player build upon a MAS 3503 chip, you ++ must set maximum bitrate to no more than 224 kpbs.
    ++
    ++
    ++
    * -c    copyright ++
    ++
    ++
    ++
    Mark the encoded file as being copyrighted. ++

    ++
    ++
    * -d    block type control ++
    ++
    ++
    ++
    Allows the left and right channels to use different block size types. ++

    ++
    ++ * -e n/5/c    de-emphasis ++
    ++
    ++
    ++

    ++ n = (none, default)
    ++ 5 = 0/15 microseconds
    ++ c = citt j.17
    ++
    ++ All this does is set a flag in the bitstream. If you have a PCM input file ++ where one of the above types of (obsolete) emphasis has been applied, you ++ can set this flag in LAME. Then the mp3 decoder should de-emphasize the output ++ during playback, although most decoders ignore this flag.
    ++ A better solution would be to apply the de-emphasis with a standalone utility ++ before encoding. ++

    ++
    ++ * -f    fast mode ++
    ++
    ++
    ++
    This switch forces the encoder to use the fast encoding mode. It will reduce ++ a lot the quality of the output. This might be useful for real-time processing ++ or streaming on slowest computers. ++

    ++
    ++ * -h    high quality ++
    ++
    ++
    ++
    Enables specialized mid/side masking thresholds to be used in jstereo mode ++ and also enables a more accurate but slightly slower quantization formula. ++ It will sound better but is about 20% slower.
    ++ Recommended setting.
    ++
    ++ In the future there may be more quality settings, ranging from 0 (highest) ++ to 9 (lowest) but right now there are just 3: 2,5 or 9. They are enabled with:
    ++
    ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++
    -f 'fast mode'quality 9
    defaultquality 5
    -h 'high quality'quality 2
    ++ ++

    ++
    ++ * --help    help ++
    ++
    ++
    ++
    Display a list of all available options. ++

    ++
    ++ * --highpass    highpass ++ filtering frequency in kHz
    ++
    ++
    ++
    Set an highpass filtering frequency. Frequencies below the specified one ++ will be cutoff. ++

    ++
    ++ * --highpass-width    width ++ of highpass filtering in kHz
    ++
    ++
    ++
    Set the width of the highpass filter. The default value is 15% of highpass ++ frequency. ++

    ++
    ++ * -k    full bandwidth ++
    ++
    ++
    ++
    Tells the encoder to use full bandwidth. By default, the encoder uses some ++ highpass filtering at low bitrates, in order to keep a good quality. Increasing ++ the bandwidth from the default setting might produces ringing artefacts at ++ low bitrates. Use with care! ++

    ++
    ++ * --lowpass    lowpass ++ filtering frequency in kHz
    ++
    ++
    ++
    Set a lowpass filtering frequency. Frequencies above the specified one ++ will be cutoff. ++

    ++
    ++ * --lowpass-width    width ++ of lowpass filtering in kHz
    ++
    ++
    ++
    Set the width of the lowpass filter. The default value is 15% of lowpass ++ frequency. ++

    ++
    ++ * -m s/j/f/m    stereo ++ mode
    ++
    ++
    ++
    Joint-stereo is the default mode for stereo files with VBR or fixed bitrates ++ of 160kbs or less. At higher fixed bitrates, the default is stereo.
    ++
    ++ stereo
    ++ In this mode, the encoder makes no use of potentially existing correlations ++ between the two input channels. It can, however, negotiate the bit demand ++ between both channel, i.e. give one channel more bits if the other contains ++ silence.
    ++
    ++ joint stereo
    ++ In this mode, the encoder will make use of a correlation between both channels. ++ The signal will be matrixed into a sum ("mid") and difference ("side") signal. ++ For quasi-mono signals, this will give a significant gain in encoding quality.
    ++ This mode does not destroy phase information like IS stereo that may be used ++ by other encoders and thus can be used to encode DOLBY ProLogic surround signals. ++
    ++
    ++ forced joint stereo
    ++ This mode will force MS joint stereo on all frames. It's slightly faster than ++ joint stereo, but it should be used only if you are sure that every frame ++ of the input file has very little stereo separation.
    ++
    ++ mono
    ++ The input will be encoded as a mono signal. If it was a stereo signal, it ++ will be downsampled to mono. The downmix is calculated as the sum of the left ++ and right channel, attenuated by 6 dB. ++

    ++
    ++
    ++
    * --mp3input    mp3 ++ input file
    ++
    ++
    ++
    Assume the input file is a MP3 file. Usefull for downsampling from one ++ mp3 to another. As an example, it can be usefull for streaming through an ++ IceCast server.
    ++ If the filename ends in ".mp3" LAME will assume it is an MP3. For stdin or ++ MP3 files which dont end in .mp3 you need to use this switch. ++

    ++
    ++
    ++
    ++
    * --noath    no ++ ATH
    ++
    ++
    ++
    Disable any use of the ATH for masking. ++

    ++
    ++
    ++
    ++
    * --nohist    disable ++ histogram display
    ++
    ++
    ++
    By default, LAME will display a bitrate histogram while producing VBR mp3 ++ files. This will disable that feature.
    ++ Histogram display might not be available on your release. ++

    ++
    ++
    ++
    ++
    * --nores    disable ++ bit reservoir
    ++
    ++
    ++
    Disable the bit reservoir. Each frame will then become independent from ++ previous ones, but the quality will be lower. ++

    ++
    ++
    ++
    ++
    * --noshort    disable ++ short blocks frames
    ++
    ++
    ++
    Encode all frames using long blocks only. This could increase quality when ++ encoding at very low bitrates, but can produce serious pre-echo artefacts. ++

    ++
    ++
    ++
    ++
    * -o    non-original ++
    ++
    ++
    ++
    Mark the encoded file as a copy. ++

    ++
    ++ * -p    error protection
    ++
    ++
    ++
    Turn on CRC error protection. ++

    ++
    ++ * --preset phone/voice/fm/tape/hifi/cd/studio     use ++ built-in preset
    ++
    ++
    ++
    Use one of the built-in presets. ++

    ++
    ++ * -r    input file is raw ++ pcm
    ++
    ++
    ++
    Assume the input file is raw pcm. Sampling rate and mono/stereo/jstereo ++ must be specified on the command line. Without -r, LAME will perform several ++ fseek()'s on the input file looking for WAV and AIFF headers.
    ++ Might not be available on your release. ++

    ++
    ++
    ++
    * --resample    output ++ sampling frequency in kHz
    ++
    ++
    ++
    n = 16, 22.05, 24, 32, 44.1, 48
    ++
    ++ Resample the input to the specified frequency. If not specified, LAME will ++ automatically resample the input when using high compression ratios. ++

    ++
    ++
    ++
    ++
    * -s 16/22.05/24/32/44.1/48    sampling ++ frequency
    ++
    ++
    ++
    Required only for raw PCM input files. Otherwise it will be determined ++ from the header. ++

    ++
    ++ * -S    silent operation ++
    ++
    ++
    ++
    Don't print progress report. ++

    ++
    ++ * -t    disable VBR tag
    ++
    ++
    ++
    Disable writing of the VBR Tag (only valid in VBR mode).
    ++ This tag in embedded in frame 0 of the MP3 file. It lets VBR aware players ++ correctly seek and compute playing times of VBR files. ++

    ++
    ++
    ++
    * -V 0...9    VBR quality ++ setting
    ++
    ++
    ++
    Enable VBR (Variable BitRate) and specifies the value of VBR quality.
    ++ default=4
    ++ 0=highest quality. ++

    ++
    ++
    ++
    ++
    * --voice    voice ++ encoding mode
    ++
    ++
    ++
    Use voice encoding mode. In the case of voice encoding, it will increase ++ the quality a low bitrates, or reduce the file size when using VBR.
    ++ However, be carefull to only use it on 44.1kHz files. ++

    ++
    ++
    ++
    ++
    * -x    swapbytes ++
    ++
    ++
    ++
    Swap bytes in the input file. For sorting out little endian/big endian ++ type problems. If your encodings sounds like static, try this first. ++
    ++ ++ +diff -r -c -N encoder/doc/man/CVS/Entries lame3.70/doc/man/CVS/Entries +*** encoder/doc/man/CVS/Entries Wed Dec 31 17:00:00 1969 +--- lame3.70/doc/man/CVS/Entries Thu Apr 6 12:50:53 2000 +*************** +*** 0 **** +--- 1,2 ---- ++ /lame.1/1.1.1.1/Wed Nov 24 08:44:31 1999//Tlame3_70 ++ D +diff -r -c -N encoder/doc/man/CVS/Repository lame3.70/doc/man/CVS/Repository +*** encoder/doc/man/CVS/Repository Wed Dec 31 17:00:00 1969 +--- lame3.70/doc/man/CVS/Repository Thu Apr 6 12:50:36 2000 +*************** +*** 0 **** +--- 1 ---- ++ lame/doc/man +diff -r -c -N encoder/doc/man/CVS/Root lame3.70/doc/man/CVS/Root +*** encoder/doc/man/CVS/Root Wed Dec 31 17:00:00 1969 +--- lame3.70/doc/man/CVS/Root Thu Apr 6 12:50:36 2000 +*************** +*** 0 **** +--- 1 ---- ++ markt@cvs.lame.sourceforge.net:/cvsroot/lame +diff -r -c -N encoder/doc/man/CVS/Tag lame3.70/doc/man/CVS/Tag +*** encoder/doc/man/CVS/Tag Wed Dec 31 17:00:00 1969 +--- lame3.70/doc/man/CVS/Tag Thu Apr 6 12:50:53 2000 +*************** +*** 0 **** +--- 1 ---- ++ Tlame3_70 +diff -r -c -N encoder/doc/man/lame.1 lame3.70/doc/man/lame.1 +*** encoder/doc/man/lame.1 Wed Dec 31 17:00:00 1969 +--- lame3.70/doc/man/lame.1 Wed Nov 24 01:44:31 1999 +*************** +*** 0 **** +--- 1,353 ---- ++ .TH lame 1 "November 8, 1999" " " "LAME audio compressor MP3 " ++ .SH NAME ++ lame \- create mp3 audio files ++ .SH SYNOPSIS ++ .nf ++ lame [options] ++ .fi ++ .SH DESCRIPTION ++ .PP ++ Lame is a program which can be used to create compressed audio files. ++ (Lame aint MP3 encoder). These audio files can be played back by popular ++ mp3 players such as mpg123. To read from stdin, use "-" for . ++ To write to stdout, use a "-" for . ++ ++ .SH OPTIONS ++ .TP ++ .BI -m " mode" ++ (s)tereo, (j)oint, (f)orce or (m)ono (default j) ++ force = force mid/side stereo on all frames. Faster if you know mid/side ++ stereo is appropriate for the input file. ++ .TP ++ .BI -b " bitrate" ++ set the bitrate, default 128kbps ++ .TP ++ .BI -s " sample-freq" ++ frequency of input file (khz) - default 44.1 ++ .TP ++ .BI --resamp " sfreq" ++ sampling frequency of output file(kHz)- default=input sfreq ++ .TP ++ .BI --mp3input ++ input file is a MP3 file ++ .TP ++ .BI --voice ++ experimental voice mode ++ .TP ++ .BI -v ++ use variable bitrate (VBR) ++ .TP ++ .BI -V " n" ++ quality setting for VBR. Default n=4, 0=high quality,bigger files. ++ 9=smaller files ++ .TP ++ .BI -b " " ++ specify minimum allowed bitrate for VBR, default 32kbs ++ .TP ++ .BI -B " " ++ specify miximum allowed bitrate for VBR, default 256kbs ++ .TP ++ .BI -t ++ disable Xing VBR informational tag ++ .TP ++ .BI --nohist ++ disable VBR histogram display ++ ++ .TP ++ .BI -h ++ slower, but (maybe) higher quality ++ .TP ++ .BI -f ++ fast mode (low quality) ++ .TP ++ .BI -k ++ disable sfb=21 cutoff ++ .TP ++ .BI -d ++ allow channels to have different blocktypes ++ .TP ++ .BI --athonly ++ only use the ATH for masking ++ .TP ++ .BI -r ++ input is raw pcm ++ .TP ++ .BI -x ++ force byte-swapping of input ++ .TP ++ .BI -a ++ autoconvert from stereo to mono file for mono encoding ++ .TP ++ .BI -e " emp" ++ de-emphasis n/5/c (default: n) ++ .TP ++ .BI -p ++ add CRC error protection (adds additional 16bits per frame to stream) ++ .TP ++ .BI -c ++ mark as copyright ++ .TP ++ .BI -o ++ mark as non-original (i.e. do not set the original bit) ++ .TP ++ .BI -S ++ don't print progress report, VBR histograms ++ .TP ++ .BI -g ++ run graphical analysis on . can also be a .mp3 file. ++ (This feature is a compile time option. Your binary may for speed reasons be ++ compiled without this.) ++ ++ ++ ++ .SH EXAMPLES ++ .LP ++ Recommended way to create a fixed 128kbs mp3 file from a wav file ++ .IP ++ .B lame -h ++ .I foo.wav foo.mp3 ++ .LP ++ Copy track 7 from a regular audio CD, using cdparanoia ++ ++ .IP ++ .B cdparanoia 7 foo.wav ; ++ .B lame -h foo.wav foo.mp3 ++ ++ .LP ++ Copy track 7 from a regular audio CD with cdda2wav, where the cdrom device is ++ device /dev/cdrom to an mp3 file foo.mp3 ++ ++ .IP ++ .B cdda2wav \-O ++ .I wav ++ .B \-t 7 \-D ++ .I /dev/cdrom ++ .I \- ++ .B | lame ++ .I \- foo.mp3 ++ ++ .IP ++ ++ .LP ++ Streaming from mono 22.05kHz raw pcm data and using sox to ++ downsample to 16kHz: ++ .IP ++ cat inputfile | sox -t raw -x -w -s -c 1 -r 22050 - ++ -t raw -x -w -s -c 1 -r 16000 - | lame -r -m m -b 24 -s 16 - - > output ++ .IP ++ ++ ++ ++ .SH BUGS ++ .PP ++ Quality is much better than encoders based on the ISO routines, ++ but still not as good as the FhG encoder. ++ .SH SEE ALSO ++ .BR mpg123 (1) , ++ .BR cdparanoia (1) , ++ .BR cdda2wav (1) , ++ .BR sox (1) ++ .SH AUTHORS ++ .nf ++ LAME originally developed by Mike Cheng ++ (www.uq.net.au/~zzmcheng) ++ GPSYCHO psycho-acoustic model by Mark Taylor ++ (www.sulaco.org/mp3). ++ mpglib by Michael Hipp ++ Manual page by William Schelter, Nils Faerber ++ .f ++  -=- MIME -=-  ++ ++ --jI8keyz6grp/JLjh ++ Content-Type: text/plain; charset=us-ascii ++ ++ Hi Mark! ++ I have just updated the man page for lame 3.53beta to the latest options ++ with some minor cosmetic changes. Find the new version as attachment... ++ CU ++ nils ++ ++ -- ++ Nils Faerber (Linux Nils) eMail: nils@unix-ag.org ++ Student of computer science http://www.si.unix-ag.org/~nils/ ++ Unix user group, University of Siegen, Germany ++ ++ Siegen ... the arctic rain forest! ++ -- ++ --jI8keyz6grp/JLjh ++ Content-Type: text/plain; charset=us-ascii ++ Content-Disposition: attachment; filename="lame.1" ++ ++ .TH lame 1 "November 8, 1999" " " "LAME audio compressor MP3 " ++ .SH NAME ++ lame \- create mp3 audio files ++ .SH SYNOPSIS ++ .nf ++ lame [options] ++ .fi ++ .SH DESCRIPTION ++ .PP ++ Lame is a program which can be used to create compressed audio files. ++ (Lame aint MP3 encoder). These audio files can be played back by popular ++ mp3 players such as mpg123. To read from stdin, use "-" for . ++ To write to stdout, use a "-" for . ++ ++ .SH OPTIONS ++ .TP ++ .BI -m " mode" ++ (s)tereo, (j)oint, (f)orce or (m)ono (default j) ++ force = force mid/side stereo on all frames. Faster if you know mid/side ++ stereo is appropriate for the input file. ++ .TP ++ .BI -b " bitrate" ++ set the bitrate, default 128kbps ++ .TP ++ .BI -s " sample-freq" ++ frequency of input file (khz) - default 44.1 ++ .TP ++ .BI --resamp " sfreq" ++ sampling frequency of output file(kHz)- default=input sfreq ++ .TP ++ .BI --mp3input ++ input file is a MP3 file ++ .TP ++ .BI --voice ++ experimental voice mode ++ .TP ++ .BI -v ++ use variable bitrate (VBR) ++ .TP ++ .BI -V " n" ++ quality setting for VBR. Default n=4, 0=high quality,bigger files. ++ 9=smaller files ++ .TP ++ .BI -b " " ++ specify minimum allowed bitrate for VBR, default 32kbs ++ .TP ++ .BI -B " " ++ specify miximum allowed bitrate for VBR, default 256kbs ++ .TP ++ .BI -t ++ disable Xing VBR informational tag ++ .TP ++ .BI --nohist ++ disable VBR histogram display ++ ++ .TP ++ .BI -h ++ slower, but (maybe) higher quality ++ .TP ++ .BI -f ++ fast mode (low quality) ++ .TP ++ .BI -k ++ disable sfb=21 cutoff ++ .TP ++ .BI -d ++ allow channels to have different blocktypes ++ .TP ++ .BI --athonly ++ only use the ATH for masking ++ .TP ++ .BI -r ++ input is raw pcm ++ .TP ++ .BI -x ++ force byte-swapping of input ++ .TP ++ .BI -a ++ autoconvert from stereo to mono file for mono encoding ++ .TP ++ .BI -e " emp" ++ de-emphasis n/5/c (default: n) ++ .TP ++ .BI -p ++ add CRC error protection (adds additional 16bits per frame to stream) ++ .TP ++ .BI -c ++ mark as copyright ++ .TP ++ .BI -o ++ mark as non-original (i.e. do not set the original bit) ++ .TP ++ .BI -S ++ don't print progress report, VBR histograms ++ .TP ++ .BI -g ++ run graphical analysis on . can also be a .mp3 file. ++ (This feature is a compile time option. Your binary may for speed reasons be ++ compiled without this.) ++ ++ ++ ++ .SH EXAMPLES ++ .LP ++ Recommended way to create a fixed 128kbs mp3 file from a wav file ++ .IP ++ .B lame -h ++ .I foo.wav foo.mp3 ++ .LP ++ Copy track 7 from a regular audio CD, using cdparanoia ++ ++ .IP ++ .B cdparanoia 7 foo.wav ; ++ .B lame -h foo.wav foo.mp3 ++ ++ .LP ++ Copy track 7 from a regular audio CD with cdda2wav, where the cdrom device is ++ device /dev/cdrom to an mp3 file foo.mp3 ++ ++ .IP ++ .B cdda2wav \-O ++ .I wav ++ .B \-t 7 \-D ++ .I /dev/cdrom ++ .I \- ++ .B | lame ++ .I \- foo.mp3 ++ ++ .IP ++ ++ .LP ++ Streaming from mono 22.05kHz raw pcm data and using sox to ++ downsample to 16kHz: ++ .IP ++ cat inputfile | sox -t raw -x -w -s -c 1 -r 22050 - ++ -t raw -x -w -s -c 1 -r 16000 - | lame -r -m m -b 24 -s 16 - - > output ++ .IP ++ ++ ++ ++ .SH BUGS ++ .PP ++ Quality is much better than encoders based on the ISO routines, ++ but still not as good as the FhG encoder. ++ .SH SEE ALSO ++ .BR mpg123 (1) , ++ .BR cdparanoia (1) , ++ .BR cdda2wav (1) , ++ .BR sox (1) ++ .SH AUTHORS ++ .nf ++ LAME originally developed by Mike Cheng ++ (www.uq.net.au/~zzmcheng) ++ GPSYCHO psycho-acoustic model by Mark Taylor ++ (www.sulaco.org/mp3). ++ mpglib by Michael Hipp ++ Manual page by William Schelter, Nils Faerber ++ .f ++ ++ --jI8keyz6grp/JLjh-- ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ +diff -r -c -N encoder/encode.c lame3.70/encode.c +*** encoder/encode.c Wed Jan 22 02:43:14 1997 +--- lame3.70/encode.c Wed Dec 31 17:00:00 1969 +*************** +*** 1,1456 **** +- /********************************************************************** +- * ISO MPEG Audio Subgroup Software Simulation Group (1996) +- * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- * +- * $Id: encode.c,v 1.1 1996/02/14 04:04:23 rowlands Exp $ +- * +- * $Log: encode.c,v $ +- * Revision 1.1 1996/02/14 04:04:23 rowlands +- * Initial revision +- * +- * Received from Mike Coleman +- **********************************************************************/ +- /********************************************************************** +- * date programmers comment * +- * 3/01/91 Douglas Wong, start of version 1.1 records * +- * Davis Pan * +- * 3/06/91 Douglas Wong rename: setup.h to endef.h * +- * efilter to enfilter * +- * ewindow to enwindow * +- * integrated "quantizer", "scalefactor",* +- * and "transmission" files * +- * update routine "window_subband" * +- * 3/31/91 Bill Aspromonte replaced read_filter by * +- * create_an_filter * +- * 5/10/91 W. Joseph Carter Ported to Macintosh and Unix. * +- * Incorporated Jean-Georges Fritsch's * +- * "bitstream.c" package. * +- * Incorporated Bill Aspromonte's * +- * filterbank coefficient matrix * +- * calculation routines and added * +- * roundoff to coincide with specs. * +- * Modified to strictly adhere to * +- * encoded bitstream specs, including * +- * "Berlin changes". * +- * Modified PCM sound file handling to * +- * process all incoming samples and fill * +- * out last encoded frame with zeros * +- * (silence) if needed. * +- * Located and fixed numerous software * +- * bugs and table data errors. * +- * 19jun91 dpwe (Aware) moved "alloc_*" reader to common.c * +- * Globals sblimit, alloc replaced by new* +- * struct 'frame_params' passed as arg. * +- * Added JOINT STEREO coding, layers I,II* +- * Affects: *_bit_allocation, * +- * subband_quantization, encode_bit_alloc* +- * sample_encoding * +- * 6/10/91 Earle Jennings modified II_subband_quantization to * +- * resolve type cast problem for MS_DOS * +- * 6/11/91 Earle Jennings modified to avoid overflow on MS_DOS * +- * in routine filter_subband * +- * 7/10/91 Earle Jennings port to MsDos from MacIntosh version * +- * 8/ 8/91 Jens Spille Change for MS-C6.00 * +- *10/ 1/91 S.I. Sudharsanan, Ported to IBM AIX platform. * +- * Don H. Lee, * +- * Peter W. Farrett * +- *10/ 3/91 Don H. Lee implemented CRC-16 error protection * +- * newly introduced function encode_CRC * +- *11/ 8/91 Kathy Wang Documentation of code * +- * All variablenames are referred to * +- * with surrounding pound (#) signs * +- * 2/11/92 W. Joseph Carter Ported new code to Macintosh. Most * +- * important fixes involved changing * +- * 16-bit ints to long or unsigned in * +- * bit alloc routines for quant of 65535 * +- * and passing proper function args. * +- * Removed "Other Joint Stereo" option * +- * and made bitrate be total channel * +- * bitrate, irrespective of the mode. * +- * Fixed many small bugs & reorganized. * +- * 6/16/92 Shaun Astarabadi Changed I_scale_factor_calc() and * +- * II_scale_factor_calc() to use scale * +- * factor 0 thru 62 only and not to * +- * encode index 63 into the bit stream. * +- * 7/27/92 Mike Li (re-)Port to MS-DOS * +- * 9/22/92 jddevine@aware.com Fixed _scale_factor_calc() defs * +- * 3/31/93 Giogio Dimino changed II_a_bit_allocation() from: * +- * if( ad > ...) to if(ad >= ...) * +- * 8/05/93 TEST changed I_a_bit_allocation() from: * +- * if( ad > ...) to if(ad >= ...) * +- * 8/02/95 mc@fivebats.com Changed audio file reading code to * +- * read samples big-endian * +- *10/15/95 mc@fivebats.com Modified get_audio() for layer3-LSF * +- **********************************************************************/ +- +- #include "common.h" +- #include "encoder.h" +- +- #ifdef MS_DOS +- extern unsigned _stklen = 16384; +- #endif +- +- +- /*=======================================================================\ +- | | +- | This segment contains all the core routines of the encoder, | +- | except for the psychoacoustic models. | +- | | +- | The user can select either one of the two psychoacoustic | +- | models. Model I is a simple tonal and noise masking threshold | +- | generator, and Model II is a more sophisticated cochlear masking | +- | threshold generator. Model I is recommended for lower complexity | +- | applications whereas Model II gives better subjective quality at low | +- | bit rates. | +- | | +- | Layers I and II of mono, stereo, and joint stereo modes are supported.| +- | Routines associated with a given layer are prefixed by "I_" for layer | +- | 1 and "II_" for layer 2. | +- \=======================================================================*/ +- +- /************************************************************************ +- * +- * read_samples() +- * +- * PURPOSE: reads the PCM samples from a file to the buffer +- * +- * SEMANTICS: +- * Reads #samples_read# number of shorts from #musicin# filepointer +- * into #sample_buffer[]#. Returns the number of samples read. +- * +- ************************************************************************/ +- +- unsigned long read_samples(musicin, sample_buffer, num_samples, frame_size) +- FILE *musicin; +- short sample_buffer[2304]; +- unsigned long num_samples, frame_size; +- { +- unsigned long samples_read; +- static unsigned long samples_to_read; +- static char init = TRUE; +- +- if (init) { +- samples_to_read = num_samples; +- init = FALSE; +- } +- if (samples_to_read >= frame_size) +- samples_read = frame_size; +- else +- samples_read = samples_to_read; +- if ((samples_read = +- fread(sample_buffer, sizeof(short), (int)samples_read, musicin)) == 0) +- printf("Hit end of audio data\n"); +- /* +- Samples are big-endian. If this is a little-endian machine +- we must swap +- */ +- if ( NativeByteOrder == order_unknown ) +- { +- NativeByteOrder = DetermineByteOrder(); +- if ( NativeByteOrder == order_unknown ) +- { +- fprintf( stderr, "byte order not determined\n" ); +- exit( 1 ); +- } +- } +- if ( NativeByteOrder == order_littleEndian ) +- SwapBytesInWords( sample_buffer, samples_read ); +- +- samples_to_read -= samples_read; +- if (samples_read < frame_size && samples_read > 0) { +- printf("Insufficient PCM input for one frame - fillout with zeros\n"); +- for (; samples_read < frame_size; sample_buffer[samples_read++] = 0); +- samples_to_read = 0; +- } +- return(samples_read); +- } +- +- /************************************************************************ +- * +- * get_audio() +- * +- * PURPOSE: reads a frame of audio data from a file to the buffer, +- * aligns the data for future processing, and separates the +- * left and right channels +- * +- * +- ************************************************************************/ +- +- unsigned long get_audio( musicin, buffer, num_samples, stereo, info ) +- FILE *musicin; +- short FAR buffer[2][1152]; +- unsigned long num_samples; +- int stereo; +- layer *info; +- { +- int j; +- short insamp[2304]; +- unsigned long samples_read; +- int lay; +- lay = info->lay; +- +- if ( (lay == 3) && (info->version == 0) ) +- { +- if ( stereo == 2 ) +- { +- samples_read = read_samples( musicin, insamp, num_samples, +- (unsigned long) 1152 ); +- for ( j = 0; j < 576; j++ ) +- { +- buffer[0][j] = insamp[2 * j]; +- buffer[1][j] = insamp[2 * j + 1]; +- } +- } +- else +- { +- samples_read = read_samples( musicin, insamp, num_samples, +- (unsigned long) 576 ); +- for ( j = 0; j < 576; j++ ) +- { +- buffer[0][j] = insamp[j]; +- buffer[1][j] = 0; +- } +- } +- } +- else +- { +- if (lay == 1){ +- if(stereo == 2){ /* layer 1, stereo */ +- samples_read = read_samples(musicin, insamp, num_samples, +- (unsigned long) 768); +- for(j=0;j<448;j++) { +- if(j<64) { +- buffer[0][j] = buffer[0][j+384]; +- buffer[1][j] = buffer[1][j+384]; +- } +- else { +- buffer[0][j] = insamp[2*j-128]; +- buffer[1][j] = insamp[2*j-127]; +- } +- } +- } +- else { /* layer 1, mono */ +- samples_read = read_samples(musicin, insamp, num_samples, +- (unsigned long) 384); +- for(j=0;j<448;j++){ +- if(j<64) { +- buffer[0][j] = buffer[0][j+384]; +- buffer[1][j] = 0; +- } +- else { +- buffer[0][j] = insamp[j-64]; +- buffer[1][j] = 0; +- } +- } +- } +- } +- else { +- if(stereo == 2){ /* layer 2 (or 3), stereo */ +- samples_read = read_samples(musicin, insamp, num_samples, +- (unsigned long) 2304); +- for(j=0;j<1152;j++) { +- buffer[0][j] = insamp[2*j]; +- buffer[1][j] = insamp[2*j+1]; +- } +- } +- else { /* layer 2 (or 3), mono */ +- samples_read = read_samples(musicin, insamp, num_samples, +- (unsigned long) 1152); +- for(j=0;j<1152;j++){ +- buffer[0][j] = insamp[j]; +- buffer[1][j] = 0; +- } +- } +- } +- } +- return(samples_read); +- } +- +- /************************************************************************ +- * +- * read_ana_window() +- * +- * PURPOSE: Reads encoder window file "enwindow" into array #ana_win# +- * +- ************************************************************************/ +- +- void read_ana_window(ana_win) +- double FAR ana_win[HAN_SIZE]; +- { +- int i,j[4]; +- FILE *fp; +- double f[4]; +- char t[150]; +- +- if (!(fp = OpenTableFile("enwindow") ) ) { +- printf("Please check analysis window table 'enwindow'\n"); +- exit(1); +- } +- for (i=0;i<512;i+=4) { +- fgets(t, 150, fp); +- sscanf(t,"C[%d] = %lf C[%d] = %lf C[%d] = %lf C[%d] = %lf\n", +- j, f,j+1,f+1,j+2,f+2,j+3,f+3); +- if (i==j[0]) { +- ana_win[i] = f[0]; +- ana_win[i+1] = f[1]; +- ana_win[i+2] = f[2]; +- ana_win[i+3] = f[3]; +- } +- else { +- printf("Check index in analysis window table\n"); +- exit(1); +- } +- fgets(t,150,fp); +- } +- fclose(fp); +- } +- +- /************************************************************************ +- * +- * window_subband() +- * +- * PURPOSE: Overlapping window on PCM samples +- * +- * SEMANTICS: +- * 32 16-bit pcm samples are scaled to fractional 2's complement and +- * concatenated to the end of the window buffer #x#. The updated window +- * buffer #x# is then windowed by the analysis window #c# to produce the +- * windowed sample #z# +- * +- ************************************************************************/ +- +- void window_subband(buffer, z, k) +- short FAR **buffer; +- double FAR z[HAN_SIZE]; +- int k; +- { +- typedef double FAR XX[2][HAN_SIZE]; +- static XX FAR *x; +- int i, j; +- static off[2] = {0,0}; +- static char init = 0; +- static double FAR *c; +- if (!init) { +- c = (double FAR *) mem_alloc(sizeof(double) * HAN_SIZE, "window"); +- read_ana_window(c); +- x = (XX FAR *) mem_alloc(sizeof(XX),"x"); +- for (i=0;i<2;i++) +- for (j=0;j= 0) +- modf(filter[i][k]+0.5, &filter[i][k]); +- else +- modf(filter[i][k]-0.5, &filter[i][k]); +- filter[i][k] *= 1e-9; +- } +- } +- +- /************************************************************************ +- * +- * filter_subband() +- * +- * PURPOSE: Calculates the analysis filter bank coefficients +- * +- * SEMANTICS: +- * The windowed samples #z# is filtered by the digital filter matrix #m# +- * to produce the subband samples #s#. This done by first selectively +- * picking out values from the windowed samples, and then multiplying +- * them by the filter matrix, producing 32 subband samples. +- * +- ************************************************************************/ +- +- void filter_subband(z,s) +- double FAR z[HAN_SIZE], s[SBLIMIT]; +- { +- double y[64]; +- int i,j; +- static char init = 0; +- typedef double MM[SBLIMIT][64]; +- static MM FAR *m; +- #ifdef MS_DOS +- long SIZE_OF_MM; +- SIZE_OF_MM = SBLIMIT*64; +- SIZE_OF_MM *= 8; +- if (!init) { +- m = (MM FAR *) mem_alloc(SIZE_OF_MM, "filter"); +- create_ana_filter(*m); +- init = 1; +- } +- #else +- if (!init) { +- m = (MM FAR *) mem_alloc(sizeof(MM), "filter"); +- create_ana_filter(*m); +- init = 1; +- } +- #endif +- for (i=0;i<64;i++) for (j=0, y[i] = 0;j<8;j++) y[i] += z[i+64*j]; +- for (i=0;iheader; +- +- putbits(bs,0xfff,12); /* syncword 12 bits */ +- put1bit(bs,info->version); /* ID 1 bit */ +- putbits(bs,4-info->lay,2); /* layer 2 bits */ +- put1bit(bs,!info->error_protection); /* bit set => no err prot */ +- putbits(bs,info->bitrate_index,4); +- putbits(bs,info->sampling_frequency,2); +- put1bit(bs,info->padding); +- put1bit(bs,info->extension); /* private_bit */ +- putbits(bs,info->mode,2); +- putbits(bs,info->mode_ext,2); +- put1bit(bs,info->copyright); +- put1bit(bs,info->original); +- putbits(bs,info->emphasis,2); +- } +- +- /************************************************************************ +- * +- * mod() +- * +- * PURPOSE: Returns the absolute value of its argument +- * +- ************************************************************************/ +- +- double mod(a) +- double a; +- { +- return (a > 0) ? a : -a; +- } +- +- /************************************************************************ +- * +- * I_combine_LR (Layer I) +- * II_combine_LR (Layer II) +- * +- * PURPOSE:Combines left and right channels into a mono channel +- * +- * SEMANTICS: The average of left and right subband samples is put into +- * #joint_sample# +- * +- * Layer I and II differ in frame length and # subbands used +- * +- ************************************************************************/ +- +- void I_combine_LR(sb_sample, joint_sample) +- double FAR sb_sample[2][3][SCALE_BLOCK][SBLIMIT]; +- double FAR joint_sample[3][SCALE_BLOCK][SBLIMIT]; +- { /* make a filtered mono for joint stereo */ +- int sb, smp; +- +- for(sb = 0; sb s[i]) +- s[i] = mod(sb_sample[k][0][j][i]); +- +- for (i=0;i=0;j--) /* $A 6/16/92 */ +- if (s[i] <= multiple[j]) { +- scalar[k][0][i] = j; +- break; +- } +- } +- } +- +- /******************************** Layer II ******************************/ +- +- void II_scale_factor_calc(sb_sample,scalar,stereo,sblimit) +- double FAR sb_sample[][3][SCALE_BLOCK][SBLIMIT]; +- unsigned int scalar[][3][SBLIMIT]; +- int stereo,sblimit; +- { +- int i,j, k,t; +- double s[SBLIMIT]; +- +- for (k=0;k s[i]) +- s[i] = mod(sb_sample[k][t][j][i]); +- +- for (i=0;i=0;j--) /* $A 6/16/92 */ +- if (s[i] <= multiple[j]) { +- scalar[k][t][i] = j; +- break; +- } +- for (i=sblimit;istereo; +- int sblimit = fr_ps->sblimit; +- +- for (k=0;k scalar[k][j][i]) max = scalar[k][j][i]; +- for (i=sblimit;istereo; +- int sblimit = fr_ps->sblimit; +- +- for (k=0;kstereo; +- int sblimit = fr_ps->sblimit; +- int dscf[2]; +- int class[2],i,j,k; +- static int pattern[5][5] = {0x123, 0x122, 0x122, 0x133, 0x123, +- 0x113, 0x111, 0x111, 0x444, 0x113, +- 0x111, 0x111, 0x111, 0x333, 0x113, +- 0x222, 0x222, 0x222, 0x333, 0x123, +- 0x123, 0x122, 0x122, 0x133, 0x123}; +- +- for (k=0;k -3 && dscf[j] <0) class[j] = 1; +- else if (dscf[j] == 0) class[j] = 2; +- else if (dscf[j] > 0 && dscf[j] < 3) class[j] = 3; +- else class[j] = 4; +- } +- switch (pattern[class[0]][class[1]]) { +- case 0x123 : scfsi[k][i] = 0; +- break; +- case 0x122 : scfsi[k][i] = 3; +- scalar[k][2][i] = scalar[k][1][i]; +- break; +- case 0x133 : scfsi[k][i] = 3; +- scalar[k][1][i] = scalar[k][2][i]; +- break; +- case 0x113 : scfsi[k][i] = 1; +- scalar[k][1][i] = scalar[k][0][i]; +- break; +- case 0x111 : scfsi[k][i] = 2; +- scalar[k][1][i] = scalar[k][2][i] = scalar[k][0][i]; +- break; +- case 0x222 : scfsi[k][i] = 2; +- scalar[k][0][i] = scalar[k][2][i] = scalar[k][1][i]; +- break; +- case 0x333 : scfsi[k][i] = 2; +- scalar[k][0][i] = scalar[k][1][i] = scalar[k][2][i]; +- break; +- case 0x444 : scfsi[k][i] = 2; +- if (scalar[k][0][i] > scalar[k][2][i]) +- scalar[k][0][i] = scalar[k][2][i]; +- scalar[k][1][i] = scalar[k][2][i] = scalar[k][0][i]; +- } +- } +- } +- +- /************************************************************************ +- * +- * I_encode_scale (Layer I) +- * II_encode_scale (Layer II) +- * +- * PURPOSE:The encoded scalar factor information is arranged and +- * queued into the output fifo to be transmitted. +- * +- * For Layer II, the three scale factors associated with +- * a given subband and channel are transmitted in accordance +- * with the scfsi, which is transmitted first. +- * +- ************************************************************************/ +- +- void I_encode_scale(scalar, bit_alloc, fr_ps, bs) +- unsigned int scalar[2][3][SBLIMIT]; +- unsigned int bit_alloc[2][SBLIMIT]; +- frame_params *fr_ps; +- Bit_stream_struc *bs; +- { +- int stereo = fr_ps->stereo; +- int sblimit = fr_ps->sblimit; +- int i,j; +- +- for (i=0;istereo; +- int sblimit = fr_ps->sblimit; +- int jsbound = fr_ps->jsbound; +- int i,j,k; +- +- for (i=0;istereo; +- int sblimit = fr_ps->sblimit; +- int jsbound = fr_ps->jsbound; +- int req_bits = 0; +- +- /* initial b_anc (header) allocation bits */ +- req_bits = 32 + 4 * ( (jsbound * stereo) + (SBLIMIT-jsbound) ); +- +- for(i=0; i= NOISY_MIN_MNR) +- break; /* we found enough bits */ +- if(stereo == 2 && i >= jsbound) /* check other JS channel */ +- for(;k<14; ++k) +- if( (-perm_smr[1-j][i] + snr[k]) >= NOISY_MIN_MNR) break; +- if(k>0) req_bits += (k+1)*SCALE_BLOCK + 6*((i>=jsbound)?stereo:1); +- } +- return req_bits; +- } +- +- /***************************** Layer II ********************************/ +- +- int II_bits_for_nonoise(perm_smr, scfsi, fr_ps) +- double FAR perm_smr[2][SBLIMIT]; +- unsigned int scfsi[2][SBLIMIT]; +- frame_params *fr_ps; +- { +- int sb,ch,ba; +- int stereo = fr_ps->stereo; +- int sblimit = fr_ps->sblimit; +- int jsbound = fr_ps->jsbound; +- al_table *alloc = fr_ps->alloc; +- int req_bits = 0, bbal = 0, berr = 0, banc = 32; +- int maxAlloc, sel_bits, sc_bits, smp_bits; +- static int sfsPerScfsi[] = { 3,2,1,2 }; /* lookup # sfs per scfsi */ +- +- /* added 92-08-11 shn */ +- if (fr_ps->header->error_protection) berr=16; else berr=0; +- +- for (sb=0; sb0)?1:0)]) +- >= NOISY_MIN_MNR) +- break; /* we found enough bits */ +- if(stereo == 2 && sb >= jsbound) /* check other JS channel */ +- for(;ba0)?1:0)]) +- >= NOISY_MIN_MNR) +- break; +- if(ba>0) { +- smp_bits = SCALE_BLOCK * ((*alloc)[sb][ba].group * (*alloc)[sb][ba].bits); +- /* scale factor bits required for subband */ +- sel_bits = 2; +- sc_bits = 6 * sfsPerScfsi[scfsi[ch][sb]]; +- if(stereo == 2 && sb >= jsbound) { +- /* each new js sb has L+R scfsis */ +- sel_bits += 2; +- sc_bits += 6 * sfsPerScfsi[scfsi[1-ch][sb]]; +- } +- req_bits += smp_bits+sel_bits+sc_bits; +- } +- } +- return req_bits; +- } +- +- /************************************************************************ +- * +- * I_main_bit_allocation (Layer I) +- * II_main_bit_allocation (Layer II) +- * +- * PURPOSE:For joint stereo mode, determines which of the 4 joint +- * stereo modes is needed. Then calls *_a_bit_allocation(), which +- * allocates bits for each of the subbands until there are no more bits +- * left, or the MNR is at the noise/no_noise threshold. +- * +- * SEMANTICS: +- * +- * For joint stereo mode, joint stereo is changed to stereo if +- * there are enough bits to encode stereo at or better than the +- * no-noise threshold (NOISY_MIN_MNR). Otherwise, the system +- * iteratively allocates less bits by using joint stereo until one +- * of the following occurs: +- * - there are no more noisy subbands (MNR >= NOISY_MIN_MNR) +- * - mode_ext has been reduced to 0, which means that all but the +- * lowest 4 subbands have been converted from stereo to joint +- * stereo, and no more subbands may be converted +- * +- * This function calls *_bits_for_nonoise() and *_a_bit_allocation(). +- * +- ************************************************************************/ +- +- void I_main_bit_allocation(perm_smr, bit_alloc, adb, fr_ps) +- double FAR perm_smr[2][SBLIMIT]; +- unsigned int bit_alloc[2][SBLIMIT]; +- int *adb; +- frame_params *fr_ps; +- { +- int noisy_sbs; +- int mode, mode_ext, lay, i; +- int rq_db, av_db = *adb; +- static int init = 0; +- +- if(init == 0) { +- /* rearrange snr for layer I */ +- snr[2] = snr[3]; +- for (i=3;i<16;i++) snr[i] = snr[i+2]; +- init = 1; +- } +- +- if((mode = fr_ps->actual_mode) == MPG_MD_JOINT_STEREO) { +- fr_ps->header->mode = MPG_MD_STEREO; +- fr_ps->header->mode_ext = 0; +- fr_ps->jsbound = fr_ps->sblimit; +- if(rq_db = I_bits_for_nonoise(perm_smr, fr_ps) > *adb) { +- fr_ps->header->mode = MPG_MD_JOINT_STEREO; +- mode_ext = 4; /* 3 is least severe reduction */ +- lay = fr_ps->header->lay; +- do { +- --mode_ext; +- fr_ps->jsbound = js_bound(lay, mode_ext); +- rq_db = I_bits_for_nonoise(perm_smr, fr_ps); +- } while( (rq_db > *adb) && (mode_ext > 0)); +- fr_ps->header->mode_ext = mode_ext; +- } /* well we either eliminated noisy sbs or mode_ext == 0 */ +- } +- noisy_sbs = I_a_bit_allocation(perm_smr, bit_alloc, adb, fr_ps); +- } +- +- /***************************** Layer II ********************************/ +- +- void II_main_bit_allocation(perm_smr, scfsi, bit_alloc, adb, fr_ps) +- double FAR perm_smr[2][SBLIMIT]; +- unsigned int scfsi[2][SBLIMIT]; +- unsigned int bit_alloc[2][SBLIMIT]; +- int *adb; +- frame_params *fr_ps; +- { +- int noisy_sbs, nn; +- int mode, mode_ext, lay; +- int rq_db, av_db = *adb; +- +- if((mode = fr_ps->actual_mode) == MPG_MD_JOINT_STEREO) { +- fr_ps->header->mode = MPG_MD_STEREO; +- fr_ps->header->mode_ext = 0; +- fr_ps->jsbound = fr_ps->sblimit; +- if((rq_db=II_bits_for_nonoise(perm_smr, scfsi, fr_ps)) > *adb) { +- fr_ps->header->mode = MPG_MD_JOINT_STEREO; +- mode_ext = 4; /* 3 is least severe reduction */ +- lay = fr_ps->header->lay; +- do { +- --mode_ext; +- fr_ps->jsbound = js_bound(lay, mode_ext); +- rq_db = II_bits_for_nonoise(perm_smr, scfsi, fr_ps); +- } while( (rq_db > *adb) && (mode_ext > 0)); +- fr_ps->header->mode_ext = mode_ext; +- } /* well we either eliminated noisy sbs or mode_ext == 0 */ +- } +- noisy_sbs = II_a_bit_allocation(perm_smr, scfsi, bit_alloc, adb, fr_ps); +- } +- +- /************************************************************************ +- * +- * I_a_bit_allocation (Layer I) +- * II_a_bit_allocation (Layer II) +- * +- * PURPOSE:Adds bits to the subbands with the lowest mask-to-noise +- * ratios, until the maximum number of bits for the subband has +- * been allocated. +- * +- * SEMANTICS: +- * 1. Find the subband and channel with the smallest MNR (#min_sb#, +- * and #min_ch#) +- * 2. Calculate the increase in bits needed if we increase the bit +- * allocation to the next higher level +- * 3. If there are enough bits available for increasing the resolution +- * in #min_sb#, #min_ch#, and the subband has not yet reached its +- * maximum allocation, update the bit allocation, MNR, and bits +- available accordingly +- * 4. Repeat until there are no more bits left, or no more available +- * subbands. (A subband is still available until the maximum +- * number of bits for the subband has been allocated, or there +- * aren't enough bits to go to the next higher resolution in the +- subband.) +- * +- ************************************************************************/ +- +- int I_a_bit_allocation(perm_smr, bit_alloc, adb, fr_ps) /* return noisy sbs */ +- double FAR perm_smr[2][SBLIMIT]; +- unsigned int bit_alloc[2][SBLIMIT]; +- int *adb; +- frame_params *fr_ps; +- { +- int i, k, smpl_bits, scale_bits, min_sb, min_ch, oth_ch; +- int bspl, bscf, ad, noisy_sbs, done = 0, bbal ; +- double mnr[2][SBLIMIT], small; +- char used[2][SBLIMIT]; +- int stereo = fr_ps->stereo; +- int sblimit = fr_ps->sblimit; +- int jsbound = fr_ps->jsbound; +- al_table *alloc = fr_ps->alloc; +- static char init= 0; +- static int banc=32, berr=0; +- +- if (!init) { +- init = 1; +- if (fr_ps->header->error_protection) berr = 16; /* added 92-08-11 shn */ +- } +- bbal = 4 * ( (jsbound * stereo) + (SBLIMIT-jsbound) ); +- *adb -= bbal + berr + banc; +- ad= *adb; +- +- for (i=0;i mnr[k][i]) { +- small = mnr[k][i]; +- min_sb = i; min_ch = k; +- } +- if(min_sb > -1) { /* there was something to find */ +- /* first step of bit allocation is biggest */ +- if (used[min_ch][min_sb]) { smpl_bits = SCALE_BLOCK; scale_bits = 0; } +- else { smpl_bits = 24; scale_bits = 6; } +- if(min_sb >= jsbound) scale_bits *= stereo; +- +- /* check to see enough bits were available for */ +- /* increasing resolution in the minimum band */ +- +- if (ad >= bspl + bscf + scale_bits + smpl_bits) { +- bspl += smpl_bits; /* bit for subband sample */ +- bscf += scale_bits; /* bit for scale factor */ +- bit_alloc[min_ch][min_sb]++; +- used[min_ch][min_sb] = 1; /* subband has bits */ +- mnr[min_ch][min_sb] = -perm_smr[min_ch][min_sb] +- + snr[bit_alloc[min_ch][min_sb]]; +- /* Check if subband has been fully allocated max bits */ +- if (bit_alloc[min_ch][min_sb] == 14 ) used[min_ch][min_sb] = 2; +- } +- else /* no room to improve this band */ +- used[min_ch][min_sb] = 2; /* for allocation anymore */ +- if(stereo == 2 && min_sb >= jsbound) { +- oth_ch = 1-min_ch; /* joint-st : fix other ch */ +- bit_alloc[oth_ch][min_sb] = bit_alloc[min_ch][min_sb]; +- used[oth_ch][min_sb] = used[min_ch][min_sb]; +- mnr[oth_ch][min_sb] = -perm_smr[oth_ch][min_sb] +- + snr[bit_alloc[oth_ch][min_sb]]; +- } +- } +- } while(min_sb>-1); /* i.e. still some sub-bands to find */ +- +- /* Calculate the number of bits left, add on to pointed var */ +- ad -= bspl+bscf; +- *adb = ad; +- +- /* see how many channels are noisy */ +- noisy_sbs = 0; small = mnr[0][0]; +- for(k=0; k mnr[k][i]) small = mnr[k][i]; +- } +- } +- return noisy_sbs; +- } +- +- /***************************** Layer II ********************************/ +- +- int II_a_bit_allocation(perm_smr, scfsi, bit_alloc, adb, fr_ps) +- double FAR perm_smr[2][SBLIMIT]; +- unsigned int scfsi[2][SBLIMIT]; +- unsigned int bit_alloc[2][SBLIMIT]; +- int *adb; +- frame_params *fr_ps; +- { +- int i, min_ch, min_sb, oth_ch, k, increment, scale, seli, ba; +- int bspl, bscf, bsel, ad, noisy_sbs, bbal=0; +- double mnr[2][SBLIMIT], small; +- char used[2][SBLIMIT]; +- int stereo = fr_ps->stereo; +- int sblimit = fr_ps->sblimit; +- int jsbound = fr_ps->jsbound; +- al_table *alloc = fr_ps->alloc; +- static char init= 0; +- static int banc=32, berr=0; +- static int sfsPerScfsi[] = { 3,2,1,2 }; /* lookup # sfs per scfsi */ +- +- if (!init) { +- init = 1; +- if (fr_ps->header->error_protection) berr=16; /* added 92-08-11 shn */ +- } +- for (i=0; i mnr[k][i]) { +- small = mnr[k][i]; +- min_sb = i; min_ch = k; +- } +- if(min_sb > -1) { /* there was something to find */ +- /* find increase in bit allocation in subband [min] */ +- increment = SCALE_BLOCK * ((*alloc)[min_sb][bit_alloc[min_ch][min_sb]+1].group * +- (*alloc)[min_sb][bit_alloc[min_ch][min_sb]+1].bits); +- if (used[min_ch][min_sb]) +- increment -= SCALE_BLOCK * ((*alloc)[min_sb][bit_alloc[min_ch][min_sb]].group* +- (*alloc)[min_sb][bit_alloc[min_ch][min_sb]].bits); +- +- /* scale factor bits required for subband [min] */ +- oth_ch = 1 - min_ch; /* above js bound, need both chans */ +- if (used[min_ch][min_sb]) scale = seli = 0; +- else { /* this channel had no bits or scfs before */ +- seli = 2; +- scale = 6 * sfsPerScfsi[scfsi[min_ch][min_sb]]; +- if(stereo == 2 && min_sb >= jsbound) { +- /* each new js sb has L+R scfsis */ +- seli += 2; +- scale += 6 * sfsPerScfsi[scfsi[oth_ch][min_sb]]; +- } +- } +- /* check to see enough bits were available for */ +- /* increasing resolution in the minimum band */ +- if (ad >= bspl + bscf + bsel + seli + scale + increment) { +- ba = ++bit_alloc[min_ch][min_sb]; /* next up alloc */ +- bspl += increment; /* bits for subband sample */ +- bscf += scale; /* bits for scale factor */ +- bsel += seli; /* bits for scfsi code */ +- used[min_ch][min_sb] = 1; /* subband has bits */ +- mnr[min_ch][min_sb] = -perm_smr[min_ch][min_sb] + +- snr[(*alloc)[min_sb][ba].quant+1]; +- /* Check if subband has been fully allocated max bits */ +- if (ba >= (1<<(*alloc)[min_sb][0].bits)-1) used[min_ch][min_sb] = 2; +- } +- else used[min_ch][min_sb] = 2; /* can't increase this alloc */ +- if(min_sb >= jsbound && stereo == 2) { +- /* above jsbound, alloc applies L+R */ +- ba = bit_alloc[oth_ch][min_sb] = bit_alloc[min_ch][min_sb]; +- used[oth_ch][min_sb] = used[min_ch][min_sb]; +- mnr[oth_ch][min_sb] = -perm_smr[oth_ch][min_sb] + +- snr[(*alloc)[min_sb][ba].quant+1]; +- } +- } +- } while(min_sb > -1); /* until could find no channel */ +- /* Calculate the number of bits left */ +- ad -= bspl+bscf+bsel; *adb = ad; +- for (i=sblimit;i mnr[k][i]) small = mnr[k][i]; +- if(mnr[k][i] < NOISY_MIN_MNR) ++noisy_sbs; /* noise is not masked */ +- +- } +- } +- return noisy_sbs; +- } +- +- /************************************************************************ +- * +- * I_subband_quantization (Layer I) +- * II_subband_quantization (Layer II) +- * +- * PURPOSE:Quantizes subband samples to appropriate number of bits +- * +- * SEMANTICS: Subband samples are divided by their scalefactors, which +- makes the quantization more efficient. The scaled samples are +- * quantized by the function a*x+b, where a and b are functions of +- * the number of quantization levels. The result is then truncated +- * to the appropriate number of bits and the MSB is inverted. +- * +- * Note that for fractional 2's complement, inverting the MSB for a +- negative number x is equivalent to adding 1 to it. +- * +- ************************************************************************/ +- +- static double a[17] = { +- 0.750000000, 0.625000000, 0.875000000, 0.562500000, 0.937500000, +- 0.968750000, 0.984375000, 0.992187500, 0.996093750, 0.998046875, +- 0.999023438, 0.999511719, 0.999755859, 0.999877930, 0.999938965, +- 0.999969482, 0.999984741 }; +- +- static double b[17] = { +- -0.250000000, -0.375000000, -0.125000000, -0.437500000, -0.062500000, +- -0.031250000, -0.015625000, -0.007812500, -0.003906250, -0.001953125, +- -0.000976563, -0.000488281, -0.000244141, -0.000122070, -0.000061035, +- -0.000030518, -0.000015259 }; +- +- void I_subband_quantization(scalar, sb_samples, j_scale, j_samps, +- bit_alloc, sbband, fr_ps) +- unsigned int scalar[2][3][SBLIMIT]; +- double FAR sb_samples[2][3][SCALE_BLOCK][SBLIMIT]; +- unsigned int j_scale[3][SBLIMIT]; +- double FAR j_samps[3][SCALE_BLOCK][SBLIMIT]; /* L+R for j-stereo if necess */ +- unsigned int bit_alloc[2][SBLIMIT]; +- unsigned int FAR sbband[2][3][SCALE_BLOCK][SBLIMIT]; +- frame_params *fr_ps; +- { +- int i, j, k, n, sig; +- int stereo = fr_ps->stereo; +- int sblimit = fr_ps->sblimit; +- int jsbound = fr_ps->jsbound; +- double d; +- static char init = 0; +- +- if (!init) { +- init = 1; +- /* rearrange quantization coef to correspond to layer I table */ +- a[1] = a[2]; b[1] = b[2]; +- for (i=2;i<15;i++) { a[i] = a[i+2]; b[i] = b[i+2]; } +- } +- for (j=0;j=jsbound) +- /* use the joint data passed in */ +- d = j_samps[0][j][i] / multiple[j_scale[0][i]]; +- else +- d = sb_samples[k][0][j][i] / multiple[scalar[k][0][i]]; +- /* scale and quantize floating point sample */ +- n = bit_alloc[k][i]; +- d = d * a[n-1] + b[n-1]; +- /* extract MSB N-1 bits from the floating point sample */ +- if (d >= 0) sig = 1; +- else { sig = 0; d += 1.0; } +- sbband[k][0][j][i] = (unsigned int) (d * (double) (1L<stereo; +- int sblimit = fr_ps->sblimit; +- int jsbound = fr_ps->jsbound; +- unsigned int stps; +- double d; +- al_table *alloc = fr_ps->alloc; +- +- for (s=0;s<3;s++) +- for (j=0;j=jsbound) /* use j-stereo samples */ +- d = j_samps[s][j][i] / multiple[j_scale[s][i]]; +- else +- d = sb_samples[k][s][j][i] / multiple[scalar[k][s][i]]; +- if (mod(d) > 1.0) +- printf("Not scaled properly %d %d %d %d\n",k,s,j,i); +- qnt = (*alloc)[i][bit_alloc[k][i]].quant; +- d = d * a[qnt] + b[qnt]; +- /* extract MSB N-1 bits from the floating point sample */ +- if (d >= 0) sig = 1; +- else { sig = 0; d += 1.0; } +- n = 0; +- #ifndef MS_DOS +- stps = (*alloc)[i][bit_alloc[k][i]].steps; +- while ((1L<stereo; +- int sblimit = fr_ps->sblimit; +- int jsbound = fr_ps->jsbound; +- +- for (i=0;istereo; +- int sblimit = fr_ps->sblimit; +- int jsbound = fr_ps->jsbound; +- al_table *alloc = fr_ps->alloc; +- +- for (i=0;istereo; +- int sblimit = fr_ps->sblimit; +- int jsbound = fr_ps->jsbound; +- +- for(j=0;jstereo; +- int sblimit = fr_ps->sblimit; +- int jsbound = fr_ps->jsbound; +- al_table *alloc = fr_ps->alloc; +- +- for (s=0;s<3;s++) +- for (j=0;jdec delay: 1056 (observed) ++ layerII enc->dec dealy: 480 (observed) + + +! polyphase 256-16 (dec or enc) = 240 +! mdct 256+32 (9*32) (dec or enc) = 288 +! total: 512+16 + +! */ + + + +! /* ENCDELAY The encoder delay. + +! Minimum allowed is MDCTDELAY (see below) +! +! The first 96 samples will be attenuated, so using a value +! less than 96 will result in lost data in the first 96-ENCDELAY +! samples. + +! suggested: 800 +! set to 1160 to sync with FhG. +! */ +! #define ENCDELAY 800 + + + +! +! /* delay of the MDCT used in mdct.c */ +! /* original ISO routiens had a delay of 528! Takehiro's routines: */ +! #define MDCTDELAY 48 +! #define FFTOFFSET (224+MDCTDELAY) +! +! /* +! Most decoders, including the one we use, have a delay of 528 samples. +! */ +! #define DECDELAY 528 +! +! +! /* number of subbands */ +! #define SBLIMIT 32 +! +! /* parition bands bands */ + #define CBANDS 63 + +! /* number of critical bands/scale factor bands where masking is computed*/ +! #define SBPSY_l 21 +! #define SBPSY_s 12 + +! /* total number of scalefactor bands encoded */ +! #define SBMAX_l 22 +! #define SBMAX_s 13 + + + +! /* FFT sizes */ +! #define BLKSIZE 1024 +! #define HBLKSIZE 513 +! #define BLKSIZE_s 256 +! #define HBLKSIZE_s 129 +! +! +! /* #define switch_pe 1800 */ +! #define NORM_TYPE 0 +! #define START_TYPE 1 +! #define SHORT_TYPE 2 +! #define STOP_TYPE 3 +! + + #endif +diff -r -c -N encoder/fft.c lame3.70/fft.c +*** encoder/fft.c Wed Dec 31 17:00:00 1969 +--- lame3.70/fft.c Thu Apr 6 12:50:49 2000 +*************** +*** 0 **** +--- 1,356 ---- ++ /* ++ ** FFT and FHT routines ++ ** Copyright 1988, 1993; Ron Mayer ++ ** ++ ** fht(fz,n); ++ ** Does a hartley transform of "n" points in the array "fz". ++ ** ++ ** NOTE: This routine uses at least 2 patented algorithms, and may be ++ ** under the restrictions of a bunch of different organizations. ++ ** Although I wrote it completely myself; it is kind of a derivative ++ ** of a routine I once authored and released under the GPL, so it ++ ** may fall under the free software foundation's restrictions; ++ ** it was worked on as a Stanford Univ project, so they claim ++ ** some rights to it; it was further optimized at work here, so ++ ** I think this company claims parts of it. The patents are ++ ** held by R. Bracewell (the FHT algorithm) and O. Buneman (the ++ ** trig generator), both at Stanford Univ. ++ ** If it were up to me, I'd say go do whatever you want with it; ++ ** but it would be polite to give credit to the following people ++ ** if you use this anywhere: ++ ** Euler - probable inventor of the fourier transform. ++ ** Gauss - probable inventor of the FFT. ++ ** Hartley - probable inventor of the hartley transform. ++ ** Buneman - for a really cool trig generator ++ ** Mayer(me) - for authoring this particular version and ++ ** including all the optimizations in one package. ++ ** Thanks, ++ ** Ron Mayer; mayer@acuson.com ++ ** and added some optimization by ++ ** Mather - idea of using lookup table ++ ** Takehiro - some dirty hack for speed up ++ */ ++ ++ #include ++ #include "util.h" ++ #include "psymodel.h" ++ #include "lame.h" ++ ++ #define TRI_SIZE (5-1) /* 1024 = 4**5 */ ++ static FLOAT costab[TRI_SIZE*2]; ++ static FLOAT window[BLKSIZE / 2], window_s[BLKSIZE_s / 2]; ++ ++ static INLINE void fht(FLOAT *fz, short n) ++ { ++ short k4; ++ FLOAT *fi, *fn, *gi; ++ FLOAT *tri; ++ ++ fn = fz + n; ++ tri = &costab[0]; ++ k4 = 4; ++ do { ++ FLOAT s1, c1; ++ short i, k1, k2, k3, kx; ++ kx = k4 >> 1; ++ k1 = k4; ++ k2 = k4 << 1; ++ k3 = k2 + k1; ++ k4 = k2 << 1; ++ fi = fz; ++ gi = fi + kx; ++ do { ++ FLOAT f0,f1,f2,f3; ++ f1 = fi[0] - fi[k1]; ++ f0 = fi[0] + fi[k1]; ++ f3 = fi[k2] - fi[k3]; ++ f2 = fi[k2] + fi[k3]; ++ fi[k2] = f0 - f2; ++ fi[0 ] = f0 + f2; ++ fi[k3] = f1 - f3; ++ fi[k1] = f1 + f3; ++ f1 = gi[0] - gi[k1]; ++ f0 = gi[0] + gi[k1]; ++ f3 = SQRT2 * gi[k3]; ++ f2 = SQRT2 * gi[k2]; ++ gi[k2] = f0 - f2; ++ gi[0 ] = f0 + f2; ++ gi[k3] = f1 - f3; ++ gi[k1] = f1 + f3; ++ gi += k4; ++ fi += k4; ++ } while (fi= 0); ++ } else if (chn == 2) { ++ do { ++ FLOAT f0,f1,f2,f3, w; ++ ++ i = rv_tbl[j << 2]; ++ ++ f0 = ms00(ch2); w = ms10(ch2); f1 = f0 - w; f0 = f0 + w; ++ f2 = ms20(ch2); w = ms30(ch2); f3 = f2 - w; f2 = f2 + w; ++ ++ x -= 4; ++ x[0] = f0 + f2; ++ x[2] = f0 - f2; ++ x[1] = f1 + f3; ++ x[3] = f1 - f3; ++ ++ f0 = ms01(ch2); w = ms11(ch2); f1 = f0 - w; f0 = f0 + w; ++ f2 = ms21(ch2); w = ms31(ch2); f3 = f2 - w; f2 = f2 + w; ++ ++ x[BLKSIZE_s / 2 + 0] = f0 + f2; ++ x[BLKSIZE_s / 2 + 2] = f0 - f2; ++ x[BLKSIZE_s / 2 + 1] = f1 + f3; ++ x[BLKSIZE_s / 2 + 3] = f1 - f3; ++ } while (--j >= 0); ++ } else { ++ do { ++ FLOAT f0,f1,f2,f3, w; ++ ++ i = rv_tbl[j << 2]; ++ ++ f0 = ms00(ch3); w = ms10(ch3); f1 = f0 - w; f0 = f0 + w; ++ f2 = ms20(ch3); w = ms30(ch3); f3 = f2 - w; f2 = f2 + w; ++ ++ x -= 4; ++ x[0] = f0 + f2; ++ x[2] = f0 - f2; ++ x[1] = f1 + f3; ++ x[3] = f1 - f3; ++ ++ f0 = ms01(ch3); w = ms11(ch3); f1 = f0 - w; f0 = f0 + w; ++ f2 = ms21(ch3); w = ms31(ch3); f3 = f2 - w; f2 = f2 + w; ++ ++ x[BLKSIZE_s / 2 + 0] = f0 + f2; ++ x[BLKSIZE_s / 2 + 2] = f0 - f2; ++ x[BLKSIZE_s / 2 + 1] = f1 + f3; ++ x[BLKSIZE_s / 2 + 3] = f1 - f3; ++ } while (--j >= 0); ++ } ++ ++ fht(x, BLKSIZE_s); ++ } ++ } ++ ++ void fft_long( ++ FLOAT x[BLKSIZE], int chn, short *buffer[2]) ++ { ++ short i,jj = BLKSIZE / 8 - 1; ++ x += BLKSIZE / 2; ++ ++ if (chn < 2) { ++ do { ++ FLOAT f0,f1,f2,f3, w; ++ ++ i = rv_tbl[jj]; ++ f0 = ml00(ch01); w = ml10(ch01); f1 = f0 - w; f0 = f0 + w; ++ f2 = ml20(ch01); w = ml30(ch01); f3 = f2 - w; f2 = f2 + w; ++ ++ x -= 4; ++ x[0] = f0 + f2; ++ x[2] = f0 - f2; ++ x[1] = f1 + f3; ++ x[3] = f1 - f3; ++ ++ f0 = ml01(ch01); w = ml11(ch01); f1 = f0 - w; f0 = f0 + w; ++ f2 = ml21(ch01); w = ml31(ch01); f3 = f2 - w; f2 = f2 + w; ++ ++ x[BLKSIZE / 2 + 0] = f0 + f2; ++ x[BLKSIZE / 2 + 2] = f0 - f2; ++ x[BLKSIZE / 2 + 1] = f1 + f3; ++ x[BLKSIZE / 2 + 3] = f1 - f3; ++ } while (--jj >= 0); ++ } else if (chn == 2) { ++ do { ++ FLOAT f0,f1,f2,f3, w; ++ ++ i = rv_tbl[jj]; ++ f0 = ml00(ch2); w = ml10(ch2); f1 = f0 - w; f0 = f0 + w; ++ f2 = ml20(ch2); w = ml30(ch2); f3 = f2 - w; f2 = f2 + w; ++ ++ x -= 4; ++ x[0] = f0 + f2; ++ x[2] = f0 - f2; ++ x[1] = f1 + f3; ++ x[3] = f1 - f3; ++ ++ f0 = ml01(ch2); w = ml11(ch2); f1 = f0 - w; f0 = f0 + w; ++ f2 = ml21(ch2); w = ml31(ch2); f3 = f2 - w; f2 = f2 + w; ++ ++ x[BLKSIZE / 2 + 0] = f0 + f2; ++ x[BLKSIZE / 2 + 2] = f0 - f2; ++ x[BLKSIZE / 2 + 1] = f1 + f3; ++ x[BLKSIZE / 2 + 3] = f1 - f3; ++ } while (--jj >= 0); ++ } else { ++ do { ++ FLOAT f0,f1,f2,f3, w; ++ ++ i = rv_tbl[jj]; ++ f0 = ml00(ch3); w = ml10(ch3); f1 = f0 - w; f0 = f0 + w; ++ f2 = ml20(ch3); w = ml30(ch3); f3 = f2 - w; f2 = f2 + w; ++ ++ x -= 4; ++ x[0] = f0 + f2; ++ x[2] = f0 - f2; ++ x[1] = f1 + f3; ++ x[3] = f1 - f3; ++ ++ f0 = ml01(ch3); w = ml11(ch3); f1 = f0 - w; f0 = f0 + w; ++ f2 = ml21(ch3); w = ml31(ch3); f3 = f2 - w; f2 = f2 + w; ++ ++ x[BLKSIZE / 2 + 0] = f0 + f2; ++ x[BLKSIZE / 2 + 2] = f0 - f2; ++ x[BLKSIZE / 2 + 1] = f1 + f3; ++ x[BLKSIZE / 2 + 3] = f1 - f3; ++ } while (--jj >= 0); ++ } ++ ++ fht(x, BLKSIZE); ++ } ++ ++ ++ void init_fft(void) ++ { ++ int i; ++ ++ FLOAT r = PI*0.125; ++ for (i = 0; i < TRI_SIZE; i++) { ++ costab[i*2 ] = cos(r); ++ costab[i*2+1] = sin(r); ++ r *= 0.25; ++ } ++ ++ /* ++ * calculate HANN window coefficients ++ */ ++ for (i = 0; i < BLKSIZE / 2; i++) ++ window[i] = 0.5 * (1.0 - cos(2.0 * PI * (i + 0.5) / BLKSIZE)); ++ for (i = 0; i < BLKSIZE_s / 2; i++) ++ window_s[i] = 0.5 * (1.0 - cos(2.0 * PI * (i + 0.5) / BLKSIZE_s)); ++ } +diff -r -c -N encoder/fft.h lame3.70/fft.h +*** encoder/fft.h Wed Dec 31 17:00:00 1969 +--- lame3.70/fft.h Sun Feb 13 11:52:59 2000 +*************** +*** 0 **** +--- 1,10 ---- ++ #ifndef FFT_H ++ #define FFT_H ++ ++ #include "encoder.h" ++ ++ void fft_long(FLOAT x_real[BLKSIZE], int, short **); ++ void fft_short(FLOAT x_real[3][BLKSIZE_s], int, short **); ++ void init_fft(void); ++ ++ #endif +diff -r -c -N encoder/formatBitstream.c lame3.70/formatBitstream.c +*** encoder/formatBitstream.c Wed Jan 22 02:43:14 1997 +--- lame3.70/formatBitstream.c Sun Feb 20 07:09:34 2000 +*************** +*** 17,40 **** + #include + #include + +- #ifndef EXIT_FAILURE +- #define EXIT_FAILURE 1 +- #endif +- + /* globals */ + static int BitCount = 0; + static int ThisFrameSize = 0; + static int BitsRemaining = 0; +! static BitsFcnPtr PutBits = NULL; + + /* forward declarations */ + static int store_side_info( BF_FrameData *frameInfo ); + static int main_data( BF_FrameData *frameInfo, BF_FrameResults *results ); + static int side_queue_elements( int *forwardFrameLength, int *forwardSILength ); +! static void free_side_queues(); +! static void WriteMainDataBits( unsigned val, +! unsigned nbits, +! BF_FrameResults *results ); + /* + BitStreamFrame is the public interface to the bitstream + formatting package. It writes one frame of main data per call. +--- 17,41 ---- + #include + #include + + /* globals */ + static int BitCount = 0; + static int ThisFrameSize = 0; + static int BitsRemaining = 0; +! +! void InitFormatBitStream(void) +! { +! BitCount = 0; +! ThisFrameSize = 0; +! BitsRemaining = 0; +! } + + /* forward declarations */ + static int store_side_info( BF_FrameData *frameInfo ); + static int main_data( BF_FrameData *frameInfo, BF_FrameResults *results ); + static int side_queue_elements( int *forwardFrameLength, int *forwardSILength ); +! static void free_side_queues(void); +! static void WriteMainDataBits( u_int val,u_int nbits,BF_FrameResults *results ); +! + /* + BitStreamFrame is the public interface to the bitstream + formatting package. It writes one frame of main data per call. +*************** +*** 49,65 **** + See formatBitstream.h for more information about the data + structures and the bitstream syntax. + */ + void + BF_BitstreamFrame( BF_FrameData *frameInfo, BF_FrameResults *results ) + { +! int elements, forwardFrameLength, forwardSILength; + + assert( frameInfo->nGranules <= MAX_GRANULES ); + assert( frameInfo->nChannels <= MAX_CHANNELS ); + +- /* get ptr to bit writing function */ +- PutBits = frameInfo->putbits; +- assert( PutBits ); + /* save SI and compute its length */ + results->SILength = store_side_info( frameInfo ); + +--- 50,64 ---- + See formatBitstream.h for more information about the data + structures and the bitstream syntax. + */ ++ static int elements, forwardFrameLength, forwardSILength; + void + BF_BitstreamFrame( BF_FrameData *frameInfo, BF_FrameResults *results ) + { +! /* int elements, forwardFrameLength, forwardSILength; */ + + assert( frameInfo->nGranules <= MAX_GRANULES ); + assert( frameInfo->nChannels <= MAX_CHANNELS ); + + /* save SI and compute its length */ + results->SILength = store_side_info( frameInfo ); + +*************** +*** 87,108 **** + void + BF_FlushBitstream( BF_FrameData *frameInfo, BF_FrameResults *results ) + { +! int elements, forwardFrameLength, forwardSILength; +! +! /* get ptr to bit writing function */ +! PutBits = frameInfo->putbits; +! assert( PutBits ); +! +! elements = side_queue_elements( &forwardFrameLength, &forwardSILength ); + + if ( elements ) + { +! int bitsRemaining = forwardFrameLength - forwardSILength; +! int wordsRemaining = bitsRemaining / 32; +! while ( wordsRemaining-- ) +! WriteMainDataBits( 0, 32, results ); +! WriteMainDataBits( 0, (bitsRemaining % 32), results ); + } + + results->mainDataLength = forwardFrameLength - forwardSILength; + results->SILength = forwardSILength; +--- 86,103 ---- + void + BF_FlushBitstream( BF_FrameData *frameInfo, BF_FrameResults *results ) + { +! /* int elements, forwardFrameLength, forwardSILength; */ + + if ( elements ) + { +! int bitsRemaining = forwardFrameLength - forwardSILength; +! int wordsRemaining = bitsRemaining / 32; +! while ( wordsRemaining-- ) { +! WriteMainDataBits( 0, 32, results ); +! } +! WriteMainDataBits( 0, (bitsRemaining % 32), results ); + } ++ + + results->mainDataLength = forwardFrameLength - forwardSILength; + results->SILength = forwardSILength; +*************** +*** 115,121 **** + BitCount = 0; + ThisFrameSize = 0; + BitsRemaining = 0; +- PutBits = NULL; + return; + } + +--- 110,115 ---- +*************** +*** 123,137 **** + BF_PartLength( BF_BitstreamPart *part ) + { + BF_BitstreamElement *ep = part->element; +! int i, bits; + +- bits = 0; + for ( i = 0; i < part->nrEntries; i++, ep++ ) +! bits += ep->length; + return bits; + } + +! + /* + The following is all private to this file + */ +--- 117,131 ---- + BF_PartLength( BF_BitstreamPart *part ) + { + BF_BitstreamElement *ep = part->element; +! u_int i; +! int bits=0; + + for ( i = 0; i < part->nrEntries; i++, ep++ ) +! bits += ep->length; + return bits; + } + +! + /* + The following is all private to this file + */ +*************** +*** 148,155 **** + BF_PartHolder *spectrumSIPH[MAX_GRANULES][MAX_CHANNELS]; + } MYSideInfo; + +! static MYSideInfo *get_side_info(); +! static int write_side_info(); + typedef int (*PartWriteFcnPtr)( BF_BitstreamPart *part, BF_FrameResults *results ); + + +--- 142,149 ---- + BF_PartHolder *spectrumSIPH[MAX_GRANULES][MAX_CHANNELS]; + } MYSideInfo; + +! static MYSideInfo *get_side_info(void); +! static int write_side_info(void); + typedef int (*PartWriteFcnPtr)( BF_BitstreamPart *part, BF_FrameResults *results ); + + +*************** +*** 157,173 **** + writePartMainData( BF_BitstreamPart *part, BF_FrameResults *results ) + { + BF_BitstreamElement *ep; +! int i, bits; + + assert( results ); + assert( part ); + +- bits = 0; + ep = part->element; + for ( i = 0; i < part->nrEntries; i++, ep++ ) + { +! WriteMainDataBits( ep->value, ep->length, results ); +! bits += ep->length; + } + return bits; + } +--- 151,167 ---- + writePartMainData( BF_BitstreamPart *part, BF_FrameResults *results ) + { + BF_BitstreamElement *ep; +! u_int i; +! int bits=0; + + assert( results ); + assert( part ); + + ep = part->element; + for ( i = 0; i < part->nrEntries; i++, ep++ ) + { +! WriteMainDataBits( ep->value, ep->length, results ); +! bits += ep->length; + } + return bits; + } +*************** +*** 176,191 **** + writePartSideInfo( BF_BitstreamPart *part, BF_FrameResults *results ) + { + BF_BitstreamElement *ep; +! int i, bits; + + assert( part ); + +- bits = 0; + ep = part->element; + for ( i = 0; i < part->nrEntries; i++, ep++ ) + { +! (*PutBits)( ep->value, ep->length ); +! bits += ep->length; + } + return bits; + } +--- 170,185 ---- + writePartSideInfo( BF_BitstreamPart *part, BF_FrameResults *results ) + { + BF_BitstreamElement *ep; +! u_int i; +! int bits=0; + + assert( part ); + + ep = part->element; + for ( i = 0; i < part->nrEntries; i++, ep++ ) + { +! putMyBits( ep->value, ep->length ); +! bits += ep->length; + } + return bits; + } +*************** +*** 216,244 **** + */ + + static void +! WriteMainDataBits( unsigned val, +! unsigned nbits, + BF_FrameResults *results ) + { + assert( nbits <= 32 ); + if ( BitCount == ThisFrameSize ) + { + BitCount = write_side_info(); + BitsRemaining = ThisFrameSize - BitCount; + } +! if ( nbits == 0 ) +! return; +! if ( nbits > BitsRemaining ) + { + unsigned extra = val >> (nbits - BitsRemaining); + nbits -= BitsRemaining; +! (*PutBits)( extra, BitsRemaining ); + BitCount = write_side_info(); + BitsRemaining = ThisFrameSize - BitCount; +! (*PutBits)( val, nbits ); + } + else +! (*PutBits)( val, nbits ); + BitCount += nbits; + BitsRemaining -= nbits; + assert( BitCount <= ThisFrameSize ); +--- 210,238 ---- + */ + + static void +! WriteMainDataBits( u_int val, +! u_int nbits, + BF_FrameResults *results ) + { + assert( nbits <= 32 ); ++ if ( nbits == 0 ) ++ return; + if ( BitCount == ThisFrameSize ) + { + BitCount = write_side_info(); + BitsRemaining = ThisFrameSize - BitCount; + } +! if ( nbits > (u_int)BitsRemaining ) + { + unsigned extra = val >> (nbits - BitsRemaining); + nbits -= BitsRemaining; +! putMyBits( extra, BitsRemaining ); + BitCount = write_side_info(); + BitsRemaining = ThisFrameSize - BitCount; +! putMyBits( val, nbits ); + } + else +! putMyBits( val, nbits ); + BitCount += nbits; + BitsRemaining -= nbits; + assert( BitCount <= ThisFrameSize ); +*************** +*** 246,254 **** + assert( (BitCount + BitsRemaining) == ThisFrameSize ); + } + +! + static int +! write_side_info() + { + MYSideInfo *si; + int bits, ch, gr; +--- 240,248 ---- + assert( (BitCount + BitsRemaining) == ThisFrameSize ); + } + +! + static int +! write_side_info(void) + { + MYSideInfo *si; + int bits, ch, gr; +*************** +*** 284,290 **** + side_queue_elements( int *frameLength, int *SILength ) + { + int elements = 0; +! side_info_link *l = side_queue_head; + + *frameLength = 0; + *SILength = 0; +--- 278,284 ---- + side_queue_elements( int *frameLength, int *SILength ) + { + int elements = 0; +! side_info_link *l; + + *frameLength = 0; + *SILength = 0; +*************** +*** 302,308 **** + store_side_info( BF_FrameData *info ) + { + int ch, gr; +! side_info_link *l = NULL; + /* obtain a side_info_link to store info */ + side_info_link *f = side_queue_free; + int bits = 0; +--- 296,302 ---- + store_side_info( BF_FrameData *info ) + { + int ch, gr; +! side_info_link *l; + /* obtain a side_info_link to store info */ + side_info_link *f = side_queue_free; + int bits = 0; +*************** +*** 318,324 **** + if ( l == NULL ) + { + fprintf( stderr, "cannot allocate side_info_link" ); +! exit( EXIT_FAILURE ); + } + l->next = NULL; + l->side_info.headerPH = BF_newPartHolder( info->header->nrEntries ); +--- 312,318 ---- + if ( l == NULL ) + { + fprintf( stderr, "cannot allocate side_info_link" ); +! exit( 1); + } + l->next = NULL; + l->side_info.headerPH = BF_newPartHolder( info->header->nrEntries ); +*************** +*** 377,383 **** + } + + static MYSideInfo* +! get_side_info() + { + side_info_link *f = side_queue_free; + side_info_link *l = side_queue_head; +--- 371,377 ---- + } + + static MYSideInfo* +! get_side_info(void) + { + side_info_link *f = side_queue_free; + side_info_link *l = side_queue_head; +*************** +*** 404,410 **** + } + + static void +! free_side_queues() + { + side_info_link *l, *next; + +--- 398,404 ---- + } + + static void +! free_side_queues(void) + { + side_info_link *l, *next; + +*************** +*** 445,457 **** + */ + BF_PartHolder *BF_newPartHolder( int max_elements ) + { +! BF_PartHolder *newPH = calloc( 1, sizeof(BF_PartHolder) ); + assert( newPH ); + newPH->max_elements = max_elements; +! newPH->part = calloc( 1, sizeof(BF_BitstreamPart) ); + assert( newPH->part ); +! newPH->part->element = calloc( max_elements, sizeof(BF_BitstreamElement) ); +! assert( newPH->part->element ); + newPH->part->nrEntries = 0; + return newPH; + } +--- 439,451 ---- + */ + BF_PartHolder *BF_newPartHolder( int max_elements ) + { +! BF_PartHolder *newPH = (BF_PartHolder*) calloc( 1, sizeof(BF_PartHolder) ); + assert( newPH ); + newPH->max_elements = max_elements; +! newPH->part = (BF_BitstreamPart*) calloc( 1, sizeof(BF_BitstreamPart) ); + assert( newPH->part ); +! newPH->part->element = (BF_BitstreamElement*) calloc( max_elements, sizeof(BF_BitstreamElement) ); +! if (max_elements>0) assert( newPH->part->element ); + newPH->part->nrEntries = 0; + return newPH; + } +*************** +*** 465,471 **** + BF_PartHolder *BF_LoadHolderFromBitstreamPart( BF_PartHolder *theHolder, BF_BitstreamPart *thePart ) + { + BF_BitstreamElement *pElem; +! int i; + + theHolder->part->nrEntries = 0; + for ( i = 0; i < thePart->nrEntries; i++ ) +--- 459,465 ---- + BF_PartHolder *BF_LoadHolderFromBitstreamPart( BF_PartHolder *theHolder, BF_BitstreamPart *thePart ) + { + BF_BitstreamElement *pElem; +! u_int i; + + theHolder->part->nrEntries = 0; + for ( i = 0; i < thePart->nrEntries; i++ ) +*************** +*** 536,548 **** + /* + Add a bit value and length to the element list in thePH + */ +! BF_PartHolder *BF_addEntry( BF_PartHolder *thePH, uint32 value, uint16 length ) + { + BF_BitstreamElement myElement; + myElement.value = value; + myElement.length = length; + if ( length ) +! return BF_addElement( thePH, &myElement ); + else +! return thePH; + } +--- 530,543 ---- + /* + Add a bit value and length to the element list in thePH + */ +! BF_PartHolder *BF_addEntry( BF_PartHolder *thePH, u_int value, u_int length ) + { + BF_BitstreamElement myElement; + myElement.value = value; + myElement.length = length; ++ + if ( length ) +! return BF_addElement( thePH, &myElement ); + else +! return thePH; + } +diff -r -c -N encoder/formatBitstream.h lame3.70/formatBitstream.h +*** encoder/formatBitstream.h Wed Jan 22 02:43:15 1997 +--- lame3.70/formatBitstream.h Sat Feb 19 06:32:30 2000 +*************** +*** 13,18 **** +--- 13,21 ---- + 1995/09/06 mc@fivebats.com created + + */ ++ ++ #include "machine.h" ++ + #ifndef MAX_CHANNELS + #define MAX_CHANNELS 2 + #endif +*************** +*** 21,37 **** + #define MAX_GRANULES 2 + #endif + +- /* Find the ANSI header for these! */ +- typedef unsigned int uint32; +- typedef unsigned short uint16; +- + /* + This is the prototype for the function pointer you must + provide to write bits to the bitstream. It should write + 'length' bits from 'value,' msb first. Bits in value are + assumed to be right-justified. + */ +! typedef void (*BitsFcnPtr)( uint32 value, uint16 length ); + + /* + A BitstreamElement contains encoded data +--- 24,36 ---- + #define MAX_GRANULES 2 + #endif + + /* + This is the prototype for the function pointer you must + provide to write bits to the bitstream. It should write + 'length' bits from 'value,' msb first. Bits in value are + assumed to be right-justified. + */ +! void putMyBits( u_int value, u_int length ); + + /* + A BitstreamElement contains encoded data +*************** +*** 41,48 **** + */ + typedef struct + { +! uint32 value; +! uint16 length; + } BF_BitstreamElement; + + /* +--- 40,47 ---- + */ + typedef struct + { +! u_int value; +! u_short length; + } BF_BitstreamElement; + + /* +*************** +*** 54,60 **** + */ + typedef struct + { +! uint32 nrEntries; + BF_BitstreamElement *element; + } BF_BitstreamPart; + +--- 53,59 ---- + */ + typedef struct + { +! u_int nrEntries; + BF_BitstreamElement *element; + } BF_BitstreamPart; + +*************** +*** 70,76 **** + + typedef struct BF_FrameData + { +- BitsFcnPtr putbits; /* your low-level bitstream function */ + int frameLength; + int nGranules; + int nChannels; +--- 69,74 ---- +*************** +*** 97,145 **** + int nextBackPtr; + } BF_FrameResults; + +- /* +- The following is a shorthand bitstream syntax for +- the type of bitstream this package will create. +- The bitstream has headers and side information that +- are placed at appropriate sections to allow framing. +- The main data is placed where it fits in a manner +- similar to layer3, which means that main data for a +- frame may be written to the bitstream before the +- frame's header and side information is written. +- +- BitstreamFrame() +- { +- Header(); +- FrameSI(); +- +- for ( ch ) +- ChannelSI(); +- +- for ( gr ) +- for ( ch ) +- SpectrumSI(); +- +- MainData(); +- } +- +- MainData() +- { +- for ( gr ) +- for ( ch ) +- { +- Scalefactors(); +- CodedData(); +- UserSpectrum(); +- } +- UserFrameData(); +- } +- +- */ + + /* + public functions in formatBitstream.c + */ + + /* count the bits in a BitstreamPart */ + int BF_PartLength( BF_BitstreamPart *part ); + +--- 95,109 ---- + int nextBackPtr; + } BF_FrameResults; + + + /* + public functions in formatBitstream.c + */ + ++ ++ /* Initialize the bitstream */ ++ void InitFormatBitStream(void); ++ + /* count the bits in a BitstreamPart */ + int BF_PartLength( BF_BitstreamPart *part ); + +*************** +*** 149,154 **** +--- 113,120 ---- + /* write any remaining frames to the bitstream, padding with zeros */ + void BF_FlushBitstream( BF_FrameData *frameInfo, BF_FrameResults *results ); + ++ ++ + typedef struct BF_PartHolder + { + int max_elements; +*************** +*** 158,165 **** + BF_PartHolder *BF_newPartHolder( int max_elements ); + BF_PartHolder *BF_resizePartHolder( BF_PartHolder *oldPH, int max_elements ); + BF_PartHolder *BF_addElement( BF_PartHolder *thePH, BF_BitstreamElement *theElement ); +! BF_PartHolder *BF_addEntry( BF_PartHolder *thePH, uint32 value, uint16 length ); + BF_PartHolder *BF_NewHolderFromBitstreamPart( BF_BitstreamPart *thePart ); + BF_PartHolder *BF_LoadHolderFromBitstreamPart( BF_PartHolder *theHolder, BF_BitstreamPart *thePart ); + BF_PartHolder *BF_freePartHolder( BF_PartHolder *thePH ); +! #endif +--- 124,132 ---- + BF_PartHolder *BF_newPartHolder( int max_elements ); + BF_PartHolder *BF_resizePartHolder( BF_PartHolder *oldPH, int max_elements ); + BF_PartHolder *BF_addElement( BF_PartHolder *thePH, BF_BitstreamElement *theElement ); +! BF_PartHolder *BF_addEntry( BF_PartHolder *thePH, u_int value, u_int length ); + BF_PartHolder *BF_NewHolderFromBitstreamPart( BF_BitstreamPart *thePart ); + BF_PartHolder *BF_LoadHolderFromBitstreamPart( BF_PartHolder *theHolder, BF_BitstreamPart *thePart ); + BF_PartHolder *BF_freePartHolder( BF_PartHolder *thePH ); +! +! #endif /* #ifndef _FORMAT_BITSTREAM_H */ +diff -r -c -N encoder/get_audio.c lame3.70/get_audio.c +*** encoder/get_audio.c Wed Dec 31 17:00:00 1969 +--- lame3.70/get_audio.c Thu Apr 6 12:50:49 2000 +*************** +*** 0 **** +--- 1,937 ---- ++ #include "util.h" ++ #include "get_audio.h" ++ #ifdef HAVEGTK ++ #include "gtkanal.h" ++ #include ++ #endif ++ ++ #if (defined LIBSNDFILE || defined LAMESNDFILE) ++ ++ #ifdef _WIN32 ++ /* needed to set stdin to binary on windoze machines */ ++ #include ++ #endif ++ ++ ++ ++ static FILE *musicin=NULL; /* input file pointer */ ++ static unsigned long num_samples; ++ static int samp_freq; ++ static int input_bitrate; ++ static int num_channels; ++ static int count_samples_carefully; ++ ++ int read_samples_pcm(lame_global_flags *gfp,short sample_buffer[2304],int frame_size, int samples_to_read); ++ int read_samples_mp3(lame_global_flags *gfp,FILE *musicin,short int mpg123pcm[2][1152],int num_chan); ++ ++ ++ ++ ++ ++ void lame_init_infile(lame_global_flags *gfp) ++ { ++ /* open the input file */ ++ count_samples_carefully=0; ++ OpenSndFile(gfp,gfp->inPath,gfp->in_samplerate,gfp->num_channels); ++ /* if GetSndSampleRate is non zero, use it to overwrite the default */ ++ if (GetSndSampleRate()) gfp->in_samplerate=GetSndSampleRate(); ++ if (GetSndChannels()) gfp->num_channels=GetSndChannels(); ++ gfp->num_samples = GetSndSamples(); ++ } ++ void lame_close_infile(lame_global_flags *gfp) ++ { ++ CloseSndFile(gfp); ++ } ++ ++ ++ ++ ++ /************************************************************************ ++ * ++ * lame_readframe() ++ * ++ * PURPOSE: reads a frame of audio data from a file to the buffer, ++ * aligns the data for future processing, and separates the ++ * left and right channels ++ * ++ * ++ ************************************************************************/ ++ int lame_readframe(lame_global_flags *gfp,short int Buffer[2][1152]) ++ { ++ int iread; ++ ++ /* note: if input is gfp->stereo and output is mono, get_audio() ++ * will return .5*(L+R) in channel 0, and nothing in channel 1. */ ++ iread = get_audio(gfp,Buffer,gfp->stereo); ++ ++ /* check to see if we overestimated/underestimated totalframes */ ++ if (iread==0) gfp->totalframes = Min(gfp->totalframes,gfp->frameNum+2); ++ if (gfp->frameNum > (gfp->totalframes-1)) gfp->totalframes = gfp->frameNum; ++ return iread; ++ } ++ ++ ++ ++ ++ ++ /************************************************************************ ++ * ++ * get_audio() ++ * ++ * PURPOSE: reads a frame of audio data from a file to the buffer, ++ * aligns the data for future processing, and separates the ++ * left and right channels ++ * ++ * ++ ************************************************************************/ ++ int get_audio(lame_global_flags *gfp,short buffer[2][1152],int stereo) ++ { ++ ++ int j; ++ short insamp[2304]; ++ int samples_read; ++ int framesize,samples_to_read; ++ static unsigned long num_samples_read; ++ unsigned long remaining; ++ int num_channels = gfp->num_channels; ++ ++ if (gfp->frameNum==0) { ++ num_samples_read=0; ++ num_samples= GetSndSamples(); ++ } ++ framesize = gfp->mode_gr*576; ++ ++ samples_to_read = framesize; ++ if (count_samples_carefully) { ++ /* if this flag has been set, then we are carefull to read ++ * exactly num_samples and no more. This is usefull for .wav and .aiff ++ * files which have id3 or other tags at the end. Note that if you ++ * are using LIBSNDFILE, this is not necessary */ ++ remaining=num_samples-Min(num_samples,num_samples_read); ++ if (remaining < (unsigned long)framesize) ++ samples_to_read = remaining; ++ } ++ ++ ++ if (gfp->input_format==sf_mp3) { ++ /* decode an mp3 file for the input */ ++ samples_read=read_samples_mp3(gfp,musicin,buffer,num_channels); ++ }else{ ++ samples_read = read_samples_pcm(gfp,insamp,num_channels*framesize,num_channels*samples_to_read); ++ samples_read /=num_channels; ++ ++ for(j=0;jgtkflag) { ++ framesize=1152; ++ if (out==576) framesize=576; ++ ++ /* add a delay of framesize-DECDELAY, which will make the total delay ++ * exactly one frame, so we can sync MP3 output with WAV input */ ++ ++ for ( ch = 0; ch < stereo; ch++ ) { ++ for ( j = 0; j < framesize-DECDELAY; j++ ) ++ pinfo->pcmdata2[ch][j] = pinfo->pcmdata2[ch][j+framesize]; ++ for ( j = 0; j < framesize; j++ ) ++ pinfo->pcmdata2[ch][j+framesize-DECDELAY] = mpg123pcm[ch][j]; ++ } ++ ++ pinfo->frameNum123 = gfp->frameNum-1; ++ pinfo->frameNum = gfp->frameNum; ++ } ++ #endif ++ if (out==-1) return 0; ++ else return out; ++ #else ++ fprintf(stderr,"Error: libmp3lame was not compiled with I/O support \n"); ++ exit(1); ++ #endif ++ } ++ #endif /* LAMESNDFILE or LIBSNDFILE */ ++ ++ ++ ++ ++ #ifdef LIBSNDFILE ++ /* ++ ** Copyright (C) 1999 Albert Faber ++ ** ++ * This library is free software; you can redistribute it and/or ++ * modify it under the terms of the GNU Library General Public ++ * License as published by the Free Software Foundation; either ++ * version 2 of the License, or (at your option) any later version. ++ * ++ * This library is distributed in the hope that it will be useful, ++ * but WITHOUT ANY WARRANTY; without even the implied warranty of ++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU ++ * Library General Public License for more details. ++ * ++ * You should have received a copy of the GNU Library General Public ++ * License along with this library; if not, write to the ++ * Free Software Foundation, Inc., 59 Temple Place - Suite 330, ++ * Boston, MA 02111-1307, USA. ++ */ ++ ++ ++ #include ++ ++ /* External references */ ++ ++ static SNDFILE* gs_pSndFileIn=NULL; ++ static SF_INFO gs_wfInfo; ++ ++ ++ unsigned long GetSndSamples(void) ++ { ++ return gs_wfInfo.samples; ++ } ++ int GetSndSampleRate(void) ++ { ++ return gs_wfInfo.samplerate; ++ } ++ ++ int GetSndChannels(void) ++ { ++ return gs_wfInfo.channels; ++ } ++ ++ void CloseSndFile(lame_global_flags *gfp) ++ { ++ if (gfp->input_format==sf_mp3) { ++ #ifndef AMIGA_MPEGA ++ if (fclose(musicin) != 0){ ++ fprintf(stderr, "Could not close audio input file\n"); ++ exit(2); ++ } ++ #endif ++ }else{ ++ if (gs_pSndFileIn) ++ { ++ if (sf_close(gs_pSndFileIn) !=0) ++ { ++ fprintf(stderr, "Could not close sound file \n"); ++ exit(2); ++ } ++ } ++ } ++ } ++ ++ ++ ++ FILE * OpenSndFile(lame_global_flags *gfp,const char* lpszFileName, int default_samp, ++ int default_channels) ++ { ++ input_bitrate=0; ++ if (gfp->input_format==sf_mp3) { ++ #ifdef AMIGA_MPEGA ++ if (-1==lame_decode_initfile(lpszFileName,&num_channels,&samp_freq,&input_bitrate,&num_samples)) { ++ fprintf(stderr,"Error reading headers in mp3 input file %s.\n", lpszFileName); ++ exit(1); ++ } ++ #endif ++ #ifdef HAVEMPGLIB ++ if ((musicin = fopen(lpszFileName, "rb")) == NULL) { ++ fprintf(stderr, "Could not find \"%s\".\n", lpszFileName); ++ exit(1); ++ } ++ if (-1==lame_decode_initfile(musicin,&num_channels,&samp_freq,&input_bitrate,&num_samples)) { ++ fprintf(stderr,"Error reading headers in mp3 input file %s.\n", lpszFileName); ++ exit(1); ++ } ++ #endif ++ gs_wfInfo.samples=num_samples; ++ gs_wfInfo.channels=num_channels; ++ gs_wfInfo.samplerate=samp_freq; ++ ++ } else { ++ ++ /* Try to open the sound file */ ++ /* set some defaults incase input is raw PCM */ ++ gs_wfInfo.seekable=(gfp->input_format!=sf_raw); /* if user specified -r, set to not seekable */ ++ gs_wfInfo.samplerate=default_samp; ++ gs_wfInfo.pcmbitwidth=16; ++ gs_wfInfo.channels=default_channels; ++ if (DetermineByteOrder()==order_littleEndian) { ++ if (gfp->swapbytes) gs_wfInfo.format=SF_FORMAT_RAW_BE; ++ else gs_wfInfo.format=SF_FORMAT_RAW_LE; ++ } else { ++ if (gfp->swapbytes) gs_wfInfo.format=SF_FORMAT_RAW_LE; ++ else gs_wfInfo.format=SF_FORMAT_RAW_BE; ++ } ++ ++ gs_pSndFileIn=sf_open_read(lpszFileName,&gs_wfInfo); ++ ++ /* Check result */ ++ if (gs_pSndFileIn==NULL) ++ { ++ sf_perror(gs_pSndFileIn); ++ fprintf(stderr, "Could not open sound file \"%s\".\n", lpszFileName); ++ exit(1); ++ } ++ ++ if ((gs_wfInfo.format==SF_FORMAT_RAW_LE) || ++ (gs_wfInfo.format==SF_FORMAT_RAW_BE)) ++ gfp->input_format=sf_raw; ++ ++ #ifdef _DEBUG_SND_FILE ++ printf("\n\nSF_INFO structure\n"); ++ printf("samplerate :%d\n",gs_wfInfo.samplerate); ++ printf("samples :%d\n",gs_wfInfo.samples); ++ printf("channels :%d\n",gs_wfInfo.channels); ++ printf("pcmbitwidth :%d\n",gs_wfInfo.pcmbitwidth); ++ printf("format :"); ++ ++ /* new formats from sbellon@sbellon.de 1/2000 */ ++ if ((gs_wfInfo.format&SF_FORMAT_TYPEMASK)==SF_FORMAT_WAV) ++ printf("Microsoft WAV format (big endian). "); ++ if ((gs_wfInfo.format&SF_FORMAT_TYPEMASK)==SF_FORMAT_AIFF) ++ printf("Apple/SGI AIFF format (little endian). "); ++ if ((gs_wfInfo.format&SF_FORMAT_TYPEMASK)==SF_FORMAT_AU) ++ printf("Sun/NeXT AU format (big endian). "); ++ if ((gs_wfInfo.format&SF_FORMAT_TYPEMASK)==SF_FORMAT_AULE) ++ printf("DEC AU format (little endian). "); ++ if ((gs_wfInfo.format&SF_FORMAT_TYPEMASK)==SF_FORMAT_RAW) ++ printf("RAW PCM data. "); ++ if ((gs_wfInfo.format&SF_FORMAT_TYPEMASK)==SF_FORMAT_PAF) ++ printf("Ensoniq PARIS file format. "); ++ if ((gs_wfInfo.format&SF_FORMAT_TYPEMASK)==SF_FORMAT_SVX) ++ printf("Amiga IFF / SVX8 / SV16 format. "); ++ if ((gs_wfInfo.format&SF_FORMAT_TYPEMASK)==SF_FORMAT_NIST) ++ printf("Sphere NIST format. "); ++ ++ if ((gs_wfInfo.format&SF_FORMAT_SUBMASK)==SF_FORMAT_PCM) ++ printf("PCM data in 8, 16, 24 or 32 bits."); ++ if ((gs_wfInfo.format&SF_FORMAT_SUBMASK)==SF_FORMAT_FLOAT) ++ printf("32 bit Intel x86 floats."); ++ if ((gs_wfInfo.format&SF_FORMAT_SUBMASK)==SF_FORMAT_ULAW) ++ printf("U-Law encoded."); ++ if ((gs_wfInfo.format&SF_FORMAT_SUBMASK)==SF_FORMAT_ALAW) ++ printf("A-Law encoded."); ++ if ((gs_wfInfo.format&SF_FORMAT_SUBMASK)==SF_FORMAT_IMA_ADPCM) ++ printf("IMA ADPCM."); ++ if ((gs_wfInfo.format&SF_FORMAT_SUBMASK)==SF_FORMAT_MS_ADPCM) ++ printf("Microsoft ADPCM."); ++ if ((gs_wfInfo.format&SF_FORMAT_SUBMASK)==SF_FORMAT_PCM_BE) ++ printf("Big endian PCM data."); ++ if ((gs_wfInfo.format&SF_FORMAT_SUBMASK)==SF_FORMAT_PCM_LE) ++ printf("Little endian PCM data."); ++ if ((gs_wfInfo.format&SF_FORMAT_SUBMASK)==SF_FORMAT_PCM_S8) ++ printf("Signed 8 bit PCM."); ++ if ((gs_wfInfo.format&SF_FORMAT_SUBMASK)==SF_FORMAT_PCM_U8) ++ printf("Unsigned 8 bit PCM."); ++ if ((gs_wfInfo.format&SF_FORMAT_SUBMASK)==SF_FORMAT_SVX_FIB) ++ printf("SVX Fibonacci Delta encoding."); ++ if ((gs_wfInfo.format&SF_FORMAT_SUBMASK)==SF_FORMAT_SVX_EXP) ++ printf("SVX Exponential Delta encoding."); ++ ++ ++ ++ ++ printf("\n"); ++ printf("pcmbitwidth :%d\n",gs_wfInfo.pcmbitwidth); ++ printf("sections :%d\n",gs_wfInfo.sections); ++ printf("seekable :\n",gs_wfInfo.seekable); ++ #endif ++ } ++ ++ if (gs_wfInfo.samples==MAX_U_32_NUM) { ++ struct stat sb; ++ /* try to figure out num_samples */ ++ if (0==stat(lpszFileName,&sb)) { ++ /* try file size, assume 2 bytes per sample */ ++ if (gfp->input_format == sf_mp3) { ++ FLOAT totalseconds = (sb.st_size*8.0/(1000.0*GetSndBitrate())); ++ gs_wfInfo.samples= totalseconds*GetSndSampleRate(); ++ }else{ ++ gs_wfInfo.samples = sb.st_size/(2*GetSndChannels()); ++ } ++ } ++ } ++ return musicin; ++ } ++ ++ ++ /************************************************************************ ++ * ++ * read_samples() ++ * ++ * PURPOSE: reads the PCM samples from a file to the buffer ++ * ++ * SEMANTICS: ++ * Reads #samples_read# number of shorts from #musicin# filepointer ++ * into #sample_buffer[]#. Returns the number of samples read. ++ * ++ ************************************************************************/ ++ ++ int read_samples_pcm(lame_global_flags *gfp,short sample_buffer[2304],int frame_size,int samples_to_read) ++ { ++ int samples_read; ++ int rcode; ++ ++ samples_read=sf_read_short(gs_pSndFileIn,sample_buffer,samples_to_read); ++ ++ rcode = samples_read; ++ if (samples_read < frame_size) ++ { ++ /*fprintf(stderr,"Insufficient PCM input for one frame - fillout with zeros\n"); ++ */ ++ if (samples_read<0) samples_read=0; ++ for (; samples_read < frame_size; sample_buffer[samples_read++] = 0); ++ } ++ ++ if (8==gs_wfInfo.pcmbitwidth) ++ for (; samples_read >= 0; sample_buffer[samples_read] = sample_buffer[samples_read--] * 256); ++ ++ return(rcode); ++ } ++ ++ ++ #endif /* ifdef LIBSNDFILE */ ++ #ifdef LAMESNDFILE ++ ++ /************************************************************************ ++ ************************************************************************ ++ ************************************************************************ ++ ************************************************************************ ++ ************************************************************************ ++ ************************************************************************ ++ * ++ * OLD ISO/LAME routines follow. Used if you dont have LIBSNDFILE ++ * or for stdin/stdout support ++ * ++ ************************************************************************ ++ ************************************************************************ ++ ************************************************************************ ++ ************************************************************************ ++ ************************************************************************ ++ ************************************************************************/ ++ ++ /* Replacement for forward fseek(,,SEEK_CUR), because fseek() fails on pipes */ ++ int fskip(FILE *sf,long num_bytes,int dummy) ++ { ++ char data[1024]; ++ int nskip = 0; ++ while (num_bytes > 0) { ++ nskip = (num_bytes>1024) ? 1024 : num_bytes; ++ num_bytes -= fread(data,(size_t)1,(size_t)nskip,sf); ++ } ++ /* return 0 if last read was successful */ ++ return num_bytes; ++ } ++ ++ ++ ++ void CloseSndFile(lame_global_flags *gfp) ++ { ++ if (fclose(musicin) != 0){ ++ fprintf(stderr, "Could not close audio input file\n"); ++ exit(2); ++ } ++ } ++ ++ ++ unsigned long GetSndSamples(void) ++ { ++ return num_samples; ++ } ++ int GetSndSampleRate(void) ++ { ++ return samp_freq; ++ } ++ ++ int GetSndChannels(void) ++ { ++ return num_channels; ++ } ++ ++ ++ FILE * OpenSndFile(lame_global_flags *gfp,const char* inPath, int default_samp, ++ int default_channels) ++ { ++ struct stat sb; ++ void parse_file_header(lame_global_flags *gfp,FILE *sf); ++ /* set the defaults from info incase we cannot determine them from file */ ++ num_samples=MAX_U_32_NUM; ++ samp_freq=default_samp; ++ num_channels = default_channels; ++ ++ if (!strcmp(inPath, "-")) { ++ /* Read from standard input. */ ++ #ifdef __EMX__ ++ _fsetmode(stdin,"b"); ++ #elif (defined __BORLANDC__) ++ setmode(_fileno(stdin), O_BINARY); ++ #elif (defined __CYGWIN__) ++ setmode(fileno(stdin), _O_BINARY); ++ #elif (defined _WIN32) ++ _setmode(_fileno(stdin), _O_BINARY); ++ #endif ++ musicin = stdin; ++ } else { ++ if ((musicin = fopen(inPath, "rb")) == NULL) { ++ fprintf(stderr, "Could not find \"%s\".\n", inPath); ++ exit(1); ++ } ++ } ++ ++ input_bitrate=0; ++ if (gfp->input_format==sf_mp3) { ++ #ifdef AMIGA_MPEGA ++ if (-1==lame_decode_initfile(inPath,&num_channels,&samp_freq,&input_bitrate,&num_samples)) { ++ fprintf(stderr,"Error reading headers in mp3 input file %s.\n", inPath); ++ exit(1); ++ } ++ #endif ++ #ifdef HAVEMPGLIB ++ if (-1==lame_decode_initfile(musicin,&num_channels,&samp_freq,&input_bitrate,&num_samples)) { ++ fprintf(stderr,"Error reading headers in mp3 input file %s.\n", inPath); ++ exit(1); ++ } ++ #endif ++ }else{ ++ if (gfp->input_format != sf_raw) { ++ parse_file_header(gfp,musicin); ++ } ++ ++ if (gfp->input_format==sf_raw) { ++ /* assume raw PCM */ ++ fprintf(stderr, "Assuming raw pcm input file"); ++ if (gfp->swapbytes==TRUE) ++ fprintf(stderr, " : Forcing byte-swapping\n"); ++ else ++ fprintf(stderr, "\n"); ++ } ++ } ++ ++ if (num_samples==MAX_U_32_NUM && musicin != stdin) { ++ /* try to figure out num_samples */ ++ if (0==stat(inPath,&sb)) { ++ /* try file size, assume 2 bytes per sample */ ++ if (gfp->input_format == sf_mp3) { ++ FLOAT totalseconds = (sb.st_size*8.0/(1000.0*GetSndBitrate())); ++ num_samples= totalseconds*GetSndSampleRate(); ++ }else{ ++ num_samples = sb.st_size/(2*GetSndChannels()); ++ } ++ } ++ } ++ return musicin; ++ } ++ ++ ++ /************************************************************************ ++ * ++ * read_samples() ++ * ++ * PURPOSE: reads the PCM samples from a file to the buffer ++ * ++ * SEMANTICS: ++ * Reads #samples_read# number of shorts from #musicin# filepointer ++ * into #sample_buffer[]#. Returns the number of samples read. ++ * ++ ************************************************************************/ ++ ++ int read_samples_pcm(lame_global_flags *gfp,short sample_buffer[2304], int frame_size,int samples_to_read) ++ { ++ int samples_read; ++ int rcode; ++ int iswav=(gfp->input_format==sf_wave); ++ ++ samples_read = fread(sample_buffer, sizeof(short), samples_to_read, musicin); ++ if (ferror(musicin)) { ++ fprintf(stderr, "Error reading input file\n"); ++ exit(2); ++ } ++ ++ /* ++ Samples are big-endian. If this is a little-endian machine ++ we must swap ++ */ ++ if ( NativeByteOrder == order_unknown ) ++ { ++ NativeByteOrder = DetermineByteOrder(); ++ if ( NativeByteOrder == order_unknown ) ++ { ++ fprintf( stderr, "byte order not determined\n" ); ++ exit( 1 ); ++ } ++ } ++ /* intel=littleEndian */ ++ if (!iswav && ( NativeByteOrder == order_littleEndian )) ++ SwapBytesInWords( sample_buffer, samples_read ); ++ ++ if (iswav && ( NativeByteOrder == order_bigEndian )) ++ SwapBytesInWords( sample_buffer, samples_read ); ++ ++ if (gfp->swapbytes==TRUE) ++ SwapBytesInWords( sample_buffer, samples_read ); ++ ++ ++ rcode=samples_read; ++ if (samples_read < frame_size) { ++ if (samples_read<0) samples_read=0; ++ /*fprintf(stderr,"Insufficient PCM input for one frame - fillout with zeros\n"); ++ */ ++ for (; samples_read < frame_size; sample_buffer[samples_read++] = 0); ++ } ++ return(rcode); ++ } ++ ++ ++ ++ #define WAV_ID_RIFF 0x52494646 /* "RIFF" */ ++ #define WAV_ID_WAVE 0x57415645 /* "WAVE" */ ++ #define WAV_ID_FMT 0x666d7420 /* "fmt " */ ++ #define WAV_ID_DATA 0x64617461 /* "data" */ ++ ++ typedef struct fmt_chunk_data_struct { ++ short format_tag; /* Format category */ ++ u_short channels; /* Number of channels */ ++ u_long samples_per_sec; /* Sampling rate */ ++ u_long avg_bytes_per_sec; /* For buffer estimation */ ++ u_short block_align; /* Data block size */ ++ u_short bits_per_sample; /* for PCM data, anyway... */ ++ } fmt_chunk_data; ++ ++ ++ ++ ++ ++ ++ /************************************************************************ ++ * ++ * wave_check ++ * ++ * PURPOSE: Checks Wave header information to make sure it is valid. ++ * Exits if not. ++ * ++ ************************************************************************/ ++ ++ static void ++ wave_check(char *file_name, fmt_chunk_data *wave_info) ++ { ++ if (wave_info->bits_per_sample != 16) { ++ fprintf(stderr, "%d-bit sample-size is not supported!\n", ++ wave_info->bits_per_sample); ++ exit(1); ++ } ++ } ++ ++ ++ /***************************************************************************** ++ * ++ * Read Microsoft Wave headers ++ * ++ * By the time we get here the first 32-bits of the file have already been ++ * read, and we're pretty sure that we're looking at a WAV file. ++ * ++ *****************************************************************************/ ++ ++ static int ++ parse_wave_header(FILE *sf) ++ { ++ fmt_chunk_data wave_info; ++ int is_wav = 0; ++ long data_length = 0, file_length, subSize = 0; ++ int loop_sanity = 0; ++ ++ memset(&wave_info, 0, sizeof(wave_info)); ++ ++ file_length = Read32BitsHighLow(sf); ++ ++ if (Read32BitsHighLow(sf) != WAV_ID_WAVE) ++ return 0; ++ ++ for (loop_sanity = 0; loop_sanity < 20; ++loop_sanity) { ++ u_int type = Read32BitsHighLow(sf); ++ ++ if (type == WAV_ID_FMT) { ++ subSize = Read32BitsLowHigh(sf); ++ if (subSize < 16) { ++ /*fprintf(stderr, ++ "'fmt' chunk too short (only %ld bytes)!", subSize); */ ++ return 0; ++ } ++ ++ wave_info.format_tag = Read16BitsLowHigh(sf); ++ subSize -= 2; ++ wave_info.channels = Read16BitsLowHigh(sf); ++ subSize -= 2; ++ wave_info.samples_per_sec = Read32BitsLowHigh(sf); ++ subSize -= 4; ++ wave_info.avg_bytes_per_sec = Read32BitsLowHigh(sf); ++ subSize -= 4; ++ wave_info.block_align = Read16BitsLowHigh(sf); ++ subSize -= 2; ++ wave_info.bits_per_sample = Read16BitsLowHigh(sf); ++ subSize -= 2; ++ ++ /* fprintf(stderr, " skipping %d bytes\n", subSize); */ ++ ++ if (subSize > 0) { ++ if (fskip(sf, (long)subSize, SEEK_CUR) != 0 ) ++ return 0; ++ }; ++ ++ } else if (type == WAV_ID_DATA) { ++ subSize = Read32BitsLowHigh(sf); ++ data_length = subSize; ++ is_wav = 1; ++ /* We've found the audio data. Read no further! */ ++ break; ++ ++ } else { ++ subSize = Read32BitsLowHigh(sf); ++ if (fskip(sf, (long) subSize, SEEK_CUR) != 0 ) return 0; ++ } ++ } ++ ++ if (is_wav) { ++ /* make sure the header is sane */ ++ wave_check("name", &wave_info); ++ ++ num_channels = wave_info.channels; ++ samp_freq = wave_info.samples_per_sec; ++ num_samples = data_length / (wave_info.channels * wave_info.bits_per_sample / 8); ++ } ++ return is_wav; ++ } ++ ++ ++ ++ /************************************************************************ ++ * ++ * aiff_check ++ * ++ * PURPOSE: Checks AIFF header information to make sure it is valid. ++ * Exits if not. ++ * ++ ************************************************************************/ ++ ++ static void ++ aiff_check2(const char *file_name, IFF_AIFF *pcm_aiff_data) ++ { ++ if (pcm_aiff_data->sampleType != IFF_ID_SSND) { ++ fprintf(stderr, "Sound data is not PCM in \"%s\".\n", file_name); ++ exit(1); ++ } ++ ++ if (pcm_aiff_data->sampleSize != sizeof(short) * BITS_IN_A_BYTE) { ++ fprintf(stderr, "Sound data is not %d bits in \"%s\".\n", ++ (unsigned int) sizeof(short) * BITS_IN_A_BYTE, file_name); ++ exit(1); ++ } ++ ++ if (pcm_aiff_data->numChannels != 1 && ++ pcm_aiff_data->numChannels != 2) { ++ fprintf(stderr, "Sound data is not mono or stereo in \"%s\".\n", ++ file_name); ++ exit(1); ++ } ++ ++ if (pcm_aiff_data->blkAlgn.blockSize != 0) { ++ fprintf(stderr, "Block size is not %d bytes in \"%s\".\n", ++ 0, file_name); ++ exit(1); ++ } ++ ++ if (pcm_aiff_data->blkAlgn.offset != 0) { ++ fprintf(stderr, "Block offset is not %d bytes in \"%s\".\n", ++ 0, file_name); ++ exit(1); ++ } ++ } ++ ++ /***************************************************************************** ++ * ++ * Read Audio Interchange File Format (AIFF) headers. ++ * ++ * By the time we get here the first 32-bits of the file have already been ++ * read, and we're pretty sure that we're looking at an AIFF file. ++ * ++ *****************************************************************************/ ++ ++ static int ++ parse_aiff_header(FILE *sf) ++ { ++ int is_aiff = 0; ++ long chunkSize = 0, subSize = 0; ++ IFF_AIFF aiff_info; ++ ++ memset(&aiff_info, 0, sizeof(aiff_info)); ++ chunkSize = Read32BitsHighLow(sf); ++ ++ if ( Read32BitsHighLow(sf) != IFF_ID_AIFF ) ++ return 0; ++ ++ while ( chunkSize > 0 ) ++ { ++ u_int type = 0; ++ chunkSize -= 4; ++ ++ type = Read32BitsHighLow(sf); ++ ++ /* fprintf(stderr, ++ "found chunk type %08x '%4.4s'\n", type, (char*)&type); */ ++ ++ /* don't use a switch here to make it easier to use 'break' for SSND */ ++ if (type == IFF_ID_COMM) { ++ subSize = Read32BitsHighLow(sf); ++ chunkSize -= subSize; ++ ++ aiff_info.numChannels = Read16BitsHighLow(sf); ++ subSize -= 2; ++ aiff_info.numSampleFrames = Read32BitsHighLow(sf); ++ subSize -= 4; ++ aiff_info.sampleSize = Read16BitsHighLow(sf); ++ subSize -= 2; ++ aiff_info.sampleRate = ReadIeeeExtendedHighLow(sf); ++ subSize -= 10; ++ ++ if (fskip(sf, (long) subSize, SEEK_CUR) != 0 ) ++ return 0; ++ ++ } else if (type == IFF_ID_SSND) { ++ subSize = Read32BitsHighLow(sf); ++ chunkSize -= subSize; ++ ++ aiff_info.blkAlgn.offset = Read32BitsHighLow(sf); ++ subSize -= 4; ++ aiff_info.blkAlgn.blockSize = Read32BitsHighLow(sf); ++ subSize -= 4; ++ ++ if (fskip(sf, aiff_info.blkAlgn.offset, SEEK_CUR) != 0 ) ++ return 0; ++ ++ aiff_info.sampleType = IFF_ID_SSND; ++ is_aiff = 1; ++ ++ /* We've found the audio data. Read no further! */ ++ break; ++ ++ } else { ++ subSize = Read32BitsHighLow(sf); ++ chunkSize -= subSize; ++ ++ if (fskip(sf, (long) subSize, SEEK_CUR) != 0 ) ++ return 0; ++ } ++ } ++ ++ /* fprintf(stderr, "Parsed AIFF %d\n", is_aiff); */ ++ if (is_aiff) { ++ /* make sure the header is sane */ ++ aiff_check2("name", &aiff_info); ++ num_channels = aiff_info.numChannels; ++ samp_freq = aiff_info.sampleRate; ++ num_samples = aiff_info.numSampleFrames; ++ } ++ return is_aiff; ++ } ++ ++ ++ ++ /************************************************************************ ++ * ++ * parse_file_header ++ * ++ * PURPOSE: Read the header from a bytestream. Try to determine whether ++ * it's a WAV file or AIFF without rewinding, since rewind ++ * doesn't work on pipes and there's a good chance we're reading ++ * from stdin (otherwise we'd probably be using libsndfile). ++ * ++ * When this function returns, the file offset will be positioned at the ++ * beginning of the sound data. ++ * ++ ************************************************************************/ ++ ++ void parse_file_header(lame_global_flags *gfp,FILE *sf) ++ { ++ u_int type = 0; ++ type = Read32BitsHighLow(sf); ++ ++ /* fprintf(stderr, ++ "First word of input stream: %08x '%4.4s'\n", type, (char*) &type); */ ++ ++ count_samples_carefully=0; ++ gfp->input_format = sf_raw; ++ ++ if (type == WAV_ID_RIFF) { ++ /* It's probably a WAV file */ ++ if (parse_wave_header(sf)) { ++ gfp->input_format = sf_wave; ++ count_samples_carefully=1; ++ } ++ ++ } else if (type == IFF_ID_FORM) { ++ /* It's probably an AIFF file */ ++ if (parse_aiff_header(sf)) { ++ gfp->input_format = sf_aiff; ++ count_samples_carefully=1; ++ } ++ } ++ if (gfp->input_format==sf_raw) { ++ /* ++ ** Assume it's raw PCM. Since the audio data is assumed to begin ++ ** at byte zero, this will unfortunately require seeking. ++ */ ++ if (fseek(sf, 0L, SEEK_SET) != 0) { ++ /* ignore errors */ ++ } ++ gfp->input_format = sf_raw; ++ } ++ } ++ #endif /* LAMESNDFILE */ ++ ++ +diff -r -c -N encoder/get_audio.h lame3.70/get_audio.h +*** encoder/get_audio.h Wed Dec 31 17:00:00 1969 +--- lame3.70/get_audio.h Thu Mar 16 12:05:26 2000 +*************** +*** 0 **** +--- 1,94 ---- ++ /* ++ ** Copyright (C) 1999 Albert Faber ++ ** ++ * This library is free software; you can redistribute it and/or ++ * modify it under the terms of the GNU Library General Public ++ * License as published by the Free Software Foundation; either ++ * version 2 of the License, or (at your option) any later version. ++ * ++ * This library is distributed in the hope that it will be useful, ++ * but WITHOUT ANY WARRANTY; without even the implied warranty of ++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU ++ * Library General Public License for more details. ++ * ++ * You should have received a copy of the GNU Library General Public ++ * License along with this library; if not, write to the ++ * Free Software Foundation, Inc., 59 Temple Place - Suite 330, ++ * Boston, MA 02111-1307, USA. ++ */ ++ ++ ++ #ifndef GET_AUDIO_H_INCLUDED ++ #define GET_AUDIO_H_INCLUDED ++ ++ #if (defined LIBSNDFILE || defined LAMESNDFILE) ++ ++ ++ /* AIFF Definitions */ ++ ++ #define IFF_ID_FORM 0x464f524d /* "FORM" */ ++ #define IFF_ID_AIFF 0x41494646 /* "AIFF" */ ++ #define IFF_ID_COMM 0x434f4d4d /* "COMM" */ ++ #define IFF_ID_SSND 0x53534e44 /* "SSND" */ ++ #define IFF_ID_MPEG 0x4d504547 /* "MPEG" */ ++ ++ ++ ++ void CloseSndFile(lame_global_flags *gfp); ++ FILE * OpenSndFile(lame_global_flags *gfp,const char* lpszFileName,int default_samp, int ++ default_chan); ++ unsigned long GetSndSamples(void); ++ int GetSndSampleRate(void); ++ int GetSndChannels(void); ++ int GetSndBitrate(void); ++ ++ ++ int get_audio(lame_global_flags *gfp,short buffer[2][1152],int stereo); ++ ++ ++ ++ #ifdef LIBSNDFILE ++ /* INCLUDE the sound library header file */ ++ #ifdef _MSC_VER ++ /* one byte alignment for WIN32 platforms */ ++ #pragma pack(push,1) ++ #include "./libsndfile/src/sndfile.h" ++ #pragma pack(pop,1) ++ #else ++ #include "sndfile.h" ++ #endif ++ ++ ++ #else ++ /***************************************************************** ++ * LAME/ISO built in audio file I/O routines ++ *******************************************************************/ ++ #include "portableio.h" ++ #include "ieeefloat.h" ++ ++ ++ typedef struct blockAlign_struct { ++ unsigned long offset; ++ unsigned long blockSize; ++ } blockAlign; ++ ++ typedef struct IFF_AIFF_struct { ++ short numChannels; ++ unsigned long numSampleFrames; ++ short sampleSize; ++ FLOAT sampleRate; ++ unsigned long sampleType; ++ blockAlign blkAlgn; ++ } IFF_AIFF; ++ ++ extern int aiff_read_headers(FILE*, IFF_AIFF*); ++ extern int aiff_seek_to_sound_data(FILE*); ++ extern int aiff_write_headers(FILE*, IFF_AIFF*); ++ extern int parse_wavheader(void); ++ extern int parse_aiff(const char fn[]); ++ extern void aiff_check(const char*, IFF_AIFF*, int*); ++ ++ ++ #endif /* ifndef _LIBSNDDLL */ ++ #endif /* ifdef LAMESNDFILE or LIBSNDFILE */ ++ #endif /* ifndef GET_AUDIO_H_INCLUDED*/ +diff -r -c -N encoder/gpkplotting.c lame3.70/gpkplotting.c +*** encoder/gpkplotting.c Wed Dec 31 17:00:00 1969 +--- lame3.70/gpkplotting.c Wed Nov 24 01:42:07 1999 +*************** +*** 0 **** +--- 1,282 ---- ++ #ifdef HAVEGTK ++ #include "gpkplotting.h" ++ #include "string.h" ++ ++ static gint num_plotwindows = 0; ++ static gint max_plotwindows = 10; ++ static GdkPixmap *pixmaps[10]; ++ static GtkWidget *pixmapboxes[10]; ++ ++ ++ ++ ++ /* compute a gdkcolor */ ++ void setcolor(GtkWidget *widget, GdkColor *color, gint red,gint green,gint blue) ++ { ++ ++ /* colors in GdkColor are taken from 0 to 65535, not 0 to 255. */ ++ color->red = red * (65535/255); ++ color->green = green * (65535/255); ++ color->blue = blue * (65535/255); ++ color->pixel = (gulong)(color->red*65536 + color->green*256 + color->blue); ++ /* find closest in colormap, if needed */ ++ gdk_color_alloc(gtk_widget_get_colormap(widget),color); ++ } ++ ++ ++ void gpk_redraw(GdkPixmap *pixmap, GtkWidget *pixmapbox) ++ { ++ /* redraw the entire pixmap */ ++ gdk_draw_pixmap(pixmapbox->window, ++ pixmapbox->style->fg_gc[GTK_WIDGET_STATE (pixmapbox)], ++ pixmap,0,0,0,0, ++ pixmapbox->allocation.width, ++ pixmapbox->allocation.height); ++ } ++ ++ ++ static GdkPixmap **findpixmap(GtkWidget *widget) ++ { ++ int i; ++ for (i=0; i=num_plotwindows) { ++ g_print("findpixmap(): bad argument widget \n"); ++ return NULL; ++ } ++ return &pixmaps[i]; ++ } ++ ++ void gpk_graph_draw(GtkWidget *widget, /* plot on this widged */ ++ int n, /* number of data points */ ++ gdouble *xcord, gdouble *ycord, /* data */ ++ gdouble xmn,gdouble ymn, /* coordinates of corners */ ++ gdouble xmx,gdouble ymx, ++ int clear, /* clear old plot first */ ++ char *title, /* add a title (only if clear=1) */ ++ GdkColor *color) ++ { ++ GdkPixmap **ppixmap; ++ GdkPoint *points; ++ int i; ++ gint16 width,height; ++ GdkFont *fixed_font; ++ GdkGC *gc; ++ ++ gc = gdk_gc_new(widget->window); ++ gdk_gc_set_foreground(gc, color); ++ ++ ++ ++ if ((ppixmap=findpixmap(widget))) { ++ width = widget->allocation.width; ++ height = widget->allocation.height; ++ ++ ++ if (clear) { ++ /* white background */ ++ gdk_draw_rectangle (*ppixmap, ++ widget->style->white_gc, ++ TRUE,0, 0,width,height); ++ /* title */ ++ #ifndef _WIN32 ++ fixed_font = gdk_font_load ("-misc-fixed-medium-r-*-*-*-100-*-*-*-*-*-*"); ++ #else ++ fixed_font = gdk_font_load ("-misc-fixed-large-r-*-*-*-100-*-*-*-*-*-*"); ++ #endif ++ ++ gdk_draw_text (*ppixmap,fixed_font, ++ widget->style->fg_gc[GTK_WIDGET_STATE (widget)], ++ 0,10,title,strlen(title)); ++ } ++ ++ ++ points = g_malloc(n*sizeof(GdkPoint)); ++ for (i=0; iwindow); ++ gdk_gc_set_foreground(gc, color); ++ ++ ++ if ((ppixmap=findpixmap(widget))) { ++ width = widget->allocation.width; ++ height = widget->allocation.height; ++ ++ ++ for (i=0; i<2 ; i++) { ++ points[i].x =.5+ ((xcord[i]-xmn)*(width-1)/(xmx-xmn)); ++ points[i].y =.5+ ((ycord[i]-ymx)*(height-1)/(ymn-ymx)); ++ } ++ width=points[1].x-points[0].x + 1; ++ height=points[1].y-points[0].y + 1; ++ gdk_draw_rectangle(*ppixmap,gc,TRUE, ++ points[0].x,points[0].y,width,height); ++ gpk_redraw(*ppixmap,widget); ++ } ++ gdk_gc_destroy(gc); ++ } ++ ++ ++ ++ void gpk_bargraph_draw(GtkWidget *widget, /* plot on this widged */ ++ int n, /* number of data points */ ++ gdouble *xcord, gdouble *ycord, /* data */ ++ gdouble xmn,gdouble ymn, /* coordinates of corners */ ++ gdouble xmx,gdouble ymx, ++ int clear, /* clear old plot first */ ++ char *title, /* add a title (only if clear=1) */ ++ int barwidth, /* bar width. 0=compute based on window size */ ++ GdkColor *color) ++ { ++ GdkPixmap **ppixmap; ++ GdkPoint points[2]; ++ int i; ++ gint16 width,height,x,y,barheight; ++ GdkFont *fixed_font; ++ GdkGC *gc; ++ ++ ++ gc = gdk_gc_new(widget->window); ++ gdk_gc_set_foreground(gc, color); ++ ++ ++ if ((ppixmap=findpixmap(widget))) { ++ width = widget->allocation.width; ++ height = widget->allocation.height; ++ ++ ++ if (clear) { ++ /* white background */ ++ gdk_draw_rectangle (*ppixmap, ++ widget->style->white_gc, ++ TRUE,0, 0,width,height); ++ /* title */ ++ #ifndef _WIN32 ++ fixed_font = gdk_font_load ("-misc-fixed-medium-r-*-*-*-100-*-*-*-*-*-*"); ++ #else ++ fixed_font = gdk_font_load ("-misc-fixed-large-r-*-*-*-100-*-*-*-*-*-*"); ++ #endif ++ ++ gdk_draw_text (*ppixmap,fixed_font, ++ widget->style->fg_gc[GTK_WIDGET_STATE (widget)], ++ 0,10,title,strlen(title)); ++ } ++ ++ ++ for (i=0; i 5 ? 5 : barwidth; ++ barwidth = barwidth < 1 ? 1 : barwidth; ++ barheight = height-1 - y; ++ /* gdk_draw_lines(*ppixmap,gc,points,2); */ ++ gdk_draw_rectangle(*ppixmap,gc,TRUE,x,y,barwidth,barheight); ++ ++ } ++ gpk_redraw(*ppixmap,widget); ++ } ++ gdk_gc_destroy(gc); ++ } ++ ++ ++ ++ ++ ++ /* Create a new backing pixmap of the appropriate size */ ++ static gint ++ configure_event (GtkWidget *widget, GdkEventConfigure *event, gpointer data) ++ { ++ GdkPixmap **ppixmap; ++ if ((ppixmap=findpixmap(widget))){ ++ if (*ppixmap) gdk_pixmap_unref(*ppixmap); ++ *ppixmap = gdk_pixmap_new(widget->window, ++ widget->allocation.width, ++ widget->allocation.height, ++ -1); ++ gdk_draw_rectangle (*ppixmap, ++ widget->style->white_gc, ++ TRUE, ++ 0, 0, ++ widget->allocation.width, ++ widget->allocation.height); ++ } ++ return TRUE; ++ } ++ ++ ++ ++ /* Redraw the screen from the backing pixmap */ ++ static gint ++ expose_event (GtkWidget *widget, GdkEventExpose *event, gpointer data) ++ { ++ GdkPixmap **ppixmap; ++ if ((ppixmap=findpixmap(widget))){ ++ gdk_draw_pixmap(widget->window, ++ widget->style->fg_gc[GTK_WIDGET_STATE (widget)], ++ *ppixmap, ++ event->area.x, event->area.y, ++ event->area.x, event->area.y, ++ event->area.width, event->area.height); ++ } ++ ++ return FALSE; ++ } ++ ++ ++ ++ ++ ++ GtkWidget *gpk_plot_new(int width, int height) ++ { ++ GtkWidget *pixmapbox; ++ ++ pixmapbox = gtk_drawing_area_new(); ++ gtk_drawing_area_size(GTK_DRAWING_AREA(pixmapbox),width,height); ++ gtk_signal_connect (GTK_OBJECT (pixmapbox), "expose_event", ++ (GtkSignalFunc) expose_event, NULL); ++ gtk_signal_connect (GTK_OBJECT(pixmapbox),"configure_event", ++ (GtkSignalFunc) configure_event, NULL); ++ gtk_widget_set_events (pixmapbox, GDK_EXPOSURE_MASK); ++ ++ if (num_plotwindows < max_plotwindows) { ++ pixmapboxes[num_plotwindows] = pixmapbox; ++ pixmaps[num_plotwindows] = NULL; ++ num_plotwindows ++; ++ } else { ++ g_print("gtk_plotarea_new(): exceeded maximum of 10 plotarea windows\n"); ++ } ++ ++ return pixmapbox; ++ } ++ ++ ++ #endif +diff -r -c -N encoder/gpkplotting.h lame3.70/gpkplotting.h +*** encoder/gpkplotting.h Wed Dec 31 17:00:00 1969 +--- lame3.70/gpkplotting.h Wed Nov 24 01:42:08 1999 +*************** +*** 0 **** +--- 1,27 ---- ++ #include ++ ++ /* allocate a graphing widget */ ++ GtkWidget *gpk_plot_new(int width,int height); ++ ++ /* graph a function in the graphing widged */ ++ void gpk_graph_draw(GtkWidget *widget, ++ int n, gdouble *xcord, gdouble *ycord, ++ gdouble xmn, gdouble ymn,gdouble xmx,gdouble ymx, ++ int clear, char * title,GdkColor *color); ++ ++ /* draw a rectangle in the graphing widget */ ++ void gpk_rectangle_draw(GtkWidget *widget, /* plot on this widged */ ++ gdouble xcord[2], gdouble ycord[2], /* corners */ ++ gdouble xmn,gdouble ymn, /* coordinates of corners */ ++ gdouble xmx,gdouble ymx, ++ GdkColor *color); /* color to use */ ++ ++ /* make a bar graph in the graphing widged */ ++ void gpk_bargraph_draw(GtkWidget *widget, ++ int n, gdouble *xcord, gdouble *ycord, ++ gdouble xmn, gdouble ymn,gdouble xmx,gdouble ymx, ++ int clear, char * title, int bwidth,GdkColor *color); ++ ++ /* set forground color */ ++ void setcolor(GtkWidget *widget, GdkColor *color,int red,int green,int blue); ++ +diff -r -c -N encoder/gtkanal.c lame3.70/gtkanal.c +*** encoder/gtkanal.c Wed Dec 31 17:00:00 1969 +--- lame3.70/gtkanal.c Thu Apr 6 12:50:49 2000 +*************** +*** 0 **** +--- 1,1410 ---- ++ #ifdef HAVEGTK ++ #include ++ #include ++ #include "gpkplotting.h" ++ #include "util.h" ++ #include "gtkanal.h" ++ #include "version.h" ++ #include "lame.h" ++ #include "tables.h" ++ #include "quantize-pvt.h" ++ #include ++ ++ int gtkflag; ++ ++ extern int makeframe(void); ++ ++ /* global variables for the state of the system */ ++ static gint idle_keepgoing; /* processing of frames is ON */ ++ static gint idle_count_max; /* number of frames to process before plotting */ ++ static gint idle_count; /* pause & plot when idle_count=idel_count_max */ ++ static gint idle_end=0; /* process all frames, stop at last frame */ ++ static gint idle_back = 0; /* set when we are displaying the old data */ ++ static int mp3done = 0; /* last frame has been read */ ++ static GtkWidget *frameprogress; /* progress bar */ ++ static GtkWidget *framecounter; /* progress counter */ ++ ++ static int subblock_draw[3] = { 1, 1, 1 }; ++ ++ /* main window */ ++ GtkWidget *window; ++ /* Backing pixmap for drawing areas */ ++ GtkWidget *pcmbox; /* PCM data plotted here */ ++ GtkWidget *winbox; /* mpg123 synthesis data plotted here */ ++ GtkWidget *enerbox[2]; /* spectrum, gr=0,1 plotted here */ ++ GtkWidget *mdctbox[2]; /* mdct coefficients gr=0,1 plotted here */ ++ GtkWidget *sfbbox[2]; /* scalefactors gr=0,1 plotted here */ ++ GtkWidget *headerbox; /* mpg123 header info shown here */ ++ ++ plotting_data *pinfo,*pplot; ++ plotting_data Pinfo[NUMPINFO]; ++ ++ struct gtkinfostruct { ++ int filetype; /* input file type 0=WAV, 1=MP3 */ ++ int msflag; /* toggle between L&R vs M&S PCM data display */ ++ int chflag; /* toggle between L & R channels */ ++ int kbflag; /* toggle between wave # and barks */ ++ int flag123; /* show mpg123 frame info, OR ISO encoder frame info */ ++ double avebits; /* running average bits per frame */ ++ int approxbits; /* (approx) bits per frame */ ++ int maxbits; /* max bits per frame used so far*/ ++ int totemph; /* total of frames with de-emphasis */ ++ int totms; /* total frames with ms_stereo */ ++ int totis; /* total frames with i_stereo */ ++ int totshort; /* total granules with short blocks */ ++ int totmix; /* total granules with mixed blocks */ ++ int pupdate; /* plot while processing, or only when needed */ ++ int sfblines; /* plot scalefactor bands in MDCT plot */ ++ int totalframes; ++ } gtkinfo; ++ ++ ++ static lame_global_flags *gfp; ++ ++ /********************************************************************** ++ * read one frame and encode it ++ **********************************************************************/ ++ int gtkmakeframe(void) ++ { ++ int iread = 0; ++ static int init=0; ++ static int mpglag; ++ static short int Buffer[2][1152]; ++ int ch,j; ++ int mp3count = 0; ++ int mp3out = 0; ++ short mpg123pcm[2][1152]; ++ char mp3buffer[LAME_MAXMP3BUFFER]; ++ ++ ++ #ifndef HAVEMPGLIB ++ fprintf(stderr,"Error: GTK frame analyzer requires MPGLIB\n"); ++ exit(1); ++ #else ++ /* even if iread=0, get_audio hit EOF and returned Buffer=all 0's. encode ++ * and decode to flush any previous buffers from the decoder */ ++ ++ pinfo->frameNum = gfp->frameNum; ++ pinfo->sampfreq=gfp->out_samplerate; ++ pinfo->framesize=576*gfp->mode_gr; ++ pinfo->stereo = gfp->stereo; ++ ++ if (gfp->input_format == sf_mp3) { ++ iread=lame_readframe(gfp,Buffer); ++ gfp->frameNum++; ++ }else { ++ while (gfp->frameNum == pinfo->frameNum) { ++ if (gfp->frameNum==0 && !init) { ++ mpglag=1; ++ lame_decode_init(); ++ } ++ if (gfp->frameNum==1) init=0; /* reset for next time frameNum==0 */ ++ iread=lame_readframe(gfp,Buffer); ++ ++ ++ mp3count=lame_encode(gfp,Buffer,mp3buffer,sizeof(mp3buffer)); /* encode frame */ ++ assert( !(mp3count > 0 && gfp->frameNum == pinfo->frameNum)); ++ /* not possible to produce mp3 data without encoding at least ++ * one frame of data which would increment gfp->frameNum */ ++ } ++ mp3out=lame_decode(mp3buffer,mp3count,mpg123pcm[0],mpg123pcm[1]); /* re-synthesis to pcm */ ++ /* mp3out = 0: need more data to decode */ ++ /* mp3out = -1: error. Lets assume 0 pcm output */ ++ /* mp3out = number of samples output */ ++ if (mp3out>0) assert(mp3out==pinfo->framesize); ++ if (mp3out!=0) { ++ /* decoded output is for frame pinfo->frameNum123 ++ * add a delay of framesize-DECDELAY, which will make the total delay ++ * exactly one frame */ ++ pinfo->frameNum123=pinfo->frameNum-mpglag; ++ for ( ch = 0; ch < pinfo->stereo; ch++ ) { ++ for ( j = 0; j < pinfo->framesize-DECDELAY; j++ ) ++ pinfo->pcmdata2[ch][j] = pinfo->pcmdata2[ch][j+pinfo->framesize]; ++ for ( j = 0; j < pinfo->framesize; j++ ) { ++ pinfo->pcmdata2[ch][j+pinfo->framesize-DECDELAY] = ++ (mp3out==-1) ? 0 : mpg123pcm[ch][j]; ++ } ++ } ++ }else{ ++ if (mpglag == MAXMPGLAG) { ++ fprintf(stderr,"READ_AHEAD set too low - not enough frame buffering.\n"); ++ fprintf(stderr,"MP3x display of input and output PCM data out of sync.\n"); ++ } ++ else mpglag++; ++ pinfo->frameNum123=-1; /* no frame output */ ++ } ++ } ++ #endif ++ return iread; ++ } ++ ++ ++ void plot_frame(void) ++ { ++ int i,j,n,ch,gr; ++ gdouble *xcord,*ycord; ++ gdouble xmx,xmn,ymx,ymn; ++ double *data,*data2,*data3; ++ char title2[80]; ++ char label[80],label2[80]; ++ char *title; ++ plotting_data *pplot1; ++ plotting_data *pplot2 = NULL; ++ ++ double en,samp; ++ int sampindex,version=0; ++ static int firstcall=1; ++ static GdkColor *barcolor,*color,*grcolor[2]; ++ static GdkColor yellow,gray,cyan,magenta,orange,pink,red,green,blue,black,oncolor,offcolor; ++ int blocktype[2][2]; ++ int headbits; ++ int mode_gr = 2; ++ ++ /* find the frame where mpg123 produced output coming from input frame ++ * pinfo. i.e.: out_frame + out_frame_lag = input_frame */ ++ for (i=1; i<=MAXMPGLAG; i++ ) { ++ if ((pplot-i)->frameNum123 == pplot->frameNum ) { ++ pplot2 = pplot-i; ++ break; ++ } ++ } ++ if (i > MAXMPGLAG) { ++ fprintf(stderr,"input/output pcm syncing problem. should not happen!\n"); ++ pplot2=pplot-1; ++ } ++ ++ ++ /* however, the PCM data is delayed by 528 samples in the encoder filterbanks. ++ * We added another 1152-528 delay to this so the PCM data is *exactly* one ++ * frame behind the header & MDCT information */ ++ pplot1 =pplot2 +1; /* back one frame for header info, MDCT */ ++ ++ /* allocate these GC's only once */ ++ if (firstcall) { ++ firstcall=0; ++ /* grcolor[0] = &magenta; */ ++ grcolor[0] = &blue; ++ grcolor[1] = &green; ++ barcolor = &gray; ++ ++ setcolor(headerbox,&oncolor,255,0,0); ++ setcolor(headerbox,&offcolor,175,175,175); ++ setcolor(pcmbox,&red,255,0,0); ++ setcolor(pcmbox,&pink,255,0,255); ++ setcolor(pcmbox,&magenta,255,0,100); ++ setcolor(pcmbox,&orange,255,127,0); ++ setcolor(pcmbox,&cyan,0,255,255); ++ setcolor(pcmbox,&green,0,255,0); ++ setcolor(pcmbox,&blue,0,0,255); ++ setcolor(pcmbox,&black,0,0,0); ++ setcolor(pcmbox,&gray,100,100,100); ++ setcolor(pcmbox,&yellow,255,255,0); ++ ++ } ++ ++ /******************************************************************* ++ * frame header info ++ *******************************************************************/ ++ if (pplot1->sampfreq) ++ samp=pplot1->sampfreq; ++ else samp=1; ++ sampindex = SmpFrqIndex((long)samp, &version); ++ ++ ch = gtkinfo.chflag; ++ ++ headbits = 32 + ((pplot1->stereo==2) ? 256 : 136); ++ gtkinfo.approxbits = (pplot1->bitrate*1000*1152.0/samp) - headbits; ++ /*font = gdk_font_load ("-misc-fixed-medium-r-*-*-*-100-*-*-*-*-*-*");*/ ++ sprintf(title2,"%3.1fkHz %ikbs ",samp/1000,pplot1->bitrate); ++ gtk_text_freeze (GTK_TEXT(headerbox)); ++ gtk_text_backward_delete(GTK_TEXT(headerbox), ++ gtk_text_get_length(GTK_TEXT(headerbox))); ++ gtk_text_set_point(GTK_TEXT(headerbox),0); ++ gtk_text_insert(GTK_TEXT(headerbox),NULL,&oncolor,NULL,title2, -1); ++ title = " mono "; ++ if (2==pplot1->stereo) title = pplot1->js ? " js " : " s "; ++ gtk_text_insert (GTK_TEXT(headerbox), NULL, &oncolor, NULL,title, -1); ++ color = pplot1->ms_stereo ? &oncolor : &offcolor ; ++ gtk_text_insert (GTK_TEXT(headerbox), NULL, color, NULL,"ms ", -1); ++ color = pplot1->i_stereo ? &oncolor : &offcolor ; ++ gtk_text_insert (GTK_TEXT(headerbox), NULL, color, NULL,"is ", -1); ++ ++ color = pplot1->crc ? &oncolor : &offcolor ; ++ gtk_text_insert (GTK_TEXT(headerbox), NULL, color, NULL,"crc ", -1); ++ color = pplot1->padding ? &oncolor : &offcolor ; ++ gtk_text_insert (GTK_TEXT(headerbox), NULL, color, NULL,"pad ", -1); ++ ++ color = pplot1->emph ? &oncolor : &offcolor ; ++ gtk_text_insert (GTK_TEXT(headerbox), NULL, color, NULL,"em ", -1); ++ ++ sprintf(title2,"c1=%i,%i ",pplot1->big_values[0][ch],pplot1->big_values[1][ch]); ++ gtk_text_insert (GTK_TEXT(headerbox), NULL, &black, NULL,title2, -1); ++ ++ color = pplot1->scfsi[ch] ? &oncolor : &offcolor ; ++ sprintf(title2,"scfsi=%i ",pplot1->scfsi[ch]); ++ gtk_text_insert (GTK_TEXT(headerbox), NULL, color, NULL,title2, -1); ++ if (gtkinfo.filetype) ++ sprintf(title2," mdb=%i %i/NA",pplot1->maindata,pplot1->totbits); ++ else ++ sprintf(title2," mdb=%i %i/%i", ++ pplot1->maindata,pplot1->totbits,pplot->resvsize); ++ gtk_text_insert (GTK_TEXT(headerbox), NULL, &oncolor, NULL,title2, -1); ++ gtk_text_thaw (GTK_TEXT(headerbox)); ++ ++ ++ ++ /******************************************************************* ++ * block type ++ *******************************************************************/ ++ for (gr = 0 ; gr < mode_gr ; gr ++) ++ if (gtkinfo.flag123) ++ blocktype[gr][ch]=pplot1->mpg123blocktype[gr][ch]; ++ else blocktype[gr][ch]=pplot->blocktype[gr][ch]; ++ ++ ++ /******************************************************************* ++ * draw the PCM data * ++ *******************************************************************/ ++ n = 1600; /* PCM frame + FFT window: 224 + 1152 + 224 */ ++ xcord = g_malloc(n*sizeof(gdouble)); ++ ycord = g_malloc(n*sizeof(gdouble)); ++ ++ ++ if (gtkinfo.msflag) ++ title=ch ? "Side Channel" : "Mid Channel"; ++ else ++ title=ch ? "Right Channel" : "Left Channel"; ++ ++ sprintf(title2,"%s mask_ratio=%3.2f %3.2f ener_ratio=%3.2f %3.2f", ++ title, ++ pplot->ms_ratio[0],pplot->ms_ratio[1], ++ pplot->ms_ener_ratio[0],pplot->ms_ener_ratio[1]); ++ ++ ++ ymn = -32767 ; ++ ymx = 32767; ++ xmn = 0; ++ xmx = 1600-1; ++ ++ /* 0 ... 224 draw in black, connecting to 224 pixel ++ * 1375 .. 1599 draw in black connecting to 1375 pixel ++ * 224 ... 1375 MP3 frame. draw in blue ++ */ ++ ++ /* draw the title */ ++ gpk_graph_draw(pcmbox,0,xcord,ycord,xmn,ymn,xmx,ymx,1,title2, ++ &black); ++ ++ ++ /* draw some hash marks dividing the frames */ ++ ycord[0] = ymx*.8; ycord[1] = ymn*.8; ++ for (gr=0 ; gr<=2; gr++) { ++ xcord[0] = 223.5 + gr*576; xcord[1] = 223.5 +gr*576; ++ gpk_rectangle_draw(pcmbox,xcord,ycord,xmn,ymn,xmx,ymx,&yellow); ++ } ++ for (gr = 0 ; gr < mode_gr ; gr++) { ++ if (blocktype[gr][ch]==2) ++ for (i=1 ; i<=2; i++) { ++ xcord[0] = 223.5+gr*576 + i*192; ++ xcord[1] = 223.5+gr*576 + i*192; ++ gpk_rectangle_draw(pcmbox,xcord,ycord,xmn,ymn,xmx,ymx,&yellow); ++ } ++ } ++ /* bars representing FFT windows */ ++ xcord[0] = 0; ycord[0] = ymn+3000; ++ xcord[1] = 1024-1; ycord[1] = ymn+1000; ++ gpk_rectangle_draw(pcmbox,xcord,ycord,xmn,ymn,xmx,ymx,grcolor[0]); ++ xcord[0] = 576; ycord[0] = ymn+2000; ++ xcord[1] = 576+1024-1; ycord[1] = ymn; ++ gpk_rectangle_draw(pcmbox,xcord,ycord,xmn,ymn,xmx,ymx,grcolor[1]); ++ ++ ++ /* plot PCM data */ ++ for (i=0; ipcmdata[0][i]-pplot->pcmdata[1][i]) : ++ .5*(pplot->pcmdata[0][i]+pplot->pcmdata[1][i]); ++ else ++ ycord[i]=pplot->pcmdata[ch][i]; ++ } ++ ++ /* skip plot if we are doing an mp3 file */ ++ if (!gtkinfo.filetype) { ++ n = 224; /* number of points on end of blue part */ ++ /* data left of frame */ ++ gpk_graph_draw(pcmbox,n+1,xcord,ycord,xmn,ymn,xmx,ymx,0,title2,&black); ++ /* data right of frame */ ++ gpk_graph_draw(pcmbox,n+1,&xcord[1152+n-1],&ycord[1152+n-1], ++ xmn,ymn,xmx,ymx,0,title2,&black); ++ /* the actual frame */ ++ gpk_graph_draw(pcmbox,1152,&xcord[n],&ycord[n],xmn,ymn,xmx,ymx,0,title2,&black); ++ } ++ ++ ++ /*******************************************************************/ ++ /* draw the PCM re-synthesis data */ ++ /*******************************************************************/ ++ n = 1152; ++ /* ++ sprintf(title2,"Re-synthesis mask_ratio=%3.2f %3.2f ener_ratio=%3.2f %3.2f", ++ pplot->ms_ratio[0],pplot->ms_ratio[1], ++ pplot->ms_ener_ratio[0],pplot->ms_ener_ratio[1]); ++ */ ++ title="Re-synthesis"; ++ ++ ++ ymn = -32767 ; ++ ymx = 32767; ++ xmn = 0; ++ xmx = 1600-1; ++ gpk_graph_draw(winbox,0,xcord,ycord, ++ xmn,ymn,xmx,ymx,1,title,&black); ++ /* draw some hash marks dividing the frames */ ++ ycord[0] = ymx*.8; ycord[1] = ymn*.8; ++ for (gr=0 ; gr<=2; gr++) { ++ xcord[0] = 223.5 + gr*576; xcord[1] = 223.5 +gr*576; ++ gpk_rectangle_draw(winbox,xcord,ycord,xmn,ymn,xmx,ymx,&yellow); ++ } ++ for (gr = 0 ; gr < 2 ; gr++) { ++ if (blocktype[gr][ch]==2) ++ for (i=1 ; i<=2; i++) { ++ xcord[0] = 223.5+gr*576 + i*192; ++ xcord[1] = 223.5+gr*576 + i*192; ++ gpk_rectangle_draw(winbox,xcord,ycord,xmn,ymn,xmx,ymx,&yellow); ++ } ++ } ++ ++ ++ ++ n = 224; ++ for (j=1152-n,i=0; i<=n; i++,j++) { ++ xcord[i] = i; ++ if (gtkinfo.msflag) ++ ycord[i] = ch ? .5*(pplot1->pcmdata2[0][j]- ++ pplot1->pcmdata2[1][j]) : ++ .5*(pplot1->pcmdata2[0][j]+pplot1->pcmdata2[1][j]); ++ else ++ ycord[i]=pplot1->pcmdata2[ch][j]; ++ } ++ gpk_graph_draw(winbox,n+1,xcord,ycord, ++ xmn,ymn,xmx,ymx,0,title,&black); ++ ++ n = 1152; ++ for (i=0; ipcmdata2[0][i]-pplot2->pcmdata2[1][i]) : ++ .5*(pplot2->pcmdata2[0][i]+pplot2->pcmdata2[1][i]); ++ else ++ ycord[i]=pplot2->pcmdata2[ch][i]; ++ } ++ gpk_graph_draw(winbox,n,xcord,ycord, ++ xmn,ymn,xmx,ymx,0,title,&black); ++ ++ ++ ++ ++ ++ /*******************************************************************/ ++ /* draw the MDCT energy spectrum */ ++ /*******************************************************************/ ++ for (gr = 0 ; gr < mode_gr ; gr ++) { ++ int bits; ++ char *blockname=""; ++ switch (blocktype[gr][ch]) { ++ case 0: blockname = "normal"; break; ++ case 1: blockname = "start"; break; ++ case 2: blockname = "short"; break; ++ case 3: blockname = "end"; break; ++ } ++ strcpy(label,blockname); ++ if (pplot1->mixed[gr][ch]) strcat(label,"(mixed)"); ++ ++ ++ ++ ++ n = 576; ++ if (gtkinfo.flag123) { ++ data = pplot1->mpg123xr[gr][0]; ++ data2 = pplot1->mpg123xr[gr][1]; ++ }else{ ++ data = pplot->xr[gr][0]; ++ data2 = pplot->xr[gr][1]; ++ } ++ ++ ++ xmn = 0; ++ xmx = n-1; ++ ymn=0; ++ ymx=11; ++ ++ /* draw title, erase old plot */ ++ if (gtkinfo.flag123) bits=pplot1->mainbits[gr][ch]; ++ else bits=pplot->LAMEmainbits[gr][ch]; ++ sprintf(title2,"MDCT%1i(%s) bits=%i q=%i ",gr,label,bits, ++ pplot1->qss[gr][ch]); ++ gpk_bargraph_draw(mdctbox[gr],0,xcord,ycord, ++ xmn,ymn,xmx,ymx,1,title2,0,barcolor); ++ ++ /* draw some hash marks showing scalefactor bands */ ++ if (gtkinfo.sfblines) { ++ int fac,nsfb, *scalefac; ++ if (blocktype[gr][ch]==SHORT_TYPE) { ++ nsfb=SBMAX_s; ++ fac=3; ++ scalefac = scalefac_band.s; ++ }else{ ++ nsfb=SBMAX_l; ++ fac=1; ++ scalefac = scalefac_band.l; ++ } ++ for (i=nsfb-7 ; i ymx) ? ycord[i] : ymx; ++ ymn=(ycord[i] < ymn) ? ycord[i] : ymn; ++ } ++ /* print the min/max ++ sprintf(title2,"MDCT%1i %5.2f %5.2f bits=%i",gr,ymn,ymx, ++ pplot1->mainbits[gr][ch]); ++ */ ++ if (gtkinfo.flag123) bits=pplot1->mainbits[gr][ch]; ++ else bits=pplot->LAMEmainbits[gr][ch]; ++ ++ ++ sprintf(title2,"MDCT%1i(%s) bits=%i q=%i ",gr,label,bits, ++ pplot1->qss[gr][ch]); ++ ++ xmn = 0; ++ xmx = n-1; ++ ymn=0; ++ ymx=11; ++ gpk_bargraph_draw(mdctbox[gr],n,xcord,ycord, ++ xmn,ymn,xmx,ymx,0,title2,0,barcolor); ++ } ++ ++ ++ ++ ++ /******************************************************************* ++ * draw the psy model energy spectrum (k space) ++ * l3psy.c computes pe, en, thm for THIS granule. ++ *******************************************************************/ ++ if (gtkinfo.kbflag){ ++ for (gr = 0 ; gr < mode_gr ; gr ++) { ++ n = HBLKSIZE; /* only show half the spectrum */ ++ ++ data = &pplot->energy[gr][ch][0]; ++ ++ ymn=9e20; ++ ymx=-9e20; ++ for (i=0; i ymx) ? ycord[i] : ymx; ++ ymn=(ycord[i] < ymn) ? ycord[i] : ymn; ++ } ++ for (en=0 , j=0; jenergy[gr][ch][j]; ++ ++ sprintf(title2,"FFT%1i pe=%4.1fK en=%5.2e ",gr, ++ pplot->pe[gr][ch]/1000,en); ++ ++ ymn = 3; ++ ymx = 15; ++ xmn = 1; ++ xmx = n; ++ gpk_bargraph_draw(enerbox[gr],n,xcord,ycord, ++ xmn,ymn,xmx,ymx,1,title2,0,barcolor); ++ ++ } ++ }else{ ++ /******************************************************************* ++ * draw the psy model energy spectrum (scalefactor bands) ++ *******************************************************************/ ++ for (gr = 0 ; gr < mode_gr ; gr ++) { ++ ++ if (blocktype[gr][ch]==2) { ++ n = 3*SBMAX_s; ++ data = &pplot->en_s[gr][ch][0]; ++ data2 = &pplot->thr_s[gr][ch][0]; ++ data3 = &pplot->xfsf_s[gr][ch][0]; ++ } else { ++ n = SBMAX_l; ++ data = &pplot->en[gr][ch][0]; ++ data2 = &pplot->thr[gr][ch][0]; ++ data3 = &pplot->xfsf[gr][ch][0]; ++ } ++ ymn=9e20; ++ ymx=-9e20; ++ for (i=0; i ymx) ? ycord[i] : ymx; ++ ymn=(ycord[i] < ymn) ? ycord[i] : ymn; ++ } ++ ++ ++ ++ /* en = max energy difference amoung the 3 short FFTs for this granule */ ++ en = pplot->ers[gr][ch]; ++ sprintf(title2,"FFT%1i pe=%4.1fK/%3.1f n=%i/%3.1f/%3.1f/%3.1f",gr, ++ pplot->pe[gr][ch]/1000,en,pplot->over[gr][ch], ++ pplot->max_noise[gr][ch], ++ pplot->over_noise[gr][ch], ++ pplot->tot_noise[gr][ch]); ++ ++ ++ ymn = 3; ++ ymx = 15; ++ xmn = 1; ++ xmx = n+1; /* a little extra because of the bar thickness */ ++ gpk_bargraph_draw(enerbox[gr],n,xcord,ycord, ++ xmn,ymn,xmx,ymx,1,title2,0,barcolor); ++ ++ ++ for (i=0; i ymx) ? ycord[i] : ymx; ++ ymn=(ycord[i] < ymn) ? ycord[i] : ymn; ++ } ++ gpk_bargraph_draw(enerbox[gr],n,xcord,ycord, ++ xmn,ymn,xmx,ymx,0,title2,3,&red); ++ ++ ++ for (i=0; i ymx) ? ycord[i] : ymx; ++ ymn=(ycord[i] < ymn) ? ycord[i] : ymn; ++ } ++ gpk_bargraph_draw(enerbox[gr],n,xcord,ycord, ++ xmn,ymn,xmx,ymx,0,title2,3,grcolor[gr]); ++ } ++ } ++ ++ /******************************************************************* ++ * draw scalefactors ++ *******************************************************************/ ++ for (gr = 0 ; gr < mode_gr ; gr ++) { ++ double ggain; ++ if (blocktype[gr][ch]==2) { ++ n = 3*SBMAX_s; ++ if (gtkinfo.flag123) data = pplot1->sfb_s[gr][ch]; ++ else data = pplot->LAMEsfb_s[gr][ch]; ++ } else { ++ n = SBMAX_l; ++ if (gtkinfo.flag123) data = pplot1->sfb[gr][ch]; ++ else data = pplot->LAMEsfb[gr][ch]; ++ } ++ ++ ymn=-1; ++ ymx=10; ++ for (i=0; i ymx) ? ycord[i] : ymx; ++ ymn=(ycord[i] < ymn) ? ycord[i] : ymn; ++ } ++ ++ if (blocktype[gr][ch]==2) { ++ sprintf(label2, ++ "SFB scale=%i %i%i%i", ++ pplot1->scalefac_scale[gr][ch], ++ pplot1->sub_gain[gr][ch][0], ++ pplot1->sub_gain[gr][ch][1], ++ pplot1->sub_gain[gr][ch][2]); ++ }else{ ++ sprintf(label2,"SFB scale=%i",pplot1->scalefac_scale[gr][ch]); ++ } ++ ++ if (gtkinfo.flag123) ggain = -(pplot1->qss[gr][ch]-210)/4.0; ++ else ggain = -(pplot->LAMEqss[gr][ch]-210)/4.0; ++ ++ sprintf(title2," gain=%4.1f",ggain); ++ strcat(label2,title2); ++ ++ xmn = 1; ++ xmx = n+1; ++ gpk_bargraph_draw(sfbbox[gr],n,xcord,ycord, ++ xmn,ymn,xmx,ymx,1,label2,0,grcolor[gr]); ++ ++ ycord[0] = ycord[1] = 0; ++ xcord[0] = 1; ++ xcord[1] = n+1; ++ gpk_rectangle_draw(sfbbox[gr],xcord,ycord,xmn,ymn,xmx,ymx,&yellow); ++ ++ ++ } ++ ++ ++ } ++ ++ ++ ++ static void update_progress(void) ++ { ++ char label[80]; ++ int tf=gfp->totalframes; ++ if (gtkinfo.totalframes>0) tf=gtkinfo.totalframes; ++ ++ sprintf(label,"Frame:%4i/%4i %6.2fs", ++ pplot->frameNum,(int)tf-1, pplot->frametime); ++ gtk_progress_set_value (GTK_PROGRESS (frameprogress), (gdouble) pplot->frameNum); ++ gtk_label_set_text(GTK_LABEL(framecounter),label); ++ } ++ ++ ++ ++ static void analyze(void) ++ { ++ if ( idle_keepgoing) { ++ idle_count = 0; ++ idle_count_max=0; ++ idle_keepgoing=0; ++ idle_end=0; ++ } ++ plot_frame(); ++ update_progress(); ++ } ++ ++ static void plotclick( GtkWidget *widget, gpointer data ) ++ { analyze(); } ++ ++ ++ ++ ++ static int frameadv1(GtkWidget *widget, gpointer data ) ++ { ++ int i; ++ if (idle_keepgoing ){ ++ if (idle_back) { ++ /* frame displayed is the old frame. to advance, just swap in new frame */ ++ idle_back--; ++ pplot = &Pinfo[READ_AHEAD+idle_back]; ++ }else{ ++ /* advance the frame by reading in a new frame */ ++ pplot = &Pinfo[READ_AHEAD]; ++ if (mp3done) { ++ /* dont try to read any more frames, and quit if "finish MP3" was selected */ ++ /* if (idle_finish) gtk_main_quit(); */ ++ idle_count_max=0; ++ idle_end=0; ++ } else { ++ /* read in the next frame */ ++ for (i=NUMPINFO-1 ; i>0 ; i--) ++ memcpy(&Pinfo[i],&Pinfo[i-1],sizeof(plotting_data)); ++ pinfo = &Pinfo[0]; ++ pinfo->num_samples = gtkmakeframe(); ++ if (pinfo->num_samples==0 && gtkinfo.totalframes==0) ++ /* allow an extra frame to flush decoder buffers */ ++ gtkinfo.totalframes = pinfo->frameNum +2; ++ ++ if (pinfo->sampfreq) ++ pinfo->frametime = (pinfo->frameNum)*1152.0/pinfo->sampfreq; ++ else pinfo->frametime=0; ++ ++ /* eof? ++ if (!pinfo->num_samples) if (idle_finish) gtk_main_quit(); ++ */ ++ ++ pinfo->totbits = 0; ++ { int gr,ch; ++ for (gr = 0 ; gr < 2 ; gr ++) ++ for (ch = 0 ; ch < 2 ; ch ++) { ++ gtkinfo.totshort += (pinfo->mpg123blocktype[gr][ch]==2); ++ gtkinfo.totmix += !(pinfo->mixed[gr][ch]==0); ++ pinfo->totbits += pinfo->mainbits[gr][ch]; ++ } ++ } ++ if (pinfo->frameNum > 0) /* start averaging at second frame */ ++ gtkinfo.avebits = (gtkinfo.avebits*((pinfo->frameNum)-1) ++ + pinfo->totbits ) /(pinfo->frameNum); ++ ++ gtkinfo.maxbits=MAX(gtkinfo.maxbits,pinfo->totbits); ++ gtkinfo.totemph += !(pinfo->emph==0); ++ gtkinfo.totms += !(pinfo->ms_stereo==0); ++ gtkinfo.totis += !(pinfo->i_stereo==0); ++ ++ if (gtkinfo.totalframes>0) ++ if (pplot->frameNum >= gtkinfo.totalframes-1) mp3done=1; ++ } ++ } ++ ++ idle_count++; ++ if (gtkinfo.pupdate) plot_frame(); ++ update_progress(); ++ if ((idle_count>=idle_count_max) && (! idle_end)) analyze(); ++ } ++ return 1; ++ } ++ ++ ++ static void frameadv( GtkWidget *widget, gpointer data ) ++ { ++ int adv; ++ ++ if (!strcmp((char *) data,"-1")) { ++ /* ignore if we've already gone back as far as possible */ ++ if (pplot->frameNum==0 || (idle_back==NUMBACK)) return; ++ idle_back++; ++ pplot = &Pinfo[READ_AHEAD+idle_back]; ++ analyze(); ++ return; ++ } ++ ++ ++ adv = 1; ++ if (!strcmp((char *) data,"1")) adv = 1; ++ if (!strcmp((char *) data,"10")) adv = 10; ++ if (!strcmp((char *) data,"100")) adv = 100; ++ if (!strcmp((char *) data,"finish")) idle_end = 1; ++ ++ ++ if (idle_keepgoing) { ++ /* already running - que up additional frame advance requests */ ++ idle_count_max += adv; ++ } ++ else { ++ /* turn on idleing */ ++ idle_count_max = adv; ++ idle_count = 0; ++ idle_keepgoing = 1; ++ } ++ } ++ ++ ++ ++ ++ /* another callback */ ++ static void delete_event( GtkWidget *widget, ++ GdkEvent *event, ++ gpointer data ) ++ { ++ gtk_main_quit (); ++ } ++ ++ ++ ++ ++ ++ ++ ++ static void channel_option (GtkWidget *widget, gpointer data) ++ { ++ long option; ++ option = (long) data; ++ switch (option) { ++ case 1: ++ gtkinfo.msflag=0; ++ gtkinfo.chflag=0; ++ break; ++ case 2: ++ gtkinfo.msflag=0; ++ gtkinfo.chflag=1; ++ break; ++ case 3: ++ gtkinfo.msflag=1; ++ gtkinfo.chflag=0; ++ break; ++ case 4: ++ gtkinfo.msflag=1; ++ gtkinfo.chflag=1; ++ } ++ analyze(); ++ } ++ static void spec_option (GtkWidget *widget, gpointer data) ++ { ++ long option; ++ option = (long) data; ++ switch (option) { ++ case 1: ++ gtkinfo.kbflag=0; ++ break; ++ case 2: ++ gtkinfo.kbflag=1; ++ break; ++ case 3: ++ gtkinfo.flag123=0; ++ break; ++ case 4: ++ gtkinfo.flag123=1; ++ break; ++ case 5: ++ gtkinfo.pupdate=1; ++ break; ++ case 6: ++ gtkinfo.pupdate=0; ++ break; ++ case 7: ++ gtkinfo.sfblines = !gtkinfo.sfblines; ++ break; ++ } ++ analyze(); ++ } ++ ++ static gint key_press_event (GtkWidget *widget, GdkEventKey *event) ++ { ++ if (event->keyval == '1') { ++ subblock_draw[0] = 1; ++ subblock_draw[1] = 0; ++ subblock_draw[2] = 0; ++ analyze(); ++ } ++ else if (event->keyval == '2') { ++ subblock_draw[0] = 0; ++ subblock_draw[1] = 1; ++ subblock_draw[2] = 0; ++ analyze(); ++ } ++ else if (event->keyval == '3') { ++ subblock_draw[0] = 0; ++ subblock_draw[1] = 0; ++ subblock_draw[2] = 1; ++ analyze(); ++ } ++ else if (event->keyval == '0') { ++ subblock_draw[0] = 1; ++ subblock_draw[1] = 1; ++ subblock_draw[2] = 1; ++ analyze(); ++ } ++ /* analyze(); */ /* dont redraw entire window for every key! */ ++ return 0; ++ } ++ ++ ++ ++ static void text_window (GtkWidget *widget, gpointer data) ++ { ++ long option; ++ GtkWidget *hbox,*vbox,*button,*box; ++ GtkWidget *textwindow,*vscrollbar; ++ char text[80]; ++ ++ option = (long) data; ++ ++ textwindow = gtk_window_new(GTK_WINDOW_DIALOG); ++ gtk_signal_connect_object (GTK_OBJECT (window), "delete_event", ++ GTK_SIGNAL_FUNC(gtk_widget_destroy), ++ GTK_OBJECT (textwindow)); ++ ++ gtk_container_set_border_width (GTK_CONTAINER (textwindow), 0); ++ vbox = gtk_vbox_new(FALSE,0); ++ hbox = gtk_hbox_new(FALSE,0); ++ ++ button = gtk_button_new_with_label ("close"); ++ gtk_signal_connect_object (GTK_OBJECT (button), "clicked", ++ GTK_SIGNAL_FUNC(gtk_widget_destroy), ++ GTK_OBJECT (textwindow)); ++ ++ box = gtk_text_new (NULL, NULL); ++ gtk_text_set_editable (GTK_TEXT (box), FALSE); ++ vscrollbar = gtk_vscrollbar_new (GTK_TEXT(box)->vadj); ++ ++ ++ switch (option) { ++ case 0: ++ gtk_window_set_title (GTK_WINDOW (textwindow), "Documentation"); ++ gtk_widget_set_usize(box,450,500); ++ gtk_text_set_word_wrap(GTK_TEXT(box),TRUE); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL, ++ "Frame header information: "\ ++ "First the bitrate, sampling frequency and mono, stereo or jstereo "\ ++ "indicators are displayed . If the bitstream is jstereo, then mid/side "\ ++ "stereo or intensity stereo may be on (indicated in red). If "\ ++ "de-emphasis is used, this is also indicated in red. The mdb value is "\ ++ "main_data_begin. The encoded data starts this many bytes *before* the "\ ++ "frame header. A large value of mdb means the bitstream has saved some "\ ++ "bits into the reservoir, which it may allocate for some future frame. "\ ++ "The two numbers after mdb are the size (in bits) used to encode the "\ ++ "MDCT coefficients for this frame, followed byt the size of the bit "\ ++ "resevoir before encoding this frame. The maximum frame size and a "\ ++ "running average are given in the Stats pull down menu. A large "\ ++ "maximum frame size indicates the bitstream has made use of the bit "\ ++ "reservoir. \n\n",-1); ++ ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL, ++ "PCM data (top graph): "\ ++ "The PCM data is plotted in black. The layer3 frame is divided into 2 "\ ++ "granules of 576 samples (marked with yellow vertical lines). In the "\ ++ "case of normal, start and stop blocks, the MDCT coefficients for each "\ ++ "granule are computed using a 1152 sample window centered over the "\ ++ "granule. In the case of short blocks, the granule is further divided "\ ++ "into 3 blocks of 192 samples (also marked with yellow vertical lines)."\ ++ "The MDCT coefficients for these blocks are computed using 384 sample "\ ++ "windows centered over the 192 sample window. (This info not available "\ ++ "when analyzing .mp3 files.) For the psycho-acoustic model, a windowed "\ ++ "FFT is computed for each granule. The range of these windows "\ ++ "is denoted by the blue and green bars.\n\n",-1); ++ ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL, ++ "PCM re-synthesis data (second graph): "\ ++ "Same as the PCM window described above. The data displayed is the "\ ++ "result of encoding and then decoding the original sample. \n\n",-1); ++ ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL, ++ "MDCT windows: "\ ++ "Shows the energy in the MDCT spectrum for granule 0 (left window) "\ ++ "and granule 1 (right window). The text also shows the blocktype "\ ++ "used, the number of bits used to encode the coefficients and the "\ ++ "number of extra bits allocated from the reservoir. The MDCT pull down "\ ++ "window will toggle between the original unquantized MDCT coefficients "\ ++ "and the compressed (quantized) coefficients.\n\n",-1); ++ ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL, ++ "FFT window: "\ ++ "The gray bars show the energy in the FFT spectrum used by the "\ ++ "psycho-acoustic model. Granule 0 is in the left window, granule 1 in "\ ++ "the right window. The green and blue bars show how much distortion is "\ ++ "allowable, as computed by the psycho-acoustic model. The red bars show "\ ++ "the actual distortion after encoding. There is one FFT for each "\ ++ "granule, computed with a 1024 Hann window centered over the "\ ++ "appropriate granule. (the range of this 1024 sample window is shown "\ ++ "by the blue and green bars in the PCM data window). The Spectrum pull "\ ++ "down window will toggle between showing the energy in equally spaced "\ ++ "frequency domain and the scale factor bands used by layer3. Finally, "\ ++ "the perceptual entropy, total energy and number of scalefactor bands "\ ++ "with audible distortion is shown. (This info not available when "\ ++ "analyzing .mp3 files.)",-1); ++ ++ break; ++ case 1: ++ /* Set the about box information */ ++ gtk_window_set_title (GTK_WINDOW (textwindow), "About"); ++ gtk_widget_set_usize(box,350,260); ++ ++ sprintf(text,"LAME version %s \nwww.sulaco.org/mp3\n\n",get_lame_version()); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ ++ sprintf(text,"psycho-acoustic model: GPSYCHO version %s\n",get_psy_version()); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ ++ sprintf(text,"frame analyzer: MP3x version %s\n\n",get_mp3x_version()); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL, ++ "decoder: mpg123/mpglib .59q \nMichael Hipp (www.mpg123.de)\n\n",-1); ++ ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL, ++ "Encoder, decoder & psy-models based on ISO\ndemonstration source. ",-1); ++ break; ++ ++ case 2: ++ gtk_window_set_title (GTK_WINDOW (textwindow), "Statistics"); ++ gtk_widget_set_usize(box,350,260); ++ sprintf(text,"frames processed so far: %i \n",Pinfo[0].frameNum+1); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ sprintf(text,"granules processed so far: %i \n\n",4*(Pinfo[0].frameNum+1)); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ sprintf(text,"mean bits/frame (approximate): %i\n", ++ gtkinfo.approxbits); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ sprintf(text,"mean bits/frame (from LAME): %i\n", ++ 4*Pinfo[0].mean_bits); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ sprintf(text,"bitsize of largest frame: %i \n",gtkinfo.maxbits); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ sprintf(text,"average bits/frame: %3.1f \n\n",gtkinfo.avebits); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ sprintf(text,"ms_stereo frames: %i \n",gtkinfo.totms); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ sprintf(text,"i_stereo frames: %i \n",gtkinfo.totis); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ sprintf(text,"de-emphasis frames: %i \n",gtkinfo.totemph); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ sprintf(text,"short block granules: %i \n",gtkinfo.totshort); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ sprintf(text,"mixed block granules: %i \n",gtkinfo.totmix); ++ gtk_text_insert(GTK_TEXT(box),NULL,NULL,NULL,text,-1); ++ break; ++ } ++ ++ ++ ++ gtk_widget_show (vscrollbar); ++ gtk_widget_show (box); ++ gtk_widget_show (vbox); ++ gtk_widget_show (hbox); ++ gtk_widget_show (button); ++ ++ gtk_box_pack_start (GTK_BOX(hbox), box, FALSE, TRUE, 0); ++ gtk_box_pack_start(GTK_BOX(hbox), vscrollbar, FALSE, FALSE, 0); ++ gtk_box_pack_start (GTK_BOX (vbox), hbox, FALSE, TRUE, 0); ++ gtk_box_pack_end (GTK_BOX (vbox), button, FALSE, TRUE, 0); ++ gtk_container_add (GTK_CONTAINER (textwindow), vbox); ++ gtk_widget_show(textwindow); ++ ++ } ++ ++ ++ ++ ++ /* #include */ ++ ++ ++ /* This is the GtkItemFactoryEntry structure used to generate new menus. ++ Item 1: The menu path. The letter after the underscore indicates an ++ accelerator key once the menu is open. ++ Item 2: The accelerator key for the entry ++ Item 3: The callback function. ++ Item 4: The callback action. This changes the parameters with ++ which the function is called. The default is 0. ++ Item 5: The item type, used to define what kind of an item it is. ++ Here are the possible values: ++ ++ NULL -> "" ++ "" -> "" ++ "" -> create a title item ++ "<Item>" -> create a simple item ++ "<CheckItem>" -> create a check item ++ "<ToggleItem>" -> create a toggle item ++ "<RadioItem>" -> create a radio item ++ <path> -> path of a radio item to link against ++ "<Separator>" -> create a separator ++ "<Branch>" -> create an item to hold sub items ++ "<LastBranch>" -> create a right justified branch ++ */ ++ ++ static GtkItemFactoryEntry menu_items[] = { ++ {"/_File", NULL, NULL, 0, "<Branch>"}, ++ /* ++ {"/File/_New", "<control>N", print_hello, 0, NULL}, ++ {"/File/_Open", "<control>O", print_hello, 0, NULL}, ++ {"/File/_Save", "<control>S", print_hello, 0, NULL}, ++ {"/File/Save _As", NULL, NULL, 0, NULL}, ++ {"/File/sep1", NULL, NULL, 0, "<Separator>"}, ++ {"/File/Quit", "<control>Q", gtk_main_quit, 0, NULL}, ++ */ ++ {"/File/_Quit", "<control>Q", delete_event, 0, NULL}, ++ ++ {"/_Plotting", NULL, NULL, 0, "<Branch>"}, ++ {"/Plotting/_While advancing" , NULL, spec_option, 5, NULL}, ++ {"/Plotting/_After advancing", NULL, spec_option, 6, NULL}, ++ ++ {"/_Channel", NULL, NULL, 0, "<Branch>"}, ++ {"/Channel/show _Left" , NULL, channel_option, 1, NULL}, ++ {"/Channel/show _Right", NULL, channel_option, 2, NULL}, ++ {"/Channel/show _Mid" , NULL, channel_option, 3, NULL}, ++ {"/Channel/show _Side", NULL, channel_option, 4, NULL}, ++ ++ {"/_Spectrum", NULL, NULL, 0, "<Branch>"}, ++ {"/Spectrum/_Scalefactor bands", NULL, spec_option, 1, NULL}, ++ {"/Spectrum/_Wave number", NULL, spec_option, 2, NULL}, ++ ++ {"/_MDCT", NULL, NULL, 0, "<Branch>"}, ++ {"/MDCT/_Original", NULL, spec_option, 3, NULL}, ++ {"/MDCT/_Compressed", NULL, spec_option, 4, NULL}, ++ {"/MDCT/_Toggle SFB lines", NULL, spec_option, 7, NULL}, ++ ++ {"/_Stats", NULL, NULL, 0, "<Branch>"}, ++ {"/Stats/_Show", NULL, text_window, 2, NULL}, ++ ++ {"/_Help", NULL, NULL, 0, "<LastBranch>"}, ++ {"/_Help/_Documentation", NULL, text_window, 0, NULL}, ++ {"/_Help/_About", NULL, text_window, 1, NULL}, ++ }; ++ ++ ++ static void get_main_menu(GtkWidget *window, GtkWidget ** menubar) { ++ int nmenu_items = sizeof(menu_items) / sizeof(menu_items[0]); ++ GtkItemFactory *item_factory; ++ GtkAccelGroup *accel_group; ++ ++ accel_group = gtk_accel_group_new(); ++ ++ /* This function initializes the item factory. ++ Param 1: The type of menu - can be GTK_TYPE_MENU_BAR, GTK_TYPE_MENU, ++ or GTK_TYPE_OPTION_MENU. ++ Param 2: The path of the menu. ++ Param 3: A pointer to a gtk_accel_group. The item factory sets up ++ the accelerator table while generating menus. ++ */ ++ ++ item_factory = gtk_item_factory_new(GTK_TYPE_MENU_BAR, "<main>", ++ accel_group); ++ ++ /* This function generates the menu items. Pass the item factory, ++ the number of items in the array, the array itself, and any ++ callback data for the the menu items. */ ++ gtk_item_factory_create_items(item_factory, nmenu_items, menu_items, NULL); ++ ++ /* Attach the new accelerator group to the window. */ ++ gtk_accel_group_attach (accel_group, GTK_OBJECT (window)); ++ ++ if (menubar) ++ /* Finally, return the actual menu bar created by the item factory. */ ++ *menubar = gtk_item_factory_get_widget(item_factory, "<main>"); ++ } ++ ++ ++ ++ ++ int gtkcontrol(lame_global_flags *gfp2) ++ { ++ /* GtkWidget is the storage type for widgets */ ++ GtkWidget *button; ++ GtkAdjustment *adj; ++ GtkWidget *mbox; /* main box */ ++ GtkWidget *box1; /* frame control buttons go */ ++ GtkWidget *box2; /* frame counters */ ++ GtkWidget *box3; /* frame header info */ ++ GtkWidget *table; /* table for all the plotting areas */ ++ GtkWidget *menubar; ++ ++ gint tableops,graphx,graphy; ++ char frameinfo[80]; ++ ++ graphx = 500; /* minimum allowed size of pixmap */ ++ graphy = 95; ++ ++ gfp=gfp2; ++ ++ /* set some global defaults/variables */ ++ gtkinfo.filetype = (gfp->input_format == sf_mp3); ++ gtkinfo.msflag=0; ++ gtkinfo.chflag=0; ++ gtkinfo.kbflag=0; ++ gtkinfo.flag123 = (gfp->input_format == sf_mp3); /* MP3 file=use mpg123 output */ ++ gtkinfo.pupdate=0; ++ gtkinfo.avebits = 0; ++ gtkinfo.maxbits = 0; ++ gtkinfo.approxbits = 0; ++ gtkinfo.totemph = 0; ++ gtkinfo.totms = 0; ++ gtkinfo.totis = 0; ++ gtkinfo.totshort = 0; ++ gtkinfo.totmix = 0; ++ gtkinfo.sfblines= 1; ++ gtkinfo.totalframes = 0; ++ ++ memset((char *) Pinfo, 0, sizeof(Pinfo)); ++ pplot = &Pinfo[READ_AHEAD]; ++ ++ strcpy(frameinfo,"MP3x: "); ++ strncat(frameinfo,gfp->inPath,70); ++ ++ window = gtk_window_new (GTK_WINDOW_TOPLEVEL); ++ gtk_window_set_title (GTK_WINDOW (window), frameinfo); ++ gtk_signal_connect (GTK_OBJECT (window), "delete_event", ++ GTK_SIGNAL_FUNC (delete_event), NULL); ++ ++ gtk_signal_connect_object (GTK_OBJECT (window), "key_press_event", ++ GTK_SIGNAL_FUNC(key_press_event), ++ GTK_OBJECT (window)); ++ ++ gtk_container_set_border_width (GTK_CONTAINER (window), 0); ++ ++ ++ mbox = gtk_vbox_new(FALSE, 0); ++ ++ ++ /* layout of mbox */ ++ box1 = gtk_hbox_new(FALSE, 0); ++ box2 = gtk_hbox_new(FALSE, 0); ++ box3 = gtk_hbox_new(FALSE, 0); ++ table = gtk_table_new (5, 2, FALSE); ++ tableops = GTK_FILL | GTK_EXPAND | GTK_SHRINK; ++ get_main_menu(window, &menubar); ++ ++ gtk_box_pack_start(GTK_BOX(mbox), menubar, FALSE, TRUE, 0); ++ gtk_box_pack_end (GTK_BOX (mbox), box1, FALSE, TRUE, 0); ++ gtk_box_pack_end(GTK_BOX (mbox),box2, FALSE, TRUE, 0); ++ gtk_box_pack_start(GTK_BOX (mbox),box3, FALSE, TRUE, 0); ++ gtk_box_pack_start (GTK_BOX (mbox), table, TRUE, TRUE, 0); ++ gtk_container_add (GTK_CONTAINER (window), mbox); ++ ++ ++ /*********************************************************************/ ++ /* stuff in box3 frame header info */ ++ /*********************************************************************/ ++ /* ++ headerbox = gtk_label_new(" "); ++ gtk_label_set_justify(GTK_LABEL(headerbox),GTK_JUSTIFY_LEFT); ++ */ ++ headerbox = gtk_text_new (NULL, NULL); ++ gtk_text_set_editable (GTK_TEXT (headerbox), FALSE); ++ gtk_widget_set_usize(headerbox,200,20); ++ gtk_widget_show (headerbox); ++ gtk_box_pack_start(GTK_BOX (box3),headerbox, TRUE, TRUE, 0); ++ ++ ++ ++ /*********************************************************************/ ++ /* stuff in box2 frame counters */ ++ /*********************************************************************/ ++ framecounter = gtk_label_new(""); ++ gtk_widget_show(framecounter); ++ gtk_box_pack_start(GTK_BOX (box2),framecounter, FALSE, TRUE, 0); ++ ++ adj = (GtkAdjustment *) gtk_adjustment_new (0, 0,(gint) gfp->totalframes-1, 0, 0, 0); ++ frameprogress = gtk_progress_bar_new_with_adjustment (adj); ++ /* Set the format of the string that can be displayed in the ++ * trough of the progress bar: ++ * %p - percentage ++ * %v - value ++ * %l - lower range value ++ * %u - upper range value */ ++ gtk_progress_set_format_string (GTK_PROGRESS (frameprogress), ++ "%p%%"); ++ gtk_progress_set_value (GTK_PROGRESS (frameprogress), (gdouble) 0); ++ gtk_progress_set_show_text (GTK_PROGRESS (frameprogress),TRUE); ++ gtk_widget_show (frameprogress); ++ gtk_box_pack_end (GTK_BOX (box2), frameprogress, FALSE, TRUE, 0); ++ ++ ++ ++ /*********************************************************************/ ++ /* stuff in box1 buttons along bottom */ ++ /*********************************************************************/ ++ button = gtk_button_new_with_label ("-1"); ++ gtk_signal_connect (GTK_OBJECT (button), "clicked", ++ GTK_SIGNAL_FUNC (frameadv), (gpointer) "-1"); ++ gtk_box_pack_start(GTK_BOX(box1), button, TRUE, TRUE, 0); ++ gtk_widget_show(button); ++ ++ button = gtk_button_new_with_label ("+1"); ++ gtk_signal_connect (GTK_OBJECT (button), "clicked", ++ GTK_SIGNAL_FUNC (frameadv), (gpointer) "1"); ++ gtk_box_pack_start(GTK_BOX(box1), button, TRUE, TRUE, 0); ++ gtk_widget_show(button); ++ ++ button = gtk_button_new_with_label ("+10"); ++ gtk_signal_connect (GTK_OBJECT (button), "clicked", ++ GTK_SIGNAL_FUNC (frameadv), (gpointer) "10"); ++ gtk_box_pack_start(GTK_BOX(box1), button, TRUE, TRUE, 0); ++ gtk_widget_show(button); ++ ++ button = gtk_button_new_with_label ("+100"); ++ gtk_signal_connect (GTK_OBJECT (button), "clicked", ++ GTK_SIGNAL_FUNC (frameadv), (gpointer) "100"); ++ gtk_box_pack_start(GTK_BOX(box1), button, TRUE, TRUE, 0); ++ gtk_widget_show(button); ++ ++ button = gtk_button_new_with_label ("last frame"); ++ gtk_signal_connect (GTK_OBJECT (button), "clicked", ++ GTK_SIGNAL_FUNC (frameadv), (gpointer) "finish"); ++ gtk_box_pack_start(GTK_BOX(box1), button, TRUE, TRUE, 0); ++ gtk_widget_show(button); ++ ++ button = gtk_button_new_with_label ("stop/plot"); ++ gtk_signal_connect (GTK_OBJECT (button), "clicked", ++ GTK_SIGNAL_FUNC (plotclick), NULL); ++ gtk_box_pack_start(GTK_BOX(box1), button, TRUE, TRUE, 0); ++ gtk_widget_show(button); ++ ++ ++ /*********************************************************************/ ++ /* stuff in table. all the plotting windows */ ++ /*********************************************************************/ ++ pcmbox = gpk_plot_new(graphx,graphy); ++ gtk_table_attach (GTK_TABLE(table),pcmbox,0,2,0,1,tableops,tableops,2,2 ); ++ gtk_widget_show (pcmbox); ++ ++ winbox = gpk_plot_new(graphy,graphy); ++ gtk_table_attach(GTK_TABLE(table),winbox,0,2,1,2,tableops,tableops,2,2); ++ gtk_widget_show (winbox); ++ ++ ++ mdctbox[0] = gpk_plot_new(graphy,graphy); ++ gtk_table_attach(GTK_TABLE(table),mdctbox[0],0,1,2,3,tableops,tableops,2,2); ++ gtk_widget_show (mdctbox[0]); ++ ++ mdctbox[1] = gpk_plot_new(graphy,graphy); ++ gtk_table_attach (GTK_TABLE(table),mdctbox[1],1,2,2,3,tableops,tableops,2,2); ++ gtk_widget_show (mdctbox[1]); ++ ++ enerbox[0] = gpk_plot_new(graphy,graphy); ++ gtk_table_attach(GTK_TABLE(table),enerbox[0],0,1,3,4,tableops,tableops,2,2); ++ gtk_widget_show (enerbox[0]); ++ ++ enerbox[1] = gpk_plot_new(graphy,graphy); ++ gtk_table_attach (GTK_TABLE(table),enerbox[1],1,2,3,4,tableops,tableops,2,2); ++ gtk_widget_show (enerbox[1]); ++ ++ sfbbox[0] = gpk_plot_new(graphy,graphy); ++ gtk_table_attach(GTK_TABLE(table),sfbbox[0],0,1,4,5,tableops,tableops,2,2); ++ gtk_widget_show (sfbbox[0]); ++ ++ sfbbox[1] = gpk_plot_new(graphy,graphy); ++ gtk_table_attach (GTK_TABLE(table),sfbbox[1],1,2,4,5,tableops,tableops,2,2); ++ gtk_widget_show (sfbbox[1]); ++ ++ ++ ++ ++ gtk_idle_add((GtkFunction) frameadv1, NULL); ++ gtk_widget_show(menubar); ++ gtk_widget_show(box2); ++ gtk_widget_show(box3); ++ gtk_widget_show(table); ++ gtk_widget_show(box1); ++ gtk_widget_show (mbox); ++ gtk_widget_show (window); /* show smallest allowed window */ ++ ++ /* make window bigger. */ ++ /* now the user will be able to shrink it, if desired */ ++ /* gtk_widget_set_usize(mbox,500,500); */ ++ /* gtk_widget_show (window); */ /* show smallest allowed window */ ++ ++ ++ ++ idle_keepgoing=1; /* processing of frames is ON */ ++ idle_count_max=READ_AHEAD+1; /* number of frames to process before plotting */ ++ idle_count=0; /* pause & plot when idle_count=idle_count_max */ ++ ++ ++ gtk_main (); ++ if (!mp3done) exit(2); ++ return(0); ++ } ++ ++ #endif ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ +diff -r -c -N encoder/gtkanal.h lame3.70/gtkanal.h +*** encoder/gtkanal.h Wed Dec 31 17:00:00 1969 +--- lame3.70/gtkanal.h Wed Mar 29 22:39:41 2000 +*************** +*** 0 **** +--- 1,68 ---- ++ #ifndef GTKANAL_DOT_H ++ #define GTKANAL_DOT_H ++ #include "lame.h" ++ #include "encoder.h" ++ ++ #define READ_AHEAD 10 /* number of frames to read ahead */ ++ #define MAXMPGLAG READ_AHEAD /* if the mpg123 lag becomes bigger than this we have to stop */ ++ #define NUMBACK 6 /* number of frames we can back up */ ++ #define NUMPINFO (NUMBACK+READ_AHEAD+1) ++ ++ ++ ++ typedef struct { ++ int frameNum; /* current frame number */ ++ int frameNum123; ++ int num_samples; /* number of pcm samples read for this frame */ ++ double frametime; /* starting time of frame, in seconds */ ++ double pcmdata[2][1600]; ++ double pcmdata2[2][1152+1152-DECDELAY]; ++ double xr[2][2][576]; ++ double mpg123xr[2][2][576]; ++ double ms_ratio[2]; ++ double ms_ener_ratio[2]; ++ ++ /* L,R, M and S values */ ++ double energy[2][4][BLKSIZE]; ++ double pe[2][4]; ++ double thr[2][4][SBMAX_l]; ++ double en[2][4][SBMAX_l]; ++ double thr_s[2][4][3*SBMAX_s]; ++ double en_s[2][4][3*SBMAX_s]; ++ double ers[2][4]; ++ ++ double sfb[2][2][SBMAX_l]; ++ double sfb_s[2][2][3*SBMAX_s]; ++ double LAMEsfb[2][2][SBMAX_l]; ++ double LAMEsfb_s[2][2][3*SBMAX_s]; ++ ++ int LAMEqss[2][2]; ++ int qss[2][2]; ++ int big_values[2][2]; ++ int sub_gain[2][2][3]; ++ ++ double xfsf[2][2][SBMAX_l]; ++ double xfsf_s[2][2][3*SBMAX_s]; ++ ++ int over[2][2]; ++ double tot_noise[2][2]; ++ double max_noise[2][2]; ++ double over_noise[2][2]; ++ int blocktype[2][2]; ++ int scalefac_scale[2][2]; ++ int mpg123blocktype[2][2]; ++ int mixed[2][2]; ++ int mainbits[2][2]; ++ int LAMEmainbits[2][2]; ++ int framesize,stereo,js,ms_stereo,i_stereo,emph,bitrate,sampfreq,maindata; ++ int crc,padding; ++ int scfsi[2],mean_bits,resvsize; ++ int totbits; ++ } plotting_data; ++ ++ ++ int gtkcontrol(lame_global_flags *gfp); ++ extern plotting_data *pinfo; ++ extern int gtkflag; ++ ++ #endif +diff -r -c -N encoder/huffman.c lame3.70/huffman.c +*** encoder/huffman.c Wed Jan 22 02:43:15 1997 +--- lame3.70/huffman.c Wed Dec 31 17:00:00 1969 +*************** +*** 1,380 **** +- /********************************************************************** +- * ISO MPEG Audio Subgroup Software Simulation Group (1996) +- * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- * +- * $Id: huffman.c,v 1.1 1996/02/14 04:04:23 rowlands Exp $ +- * +- * $Log: huffman.c,v $ +- * Revision 1.1 1996/02/14 04:04:23 rowlands +- * Initial revision +- * +- * Received from Mike Coleman +- **********************************************************************/ +- /********************************************************************** +- * changes made since last update: * +- * date programmers comment * +- *27.2.92 F.O.Witte (ITT Intermetall) * +- * email: otto.witte@itt-sc.de * +- * tel: ++49 (761)517-125 * +- * fax: ++49 (761)517-880 * +- *12.6.92 J. Pineda Added sign bit to decoder. * +- * 08/24/93 M. Iwadare Changed for 1 pass decoding. * +- *--------------------------------------------------------------------* +- * 7/14/94 Juergen Koller Bug fixes in Layer III code * +- *********************************************************************/ +- #include <stdlib.h> +- #include "common.h" +- #include "huffman.h" +- +- HUFFBITS dmask = 1 << (sizeof(HUFFBITS)*8-1); +- unsigned int hs = sizeof(HUFFBITS)*8; +- +- struct huffcodetab ht[HTN]; /* array of all huffcodtable headers */ +- /* 0..31 Huffman code table 0..31 */ +- /* 32,33 count1-tables */ +- +- /* read the huffman encode table */ +- int read_huffcodetab(fi) +- FILE *fi; +- { +- +- char line[100],command[40],huffdata[40]; +- unsigned int t,i,j,k,nn,x,y,n=0; +- unsigned int xl, yl, len; +- HUFFBITS h; +- int hsize; +- +- hsize = sizeof(HUFFBITS)*8; +- do { +- fgets(line,99,fi); +- } while ((line[0] == '#') || (line[0] < ' ') ); +- +- do { +- while ((line[0]=='#') || (line[0] < ' ')) { +- fgets(line,99,fi); +- } +- +- sscanf(line,"%s %s %u %u %u",command,ht[n].tablename, +- &xl,&yl,&ht[n].linbits); +- if (strcmp(command,".end")==0) +- return n; +- else if (strcmp(command,".table")!=0) { +- fprintf(stderr,"huffman table %u data corrupted\n",n); +- return -1; +- } +- ht[n].linmax = (1<<ht[n].linbits)-1; +- +- sscanf(ht[n].tablename,"%u",&nn); +- if (nn != n) { +- fprintf(stderr,"wrong table number %u\n",n); +- return(-2); +- } +- +- ht[n].xlen = xl; +- ht[n].ylen = yl; +- +- do { +- fgets(line,99,fi); +- } while ((line[0] == '#') || (line[0] < ' ')); +- +- sscanf(line,"%s %u",command,&t); +- if (strcmp(command,".reference")==0) { +- ht[n].ref = t; +- ht[n].table = ht[t].table; +- ht[n].hlen = ht[t].hlen; +- if ( (xl != ht[t].xlen) || +- (yl != ht[t].ylen) ) { +- fprintf(stderr,"wrong table %u reference\n",n); +- return (-3); +- }; +- do { +- fgets(line,99,fi); +- } while ((line[0] == '#') || (line[0] < ' ') ); +- } +- else { +- ht[n].ref = -1; +- ht[n].table=(HUFFBITS *) calloc(xl*yl,sizeof(HUFFBITS)); +- if (ht[n].table == NULL) { +- fprintf(stderr,"unsufficient heap error\n"); +- return (-4); +- } +- ht[n].hlen=(unsigned char *) calloc(xl*yl,sizeof(unsigned char)); +- if (ht[n].hlen == NULL) { +- fprintf(stderr,"unsufficient heap error\n"); +- return (-4); +- } +- for (i=0; i<xl; i++) { +- for (j=0;j<yl; j++) { +- if (xl>1) +- sscanf(line,"%u %u %u %s",&x, &y, &len,huffdata); +- else +- sscanf(line,"%u %u %s",&x,&len,huffdata); +- h=0;k=0; +- while (huffdata[k]) { +- h <<= 1; +- if (huffdata[k] == '1') +- h++; +- else if (huffdata[k] != '0'){ +- fprintf(stderr,"huffman-table %u bit error\n",n); +- return (-5); +- }; +- k++; +- }; +- if (k != len) { +- fprintf(stderr, +- "warning: wrong codelen in table %u, pos [%2u][%2u]\n", +- n,i,j); +- }; +- ht[n].table[i*xl+j] = h; +- ht[n].hlen[i*xl+j] = (unsigned char) len; +- do { +- fgets(line,99,fi); +- } while ((line[0] == '#') || (line[0] < ' ')); +- } +- } +- } +- n++; +- } while (1); +- } +- +- /* read the huffman decoder table */ +- int read_decoder_table(fi) +- FILE *fi; +- { +- int n,i,nn,t; +- unsigned int v0,v1; +- char command[100],line[100]; +- for (n=0;n<HTN;n++) { +- /* .table number treelen xlen ylen linbits */ +- do { +- fgets(line,99,fi); +- } while ((line[0] == '#') || (line[0] < ' ')); +- +- sscanf(line,"%s %s %u %u %u %u",command,ht[n].tablename, +- &ht[n].treelen, &ht[n].xlen, &ht[n].ylen, &ht[n].linbits); +- if (strcmp(command,".end")==0) +- return n; +- else if (strcmp(command,".table")!=0) { +- fprintf(stderr,"huffman table %u data corrupted\n",n); +- return -1; +- } +- ht[n].linmax = (1<<ht[n].linbits)-1; +- +- sscanf(ht[n].tablename,"%u",&nn); +- if (nn != n) { +- fprintf(stderr,"wrong table number %u\n",n); +- return(-2); +- } +- do { +- fgets(line,99,fi); +- } while ((line[0] == '#') || (line[0] < ' ')); +- +- sscanf(line,"%s %u",command,&t); +- if (strcmp(command,".reference")==0) { +- ht[n].ref = t; +- ht[n].val = ht[t].val; +- ht[n].treelen = ht[t].treelen; +- if ( (ht[n].xlen != ht[t].xlen) || +- (ht[n].ylen != ht[t].ylen) ) { +- fprintf(stderr,"wrong table %u reference\n",n); +- return (-3); +- }; +- while ((line[0] == '#') || (line[0] < ' ') ) { +- fgets(line,99,fi); +- } +- } +- else if (strcmp(command,".treedata")==0) { +- ht[n].ref = -1; +- if ( ht[n].treelen ) +- { +- ht[n].val = (unsigned char (*)[2]) +- calloc(2*(ht[n].treelen),sizeof(unsigned char)); +- if (ht[n].val == NULL) { +- fprintf(stderr, "heaperror at table %d\n",n); +- exit (-10); +- } +- } +- else +- ht[n].val = NULL; +- for (i=0;i<ht[n].treelen; i++) { +- fscanf(fi,"%x %x",&v0, &v1); +- ht[n].val[i][0]=(unsigned char)v0; +- ht[n].val[i][1]=(unsigned char)v1; +- } +- fgets(line,99,fi); /* read the rest of the line */ +- } +- else { +- fprintf(stderr,"huffman decodertable error at table %d\n",n); +- } +- } +- return n; +- } +- +- +- /* do the huffman coding, */ +- /* note! for counta,countb - the 4 bit value is passed in y, set x to 0 */ +- /* return value: 0-no error, 1 decode error */ +- void huffman_coder( x, y, h, bs) +- unsigned int x; /* x-value */ +- unsigned int y; /* y-value */ +- struct huffcodetab *h; /* pointer to huffman code record */ +- Bit_stream_struc *bs; /* pointer to open write bitstream */ +- { +- HUFFBITS huffbits; /* data left aligned */ +- HUFFBITS linbitsX; +- HUFFBITS linbitsY; +- unsigned int len; +- unsigned int xl1 = h->xlen-1; +- unsigned int yl1 = h->ylen-1; +- linbitsX = 0; +- linbitsY = 0; +- if (h->table == NULL) return; +- if (((x < xl1) || (xl1==0)) && (y < yl1)) { +- huffbits = h->table[x*(h->xlen)+y]; +- len = h->hlen[x*(h->xlen)+y]; +- putbits(bs,huffbits,len); +- return; +- } +- else if (x >= xl1) { +- linbitsX = x-xl1; +- if (linbitsX > h->linmax) { +- fprintf(stderr,"warning: Huffman X table overflow\n"); +- linbitsX= h->linmax; +- }; +- if (y >= yl1) { +- huffbits = h->table[(h->ylen)*(h->xlen)-1]; +- len = h->hlen[(h->ylen)*(h->xlen)-1]; +- putbits(bs,huffbits,len); +- linbitsY = y-yl1; +- if (linbitsY > h->linmax) { +- fprintf(stderr,"warning: Huffman Y table overflow\n"); +- linbitsY = h->linmax; +- }; +- if (h->linbits) { +- putbits(bs,linbitsX,h->linbits); +- putbits(bs,linbitsY,h->linbits); +- } +- } +- else { /* x>= h->xlen, y<h->ylen */ +- huffbits = h->table[(h->ylen)*xl1+y]; +- len = h->hlen[(h->ylen)*xl1+y]; +- putbits(bs,huffbits,len); +- if (h->linbits) { +- putbits(bs,linbitsX,h->linbits); +- } +- } +- } +- else { /* ((x < h->xlen) && (y>=h->ylen)) */ +- huffbits = h->table[(h->ylen)*x+yl1]; +- len = h->hlen[(h->ylen)*x+yl1]; +- putbits(bs,huffbits,len); +- linbitsY = y-yl1; +- if (linbitsY > h->linmax) { +- fprintf(stderr,"warning: Huffman Y table overflow\n"); +- linbitsY = h->linmax; +- }; +- if (h->linbits) { +- putbits(bs,linbitsY,h->linbits); +- } +- } +- } +- +- /* do the huffman-decoding */ +- /* note! for counta,countb -the 4 bit value is returned in y, discard x */ +- int huffman_decoder(h, x, y, v, w) +- struct huffcodetab *h; /* pointer to huffman code record */ +- /* unsigned */ int *x; /* returns decoded x value */ +- /* unsigned */ int *y; /* returns decoded y value */ +- int *v; +- int *w; +- { +- HUFFBITS level; +- int point = 0; +- int error = 1; +- level = dmask; +- if (h->val == NULL) return 2; +- +- /* table 0 needs no bits */ +- if ( h->treelen == 0) +- { *x = *y = 0; +- return 0; +- } +- +- +- /* Lookup in Huffman table. */ +- +- do { +- if (h->val[point][0]==0) { /*end of tree*/ +- *x = h->val[point][1] >> 4; +- *y = h->val[point][1] & 0xf; +- +- error = 0; +- break; +- } +- if (hget1bit()) { +- while (h->val[point][1] >= MXOFF) point += h->val[point][1]; +- point += h->val[point][1]; +- } +- else { +- while (h->val[point][0] >= MXOFF) point += h->val[point][0]; +- point += h->val[point][0]; +- } +- level >>= 1; +- } while (level || (point < ht->treelen) ); +- +- /* Check for error. */ +- +- if (error) { /* set x and y to a medium value as a simple concealment */ +- printf("Illegal Huffman code in data.\n"); +- *x = (h->xlen-1 << 1); +- *y = (h->ylen-1 << 1); +- } +- +- /* Process sign encodings for quadruples tables. */ +- +- if (h->tablename[0] == '3' +- && (h->tablename[1] == '2' || h->tablename[1] == '3')) { +- *v = (*y>>3) & 1; +- *w = (*y>>2) & 1; +- *x = (*y>>1) & 1; +- *y = *y & 1; +- +- /* v, w, x and y are reversed in the bitstream. +- switch them around to make test bistream work. */ +- +- /* {int i=*v; *v=*y; *y=i; i=*w; *w=*x; *x=i;} MI */ +- +- if (*v) +- if (hget1bit() == 1) *v = -*v; +- if (*w) +- if (hget1bit() == 1) *w = -*w; +- if (*x) +- if (hget1bit() == 1) *x = -*x; +- if (*y) +- if (hget1bit() == 1) *y = -*y; +- } +- +- /* Process sign and escape encodings for dual tables. */ +- +- else { +- +- /* x and y are reversed in the test bitstream. +- Reverse x and y here to make test bitstream work. */ +- +- /* removed 11/11/92 -ag +- {int i=*x; *x=*y; *y=i;} +- */ +- if (h->linbits) +- if ((h->xlen-1) == *x) +- *x += hgetbits(h->linbits); +- if (*x) +- if (hget1bit() == 1) *x = -*x; +- if (h->linbits) +- if ((h->ylen-1) == *y) +- *y += hgetbits(h->linbits); +- if (*y) +- if (hget1bit() == 1) *y = -*y; +- } +- +- return error; +- } +--- 0 ---- +diff -r -c -N encoder/huffman.h lame3.70/huffman.h +*** encoder/huffman.h Wed Jan 22 02:43:15 1997 +--- lame3.70/huffman.h Wed Dec 31 17:00:00 1969 +*************** +*** 1,60 **** +- /********************************************************************** +- * ISO MPEG Audio Subgroup Software Simulation Group (1996) +- * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- * +- * $Id: huffman.h,v 1.1 1996/02/14 04:04:23 rowlands Exp $ +- * +- * $Log: huffman.h,v $ +- * Revision 1.1 1996/02/14 04:04:23 rowlands +- * Initial revision +- * +- * Received from Mike Coleman +- **********************************************************************/ +- /********************************************************************** +- * date programmers comment * +- * 27.2.92 F.O.Witte (ITT Intermetall) * +- * 8/24/93 M. Iwadare Changed for 1 pass decoding. * +- * 7/14/94 J. Koller useless 'typedef' before huffcodetab * +- * removed * +- *********************************************************************/ +- +- #define HUFFBITS unsigned long int +- #define HTN 34 +- #define MXOFF 250 +- +- struct huffcodetab { +- char tablename[3]; /*string, containing table_description */ +- unsigned int xlen; /*max. x-index+ */ +- unsigned int ylen; /*max. y-index+ */ +- unsigned int linbits; /*number of linbits */ +- unsigned int linmax; /*max number to be stored in linbits */ +- int ref; /*a positive value indicates a reference*/ +- HUFFBITS *table; /*pointer to array[xlen][ylen] */ +- unsigned char *hlen; /*pointer to array[xlen][ylen] */ +- unsigned char(*val)[2];/*decoder tree */ +- unsigned int treelen; /*length of decoder tree */ +- }; +- +- extern struct huffcodetab ht[HTN];/* global memory block */ +- /* array of all huffcodtable headers */ +- /* 0..31 Huffman code table 0..31 */ +- /* 32,33 count1-tables */ +- #ifdef PROTO_ARGS +- +- extern int read_huffcodetab(FILE *); +- extern int read_decoder_table(FILE *); +- +- extern void huffman_coder(unsigned int, unsigned int, +- struct huffcodetab *, Bit_stream_struc *); +- +- extern int huffman_decoder(struct huffcodetab *, +- /* unsigned */ int *, /* unsigned */ int*, int*, int*); +- +- #else +- +- extern int read_huffcodetab(); +- extern int read_decoder_table(); +- extern void huffman_coder(); +- extern int huffman_decoder(); +- +- #endif +--- 0 ---- +diff -r -c -N encoder/i386/CVS/Entries lame3.70/i386/CVS/Entries +*** encoder/i386/CVS/Entries Wed Dec 31 17:00:00 1969 +--- lame3.70/i386/CVS/Entries Thu Apr 6 12:50:53 2000 +*************** +*** 0 **** +--- 1,6 ---- ++ /fft.nas/1.2/Wed Feb 2 14:58:41 2000//Tlame3_70 ++ /fft3dn.nas/1.1/Sun Jan 30 22:53:49 2000//Tlame3_70 ++ /fftfpu.nas/1.1/Sun Jan 30 22:53:49 2000//Tlame3_70 ++ /fftsse.nas/1.1/Sun Jan 30 22:53:49 2000//Tlame3_70 ++ /ffttbl.nas/1.1/Sun Jan 30 22:53:49 2000//Tlame3_70 ++ D +diff -r -c -N encoder/i386/CVS/Repository lame3.70/i386/CVS/Repository +*** encoder/i386/CVS/Repository Wed Dec 31 17:00:00 1969 +--- lame3.70/i386/CVS/Repository Thu Apr 6 12:50:36 2000 +*************** +*** 0 **** +--- 1 ---- ++ lame/i386 +diff -r -c -N encoder/i386/CVS/Root lame3.70/i386/CVS/Root +*** encoder/i386/CVS/Root Wed Dec 31 17:00:00 1969 +--- lame3.70/i386/CVS/Root Thu Apr 6 12:50:36 2000 +*************** +*** 0 **** +--- 1 ---- ++ markt@cvs.lame.sourceforge.net:/cvsroot/lame +diff -r -c -N encoder/i386/CVS/Tag lame3.70/i386/CVS/Tag +*** encoder/i386/CVS/Tag Wed Dec 31 17:00:00 1969 +--- lame3.70/i386/CVS/Tag Thu Apr 6 12:50:53 2000 +*************** +*** 0 **** +--- 1 ---- ++ Tlame3_70 +diff -r -c -N encoder/i386/fft.nas lame3.70/i386/fft.nas +*** encoder/i386/fft.nas Wed Dec 31 17:00:00 1969 +--- lame3.70/i386/fft.nas Wed Feb 2 07:58:41 2000 +*************** +*** 0 **** +--- 1,267 ---- ++ ++ ; for new GOGO-no-coda (1999/09) ++ ; Copyright (C) 1999 shigeo ++ ; special thanks to Keiichi SAKAI, URURI ++ %include "nasm.h" ++ ++ globaldef fht_3DN ++ globaldef fht ++ externdef costab_fft ++ externdef sintab_fft ++ externdef gray_index ++ ++ segment_data ++ align 16 ++ D_MSB1_0 dd 0 ,0x80000000 ++ D_SQRT2 dd 1.414213562,1.414213562 ++ t_s0 dd 0 ;[ t_c:t_s] ++ t_c0 dd 0 ++ t_c1 dd 0 ;[-t_s:t_c] ++ t_s1 dd 0 ++ D_s1c1 dd 0, 0 ++ D_Mc1s1 dd 0, 0 ++ D_s2c2 dd 0, 0 ++ D_Mc2s2 dd 0, 0 ++ D_0_1 dd 1.0, 0.0 ++ S_05 DD 0.5 ++ S_00005 DD 0.0005 ++ fht dd 0 ;�ؿ��ݥ��� ++ ++ segment_code ++ ++ ;************************************************************************ ++ ++ ; by shigeo ++ ; 99/08/16 ++ ; 23000clk �ɤ��ä��� ++ ; 18500clk bit reversal from gogo1 by URURI ++ ++ ;void fht(float *fz, int n); ++ align 16 ++ fht_3DN: ++ push ebx ++ push esi ++ push edi ++ push ebp ++ %assign _P 4*4 ++ ;�ޤ��ǽ�Υ롼��... ��fht()�γ��ذ�ư ++ ++ mov esi,[esp+_P+4] ;esi=fz ++ mov ecx,[esp+_P+8] ;ecx=n ++ ++ ;�ᥤ��롼�� ++ movq mm7,[D_MSB1_0] ;mm7=[1<<31:0] ++ ++ %assign LOCAL_STACK 16 ++ sub esp,LOCAL_STACK ++ %assign _P (_P+LOCAL_STACK) ++ xor eax,eax ++ mov [esp],eax ;k=0 ++ %define k dword [esp] ++ %define kx dword [esp+4] ++ %define fn dword [esp+8] ++ ++ .lp30: ;k=0; do{ ++ mov ecx,k ++ add ecx,2 ++ mov k,ecx ++ mov eax,1 ++ shl eax,cl ;eax=k1 = 1<<k ++ lea ebx,[eax+eax] ;ebx=k2 = k1*2 ++ lea ecx,[eax+eax*2] ;ecx=k3 = k2 + k1 = k1*3 ++ lea edx,[ebx+ebx] ;edx=k4 = k1*4 ++ mov esi,eax ++ shr esi,1 ;esi=kx=k1>>1 ++ mov kx,esi ;��¸(��ǻȤ�) ++ mov edi,[esp+_P+4] ;edi=fi=fz ++ lea ebp,[edi+esi*4] ;ebp=gi=fz+kx ++ mov esi,[esp+_P+8] ;esi=n ++ lea esi,[edi+esi*4] ;esi=fn=fz+n ++ movq mm6,[D_SQRT2] ;mm6=[��2:��2] ++ ++ .lp31: ;fn=fz+n; do{ FLOAT g0,f0,f1,... ++ movd mm0,[edi] ;mm0=[0:fi[ 0]] ++ movd mm1,[edi+eax*4] ;mm1=[0:fi[k1]] ++ punpckldq mm0,mm0 ;mm0=[fi_0 :fi_0 ] ++ punpckldq mm1,mm1 ;mm1=[fi_k1:fi_k1] ++ movd mm2,[edi+ebx*4] ++ movd mm3,[edi+ecx*4] ++ punpckldq mm2,mm2 ;mm2=[fi_k2:fi_k2] ++ punpckldq mm3,mm3 ;mm3=[fi_k3:fi_k3] ++ pxor mm1,mm7 ;mm1=[-fi_k1:fi_k1] ++ pxor mm3,mm7 ;mm3=[-fi_k3:fi_k3] ++ pfadd mm0,mm1 ;mm0=[f1:f0]=[fi_0 -fi_k1 : fi_0 +fi_k1] ++ pfadd mm2,mm3 ;mm2=[f3:f2]=[fi_k2-fi_k3 : fi_k2+fi_k3] ++ movq mm3,mm0 ;mm3=[f1:f0] ++ pfadd mm0,mm2 ;mm0=[f1+f3:f0+f2] ++ movd [edi],mm0 ;fi[0]=f0+f2 ++ psrlq mm0,32 ;mm0=[0:f1+f3] ++ pfsub mm3,mm2 ;mm3=[f1-f3:f0-f2] ++ movd [edi+eax*4],mm0 ;fi[k1]=f1+f3 ++ movd [edi+ebx*4],mm3 ;fi[k2]=f0-f2 ++ psrlq mm3,32 ;mm3=[0:f1-f3] ++ movd [edi+ecx*4],mm3 ;fi[k3]=f1-f3 ++ ++ movd mm0,[ebp] ;mm0=[0:gi_0] ++ movd mm1,[ebp+eax*4] ;mm1=[0:gi_k1] ++ punpckldq mm0,mm0 ;mm0=[gi_0 :gi_0 ] ++ punpckldq mm1,mm1 ;mm1=[gi_k1:gi_k1] ++ movd mm2,[ebp+ebx*4] ;mm2=[0:gi_k2] ++ pxor mm1,mm7 ;mm1=[-gi_k1:gi_k1] ++ punpckldq mm2,[ebp+ecx*4] ;mm2=[gi_k3:gi_k2] ++ pfadd mm0,mm1 ;mm0=[g1:g0]=[gi_0 -gi_k1:gi_0 +gi_k1] ++ pfmul mm2,mm6 ;mm2=[g3:g2]=sqrt2 * [gi_k3:gi_k2] ++ movq mm1,mm0 ;mm1=[g1:g0] ++ pfadd mm0,mm2 ;mm0=[g1+g3:g0+g2] ++ movd [ebp],mm0 ;gi[0]=g0+g2 ++ psrlq mm0,32 ;mm0=[0:g1+g3] ++ pfsub mm1,mm2 ;mm1=[g1-g3:g0-g2] ++ movd [ebp+eax*4],mm0 ;gi[k1]=g1+g3 ++ movd [ebp+ebx*4],mm1 ;gi[k2]=g0-g2 ++ psrlq mm1,32 ;mm1=[0:g1-g3] ++ movd [ebp+ecx*4],mm1 ;gi[k3]=g1-g3 ++ lea edi,[edi+edx*4] ;fi += k4 ++ lea ebp,[ebp+edx*4] ;gi += k4 ++ cmp edi,esi ++ jc near .lp31 ;}while(fi<fn); ++ ++ ; �����ޤǤ�¿ʬO.K. ++ ++ mov fn,esi ;fn=fz+n ++ ;�����ͤϰ���³���Ȥ� ++ ;eax=k1,ebx=k2,ecx=k3,edx=k4 ++ ++ mov edi,k ++ lea ebp,[costab_fft+edi*4] ++ mov ebp,[ebp] ;ebp=t_c ++ mov [t_c0],ebp ++ mov [t_c1],ebp ;t_c ++ lea ebp,[sintab_fft+edi*4] ++ mov ebp,[ebp] ;ebx=t_s ++ mov [t_s0],ebp ++ xor ebp,0x80000000 ++ mov [t_s1],ebp ;-t_s ++ ++ movq mm1,[D_0_1] ;mm1=[0:1] ++ movq [D_s1c1],mm1 ;mm1=[s1:c1] ++ mov esi,1 ;esi=i=1 ++ ++ .lp32: ; for(i=1;i<kx;i++){ ++ movq mm0,[D_s1c1] ;mm1=[s1:t]=[s1:c1] ++ movq mm2,mm0 ++ pfmul mm0,[t_c1] ;mm0=[-s1*t_s: t*t_c] ++ pfmul mm2,[t_s0] ;mm2=[ s1*t_c: t*t_s] ++ pfacc mm0,mm2 ;mm0=[s1:c1]=[ s1*t_c+t*t_s:-s1*t_s+t*t_c] ++ movq mm2,mm0 ;mm2=[s1:c1] ++ movq [D_s1c1],mm0 ;��¸ ++ movq mm6,mm2 ++ punpckldq mm5,mm6 ++ punpckhdq mm6,mm5 ;mm6=[ c1:s1] ++ pxor mm6,mm7 ;mm6=[-c1:s1] ++ movq [D_Mc1s1],mm6 ;��¸ ++ pfmul mm2,mm2 ;mm2=[s1*s1:c1*c1] ++ movq mm3,mm0 ;mm3=[s1:c1] ++ pxor mm2,mm7 ;mm2=[-s1*s1:c1*c1] ++ psrlq mm3,32 ;mm3=[ 0:s1] ++ pfacc mm2,mm2 ;mm2=[c2:c2]=[c1*c1-s1*s1:<] ++ pfmul mm0,mm3 ;mm0=[ 0:c1*s1] ++ pfadd mm0,mm0 ;mm0=[0:s2]=[ 0:2*c1*s1] ++ punpckldq mm2,mm0 ;mm2=[s2:c2] ++ movq [D_s2c2],mm2 ;��¸ ++ ++ punpckldq mm0,mm2 ++ punpckhdq mm2,mm0 ;mm2=[c2:s2] ++ pxor mm2,mm7 ;mm2=[-c2:s2] ++ movq [D_Mc2s2],mm2 ;��¸ ++ ++ mov edi,[esp+_P+4] ;edi=fz ++ lea edi,[edi+esi*4] ;edi=fz+i ++ ++ mov ebp,[esp+_P+4] ;ebp=fz ++ neg esi ;esi=-i ++ lea ebp,[ebp+eax*4] ;ebp=fz+k1 ++ lea ebp,[ebp+esi*4] ;ebp=gi=fz+k1-i ++ neg esi ;esi=i ++ ++ .lp33: ; do{ FLOAT a,b,g0,f0,f1,g1,f2,g2,f3,g3; ++ ++ movd mm0,[edi+eax*4] ;mm0=[0:fi_k1] ++ punpckldq mm0,[ebp+eax*4] ;mm0=[gi_k1:fi_k1] ++ movq mm1,mm0 ++ pfmul mm0,[D_s2c2] ;mm0=[ s2*gi_k1:c2*fi_k1] ++ pfmul mm1,[D_Mc2s2] ;mm1=[-c2*gi_k1:s2*fi_k1] ++ pfacc mm0,mm1 ;mm0=[b:a] ++ movd mm4,[edi] ;mm4=[0:fi_0] ++ movq mm3,mm0 ;mm3=[b:a] ++ punpckldq mm4,[ebp] ;mm4=[gi_0:fi_0] ++ pfadd mm3,mm4 ;mm3=[g0:f0]=[gi_0+b:fi_0+a] ++ pfsub mm4,mm0 ;mm4=[g1:f1]=[gi_0-b:fi_0-a] ++ ++ movd mm0,[edi+ecx*4] ;mm0=[0:fi_k3] ++ punpckldq mm0,[ebp+ecx*4] ;mm0=[gi_k3:fi_k3] ++ movq mm1,mm0 ++ pfmul mm0,[D_s2c2] ;mm0=[ s2*gi_k3:c2*fi_k3] ++ pfmul mm1,[D_Mc2s2] ;mm1=[-c2*gi_k3:s2*fi_k3] ++ pfacc mm0,mm1 ;mm0=[b:a] ++ movd mm5,[edi+ebx*4] ;mm5=[0:fi_k2] ++ movq mm6,mm0 ;mm6=[b:a] ++ punpckldq mm5,[ebp+ebx*4] ;mm5=[gi_k2:fi_k2] ++ pfadd mm6,mm5 ;mm6=[g2:f2]=[gi_k2+b:fi_k2+a] ++ pfsub mm5,mm0 ;mm5=[g3:f3]=[gi_k2-b:fi_k2-a] ++ ++ punpckldq mm1,mm6 ;mm1=[f2:*] ++ movq mm0,[D_s1c1] ;mm0=[s1:c1] ++ punpckhdq mm1,mm5 ;mm1=[g3:f2] ++ pfmul mm0,mm1 ;mm0=[ s1*g3:c1*f2] ++ movq mm2,[D_Mc1s1] ;mm2=[-c1:s1] ++ pfmul mm2,mm1 ;mm2=[-c1*g3:s1*f2] ++ pfacc mm0,mm2 ;mm0=[b:a] ++ ++ punpckldq mm1,mm3 ;mm1=[f0:*] ++ punpckhdq mm1,mm4 ;mm1=[g1:f0] ++ movq mm2,mm0 ;mm2=[b:a] ++ pfadd mm0,mm1 ;mm0=[g1+b:f0+a] ++ pfsubr mm2,mm1 ;mm2=[g1-b:f0-a] ++ movd [edi],mm0 ;fi[0]=f0+a ++ psrlq mm0,32 ;mm0=[0:g1+b] ++ movd [edi+ebx*4],mm2 ;fi[k2]=f0-a ++ psrlq mm2,32 ;mm2=[0:g1-b] ++ movd [ebp+eax*4],mm0 ;gi[k1]=g1+b ++ movd [ebp+ecx*4],mm2 ;gi[k3]=g1-b ++ psrlq mm6,32 ;mm6=[0:g2] ++ movq mm0,[D_s1c1] ;mm0=[s1:c1] ++ punpckldq mm5,mm6 ;mm5=[g2:f3] ++ pfmul mm0,mm5 ;mm0=[g2* s1:f3*c1] ++ pfmul mm5,[D_Mc1s1] ;mm5=[g2*-c1:f3*s1] ++ pfacc mm0,mm5 ;mm0=[-b:a] ++ psrlq mm3,32 ;mm3=[0:g0] ++ movq mm1,mm0 ;mm1=[-b:a] ++ punpckldq mm3,mm4 ;mm3=[f1:g0] ++ pfadd mm0,mm3 ;mm0=[f1-b:g0+a] ++ pfsubr mm1,mm3 ;mm1=[f1+b:g0-a] ++ movd [ebp],mm0 ;gi[0]=g0+a ++ psrlq mm0,32 ;mm0=[0:f1-b] ++ movd [ebp+ebx*4],mm1 ;gi[k2]=g0-a ++ psrlq mm1,32 ;mm1=[0:f1+b] ++ movd [edi+ecx*4],mm0 ;fi[k3]=f1-b ++ movd [edi+eax*4],mm1 ;fi[k1]=f1+b ++ ++ lea edi,[edi+edx*4] ;fi += k4 ++ lea ebp,[ebp+edx*4] ;gi += k4 ++ cmp edi,fn ++ jc near .lp33 ;}while(fi<fn) ++ inc esi ++ cmp esi,kx ++ jnz near .lp32 ;} ++ cmp edx,[esp+_P+8] ++ jnz near .lp30 ;}while(k4<n) ++ ++ ++ .exit: ++ add esp,LOCAL_STACK ++ femms ++ pop ebp ++ pop edi ++ pop esi ++ pop ebx ++ ret +diff -r -c -N encoder/i386/fft3dn.nas lame3.70/i386/fft3dn.nas +*** encoder/i386/fft3dn.nas Wed Dec 31 17:00:00 1969 +--- lame3.70/i386/fft3dn.nas Sun Jan 30 15:53:49 2000 +*************** +*** 0 **** +--- 1,672 ---- ++ ; back port from GOGO-no coda 2.24b by Takehiro TOMINAGA ++ ++ ; GOGO-no-coda ++ ; Copyright (C) 1999 shigeo ++ ; special thanks to URURI ++ ++ %include "nasm.h" ++ ++ externdef costab_fft ++ externdef sintab_fft ++ ++ segment_data ++ align 32 ++ D_1_41421 dd 1.41421356 , 1.41421356 ++ D_1_0 dd 1.0 , 1.0 ++ D_0_5 dd 0.5 , 0.5 ++ D_0_25 dd 0.25 , 0.25 ++ D_0_02236 dd 0.02236067 , 0.02236067 ++ D_0_0005 dd 0.0005 , 0.0005 ++ D_0_0 dd 0.0 , 0.0 ++ ++ D_1_0_D_0_0 dd 0.0 , 1.0 ++ D_0_0_D_1_0 dd 1.0 , 0.0 ++ ++ D_MSB1_0 dd 0x00000000 , 0x80000000 ++ D_MSB1_1 dd 0x80000000 , 0x80000000 ++ D_MSB0_1 dd 0x80000000 , 0x00000000 ++ ++ segment_code ++ ++ ;void fht_3DN2(float *fz, int n); ++ proc fht_3DN2 ++ ++ %$fz arg 4 ++ %$n arg 4 ++ ++ %$k local 4 ++ ++ %$Ps2_Pc2 local 8 ++ %$Mc2_Ps2 local 8 ++ ++ %$t_s local 8 ++ %$t_c local 8 ++ alloc ++ ++ femms ++ pushd ebp, ebx, esi, edi ++ ++ fht_3DN_1st_part: ++ ++ fht_3DN_2nd_part: ++ ++ fht_3DN_3rd_part: ++ ++ .do_init: ++ mov r3, 16 ;k1*fsize = 4*fsize = k4 ++ mov r4, 8 ;kx = k1/2 ++ mov r2, 48 ;k3*fsize ++ mov dword [sp(%$k)], 2 ;k = 2 ++ mov r0, [sp(%$fz)] ;fi ++ lea r1, [r0+8] ;gi = fi + kx ++ jmp .do ++ ++ align 16 ++ .do: ++ pmov mm6, [D_MSB1_0] ;MSB1_0 ++ pmov mm7, [D_1_41421] ++ ++ .do2: ++ ;f ++ pmov mm1, [r0+r3] ;fi1 ++ pmov mm4, [r0+r2] ;fi3 ++ pmov mm0, [r0] ;fi0 ++ pmov mm3, [r0+r3*2] ;fi2 ++ ++ pupldq mm1, mm1 ++ pupldq mm4, mm4 ++ pupldq mm0, mm0 ;fi0 | fi0 ++ pupldq mm3, mm3 ;fi2 | fi2 ++ ++ pxor mm1, mm6 ;-fi1 | fi1 ++ pxor mm4, mm6 ;-fi3 | fi3 ++ ++ pfadd mm0, mm1 ;f1 | f0 ++ pfadd mm3, mm4 ;f3 | f2 ++ ++ pmov mm4, mm0 ++ pfadd mm0, mm3 ;fi1 | fi0 ++ pfsub mm4, mm3 ;fi3 | fi2 ++ ++ pmovd [r0], mm0 ;fi[0] ++ puphdq mm0, mm0 ++ pmovd [r0+r3*2], mm4 ;fi[k2] ++ puphdq mm4, mm4 ++ ++ pmovd [r0+r3], mm0 ;fi[k1] ++ pmovd [r0+r2], mm4 ;fi[k3] ++ ++ ;g ++ pmov mm1, [r1+r3] ;gi1 ++ pmov mm0, [r1] ;gi0 ++ pmov mm3, [r1+r3*2] ;gi2 ++ pmov mm5, [r1+r2] ;gi3 ++ ++ pupldq mm1, mm1 ++ pupldq mm0, mm0 ;gi0 | gi0 ++ pupldq mm3, mm5 ;gi3 | gi2 ++ ++ pxor mm1, mm6 ;-gi1 | gi1 ++ ++ pfadd mm0, mm1 ;g1 | g0 ++ pfmul mm3, mm7 ;g3 | g2 ++ ++ pmov mm4, mm0 ++ pfadd mm0, mm3 ;gi1 | gi0 ++ pfsub mm4, mm3 ;gi3 | gi2 ++ ++ pmovd [r1], mm0 ;gi[0] ++ puphdq mm0, mm0 ++ pmovd [r1+r3*2], mm4 ;gi[k2] ++ puphdq mm4, mm4 ++ ++ pmovd [r1+r3], mm0 ;gi[k1] ++ pmovd [r1+r2], mm4 ;gi[k3] ++ ++ lea r0, [r0+r3*4] ++ lea r1, [r1+r3*4] ++ cmp r0, r6 ++ jb near .do2 ++ ++ ++ mov r0, [sp(%$k)] ++ pmov mm0, [costab_fft +r0*4] ++ pmov mm1, [sintab_fft +r0*4] ++ pupldq mm0, mm0 ++ pupldq mm1, mm1 ++ pmov mm6, [D_1_0_D_0_0] ;c1 | s1 ++ pmov mm7, [D_0_0_D_1_0] ;-s1 | c1 ++ pmov [sp(%$t_c)], mm0 ++ pmov [sp(%$t_s)], mm1 ++ ++ .for_init: ++ mov r5, 4 ;i = 1*fsize ++ jmp .for ++ ++ align 16 ++ .for: ++ pfmul mm6, [sp(%$t_c)] ;c1*t_c | s1*t_c ++ pfmul mm7, [sp(%$t_s)] ;-s1*t_s | c1*t_s ++ ++ pfadd mm6, mm7 ;c1 | s1 ++ pmov mm7, [D_MSB0_1] ++ ++ pmov mm1, mm6 ++ pxor mm7, mm6 ;c1 | -s1 ++ ++ puphdq mm1, mm1 ;c1 ++ pmov mm0, mm7 ++ ++ pupldq mm2, mm7 ++ pfmul mm0, mm6 ;c1*c1 | -s1*s1 ++ pfmul mm1, mm6 ;c1*s1 ++ ++ puphdq mm7, mm2 ;-s1 | c1 ++ pfacc mm0, mm0 ;c2 ++ pfadd mm1, mm1 ;s2 = 2*c1*s1 ++ ++ pupldq mm1, mm0 ;c2 | s2 ++ pupldq mm0, mm1 ;s2 | c2 ++ ++ pxor mm1, [D_MSB1_0] ;-c2 | s2 ++ ++ pmov [sp(%$Ps2_Pc2)], mm0 ++ pmov [sp(%$Mc2_Ps2)], mm1 ++ ++ mov r0, [sp(%$fz)] ++ mov r1, [sp(%$fz)] ++ add r0, r5 ;r0 = fi ++ add r1, r3 ++ sub r1, r5 ;r1 = gi ++ jmp .do3 ++ ++ align 16 ++ .do3: ++ pmov mm2, [r0+r3] ++ pmov mm4, [r1+r3] ++ pmov mm3, [r0+r2] ++ pmov mm5, [r1+r2] ++ ++ pupldq mm2, mm2 ++ pupldq mm4, mm4 ++ pupldq mm3, mm3 ++ pupldq mm5, mm5 ++ ++ pmov mm0, [sp(%$Ps2_Pc2)] ++ pmov mm1, [sp(%$Mc2_Ps2)] ++ ++ pfmul mm2, mm0 ;s2 * fi1 | c2 * fi1 ++ pfmul mm4, mm1 ;-c2 * gi1 | s2 * gi1 ++ pfmul mm3, mm0 ;s2 * fi3 | c2 * fi3 ++ pfmul mm5, mm1 ;-c2 * gi3 | s2 * gi3 ++ ++ pfadd mm2, mm4 ;b | a ++ pfadd mm3, mm5 ;d | c ++ ++ pmov mm0, [r0] ++ pmov mm4, [r1] ++ pmov mm1, [r0+r3*2] ++ pmov mm5, [r1+r3*2] ++ ++ pupldq mm0, mm4 ;gi0 | fi0 ++ pupldq mm1, mm5 ;gi2 | fi2 ++ ++ pmov mm4, mm2 ++ pmov mm5, mm3 ++ ++ pfadd mm2, mm0 ;g0 | f0 ++ pfadd mm3, mm1 ;g2 | f2 ++ ++ pfsub mm0, mm4 ;g1 | f1 ++ pfsub mm1, mm5 ;g3 | f3 ++ ++ ++ pmov mm4, mm3 ++ pmov mm5, mm1 ++ ++ pupldq mm4, mm4 ;f2 | f2 ++ puphdq mm5, mm5 ;g3 | g3 ++ puphdq mm3, mm3 ;g2 | g2 ++ pupldq mm1, mm1 ;f3 | f3 ++ ++ pfmul mm4, mm6 ;f2 * c1 | f2 * s1 ++ pfmul mm5, mm7 ;g3 * -s1 | g3 * c1 ++ pfmul mm3, mm6 ;g2 * c1 | g2 * s1 ++ pfmul mm1, mm7 ;f3 * -s1 | f3 * c1 ++ ++ pfsub mm4, mm5 ;a | b ++ pfadd mm3, mm1 ;d | c ++ ++ pmov mm5, mm2 ++ pmov mm1, mm0 ++ ++ pupldq mm2, mm2 ;f0 | f0 ++ pupldq mm0, mm0 ;f1 | f1 ++ ++ puphdq mm1, mm2 ;f0 | g1 ++ puphdq mm5, mm0 ;f1 | g0 ++ ++ pmov mm2, mm4 ++ pmov mm0, mm3 ++ ++ pfadd mm4, mm1 ;fi0 | gi1 ++ pfadd mm3, mm5 ;fi1 | gi0 ++ pfsub mm1, mm2 ;fi2 | gi3 ++ pfsub mm5, mm0 ;fi3 | gi2 ++ ++ pmovd [r1+r3], mm4 ;gi[k1] ++ puphdq mm4, mm4 ++ pmovd [r1], mm3 ;gi[0] ++ puphdq mm3, mm3 ++ pmovd [r1+r2], mm1 ;gi[k3] ++ puphdq mm1, mm1 ++ pmovd [r1+r3*2], mm5 ;gi[k2] ++ puphdq mm5, mm5 ++ ++ pmovd [r0], mm4 ;fi[0] ++ pmovd [r0+r3], mm3 ;fi[k1] ++ pmovd [r0+r3*2], mm1 ;fi[k2] ++ pmovd [r0+r2], mm5 ;fi[k3] ++ ++ lea r0, [r0+r3*4] ++ lea r1, [r1+r3*4] ++ cmp r0, r6 ++ jb near .do3 ++ ++ add r5, 4 ++ cmp r5, r4 ++ jb near .for ++ ++ cmp r3, [sp(%$n)] ++ jae .exit ++ ++ add dword [sp(%$k)], 2 ;k += 2; ++ lea r3, [r3*4] ;k1 *= 4 ++ lea r2, [r2*4] ;k3 *= 4 ++ lea r4, [r4*4] ;kx *= 4 ++ mov r0, [sp(%$fz)] ;fi ++ lea r1, [r0+r4] ;gi = fi + kx ++ jmp .do ++ ++ .exit: ++ femms ++ popd ebp, ebx, esi, edi ++ endproc ++ ++ ;*********************************************************************** ++ %ifdef USE_E3DN ++ ++ ;void fht_E3DN(float *fz, int n); ++ proc fht_E3DN ++ ++ %$fz arg 4 ++ %$n arg 4 ++ ++ %$k local 4 ++ ++ %$Ps2_Pc2 local 8 ++ %$Mc2_Ps2 local 8 ++ ++ %$t_s local 8 ++ %$t_c local 8 ++ alloc ++ ++ femms ++ pushd ebp, ebx, esi, edi ++ ++ fht_E3DN_1st_part: ++ ++ fht_E3DN_2nd_part: ++ ++ fht_E3DN_3rd_part: ++ ++ .do_init: ++ mov r3, 16 ;k1*fsize = 4*fsize = k4 ++ mov r4, 8 ;kx = k1/2 ++ mov r2, 48 ;k3*fsize ++ mov dword [sp(%$k)], 2 ;k = 2 ++ mov r0, [sp(%$fz)] ;fi ++ lea r1, [r0+8] ;gi = fi + kx ++ jmp .do ++ ++ align 16 ++ .do: ++ pmov mm7, [D_1_41421] ++ ++ .do2: ++ pmov mm0, [r0] ;fi0 ++ pupldq mm0, [r0+r3] ;fi1 | fi0 ++ pmov mm1, [r0+r3*2] ;fi2 ++ pupldq mm1, [r0+r2] ;fi3 | fi2 ++ pmov mm3, [r1] ;gi0 ++ pupldq mm3, [r1+r3] ;gi1 | gi0 ++ pmov mm4, [r1+r2] ;gi3 ++ pupldq mm4, [r1+r3*2] ;gi2 | gi3 ++ ++ pfpnacc mm0, mm0 ;f0 | f1 ++ pfpnacc mm1, mm1 ;f2 | f3 ++ pfpnacc mm3, mm3 ;g0 | g1 ++ pfmul mm4, mm7 ;g2 | g3 ++ ++ pmov mm2, mm0 ++ pfadd mm0, mm1 ;fi0 | fi1 ++ pfsub mm2, mm1 ;fi2 | fi3 ++ pmov mm5, mm3 ++ pfadd mm3, mm4 ;gi0 | gi1 ++ pfsub mm5, mm4 ;gi2 | gi3 ++ ++ pmovd [r0+r3], mm0 ;fi[k1] ++ puphdq mm0, mm0 ++ pmovd [r0+r2], mm2 ;fi[k3] ++ puphdq mm2, mm2 ++ pmovd [r1+r3], mm3 ;gi[k1] ++ puphdq mm3, mm3 ++ pmovd [r1+r2], mm5 ;gi[k3] ++ puphdq mm5, mm5 ++ ++ pmovd [r0], mm0 ;fi[0] ++ pmovd [r0+r3*2], mm2 ;fi[k2] ++ pmovd [r1], mm3 ;gi[0] ++ pmovd [r1+r3*2], mm5 ;gi[k2] ++ ++ lea r0, [r0+r3*4] ++ lea r1, [r1+r3*4] ++ cmp r0, r6 ++ jb near .do2 ++ ++ ++ mov r0, [sp(%$k)] ++ pmov mm0, [costab_fft +r0*4] ++ pmov mm1, [sintab_fft +r0*4] ++ pupldq mm0, mm0 ++ pupldq mm1, mm1 ++ pmov mm6, [D_1_0_D_0_0] ;c1 | s1 ++ pmov mm7, [D_0_0_D_1_0] ;-s1 | c1 ++ pmov [sp(%$t_c)], mm0 ++ pmov [sp(%$t_s)], mm1 ++ ++ .for_init: ++ mov r5, 4 ;i = 1*fsize ++ jmp .for ++ ++ align 16 ++ .for: ++ pfmul mm6, [sp(%$t_c)] ;c1*t_c | s1*t_c ++ pfmul mm7, [sp(%$t_s)] ;-s1*t_s | c1*t_s ++ ++ pfadd mm6, mm7 ;c1 | s1 ++ pmov mm7, [D_MSB0_1] ++ ++ pswapd mm1, mm6 ;s1 | c1 ++ pswapd mm0, mm6 ++ pxor mm7, mm6 ;c1 | -s1 ++ ++ pfmul mm1, mm6 ;c1*s1 | c1*s1 ++ pfmul mm0, mm0 ;s1*s1 | c1*c1 ++ pswapd mm7, mm7 ;-s1 | c1 ++ ++ pfpnacc mm0, mm1 ;s2 = 2*c1*s1 | c2 = c1*c1-s1*s1 ++ pswapd mm1, mm0 ;c2 | s2 ++ pxor mm1, [D_MSB1_0] ;-c2 | s2 ++ ++ pmov [sp(%$Ps2_Pc2)], mm0 ++ pmov [sp(%$Mc2_Ps2)], mm1 ++ ++ mov r0, [sp(%$fz)] ++ mov r1, [sp(%$fz)] ++ add r0, r5 ;r0 = fi ++ add r1, r3 ++ sub r1, r5 ;r1 = gi ++ jmp .do3 ++ ++ align 16 ++ .do3: ++ pmov mm0, [r0+r2] ++ pmov mm2, [r1+r2] ++ pmov mm1, [r0+r3] ++ pmov mm3, [r1+r3] ++ ++ pupldq mm0, mm0 ++ pupldq mm2, mm2 ++ pupldq mm1, mm1 ++ pupldq mm3, mm3 ++ ++ pmov mm4, [sp(%$Ps2_Pc2)] ++ pmov mm5, [sp(%$Mc2_Ps2)] ++ ++ pfmul mm0, mm4 ;s2 * fi3 | c2 * fi3 ++ pfmul mm2, mm5 ;-c2 * gi3 | s2 * gi3 ++ pfmul mm1, mm4 ;s2 * fi1 | c2 * fi1 ++ pfmul mm3, mm5 ;-c2 * gi1 | s2 * gi1 ++ ++ pfadd mm0, mm2 ;d | c ++ pfadd mm1, mm3 ;b | a ++ ++ pmov mm2, [r0+r3*2] ;fi2 ++ pupldq mm3, [r1+r3*2] ;gi2 | - ++ pmov mm4, [r0] ;fi0 ++ pupldq mm5, [r1] ;gi0 | - ++ ++ pupldq mm2, mm0 ;c | fi2 ++ puphdq mm3, mm0 ;d | gi2 ++ pupldq mm4, mm1 ;a | fi0 ++ puphdq mm5, mm1 ;b | gi0 ++ ++ pfpnacc mm2, mm2 ;f2 | f3 ++ pfpnacc mm3, mm3 ;g2 | g3 ++ pfpnacc mm4, mm4 ;f0 | f1 ++ pfpnacc mm5, mm5 ;g0 | g1 ++ ++ pmov mm0, mm2 ++ pmov mm1, mm3 ++ pupldq mm2, mm2 ;f3 | f3 ++ pupldq mm3, mm3 ;g3 | g3 ++ puphdq mm0, mm0 ;f2 | f2 ++ puphdq mm1, mm1 ;g2 | g2 ++ ++ pswapd mm4, mm4 ;f1 | f0 ++ pswapd mm5, mm5 ;g1 | g0 ++ ++ pfmul mm0, mm7 ;f2 *-s1 | f2 * c1 ++ pfmul mm3, mm6 ;g3 * c1 | g3 * s1 ++ pfmul mm1, mm6 ;g2 * c1 | g2 * s1 ++ pfmul mm2, mm7 ;f3 *-s1 | f3 * c1 ++ ++ pfadd mm0, mm3 ;-b | a ++ pfadd mm1, mm2 ; d | c ++ ++ pmov mm2, mm5 ++ pmov mm3, mm4 ++ pupldq mm4, mm0 ; a | f0 ++ pupldq mm5, mm1 ; c | g0 ++ puphdq mm2, mm0 ;-b | g1 ++ puphdq mm3, mm1 ; d | f1 ++ ++ pfpnacc mm4, mm4 ;fi0 | fi2 ++ pfpnacc mm5, mm5 ;gi0 | gi2 ++ pfpnacc mm2, mm2 ;gi3 | gi1 ++ pfpnacc mm3, mm3 ;fi1 | fi3 ++ ++ pmovd [r0+r3*2], mm4 ;fi[k2] ++ puphdq mm4, mm4 ++ pmovd [r1+r3*2], mm5 ;gi[k2] ++ puphdq mm5, mm5 ++ pmovd [r1+r3], mm2 ;gi[k1] ++ puphdq mm2, mm2 ++ pmovd [r0+r2], mm3 ;fi[k3] ++ puphdq mm3, mm3 ++ ++ pmovd [r0], mm4 ;fi[0] ++ pmovd [r1], mm5 ;gi[0] ++ pmovd [r1+r2], mm2 ;gi[k3] ++ pmovd [r0+r3], mm3 ;fi[k1] ++ ++ lea r0, [r0+r3*4] ++ lea r1, [r1+r3*4] ++ cmp r0, r6 ++ jb near .do3 ++ ++ add r5, 4 ++ cmp r5, r4 ++ jb near .for ++ ++ cmp r3, [sp(%$n)] ++ jae .exit ++ ++ add dword [sp(%$k)], 2 ;k += 2; ++ lea r3, [r3*4] ;k1 *= 4 ++ lea r2, [r2*4] ;k3 *= 4 ++ lea r4, [r4*4] ;kx *= 4 ++ mov r0, [sp(%$fz)] ;fi ++ lea r1, [r0+r4] ;gi = fi + kx ++ jmp .do ++ ++ .exit: ++ femms ++ popd ebp, ebx, esi, edi ++ endproc ++ ++ %endif ++ ++ ;*********************************************************************** ++ ++ ;void fft_side_3DN(float in[2][1024], int s, float *ret); /* s = MSFREQ ��4���ܿ����ꡣ��������20���ä� */ ++ proc fft_side_3DN ++ ++ %$in arg 4 ++ %$s arg 4 ++ %$pret arg 4 ++ ++ femms ++ pushd ebx, esi ++ ++ .for_init: ++ mov r0, [sp(%$in)] ;r0 = &in[0][0] ++ lea r1, [r0+fsizen(1024)] ;r1 = &in[1][0] ++ ++ mov r2, [sp(%$s)] ++ mov r3, fsizen(1023) ++ shl r2, 2 ;r2 = s * fsize ++ sub r3, r2 ;r3 = (1023-s) * fsize ++ ++ mov r4, fsizen(512) ;r4 = 512 * fsize ++ pxor mm7, mm7 ++ jmp .for ++ ++ align 16 ++ .for: ++ pmov mm0, [r0+r3] ++ pmov mm1, [r1+r3] ++ pmov mm2, [r0+r2] ++ pfsub mm0, mm1 ++ pmov mm3, [r1+r2] ++ pfsub mm2, mm3 ++ pmov mm4, [r0+r3-fsizen(2)] ++ pfmul mm0, mm0 ++ pmov mm5, [r1+r3-fsizen(2)] ++ pfmul mm2, mm2 ++ pmov mm1, [r0+r2+fsizen(2)] ++ pupldq mm6, mm0 ++ pmov mm3, [r1+r2+fsizen(2)] ++ pfadd mm7, mm2 ++ pfsub mm4, mm5 ++ puphdq mm0, mm6 ++ pfsub mm1, mm3 ++ pfadd mm7, mm0 ++ ++ pfmul mm4, mm4 ++ add r2, fsizen(4) ++ pfmul mm1, mm1 ++ sub r3, fsizen(4) ++ pupldq mm6, mm4 ++ pfadd mm7, mm1 ++ cmp r2, r4 ++ puphdq mm4, mm6 ++ pfadd mm7, mm4 ++ jb .for ++ ++ pmov mm0, [r0+r2] ++ pmov mm1, [r1+r2] ++ pmov mm2, [D_0_25] ++ pfsub mm0, mm1 ++ pmov mm3, [D_0_5] ++ pfacc mm7, mm7 ++ pfmul mm0, mm0 ++ pfmul mm7, mm2 ++ pfmul mm0, mm3 ++ mov r0, [sp(%$pret)] ++ pfadd mm0, mm7 ++ ++ pmovd [r0], mm0 ++ .exit: ++ femms ++ popd ebx, esi ++ endproc ++ ++ ;*********************************************************************** ++ %ifdef USE_E3DN ++ ++ ;void fft_side_E3DN(float in[2][1024], int s, float *ret); /* s = MSFREQ ��4���ܿ����ꡣ��������20���ä� */ ++ proc fft_side_E3DN ++ ++ %$in arg 4 ++ %$s arg 4 ++ %$pret arg 4 ++ ++ femms ++ pushd ebx, esi ++ ++ .for_init: ++ mov r0, [sp(%$in)] ;r0 = &in[0][0] ++ lea r1, [r0+fsizen(1024)] ;r1 = &in[1][0] ++ ++ mov r2, [sp(%$s)] ++ mov r3, fsizen(1023) ++ shl r2, 2 ;r2 = s * fsize ++ sub r3, r2 ;r3 = (1023-s) * fsize ++ ++ mov r4, fsizen(512) ;r4 = 512 * fsize ++ pxor mm7, mm7 ++ jmp .for ++ ++ align 16 ++ .for: ++ pmov mm0, [r0+r3] ++ pfsub mm0, [r1+r3] ++ pmov mm4, [r0+r3-fsizen(2)] ++ pfsub mm4, [r1+r3-fsizen(2)] ++ pmov mm2, [r0+r2] ++ pfsub mm2, [r1+r2] ++ pmov mm1, [r0+r2+fsizen(2)] ++ pfsub mm1, [r1+r2+fsizen(2)] ++ pfmul mm0, mm0 ++ pfmul mm4, mm4 ++ pfmul mm2, mm2 ++ pfmul mm1, mm1 ++ pswapd mm0, mm0 ++ pswapd mm4, mm4 ++ pfadd mm7, mm2 ++ add r2, fsizen(4) ++ pfadd mm7, mm1 ++ sub r3, fsizen(4) ++ pfadd mm7, mm0 ++ cmp r2, r4 ++ pfadd mm7, mm4 ++ jb .for ++ ++ pmov mm0, [r0+r2] ++ pfsub mm0, [r1+r2] ++ pfacc mm7, mm7 ++ pfmul mm0, mm0 ++ pfmul mm7, [D_0_25] ++ pfmul mm0, [D_0_5] ++ mov r0, [sp(%$pret)] ++ pfadd mm0, mm7 ++ ++ pmovd [r0], mm0 ++ .exit: ++ femms ++ popd ebx, esi ++ endproc ++ ++ %endif +diff -r -c -N encoder/i386/fftfpu.nas lame3.70/i386/fftfpu.nas +*** encoder/i386/fftfpu.nas Wed Dec 31 17:00:00 1969 +--- lame3.70/i386/fftfpu.nas Sun Jan 30 15:53:49 2000 +*************** +*** 0 **** +--- 1,619 ---- ++ ; back port from GOGO-no coda 2.24b by Takehiro TOMINAGA ++ ++ ; GOGO-no-coda ++ ; Copyright (C) 1999 shigeo ++ ; special thanks to URURI ++ ++ %include "nasm.h" ++ ++ externdef costab_fft ++ externdef sintab_fft ++ ++ segment_data ++ align 32 ++ D_1_41421 dd 1.41421356 ++ D_1_0 dd 1.0 ++ D_0_5 dd 0.5 ++ D_0_25 dd 0.25 ++ D_0_0005 dd 0.0005 ++ D_0_0 dd 0.0 ++ ++ segment_code ++ ++ ;void fht(float *fz, int n); ++ proc fht_FPU ++ ++ %$fz arg 4 ++ %$n arg 4 ++ ++ %$k local 4 ++ ++ %$f0 local 4 ++ %$f1 local 4 ++ %$f2 local 4 ++ %$f3 local 4 ++ ++ %$g0 local 4 ++ %$g1 local 4 ++ %$g2 local 4 ++ %$g3 local 4 ++ ++ %$s1 local 4 ++ %$c1 local 4 ++ %$s2 local 4 ++ %$c2 local 4 ++ ++ %$t_s local 4 ++ %$t_c local 4 ++ alloc ++ ++ pushd ebp, ebx, esi, edi ++ ++ fht_FPU_1st_part: ++ ++ fht_FPU_2nd_part: ++ ++ fht_FPU_3rd_part: ++ ++ .do_init: ++ mov r3, 16 ;k1*fsize = 4*fsize = k4 ++ mov r4, 8 ;kx = k1/2 ++ mov r2, 48 ;k3*fsize ++ mov dword [sp(%$k)], 2 ;k = 2 ++ mov r0, [sp(%$fz)] ;fi ++ lea r1, [r0+8] ;gi = fi + kx ++ ++ .do: ++ .do2: ++ ;f ++ fld dword [r0] ++ fsub dword [r0+r3] ++ ++ fld dword [r0] ++ fadd dword [r0+r3] ++ ++ fld dword [r0+r3*2] ++ fsub dword [r0+r2] ++ ++ fld dword [r0+r3*2] ++ fadd dword [r0+r2] ;f2 f3 f0 f1 ++ ++ fld st2 ;f0 f2 f3 f0 f1 ++ fadd st0, st1 ++ fstp dword [r0] ;fi[0] ++ ++ fld st3 ;f1 f2 f3 f0 f1 ++ fadd st0, st2 ++ fstp dword [r0+r3] ;fi[k1] ++ ++ fsubr st0, st2 ;f0-f2 f3 f0 f1 ++ fstp dword [r0+r3*2] ;fi[k2] ++ ++ fsubr st0, st2 ;f1-f3 f0 f1 ++ fstp dword [r0+r2] ;fi[k3] ++ fcompp ++ ++ ;g ++ fld dword [r1] ++ fsub dword [r1+r3] ++ ++ fld dword [r1] ++ fadd dword [r1+r3] ++ ++ fld dword [D_1_41421] ++ fmul dword [r1+r2] ++ ++ fld dword [D_1_41421] ++ fmul dword [r1+r3*2] ;g2 g3 g0 g1 ++ ++ fld st2 ;g0 g2 g3 g0 g1 ++ fadd st0, st1 ++ fstp dword [r1] ;gi[0] ++ ++ fld st3 ;g1 g2 g3 g0 g1 ++ fadd st0, st2 ++ fstp dword [r1+r3] ;gi[k1] ++ ++ fsubr st0, st2 ;g0-g2 g3 g0 g1 ++ fstp dword [r1+r3*2] ;gi[k2] ++ ++ fsubr st0, st2 ;g1-g3 g0 g1 ++ fstp dword [r1+r2] ;gi[k3] ++ fcompp ++ ++ lea r0, [r0+r3*4] ++ lea r1, [r1+r3*4] ++ cmp r0, r6 ++ jb .do2 ++ ++ ++ mov r0, [sp(%$k)] ++ fld dword [costab_fft +r0*4] ++ fstp dword [sp(%$t_c)] ++ fld dword [sintab_fft +r0*4] ++ fstp dword [sp(%$t_s)] ++ fld dword [D_1_0] ++ fstp dword [sp(%$c1)] ++ fld dword [D_0_0] ++ fstp dword [sp(%$s1)] ++ ++ .for_init: ++ mov r5, 4 ;i = 1*fsize ++ ++ .for: ++ fld dword [sp(%$c1)] ++ fmul dword [sp(%$t_c)] ++ fld dword [sp(%$s1)] ++ fmul dword [sp(%$t_s)] ++ fsubp st1, st0 ;c1 ++ ++ fld dword [sp(%$c1)] ++ fmul dword [sp(%$t_s)] ++ fld dword [sp(%$s1)] ++ fmul dword [sp(%$t_c)] ++ faddp st1, st0 ;s1 c1 ++ ++ fld st1 ++ fmul st0, st0 ;c1c1 s1 c1 ++ fld st1 ++ fmul st0, st0 ;s1s1 c1c1 s1 c1 ++ fsubp st1, st0 ;c2 s1 c1 ++ fstp dword [sp(%$c2)] ;s1 c1 ++ ++ fld st1 ;c1 s1 c1 ++ fmul st0, st1 ;c1s1 s1 c1 ++ fadd st0, st0 ;s2 s1 c1 ++ fstp dword [sp(%$s2)] ;s1 c1 ++ ++ fstp dword [sp(%$s1)] ;c1 ++ fstp dword [sp(%$c1)] ; ++ ++ mov r0, [sp(%$fz)] ++ add r0, r5 ;r0 = fi ++ mov r1, [sp(%$fz)] ++ add r1, r3 ++ sub r1, r5 ;r1 = gi ++ ++ .do3: ++ fld dword [sp(%$s2)] ++ fmul dword [r0+r3] ++ fld dword [sp(%$c2)] ++ fmul dword [r1+r3] ++ fsubp st1, st0 ;b = s2*fi[k1] - c2*gi[k1] ++ ++ fld dword [sp(%$c2)] ++ fmul dword [r0+r3] ++ fld dword [sp(%$s2)] ++ fmul dword [r1+r3] ++ faddp st1, st0 ;a = c2*fi[k1] + s2*gi[k1] b ++ ++ fld dword [r0] ++ fsub st0, st1 ;f1 a b ++ fstp dword [sp(%$f1)] ;a b ++ ++ fadd dword [r0] ;f0 b ++ fstp dword [sp(%$f0)] ;b ++ ++ fld dword [r1] ++ fsub st0, st1 ;g1 b ++ fstp dword [sp(%$g1)] ;b ++ ++ fadd dword [r1] ;g0 ++ fstp dword [sp(%$g0)] ; ++ ++ ++ fld dword [sp(%$s2)] ++ fmul dword [r0+r2] ++ fld dword [sp(%$c2)] ++ fmul dword [r1+r2] ++ fsubp st1, st0 ;b = s2*fi[k3] - c2*gi[k3] ++ ++ fld dword [sp(%$c2)] ++ fmul dword [r0+r2] ++ fld dword [sp(%$s2)] ++ fmul dword [r1+r2] ++ faddp st1, st0 ;a = c2*fi[k3] + s2*gi[k3] b ++ ++ fld dword [r0+r3*2] ++ fsub st0, st1 ;f3 a b ++ fstp dword [sp(%$f3)] ;a b ++ ++ fadd dword [r0+r3*2] ;f2 b ++ fstp dword [sp(%$f2)] ;b ++ ++ fld dword [r1+r3*2] ++ fsub st0, st1 ;g3 b ++ fstp dword [sp(%$g3)] ;b ++ ++ fadd dword [r1+r3*2] ;g2 ++ fstp dword [sp(%$g2)] ; ++ ++ ++ fld dword [sp(%$s1)] ++ fmul dword [sp(%$f2)] ++ fld dword [sp(%$c1)] ++ fmul dword [sp(%$g3)] ++ fsubp st1, st0 ;b = s1*f2 - c1*g3 ++ ++ fld dword [sp(%$c1)] ++ fmul dword [sp(%$f2)] ++ fld dword [sp(%$s1)] ++ fmul dword [sp(%$g3)] ++ faddp st1, st0 ;a = c1*f2 + s1*g3 b ++ ++ fld dword [sp(%$f0)] ++ fsub st0, st1 ;fi[k2] a b ++ fstp dword [r0+r3*2] ++ ++ fadd dword [sp(%$f0)] ;fi[0] b ++ fstp dword [r0] ++ ++ fld dword [sp(%$g1)] ++ fsub st0, st1 ;gi[k3] b ++ fstp dword [r1+r2] ++ ++ fadd dword [sp(%$g1)] ;gi[k1] ++ fstp dword [r1+r3] ++ ++ ++ fld dword [sp(%$c1)] ++ fmul dword [sp(%$g2)] ++ fld dword [sp(%$s1)] ++ fmul dword [sp(%$f3)] ++ fsubp st1, st0 ;b = c1*g2 - s1*f3 ++ ++ fld dword [sp(%$s1)] ++ fmul dword [sp(%$g2)] ++ fld dword [sp(%$c1)] ++ fmul dword [sp(%$f3)] ++ faddp st1, st0 ;a = s1*g2 + c1*f3 b ++ ++ fld dword [sp(%$g0)] ++ fsub st0, st1 ;gi[k2] a b ++ fstp dword [r1+r3*2] ++ ++ fadd dword [sp(%$g0)] ;gi[0] b ++ fstp dword [r1] ++ ++ fld dword [sp(%$f1)] ++ fsub st0, st1 ;fi[k3] b ++ fstp dword [r0+r2] ++ ++ fadd dword [sp(%$f1)] ;fi[k1] ++ fstp dword [r0+r3] ++ ++ ++ lea r0, [r0+r3*4] ++ lea r1, [r1+r3*4] ++ cmp r0, r6 ++ jb near .do3 ++ ++ add r5, 4 ++ cmp r5, r4 ++ jb near .for ++ ++ cmp r3, [sp(%$n)] ++ jae .exit ++ ++ add dword [sp(%$k)], 2 ;k += 2; ++ lea r3, [r3*4] ;k1 *= 4 ++ lea r2, [r2*4] ;k3 *= 4 ++ lea r4, [r4*4] ;kx *= 4 ++ mov r0, [sp(%$fz)] ;fi ++ lea r1, [r0+r4] ;gi = fi + kx ++ jmp .do ++ ++ .exit: ++ popd ebp, ebx, esi, edi ++ endproc ++ ++ ;************************************************************* ++ ++ ;void fht_FPU_FXCH(float *fz, int n); ++ proc fht_FPU_FXCH ++ ++ %$fz arg 4 ++ %$n arg 4 ++ ++ %$k local 4 ++ ++ %$f0 local 4 ++ %$f1 local 4 ++ %$f2 local 4 ++ %$f3 local 4 ++ ++ %$g0 local 4 ++ %$g1 local 4 ++ %$g2 local 4 ++ %$g3 local 4 ++ ++ %$s1 local 4 ++ %$c1 local 4 ++ %$s2 local 4 ++ %$c2 local 4 ++ ++ %$t_s local 4 ++ %$t_c local 4 ++ alloc ++ ++ pushd ebp, ebx, esi, edi ++ ++ fht_FPU_FXCH_1st_part: ++ ++ fht_FPU_FXCH_2nd_part: ++ ++ fht_FPU_FXCH_3rd_part: ++ ++ .do_init: ++ mov r3, 16 ;k1*fsize = 4*fsize = k4 ++ mov r4, 8 ;kx = k1/2 ++ mov r2, 48 ;k3*fsize ++ mov dword [sp(%$k)], 2 ;k = 2 ++ mov r0, [sp(%$fz)] ;fi ++ lea r1, [r0+8] ;gi = fi + kx ++ ++ .do: ++ .do2: ++ ;f ++ fld dword [r0] ++ fsub dword [r0+r3] ++ fld dword [r0] ++ fadd dword [r0+r3] ++ ++ fld dword [r0+r3*2] ++ fsub dword [r0+r2] ++ fld dword [r0+r3*2] ++ fadd dword [r0+r2] ;f2 f3 f0 f1 ++ ++ fld st3 ++ fld st3 ++ fxch st5 ++ fadd st0, st3 ++ fxch st4 ++ fadd st0, st2 ++ fxch st3 ++ fsubp st1, st0 ++ fxch st1 ++ fsubp st4, st0 ++ fxch st2 ++ ++ fstp dword [r0+r3] ;fi[k1] ++ fstp dword [r0] ;fi[0] ++ fstp dword [r0+r2] ;fi[k3] ++ fstp dword [r0+r3*2] ;fi[k2] ++ ++ ;g ++ fld dword [r1] ++ fsub dword [r1+r3] ++ fld dword [r1] ++ fadd dword [r1+r3] ++ ++ fld dword [D_1_41421] ++ fmul dword [r1+r2] ++ fld dword [D_1_41421] ++ fmul dword [r1+r3*2] ;g2 g3 g0 g1 ++ ++ fld st3 ++ fld st3 ++ fxch st5 ++ fadd st0, st3 ++ fxch st4 ++ fadd st0, st2 ++ fxch st3 ++ fsubp st1, st0 ++ fxch st1 ++ fsubp st4, st0 ++ fxch st2 ++ ++ fstp dword [r1+r3] ;gi[k1] ++ fstp dword [r1] ;gi[0] ++ fstp dword [r1+r2] ;gi[k3] ++ fstp dword [r1+r3*2] ;gi[k2] ++ ++ lea r0, [r0+r3*4] ++ lea r1, [r1+r3*4] ++ cmp r0, r6 ++ jb .do2 ++ ++ ++ mov r0, [sp(%$k)] ++ fld dword [costab_fft +r0*4] ++ fld dword [sintab_fft +r0*4] ++ fld dword [D_1_0] ++ fld dword [D_0_0] ++ fxch st3 ++ fstp dword [sp(%$t_c)] ++ fxch st1 ++ fstp dword [sp(%$t_s)] ++ fstp dword [sp(%$c1)] ++ fstp dword [sp(%$s1)] ++ ++ .for_init: ++ mov r5, 4 ;i = 1*fsize ++ ++ .for: ++ fld dword [sp(%$c1)] ++ fmul dword [sp(%$t_c)] ++ fld dword [sp(%$s1)] ++ fmul dword [sp(%$t_s)] ++ ++ fld dword [sp(%$c1)] ++ fmul dword [sp(%$t_s)] ++ fld dword [sp(%$s1)] ++ fmul dword [sp(%$t_c)] ++ fxch st2 ++ fsubp st3, st0 ;c1 ++ faddp st1, st0 ;s1 c1 ++ ++ fld st1 ++ fxch st2 ++ fmul st0, st0 ;c1c1 s1 c1 ++ fld st1 ++ fxch st2 ++ fmul st0, st0 ;s1s1 c1c1 s1 c1 ++ ++ fxch st3 ++ fst dword [sp(%$c1)] ;c1 ++ fxch st2 ++ fst dword [sp(%$s1)] ;s1 c1c1 c1 s1s1 ++ ++ fmulp st2, st0 ++ fsubrp st2, st0 ++ fadd st0, st0 ;s2 c2 ++ fxch st1 ++ fstp dword [sp(%$c2)] ++ fstp dword [sp(%$s2)] ++ ++ mov r0, [sp(%$fz)] ++ mov r1, [sp(%$fz)] ++ add r0, r5 ;r0 = fi ++ add r1, r3 ++ sub r1, r5 ;r1 = gi ++ ++ .do3: ++ fld dword [sp(%$s2)] ++ fmul dword [r0+r3] ++ fld dword [sp(%$c2)] ++ fmul dword [r1+r3] ++ ++ fld dword [sp(%$c2)] ++ fmul dword [r0+r3] ++ fld dword [sp(%$s2)] ++ fmul dword [r1+r3] ++ fxch st2 ++ fsubp st3, st0 ;b = s2*fi[k1] - c2*gi[k1] ++ faddp st1, st0 ;a = c2*fi[k1] + s2*gi[k1] b ++ ++ fld dword [r1] ++ fsub st0, st2 ;g1 a b ++ fxch st2 ++ fadd dword [r1] ;g0 a g1 ++ ++ fld dword [r0] ++ fsub st0, st2 ;f1 g0 a g1 ++ fxch st2 ++ fadd dword [r0] ;f0 g0 f1 g1 ++ ++ fxch st3 ++ fstp dword [sp(%$g1)] ++ fstp dword [sp(%$g0)] ++ fstp dword [sp(%$f1)] ++ fstp dword [sp(%$f0)] ++ ++ ++ fld dword [sp(%$s2)] ++ fmul dword [r0+r2] ++ fld dword [sp(%$c2)] ++ fmul dword [r1+r2] ++ ++ fld dword [sp(%$c2)] ++ fmul dword [r0+r2] ++ fld dword [sp(%$s2)] ++ fmul dword [r1+r2] ++ fxch st2 ++ fsubp st3, st0 ;b = s2*fi[k3] - c2*gi[k3] ++ faddp st1, st0 ;a = c2*fi[k3] + s2*gi[k3] b ++ ++ ++ fld dword [r1+r3*2] ++ fsub st0, st2 ;g3 a b ++ fxch st2 ++ fadd dword [r1+r3*2] ;g2 a g3 ++ ++ fld dword [r0+r3*2] ++ fsub st0, st2 ;f3 g2 a g3 ++ fxch st2 ++ fadd dword [r0+r3*2] ;f2 g2 f3 g3 ++ ++ fxch st3 ++ fstp dword [sp(%$g3)] ++ fstp dword [sp(%$g2)] ++ fstp dword [sp(%$f3)] ++ fstp dword [sp(%$f2)] ++ ++ ++ fld dword [sp(%$s1)] ++ fmul dword [sp(%$f2)] ++ fld dword [sp(%$c1)] ++ fmul dword [sp(%$g3)] ++ ++ fld dword [sp(%$c1)] ++ fmul dword [sp(%$f2)] ++ fld dword [sp(%$s1)] ++ fmul dword [sp(%$g3)] ++ fxch st2 ++ fsubp st3, st0 ;b = s1*f2 - c1*g3 ++ faddp st1, st0 ;a = c1*f2 + s1*g3 b ++ ++ fld dword [sp(%$g1)] ++ fsub st0, st2 ;gi[k3] a b ++ fxch st2 ++ fadd dword [sp(%$g1)] ;gi[k1] a gi[k3] ++ ++ fld dword [sp(%$f0)] ++ fsub st0, st2 ;fi[k2] gi[k1] a gi[k3] ++ fxch st2 ++ fadd dword [sp(%$f0)] ;fi[0] gi[k1] fi[k2] gi[k3] ++ ++ fxch st3 ++ fstp dword [r1+r2] ++ fstp dword [r1+r3] ++ fstp dword [r0+r3*2] ++ fstp dword [r0] ++ ++ ++ fld dword [sp(%$c1)] ++ fmul dword [sp(%$g2)] ++ fld dword [sp(%$s1)] ++ fmul dword [sp(%$f3)] ++ ++ fld dword [sp(%$s1)] ++ fmul dword [sp(%$g2)] ++ fld dword [sp(%$c1)] ++ fmul dword [sp(%$f3)] ++ fxch st2 ++ fsubp st3, st0 ;b = c1*g2 - s1*f3 ++ faddp st1, st0 ;a = s1*g2 + c1*f3 b ++ ++ fld dword [sp(%$f1)] ++ fsub st0, st2 ;fi[k3] a b ++ fxch st2 ++ fadd dword [sp(%$f1)] ;fi[k1] a fi[k3] ++ ++ fld dword [sp(%$g0)] ++ fsub st0, st2 ;gi[k2] fi[k1] a fi[k3] ++ fxch st2 ++ fadd dword [sp(%$g0)] ;gi[0] fi[k1] gi[k2] fi[k3] ++ ++ fxch st3 ++ fstp dword [r0+r2] ++ fstp dword [r0+r3] ++ fstp dword [r1+r3*2] ++ fstp dword [r1] ++ ++ ++ lea r0, [r0+r3*4] ++ lea r1, [r1+r3*4] ++ cmp r0, r6 ++ jb near .do3 ++ ++ add r5, 4 ++ cmp r5, r4 ++ jb near .for ++ ++ cmp r3, [sp(%$n)] ++ jae .exit ++ ++ add dword [sp(%$k)], 2 ;k += 2; ++ lea r3, [r3*4] ;k1 *= 4 ++ lea r2, [r2*4] ;k3 *= 4 ++ lea r4, [r4*4] ;kx *= 4 ++ mov r0, [sp(%$fz)] ;fi ++ lea r1, [r0+r4] ;gi = fi + kx ++ jmp .do ++ ++ .exit: ++ popd ebp, ebx, esi, edi ++ endproc ++ ++ end +diff -r -c -N encoder/i386/fftsse.nas lame3.70/i386/fftsse.nas +*** encoder/i386/fftsse.nas Wed Dec 31 17:00:00 1969 +--- lame3.70/i386/fftsse.nas Sun Jan 30 15:53:49 2000 +*************** +*** 0 **** +--- 1,538 ---- ++ ; back port from GOGO-no coda 2.24b by Takehiro TOMINAGA ++ ++ ; GOGO-no-coda ++ ; Copyright (C) 1999 shigeo ++ ; special thanks to Keiichi SAKAI ++ ++ %include "nasm.h" ++ ++ globaldef fht_SSE ++ globaldef fft_side_SSE ++ externdef costab_fft ++ externdef sintab_fft ++ ++ segment_data ++ align 16 ++ Q_MMPP dd 0x0,0x0,0x80000000,0x80000000 ++ Q_MPMP dd 0x0,0x80000000,0x0,0x80000000 ++ Q_002 dd 0.02236068, 0.02236068, 0.02236068, 0.02236068 ++ D_SQRT2 dd 1.414213562,1.414213562 ++ S_025 dd 0.25 ++ S_05 DD 0.5 ++ S_00005 DD 0.0005 ++ ++ segment_code ++ ;------------------------------------------------------------------------ ++ ; by K. SAKAI ++ ; 99/08/18 PIII 23k[clk] ++ ; 99/08/19 ̿�������촹�� PIII 22k[clk] ++ ; 99/08/20 bit reversal ����夫��ܿ����� PIII 17k[clk] ++ ; 99/08/23 ���� unroll PIII 14k[clk] ++ ; 99/11/12 clean up ++ ; ++ ;void fht_SSE(float *fz, int n); ++ align 16 ++ fht_SSE: ++ push ebx ++ push esi ++ push edi ++ push ebp ++ %assign _P 4*4 ++ ++ ;2���ܤΥ롼�� ++ mov eax,[esp+_P+4] ;eax=fz ++ mov ebp,[esp+_P+8] ;=n ++ shl ebp,2 ++ add ebp,eax ; fn = fz + n, ���δؿ���λ�ޤ����� ++ ++ xor ecx,ecx ; ecx=k=0 ++ xor eax,eax ++ mov al,4 ; =k1=1*(sizeof float) // 4, 16, 64, 256,... ++ xor edx,edx ++ mov dl,12 ; =k3=3*k1 ++ jmp short .lp2 ++ ++ align 16 ++ .lp2: ; do{ ++ add cl,2 ; k += 2; ++ shl eax,2 ++ shl edx,2 ++ ++ mov esi,[esp+_P+4] ;esi=fi=fz ++ mov edi,eax ++ shr edi,1 ++ add edi,esi ; edi=gi=fi+ki/2 ++ ++ ; ��������2���󤷤����ԤǤ��ʤ���ʬ��FPU�Τۤ���®���� ++ movss xmm7,[D_SQRT2] ++ jmp short .lp20 ++ ++ align 16 ++ .lp20: ; do{ ++ ; f0 = fi[0 ] + fi[k1]; ++ ; f2 = fi[k2] + fi[k3]; ++ ; f1 = fi[0 ] - fi[k1]; ++ ; f3 = fi[k2] - fi[k3]; ++ ; fi[0 ] = f0 + f2; ++ ; fi[k1] = f1 + f3; ++ ; fi[k2] = f0 - f2; ++ ; fi[k3] = f1 - f3; ++ fld dword [esi] ++ fadd dword [esi+eax] ++ fld dword [esi+eax*2] ++ fadd dword [esi+edx] ++ ++ fld dword [esi] ++ fsub dword [esi+eax] ++ fld dword [esi+eax*2] ++ fsub dword [esi+edx] ++ ++ fld st1 ++ fadd st0,st1 ++ fstp dword [esi+eax] ++ fsubp st1,st0 ++ fstp dword [esi+edx] ++ ++ fld st1 ++ fadd st0,st1 ++ fstp dword [esi] ++ fsubp st1,st0 ++ fstp dword [esi+eax*2] ++ ++ lea esi,[esi + eax*4] ; = fi += (k1 * 4); ++ ; add esi,eax ++ ; add esi,edx ++ ; g0 = gi[0 ] + gi[k1]; ++ ; g2 = SQRT2 * gi[k2]; ++ ; g1 = gi[0 ] - gi[k1]; ++ ; g3 = SQRT2 * gi[k3]; ++ ; gi[0 ] = g0 + g2; ++ ; gi[k2] = g0 - g2; ++ ; gi[k1] = g1 + g3; ++ ; gi[k3] = g1 - g3; ++ fld dword [edi] ++ fadd dword [edi+eax] ++ fld dword [D_SQRT2] ++ fmul dword [edi+eax*2] ++ ++ fld dword [edi] ++ fsub dword [edi+eax] ++ fld dword [D_SQRT2] ++ fmul dword [edi+edx] ++ ++ fld st1 ++ fadd st0,st1 ++ fstp dword [edi+eax] ++ fsubp st1,st0 ++ fstp dword [edi+edx] ++ ++ fld st1 ++ fadd st0,st1 ++ fstp dword [edi] ++ fsubp st1,st0 ++ fstp dword [edi+eax*2] ++ ++ lea edi,[edi + eax*4] ; = gi += (k1 * 4); ++ cmp esi,ebp ++ jl near .lp20 ; while (fi<fn); ++ ++ ; i = 1; //for (i=1;i<kx;i++){ ++ ; c1 = 1.0*t_c - 0.0*t_s; ++ ; s1 = 0.0*t_c + 1.0*t_s; ++ movss xmm6,[costab_fft + ecx*4] ++ movss xmm1,[sintab_fft + ecx*4] ++ shufps xmm6,xmm1,0x00 ; = {s1, s1, c1, c1} ++ shufps xmm6,xmm6,0x28 ; = {+c1, +s1, +s1, +c1} ++ ; c2 = c1*c1 - s1*s1; ++ ; s2 = c1*s1 + s1*c1; ++ movaps xmm4,xmm6 ++ movaps xmm7,xmm6 ++ unpcklps xmm4,xmm4 ; = {s1, s1, c1, c1} ++ shufps xmm7,xmm7,0x41 ++ mulps xmm4,xmm6 ; = {s1*c1, s1*s1, c1*s1, c1*c1} ++ xorps xmm7,[Q_MMPP] ; = {-s1, -c1, +c1, +s1} ++ movhlps xmm3,xmm4 ++ xorps xmm3,[Q_MPMP] ++ subps xmm4,xmm3 ; = {--, --, s2, c2} ++ movlhps xmm4,xmm4 ; = {+s2, +c2, +s2, +c2} ++ movaps xmm5,xmm4 ++ shufps xmm5,xmm5,0x11 ++ xorps xmm5,[Q_MPMP] ; = {-c2, +s2, -c2, +s2} ++ mov esi,[esp+_P+4] ; = fz ++ lea edi,[esi + eax - 4] ; edi = gi = fz +k1-i ++ add esi,4 ; esi = fi = fz + i ++ jmp short .lp21 ++ ++ align 16 ++ .lp21: ; do{ ++ ; a = c2*fi[k1] + s2*gi[k1]; ++ ; b = s2*fi[k1] - c2*gi[k1]; ++ ; c = c2*fi[k3] + s2*gi[k3]; ++ ; d = s2*fi[k3] - c2*gi[k3]; ++ ; f0 = fi[0 ] + a; ++ ; g0 = gi[0 ] + b; ++ ; f2 = fi[k1 * 2] + c; ++ ; g2 = gi[k1 * 2] + d; ++ ; f1 = fi[0 ] - a; ++ ; g1 = gi[0 ] - b; ++ ; f3 = fi[k1 * 2] - c; ++ ; g3 = gi[k1 * 2] - d; ++ ++ movss xmm0,[esi + eax] ; = fi[k1] ++ movss xmm2,[esi + edx] ; = fi[k3] ++ shufps xmm0,xmm2,0x00 ; = {fi[k3], fi[k3], fi[k1], fi[k1]} ++ movss xmm1,[edi + eax] ; = fi[k1] ++ movss xmm3,[edi + edx] ; = fi[k3] ++ shufps xmm1,xmm3,0x00 ; = {gi[k3], gi[k3], gi[k1], gi[k1]} ++ movss xmm2,[esi] ; = fi[0] ++ mulps xmm0,xmm4 ; *= {+s2, +c2, +s2, +c2} ++ movss xmm3,[esi + eax*2] ; = fi[k2] ++ unpcklps xmm2,xmm3 ; = {--, --, fi[k2], fi[0]} ++ mulps xmm1,xmm5 ; *= {-c2, +s2, -c2, +s2} ++ movss xmm3,[edi + eax*2] ; = gi[k2] ++ addps xmm0,xmm1 ; = {d, c, b, a} ++ movss xmm1,[edi] ; = gi[0] ++ unpcklps xmm1,xmm3 ; = {--, --, gi[k2], gi[0]} ++ unpcklps xmm2,xmm1 ; = {gi[k2], fi[k2], gi[0], fi[0]} ++ movaps xmm1,xmm2 ++ addps xmm1,xmm0 ; = {g2, f2, g0, f0} ++ subps xmm2,xmm0 ; = {g3, f3, g1, f1} ++ ++ ; a = c1*f2 + s1*g3; ++ ; c = s1*g2 + c1*f3; ++ ; b = s1*f2 - c1*g3; ++ ; d = c1*g2 - s1*f3; ++ ; fi[0 ] = f0 + a; ++ ; gi[0 ] = g0 + c; ++ ; gi[k1] = g1 + b; ++ ; fi[k1] = f1 + d; ++ ; fi[k1 * 2] = f0 - a; ++ ; gi[k1 * 2] = g0 - c; ++ ; gi[k3] = g1 - b; ++ ; fi[k3] = f1 - d; ++ movaps xmm3,xmm1 ++ movhlps xmm1,xmm1 ; = {g2, f2, g2, f2} ++ shufps xmm3,xmm2,0x14 ; = {f1, g1, g0, f0} ++ mulps xmm1,xmm6 ; *= {+c1, +s1, +s1, +c1} ++ shufps xmm2,xmm2,0xBB ; = {f3, g3, f3, g3} ++ mulps xmm2,xmm7 ; *= {-s1, -c1, +c1, +s1} ++ addps xmm1,xmm2 ; = {d, b, c, a} ++ movaps xmm2,xmm3 ++ addps xmm3,xmm1 ; = {fi[k1], gi[k1], gi[0], fi[0]} ++ subps xmm2,xmm1 ; = {fi[k3], gi[k3], gi[k1*2], fi[k1*2]} ++ movhlps xmm0,xmm3 ++ movss [esi],xmm3 ++ shufps xmm3,xmm3,0x55 ++ movss [edi+eax],xmm0 ++ shufps xmm0,xmm0,0x55 ++ movss [edi],xmm3 ++ movss [esi+eax],xmm0 ++ movhlps xmm0,xmm2 ++ movss [esi+eax*2],xmm2 ++ shufps xmm2,xmm2,0x55 ++ movss [edi+edx],xmm0 ++ shufps xmm0,xmm0,0x55 ++ movss [edi+eax*2],xmm2 ++ lea edi,[edi + eax*4] ; gi += (k1 * 4); ++ movss [esi+edx],xmm0 ++ lea esi,[esi + eax*4] ; fi += (k1 * 4); ++ cmp esi,ebp ++ jl near .lp21 ; while (fi<fn); ++ ; unroll����do loop��43+4̿�� ++ ++ ; ������ǤϤʤ�for�롼�פ�unrolling���� ++ ; kx= 2, 8, 32, 128 ++ ; k4= 16, 64, 256, 1024 ++ ; 0, 6/2,30/2,126/2 ++ ; at here ++ ; xmm6 = {--, --, s1, c1} ++ ; c3 = c1; s3 = s1; ++ xor ebx,ebx ++ mov bl,4*4 ; = i = 4 ++ cmp ebx,eax ; i < k1 ++ jnl near .F22 ++ ++ shufps xmm6,xmm6,0x14 ; = {c1, s1, s1, c1} ++ jmp short .F220 ++ ++ align 16 ++ ; for (i=4;i<k1;i+=4){ // for (i=2;i<k1/2;i+=2){ ++ .lp22: ++ shufps xmm6,xmm6,0x69 ; xmm6 = {c3, s3, s3, c3} ++ .F220: ++ ; at here, xmm6 is {c3, s3, s3, c3} ++ ; c1 = c3*t_c - s3*t_s; ++ ; s1 = c3*t_s + s3*t_c; ++ movss xmm0,[costab_fft + ecx*4] ++ movss xmm1,[sintab_fft + ecx*4] ++ shufps xmm0,xmm1,0x00 ; = {t_s, t_s, t_c, t_c} ++ mulps xmm6,xmm0 ++ movhlps xmm4,xmm6 ++ xorps xmm4,[Q_MPMP] ++ subps xmm6,xmm4 ; = {--, --, s1, c1} ++ ++ ; c3 = c1*t_c - s1*t_s; ++ ; s3 = s1*t_c + c1*t_s; ++ shufps xmm6,xmm6,0x14 ; = {c1, s1, s1, c1} ++ mulps xmm0,xmm6 ++ movhlps xmm3,xmm0 ++ xorps xmm3,[Q_MPMP] ++ subps xmm0,xmm3 ; = {--, --, s3, c3} ++ ++ unpcklps xmm6,xmm0 ; xmm6 = {s3, s1, c3, c1} ++ shufps xmm6,xmm6,0xB4 ; xmm6 = {s1, s3, c3, c1} ++ ++ ; c2 = c1*c1 - s1*s1; ++ ; c4 = c3*c3 - s3*s3; ++ ; s4 = s3*c3 + s3*c3; ++ ; s2 = s1*c1 + s1*c1; ++ movaps xmm7,xmm6 ++ movaps xmm4,xmm6 ++ shufps xmm7,xmm7,0x14 ++ shufps xmm4,xmm4,0xEB ++ xorps xmm4,[Q_MMPP] ; = {-c3,-c1, s3, s1} ++ mulps xmm7,xmm6 ++ mulps xmm4,xmm6 ++ shufps xmm4,xmm4,0x1B ++ addps xmm7,xmm4 ; xmm7 = {s2, s4, c4, c2} ++ ++ ; fi = fz +i; ++ ; gi = fz +k1-i; ++ mov edi,[esp+_P+4] ; = fz ++ mov esi,ebx ++ shr esi,1 ++ sub edi,esi ; edi = fz - i/2 ++ lea esi,[edi + ebx] ; esi = fi = fz +i/2 ++ add edi,eax ; edi = gi = fz +k1-i/2 ++ sub edi,4 ++ ; do{ ++ .lp220: ++ ; unroll���do loop��51+4̿�� ++ ; a = c2*fi[k1 ] + s2*gi[k1 ]; ++ ; e = c4*fi[k1+1] + s4*gi[k1-1]; ++ ; f = s4*fi[k1+1] - c4*gi[k1-1]; ++ ; b = s2*fi[k1 ] - c2*gi[k1 ]; ++ ; c = c2*fi[k3 ] + s2*gi[k3 ]; ++ ; g = c4*fi[k3+1] + s4*gi[k3-1]; ++ ; h = s4*fi[k3+1] - c4*gi[k3-1]; ++ ; d = s2*fi[k3 ] - c2*gi[k3 ]; ++ ++ movaps xmm4,xmm7 ; xmm7 = {s2, s4, c4, c2} ++ shufps xmm4,xmm4,0x1B ++ xorps xmm4,[Q_MMPP] ++ movlps xmm0,[esi+eax] ++ movlps xmm1,[edi+eax] ++ movlps xmm2,[esi+edx] ++ movlps xmm3,[edi+edx] ++ shufps xmm0,xmm0,0x14 ++ shufps xmm1,xmm1,0x41 ++ shufps xmm2,xmm2,0x14 ++ shufps xmm3,xmm3,0x41 ++ mulps xmm0,xmm7 ++ mulps xmm1,xmm4 ++ mulps xmm2,xmm7 ++ mulps xmm3,xmm4 ++ addps xmm0,xmm1 ; xmm0 = {b, f, e, a} ++ addps xmm2,xmm3 ; xmm2 = {d, h, g, c} ++ ;17 ++ ++ ; f0 = fi[0 ] + a; ++ ; f4 = fi[0 +1] + e; ++ ; g4 = gi[0 -1] + f; ++ ; g0 = gi[0 ] + b; ++ ; f1 = fi[0 ] - a; ++ ; f5 = fi[0 +1] - e; ++ ; g5 = gi[0 -1] - f; ++ ; g1 = gi[0 ] - b; ++ ; f2 = fi[k2 ] + c; ++ ; f6 = fi[k2+1] + g; ++ ; g6 = gi[k2-1] + h; ++ ; g2 = gi[k2 ] + d; ++ ; f3 = fi[k2 ] - c; ++ ; f7 = fi[k2+1] - g; ++ ; g7 = gi[k2-1] - h; ++ ; g3 = gi[k2 ] - d; ++ movlps xmm4,[esi ] ++ movhps xmm4,[edi ] ++ movaps xmm1,xmm4 ++ subps xmm1,xmm0 ; xmm1 = {g1, g5, f5, f1} ++ movlps xmm5,[esi+eax*2] ++ movhps xmm5,[edi+eax*2] ++ movaps xmm3,xmm5 ++ subps xmm3,xmm2 ; xmm3 = {g3, g7, f7, f3} ++ addps xmm0,xmm4 ; xmm0 = {g0, g4, f4, f0} ++ addps xmm2,xmm5 ; xmm2 = {g2, g6, f6, f2} ++ ;10 ++ ++ ; a = c1*f2 + s1*g3; ��*�� + ��*�� ++ ; e = c3*f6 + s3*g7; ++ ; g = s3*g6 + c3*f7; ++ ; c = s1*g2 + c1*f3; ++ ; d = c1*g2 - s1*f3; ��*�� - ��*�� ++ ; h = c3*g6 - s3*f7; ++ ; f = s3*f6 - c3*g7; ++ ; b = s1*f2 - c1*g3; ++ ++ movaps xmm5,xmm6 ; xmm6 = {s1, s3, c3, c1} ++ shufps xmm5,xmm5,0x1B ; = {c1, c3, s3, s1} ++ movaps xmm4,xmm2 ++ mulps xmm4,xmm6 ++ shufps xmm2,xmm2,0x1B ; xmm2 = {f2, f6, g6, g2} ++ mulps xmm2,xmm6 ++ mulps xmm5,xmm3 ++ mulps xmm3,xmm6 ++ shufps xmm3,xmm3,0x1B ++ addps xmm4,xmm3 ; = {c, g, e, a} ++ subps xmm2,xmm5 ; = {b, f, h, d} ++ ;10 ++ ++ ; fi[0 ] = f0 + a; ++ ; fi[0 +1] = f4 + e; ++ ; gi[0 -1] = g4 + g; ++ ; gi[0 ] = g0 + c; ++ ; fi[k2 ] = f0 - a; ++ ; fi[k2+1] = f4 - e; ++ ; gi[k2-1] = g4 - g; ++ ; gi[k2 ] = g0 - c; ++ ; fi[k1 ] = f1 + d; ++ ; fi[k1+1] = f5 + h; ++ ; gi[k1-1] = g5 + f; ++ ; gi[k1 ] = g1 + b; ++ ; fi[k3 ] = f1 - d; ++ ; fi[k3+1] = f5 - h; ++ ; gi[k3-1] = g5 - f; ++ ; gi[k3 ] = g1 - b; ++ movaps xmm3,xmm0 ++ subps xmm0,xmm4 ++ movlps [esi+eax*2],xmm0 ++ movhps [edi+eax*2],xmm0 ++ addps xmm4,xmm3 ++ movlps [esi ],xmm4 ++ movhps [edi ],xmm4 ++ ++ movaps xmm5,xmm1 ++ subps xmm1,xmm2 ++ movlps [esi+edx ],xmm1 ++ movhps [edi+edx ],xmm1 ++ addps xmm2,xmm5 ++ movlps [esi+eax ],xmm2 ++ movhps [edi+eax ],xmm2 ++ ; 14 ++ ; gi += k4; ++ ; fi += k4; ++ lea edi,[edi + eax*4] ; gi += (k1 * 4); ++ lea esi,[esi + eax*4] ; fi += (k1 * 4); ++ cmp esi,ebp ++ jl near .lp220 ; while (fi<fn); ++ ; } while (fi<fn); ++ ++ add ebx,4*4 ; i+= 4 ++ cmp ebx,eax ; i < k1 ++ jl near .lp22 ++ ; } ++ .F22: ++ ++ cmp eax,[esp+_P+8] ; while ((k1 * 4)<n); ++ jl near .lp2 ++ ++ pop ebp ++ pop edi ++ pop esi ++ pop ebx ++ ret ++ ++ ;------------------------------------------------------------------------ ++ ; 99/11/12 Initial version for SSE by K. SAKAI, 4300clk@P3 ++ ; This routine is very slow when wsamp_r_int is not aligned to 16byte boundary. ++ ; ++ ;void fft_side_SSE( float in[2][1024], int s, float *ret) ++ ; energy = (in[0][512] - in[1][512])^2; ++ ; energy = (in[0][1024-s] - in[1][1024-s])^2; ++ ; for (i=s,j=1024-s;i<512;i++,j--){ ++ ; a = in[0][i] - in[1][i]; ++ ; energy += a*a; ++ ; b = in[0][j-1] - in[1][j-1]; ++ ; energy += b*b; ++ ; } ++ ; *ret = energy * 0.25; ++ ++ align 16 ++ fft_side_SSE: ++ mov ecx,[esp+8] ; = i = s ++ mov edx,1024 ++ sub edx,ecx ; = j ++ mov eax,[esp+4] ; = in ++ movss xmm7,[eax+1024*0*4+512*4] ++ movss xmm1,[eax+1024*1*4+512*4] ++ subss xmm7,xmm1 ++ mulss xmm7,xmm7 ++ movss xmm2,[eax+1024*0*4+edx*4] ++ movss xmm3,[eax+1024*1*4+edx*4] ++ subss xmm2,xmm3 ++ mulss xmm2,xmm2 ++ addss xmm7,xmm2 ++ ++ test cl,1 ++ jz .even ++ ++ .odd: dec edx ++ movss xmm0,[eax+1024*0*4+ecx*4] ++ movss xmm1,[eax+1024*1*4+ecx*4] ++ inc ecx ++ movss xmm2,[eax+1024*0*4+edx*4] ++ movss xmm3,[eax+1024*1*4+edx*4] ++ cmp ecx,edx ++ subss xmm0,xmm1 ++ subss xmm2,xmm3 ++ mulss xmm0,xmm0 ++ mulss xmm2,xmm2 ++ addss xmm7,xmm0 ++ addss xmm7,xmm2 ++ je near .exit1 ++ ++ .even: test cl,2 ++ jz .f0 ++ sub edx,2 ++ movlps xmm0,[eax+1024*0*4+ecx*4] ++ movlps xmm1,[eax+1024*1*4+ecx*4] ++ add ecx,2 ++ movhps xmm0,[eax+1024*0*4+edx*4] ++ movhps xmm1,[eax+1024*1*4+edx*4] ++ cmp ecx,edx ++ subps xmm0,xmm1 ++ mulps xmm0,xmm0 ++ addps xmm7,xmm0 ++ je .exit4 ++ jmp short .f0 ++ ++ align 16 ++ .f0: ++ .lp0: ++ sub edx,4 ++ movaps xmm0,[eax+1024*0*4+ecx*4] ++ movaps xmm1,[eax+1024*1*4+ecx*4] ++ add ecx,4 ++ subps xmm0,xmm1 ++ mulps xmm0,xmm0 ++ addps xmm7,xmm0 ++ movaps xmm2,[eax+1024*0*4+edx*4] ++ movaps xmm3,[eax+1024*1*4+edx*4] ++ cmp ecx,edx ++ subps xmm2,xmm3 ++ mulps xmm2,xmm2 ++ addps xmm7,xmm2 ++ jne .lp0 ++ ++ .exit4: movhlps xmm6,xmm7 ++ addps xmm7,xmm6 ++ movaps xmm6,xmm7 ++ shufps xmm6,xmm6,01010101B ++ addss xmm7,xmm6 ++ ++ .exit1: mulss xmm7,[S_025] ++ mov eax,[esp+12] ++ movss [eax],xmm7 ++ ret ++ ++ ++ end +diff -r -c -N encoder/i386/ffttbl.nas lame3.70/i386/ffttbl.nas +*** encoder/i386/ffttbl.nas Wed Dec 31 17:00:00 1969 +--- lame3.70/i386/ffttbl.nas Sun Jan 30 15:53:49 2000 +*************** +*** 0 **** +--- 1,78 ---- ++ ++ ; for new GOGO-no-coda (1999/09) ++ ; Copyright (C) 1999 shigeo ++ ; special thanks to URURI, Keiichi SAKAI ++ ++ ++ ; fft sin,cos,gray�ơ��֥� ++ ; 99/08/21 ++ ; 99/09/01(�����)���ӥå�ȿž�ơ��֥��� ++ ; 99/09/14 ���쥤�����ɥ���ǥå���ɽ�������Х벽 ++ ++ %include "nasm.h" ++ ++ globaldef costab_fft ++ globaldef sintab_fft ++ globaldef gray_index ++ ++ segment_data ++ align 16 ++ ++ costab_fft: ++ dd 0.000000000000 ++ dd 0.707106781187 ++ dd 0.923879532511 ++ dd 0.980785280403 ++ dd 0.995184726672 ++ dd 0.998795456205 ++ dd 0.999698818696 ++ dd 0.999924701839 ++ dd 0.999981175283 ++ dd 0.999995293810 ++ dd 0.999998823452 ++ dd 0.999999705863 ++ dd 0.999999926466 ++ dd 0.999999981616 ++ dd 0.999999995404 ++ dd 0.999999998851 ++ ++ sintab_fft: ++ dd 1.000000000000 ++ dd 0.707106781187 ++ dd 0.382683432365 ++ dd 0.195090322016 ++ dd 0.098017140330 ++ dd 0.049067674327 ++ dd 0.024541228523 ++ dd 0.012271538286 ++ dd 0.006135884649 ++ dd 0.003067956763 ++ dd 0.001533980186 ++ dd 0.000766990319 ++ dd 0.000383495188 ++ dd 0.000191747597 ++ dd 0.000095873799 ++ dd 0.000047936900 ++ ++ align 16 ++ gray_index: ++ db 2, 3, 2, 4, 2, 3, 2, 5, 2, 3, 2, 4, 2, 3, 2, 6 ++ db 2, 3, 2, 4, 2, 3, 2, 5, 2, 3, 2, 4, 2, 3, 2, 7 ++ db 2, 3, 2, 4, 2, 3, 2, 5, 2, 3, 2, 4, 2, 3, 2, 6 ++ db 2, 3, 2, 4, 2, 3, 2, 5, 2, 3, 2, 4, 2, 3, 2, 8 ++ db 2, 3, 2, 4, 2, 3, 2, 5, 2, 3, 2, 4, 2, 3, 2, 6 ++ db 2, 3, 2, 4, 2, 3, 2, 5, 2, 3, 2, 4, 2, 3, 2, 7 ++ db 2, 3, 2, 4, 2, 3, 2, 5, 2, 3, 2, 4, 2, 3, 2, 6 ++ db 2, 3, 2, 4, 2, 3, 2, 5, 2, 3, 2, 4, 2, 3, 2, 9 ++ db 2, 3, 2, 4, 2, 3, 2, 5, 2, 3, 2, 4, 2, 3, 2, 6 ++ db 2, 3, 2, 4, 2, 3, 2, 5, 2, 3, 2, 4, 2, 3, 2, 7 ++ db 2, 3, 2, 4, 2, 3, 2, 5, 2, 3, 2, 4, 2, 3, 2, 6 ++ db 2, 3, 2, 4, 2, 3, 2, 5, 2, 3, 2, 4, 2, 3, 2, 8 ++ db 2, 3, 2, 4, 2, 3, 2, 5, 2, 3, 2, 4, 2, 3, 2, 6 ++ db 2, 3, 2, 4, 2, 3, 2, 5, 2, 3, 2, 4, 2, 3, 2, 7 ++ db 2, 3, 2, 4, 2, 3, 2, 5, 2, 3, 2, 4, 2, 3, 2, 6 ++ db 2, 3, 2, 4, 2, 3, 2, 5, 2, 3, 2, 4, 2, 3, 2, 10 ++ ++ segment_code ++ ++ end +diff -r -c -N encoder/id3tag.c lame3.70/id3tag.c +*** encoder/id3tag.c Wed Dec 31 17:00:00 1969 +--- lame3.70/id3tag.c Sun Mar 5 11:37:28 2000 +*************** +*** 0 **** +--- 1,112 ---- ++ /* ++ * functions for writing ID3 tags in LAME ++ * ++ * text functions stolen from mp3info by Ricardo Cerqueira <rmc@rccn.net> ++ * adapted for LAME by Conrad Sanderson <c.sanderson@me.gu.edu.au> ++ * ++ * ++ */ ++ ++ #include <stdio.h> ++ #include <string.h> ++ #include "id3tag.h" ++ ID3TAGDATA id3tag; ++ ++ /* ++ * If "string" is shorter than "length", pad it with ' ' (spaces) ++ */ ++ ++ static void id3_pad(char *string, int length) { ++ int l; l=strlen(string); ++ ++ while(l<length) { string[l] = ' '; l++; } ++ string[l]='\0'; ++ } ++ ++ ++ /* ++ * initialize temporary fields ++ */ ++ ++ void id3_inittag(ID3TAGDATA *tag) { ++ strcpy( tag->title, ""); ++ strcpy( tag->artist, ""); ++ strcpy( tag->album, ""); ++ strcpy( tag->year, ""); ++ strcpy( tag->comment, ""); ++ strcpy( tag->genre, "�"); /* unset genre */ ++ tag->track = 0; ++ ++ tag->valid = 0; /* not ready for writing*/ ++ } ++ ++ /* ++ * build an ID3 tag from temporary fields ++ */ ++ ++ void id3_buildtag(ID3TAGDATA *tag) { ++ strcpy(tag->tagtext,"TAG"); ++ ++ id3_pad( tag->title, 30); strncat( tag->tagtext, tag->title,30); ++ id3_pad( tag->artist, 30); strncat( tag->tagtext, tag->artist,30); ++ id3_pad( tag->album, 30); strncat( tag->tagtext, tag->album,30); ++ id3_pad( tag->year, 4); strncat( tag->tagtext, tag->year,4); ++ id3_pad( tag->comment, 30); strncat( tag->tagtext, tag->comment,30); ++ id3_pad( tag->genre, 1); strncat( tag->tagtext, tag->genre,1); ++ ++ if( tag->track != 0 ) { ++ tag->tagtext[125] = '\0'; ++ tag->tagtext[126] = tag->track; ++ } ++ tag->valid = 1; /* ready for writing*/ ++ } ++ ++ /* ++ * write ID3 tag ++ */ ++ ++ int id3_writetag(char* filename, ID3TAGDATA *tag) { ++ FILE* f; ++ if( ! tag->valid ) return -1; ++ ++ f=fopen(filename,"rb+"); if(!f) return -1; ++ ++ fseek(f,0,SEEK_END); fwrite(tag->tagtext,1,128,f); ++ fclose(f); return 0; ++ } ++ ++ ++ ++ ++ ++ int genre_last=147; ++ char *genre_list[]={ ++ "Blues", "Classic Rock", "Country", "Dance", "Disco", "Funk", ++ "Grunge", "Hip-Hop", "Jazz", "Metal", "New Age", "Oldies", ++ "Other", "Pop", "R&B", "Rap", "Reggae", "Rock", ++ "Techno", "Industrial", "Alternative", "Ska", "Death Metal", "Pranks", ++ "Soundtrack", "Euro-Techno", "Ambient", "Trip-Hop", "Vocal", "Jazz+Funk", ++ "Fusion", "Trance", "Classical", "Instrumental", "Acid", "House", ++ "Game", "Sound Clip", "Gospel", "Noise", "AlternRock", "Bass", ++ "Soul", "Punk", "Space", "Meditative", "Instrumental Pop", "Instrumental Rock", ++ "Ethnic", "Gothic", "Darkwave", "Techno-Industrial", "Electronic", "Pop-Folk", ++ "Eurodance", "Dream", "Southern Rock", "Comedy", "Cult", "Gangsta", ++ "Top 40", "Christian Rap", "Pop/Funk", "Jungle", "Native American", "Cabaret", ++ "New Wave", "Psychadelic", "Rave", "Showtunes", "Trailer", "Lo-Fi", ++ "Tribal", "Acid Punk", "Acid Jazz", "Polka", "Retro", "Musical", ++ "Rock & Roll", "Hard Rock", "Folk", "Folk/Rock", "National Folk", "Swing", ++ "Fast-Fusion", "Bebob", "Latin", "Revival", "Celtic", "Bluegrass", "Avantgarde", ++ "Gothic Rock", "Progressive Rock", "Psychedelic Rock", "Symphonic Rock", "Slow Rock", "Big Band", ++ "Chorus", "Easy Listening", "Acoustic", "Humour", "Speech", "Chanson", ++ "Opera", "Chamber Music", "Sonata", "Symphony", "Booty Bass", "Primus", ++ "Porn Groove", "Satire", "Slow Jam", "Club", "Tango", "Samba", ++ "Folklore", "Ballad", "Power Ballad", "Rhythmic Soul", "Freestyle", "Duet", ++ "Punk Rock", "Drum Solo", "A capella", "Euro-House", "Dance Hall", ++ "Goa", "Drum & Bass", "Club House", "Hardcore", "Terror", ++ "Indie", "BritPop", "NegerPunk", "Polsk Punk", "Beat", ++ "Christian Gangsta", "Heavy Metal", "Black Metal", "Crossover", "Contemporary C", ++ "Christian Rock", "Merengue", "Salsa", "Thrash Metal", "Anime", "JPop", ++ "SynthPop", ++ }; ++ ++ +diff -r -c -N encoder/id3tag.h lame3.70/id3tag.h +*** encoder/id3tag.h Wed Dec 31 17:00:00 1969 +--- lame3.70/id3tag.h Mon Mar 6 12:53:03 2000 +*************** +*** 0 **** +--- 1,29 ---- ++ #ifndef ID3TAG_H_INCLUDED ++ #define ID3TAG_H_INCLUDED ++ typedef struct ++ { ++ int used; ++ int valid; ++ char title[31]; ++ char artist[31]; ++ char album[31]; ++ char year[5]; ++ char comment[31]; ++ char tagtext[128]; ++ char genre[1]; ++ unsigned char track; ++ ++ } ID3TAGDATA; ++ ++ void id3_inittag(ID3TAGDATA *tag); ++ void id3_buildtag(ID3TAGDATA *tag); ++ int id3_writetag(char* filename, ID3TAGDATA *tag); ++ ++ ++ /* ++ * Array of all possible music genre. Grabbed from id3ed ++ */ ++ extern ID3TAGDATA id3tag; /* id3tag info */ ++ extern int genre_last; ++ extern char *genre_list[]; ++ #endif +diff -r -c -N encoder/ieeefloat.c lame3.70/ieeefloat.c +*** encoder/ieeefloat.c Wed Jan 22 02:43:17 1997 +--- lame3.70/ieeefloat.c Mon Feb 21 16:05:05 2000 +*************** +*** 41,54 **** + * conversions, and accommodated conversions involving +/- infinity, + * NaN's, and denormalized numbers. + * +! * $Id: ieeefloat.c,v 1.1 1993/06/11 17:45:46 malcolm Exp $ + * + * $Log: ieeefloat.c,v $ + * Revision 1.1 1993/06/11 17:45:46 malcolm + * Initial revision + * + */ + + #include <stdio.h> + #include <math.h> + #include "ieeefloat.h" +--- 41,65 ---- + * conversions, and accommodated conversions involving +/- infinity, + * NaN's, and denormalized numbers. + * +! * $Id: ieeefloat.c,v 1.3 2000/02/21 23:05:05 markt Exp $ + * + * $Log: ieeefloat.c,v $ ++ * Revision 1.3 2000/02/21 23:05:05 markt ++ * some 64bit DEC Alpha patches ++ * ++ * Revision 1.2 2000/02/19 13:32:30 afaber ++ * Fixed many warning messages when compiling with MSVC ++ * ++ * Revision 1.1.1.1 1999/11/24 08:42:58 markt ++ * initial checkin of LAME ++ * Starting with LAME 3.57beta with some modifications ++ * + * Revision 1.1 1993/06/11 17:45:46 malcolm + * Initial revision + * + */ + ++ #include <limits.h> + #include <stdio.h> + #include <math.h> + #include "ieeefloat.h" +*************** +*** 82,89 **** + + + defdouble +! ConvertFromIeeeSingle(bytes) +! char* bytes; + { + defdouble f; + long mantissa, expon; +--- 93,99 ---- + + + defdouble +! ConvertFromIeeeSingle(char* bytes) + { + defdouble f; + long mantissa, expon; +*************** +*** 106,121 **** + else { + if (expon == 0) { /* Denormalized number */ + mantissa = (bits & 0x7fffff); +! f = ldexp((defdouble)mantissa, expon - SEXP_OFFSET - SEXP_POSITION + 1); + } + else { /* Normalized number */ + mantissa = (bits & 0x7fffff) + 0x800000; /* Insert hidden bit */ +! f = ldexp((defdouble)mantissa, expon - SEXP_OFFSET - SEXP_POSITION); + } + } + } + +! if (bits & 0x80000000) + return -f; + else + return f; +--- 116,131 ---- + else { + if (expon == 0) { /* Denormalized number */ + mantissa = (bits & 0x7fffff); +! f = ldexp((defdouble) mantissa, (int) (expon - SEXP_OFFSET - SEXP_POSITION + 1)); + } + else { /* Normalized number */ + mantissa = (bits & 0x7fffff) + 0x800000; /* Insert hidden bit */ +! f = ldexp((defdouble) mantissa, (int) (expon - SEXP_OFFSET - SEXP_POSITION)); + } + } + } + +! if (bits & LONG_MIN) + return -f; + else + return f; +*************** +*** 126,140 **** + + + void +! ConvertToIeeeSingle(num, bytes) +! defdouble num; +! char* bytes; + { + long sign; + register long bits; + + if (num < 0) { /* Can't distinguish a negative zero */ +! sign = 0x80000000; + num *= -1; + } else { + sign = 0; +--- 136,148 ---- + + + void +! ConvertToIeeeSingle(defdouble num, char* bytes) + { + long sign; + register long bits; + + if (num < 0) { /* Can't distinguish a negative zero */ +! sign = LONG_MIN; + num *= -1; + } else { + sign = 0; +*************** +*** 177,186 **** + } + } + +! bytes[0] = bits >> 24; /* Copy to byte string */ +! bytes[1] = bits >> 16; +! bytes[2] = bits >> 8; +! bytes[3] = bits; + } + + +--- 185,194 ---- + } + } + +! bytes[0] = (char)(bits >> 24); /* Copy to byte string */ +! bytes[1] = (char)(bits >> 16); +! bytes[2] = (char)(bits >> 8); +! bytes[3] = (char)(bits); + } + + +*************** +*** 195,202 **** + + + defdouble +! ConvertFromIeeeDouble(bytes) +! char* bytes; + { + defdouble f; + long mantissa, expon; +--- 203,209 ---- + + + defdouble +! ConvertFromIeeeDouble(char* bytes) + { + defdouble f; + long mantissa, expon; +*************** +*** 223,235 **** + else { + if (expon == 0) { /* Denormalized number */ + mantissa = (first & 0x000FFFFF); +! f = ldexp((defdouble)mantissa, expon - DEXP_OFFSET - DEXP_POSITION + 1); +! f += ldexp(UnsignedToFloat(second), expon - DEXP_OFFSET - DEXP_POSITION + 1 - 32); + } + else { /* Normalized number */ + mantissa = (first & 0x000FFFFF) + 0x00100000; /* Insert hidden bit */ +! f = ldexp((defdouble)mantissa, expon - DEXP_OFFSET - DEXP_POSITION); +! f += ldexp(UnsignedToFloat(second), expon - DEXP_OFFSET - DEXP_POSITION - 32); + } + } + } +--- 230,242 ---- + else { + if (expon == 0) { /* Denormalized number */ + mantissa = (first & 0x000FFFFF); +! f = ldexp((defdouble) mantissa, (int) (expon - DEXP_OFFSET - DEXP_POSITION + 1)); +! f += ldexp(UnsignedToFloat(second), (int) (expon - DEXP_OFFSET - DEXP_POSITION + 1 - 32)); + } + else { /* Normalized number */ + mantissa = (first & 0x000FFFFF) + 0x00100000; /* Insert hidden bit */ +! f = ldexp((defdouble) mantissa, (int) (expon - DEXP_OFFSET - DEXP_POSITION)); +! f += ldexp(UnsignedToFloat(second), (int) (expon - DEXP_OFFSET - DEXP_POSITION - 32)); + } + } + } +*************** +*** 245,259 **** + + + void +! ConvertToIeeeDouble(num, bytes) +! defdouble num; +! char *bytes; + { + long sign; + long first, second; + + if (num < 0) { /* Can't distinguish a negative zero */ +! sign = 0x80000000; + num *= -1; + } else { + sign = 0; +--- 252,264 ---- + + + void +! ConvertToIeeeDouble(defdouble num, char *bytes) + { + long sign; + long first, second; + + if (num < 0) { /* Can't distinguish a negative zero */ +! sign = LONG_MIN; + num *= -1; + } else { + sign = 0; +*************** +*** 310,323 **** + } + } + +! bytes[0] = first >> 24; +! bytes[1] = first >> 16; +! bytes[2] = first >> 8; +! bytes[3] = first; +! bytes[4] = second >> 24; +! bytes[5] = second >> 16; +! bytes[6] = second >> 8; +! bytes[7] = second; + } + + +--- 315,328 ---- + } + } + +! bytes[0] = (char)(first >> 24); +! bytes[1] = (char)(first >> 16); +! bytes[2] = (char)(first >> 8); +! bytes[3] = (char)(first); +! bytes[4] = (char)(second >> 24); +! bytes[5] = (char)(second >> 16); +! bytes[6] = (char)(second >> 8); +! bytes[7] = (char)(second); + } + + +*************** +*** 326,333 **** + ****************************************************************/ + + defdouble +! ConvertFromIeeeExtended(bytes) +! char* bytes; + { + defdouble f; + long expon; +--- 331,337 ---- + ****************************************************************/ + + defdouble +! ConvertFromIeeeExtended(char* bytes) + { + defdouble f; + long expon; +*************** +*** 359,366 **** + } + else { + expon -= 16383; +! f = ldexp(UnsignedToFloat(hiMant), expon-=31); +! f += ldexp(UnsignedToFloat(loMant), expon-=32); + } + } + +--- 363,370 ---- + } + else { + expon -= 16383; +! f = ldexp(UnsignedToFloat(hiMant), (int) (expon -= 31)); +! f += ldexp(UnsignedToFloat(loMant), (int) (expon -= 32)); + } + } + +*************** +*** 375,383 **** + + + void +! ConvertToIeeeExtended(num, bytes) +! defdouble num; +! char *bytes; + { + int sign; + int expon; +--- 379,385 ---- + + + void +! ConvertToIeeeExtended(defdouble num, char *bytes) + { + int sign; + int expon; +*************** +*** 413,426 **** + + bytes[0] = expon >> 8; + bytes[1] = expon; +! bytes[2] = hiMant >> 24; +! bytes[3] = hiMant >> 16; +! bytes[4] = hiMant >> 8; +! bytes[5] = hiMant; +! bytes[6] = loMant >> 24; +! bytes[7] = loMant >> 16; +! bytes[8] = loMant >> 8; +! bytes[9] = loMant; + } + + /**************************************************************** +--- 415,428 ---- + + bytes[0] = expon >> 8; + bytes[1] = expon; +! bytes[2] = (char)(hiMant >> 24); +! bytes[3] = (char)(hiMant >> 16); +! bytes[4] = (char)(hiMant >> 8); +! bytes[5] = (char)(hiMant); +! bytes[6] = (char)(loMant >> 24); +! bytes[7] = (char)(loMant >> 16); +! bytes[8] = (char)(loMant >> 8); +! bytes[9] = (char)(loMant); + } + + /**************************************************************** +*************** +*** 467,474 **** + + + int +! GetHexValue(x) +! register int x; + { + x &= 0x7F; + +--- 469,475 ---- + + + int +! GetHexValue(register int x) + { + x &= 0x7F; + +*************** +*** 486,493 **** + + + void +! Hex2Bytes(hex, bytes) +! register char *hex, *bytes; + { + for ( ; *hex; hex += 2) { + *bytes++ = (GetHexValue(hex[0]) << 4) | GetHexValue(hex[1]); +--- 487,493 ---- + + + void +! Hex2Bytes(register char *hex, register char *bytes) + { + for ( ; *hex; hex += 2) { + *bytes++ = (GetHexValue(hex[0]) << 4) | GetHexValue(hex[1]); +*************** +*** 498,505 **** + + + int +! GetHexSymbol(x) +! register int x; + { + x &= 0xF; + if (x <= 9) +--- 498,504 ---- + + + int +! GetHexSymbol(register int x) + { + x &= 0xF; + if (x <= 9) +*************** +*** 511,519 **** + + + void +! Bytes2Hex(bytes, hex, nBytes) +! register char *bytes, *hex; +! register int nBytes; + { + for ( ; nBytes--; bytes++) { + *hex++ = GetHexSymbol(*bytes >> 4); +--- 510,516 ---- + + + void +! Bytes2Hex(register char *bytes, register char *hex, register int nBytes) + { + for ( ; nBytes--; bytes++) { + *hex++ = GetHexSymbol(*bytes >> 4); +*************** +*** 524,532 **** + + + void +! MaybeSwapBytes(bytes, nBytes) +! char* bytes; +! int nBytes; + { + #ifdef LITTLE_ENDIAN + register char *p, *q, t; +--- 521,527 ---- + + + void +! MaybeSwapBytes(char* bytes, int nBytes) + { + #ifdef LITTLE_ENDIAN + register char *p, *q, t; +*************** +*** 543,550 **** + + + float +! MachineIEEESingle(bytes) +! char* bytes; + { + float t; + MaybeSwapBytes(bytes, 4); +--- 538,544 ---- + + + float +! MachineIEEESingle(char* bytes) + { + float t; + MaybeSwapBytes(bytes, 4); +*************** +*** 555,562 **** + + + Double +! MachineIEEEDouble(bytes) +! char* bytes; + { + Double t; + MaybeSwapBytes(bytes, 8); +--- 549,555 ---- + + + Double +! MachineIEEEDouble(char* bytes) + { + Double t; + MaybeSwapBytes(bytes, 8); +*************** +*** 567,574 **** + + + void +! TestFromIeeeSingle(hex) +! char *hex; + { + defdouble f; + union SParts p; +--- 560,566 ---- + + + void +! TestFromIeeeSingle(char *hex) + { + defdouble f; + union SParts p; +*************** +*** 589,596 **** + + + void +! TestToIeeeSingle(f) +! defdouble f; + { + union SParts p; + char bytes[4]; +--- 581,587 ---- + + + void +! TestToIeeeSingle(defdouble f) + { + union SParts p; + char bytes[4]; +*************** +*** 613,620 **** + + + void +! TestFromIeeeDouble(hex) +! char *hex; + { + defdouble f; + union DParts p; +--- 604,610 ---- + + + void +! TestFromIeeeDouble(char *hex) + { + defdouble f; + union DParts p; +*************** +*** 636,643 **** + } + + void +! TestToIeeeDouble(f) +! defdouble f; + { + union DParts p; + char bytes[8]; +--- 626,632 ---- + } + + void +! TestToIeeeDouble(defdouble f) + { + union DParts p; + char bytes[8]; +*************** +*** 663,670 **** + + + void +! TestFromIeeeExtended(hex) +! char *hex; + { + defdouble f; + union EParts p; +--- 652,658 ---- + + + void +! TestFromIeeeExtended(char *hex) + { + defdouble f; + union EParts p; +*************** +*** 701,708 **** + + + void +! TestToIeeeExtended(f) +! defdouble f; + { + char bytes[12]; + char hex[24+1]; +--- 689,695 ---- + + + void +! TestToIeeeExtended(defdouble f) + { + char bytes[12]; + char hex[24+1]; +*************** +*** 736,751 **** + + #include <signal.h> + +! void SignalFPE(i, j) +! int i; +! void (*j)(); + { + printf("[Floating Point Interrupt Caught.]\n", i, j); + signal(SIGFPE, SignalFPE); + } + + void +! main() + { + long d[3]; + char bytes[12]; +--- 723,736 ---- + + #include <signal.h> + +! void SignalFPE(int i, void (*j)()) + { + printf("[Floating Point Interrupt Caught.]\n", i, j); + signal(SIGFPE, SignalFPE); + } + + void +! main(void) + { + long d[3]; + char bytes[12]; +diff -r -c -N encoder/ieeefloat.h lame3.70/ieeefloat.h +*** encoder/ieeefloat.h Wed Jan 22 02:43:17 1997 +--- lame3.70/ieeefloat.h Wed Nov 24 01:42:58 1999 +*************** +*** 43,51 **** + * conversions, and accommodated conversions involving +/- infinity, + * NaN's, and denormalized numbers. + * +! * $Id: ieeefloat.h,v 1.1 1993/06/11 17:45:46 malcolm Exp $ + * + * $Log: ieeefloat.h,v $ + * Revision 1.1 1993/06/11 17:45:46 malcolm + * Initial revision + * +--- 43,55 ---- + * conversions, and accommodated conversions involving +/- infinity, + * NaN's, and denormalized numbers. + * +! * $Id: ieeefloat.h,v 1.1.1.1 1999/11/24 08:42:58 markt Exp $ + * + * $Log: ieeefloat.h,v $ ++ * Revision 1.1.1.1 1999/11/24 08:42:58 markt ++ * initial checkin of LAME ++ * Starting with LAME 3.57beta with some modifications ++ * + * Revision 1.1 1993/06/11 17:45:46 malcolm + * Initial revision + * +diff -r -c -N encoder/l3bitstream-pvt.h lame3.70/l3bitstream-pvt.h +*** encoder/l3bitstream-pvt.h Wed Jan 22 02:43:15 1997 +--- lame3.70/l3bitstream-pvt.h Thu Apr 6 12:50:49 2000 +*************** +*** 2,10 **** + * ISO MPEG Audio Subgroup Software Simulation Group (1996) + * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension + * +! * $Id: l3bitstream-pvt.h,v 1.1 1996/02/14 04:04:23 rowlands Exp $ + * + * $Log: l3bitstream-pvt.h,v $ + * Revision 1.1 1996/02/14 04:04:23 rowlands + * Initial revision + * +--- 2,23 ---- + * ISO MPEG Audio Subgroup Software Simulation Group (1996) + * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension + * +! * $Id: l3bitstream-pvt.h,v 1.4 2000/03/21 23:02:17 markt Exp $ + * + * $Log: l3bitstream-pvt.h,v $ ++ * Revision 1.4 2000/03/21 23:02:17 markt ++ * replaced all "gf." by gfp-> ++ * ++ * Revision 1.3 2000/02/01 11:26:32 takehiro ++ * scalefactor's structure changed ++ * ++ * Revision 1.2 1999/12/09 00:44:34 cisc ++ * Removed write_ancillary_data() prototype. (No longer used) ++ * ++ * Revision 1.1.1.1 1999/11/24 08:42:59 markt ++ * initial checkin of LAME ++ * Starting with LAME 3.57beta with some modifications ++ * + * Revision 1.1 1996/02/14 04:04:23 rowlands + * Initial revision + * +*************** +*** 14,26 **** + #ifndef L3BITSTREAM_PVT_H + #define L3BITSTREAM_PVT_H + +! static int encodeSideInfo( III_side_info_t *si ); + +! static void encodeMainData( int l3_enc[2][2][576], + III_side_info_t *si, +! III_scalefac_t *scalefac ); +! +! static void write_ancillary_data( char *theData, int lengthInBits ); + + static void drain_into_ancillary_data( int lengthInBits ); + +--- 27,38 ---- + #ifndef L3BITSTREAM_PVT_H + #define L3BITSTREAM_PVT_H + +! static int encodeSideInfo( lame_global_flags *gfp,III_side_info_t *si ); + +! static void encodeMainData( lame_global_flags *gfp, +! int l3_enc[2][2][576], + III_side_info_t *si, +! III_scalefac_t scalefac[2][2] ); + + static void drain_into_ancillary_data( int lengthInBits ); + +diff -r -c -N encoder/l3bitstream.c lame3.70/l3bitstream.c +*** encoder/l3bitstream.c Wed Jan 22 02:43:15 1997 +--- lame3.70/l3bitstream.c Thu Apr 6 12:50:49 2000 +*************** +*** 2,14 **** + * ISO MPEG Audio Subgroup Software Simulation Group (1996) + * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension + * +- * $Id: l3bitstream.c,v 1.1 1996/02/14 04:04:23 rowlands Exp $ +- * +- * $Log: l3bitstream.c,v $ +- * Revision 1.1 1996/02/14 04:04:23 rowlands +- * Initial revision +- * +- * Received from Mike Coleman + **********************************************************************/ + /* + Revision History: +--- 2,7 ---- +*************** +*** 20,36 **** + */ + + #include <stdlib.h> + #include "l3bitstream.h" /* the public interface */ +! #include "l3psy.h" +! #include "mdct.h" +! #include "loop.h" + #include "formatBitstream.h" +! #include "huffman.h" + #include <assert.h> + #include "l3bitstream-pvt.h" + +- static int stereo = 1; +- static frame_params *fr_ps = NULL; + static Bit_stream_struc *bs = NULL; + + BF_FrameData *frameData = NULL; +--- 13,28 ---- + */ + + #include <stdlib.h> ++ #include "lame.h" + #include "l3bitstream.h" /* the public interface */ +! #include "encoder.h" +! #include "quantize.h" +! #include "quantize-pvt.h" + #include "formatBitstream.h" +! #include "tables.h" + #include <assert.h> + #include "l3bitstream-pvt.h" + + static Bit_stream_struc *bs = NULL; + + BF_FrameData *frameData = NULL; +*************** +*** 48,54 **** + BF_PartHolder *userFrameDataPH; + + +! void putMyBits( uint32 val, uint16 len ) + { + putbits( bs, val, len ); + } +--- 40,46 ---- + BF_PartHolder *userFrameDataPH; + + +! void putMyBits( u_int val, u_int len ) + { + putbits( bs, val, len ); + } +*************** +*** 65,99 **** + */ + + void +! III_format_bitstream( int bitsPerFrame, +! frame_params *in_fr_ps, + int l3_enc[2][2][576], + III_side_info_t *l3_side, +! III_scalefac_t *scalefac, +! Bit_stream_struc *in_bs, +! double (*xr)[2][576], +! char *ancillary, +! int ancillary_bits ) + { +! int gr, ch, i, mode_gr; +! fr_ps = in_fr_ps; + bs = in_bs; +! stereo = fr_ps->stereo; +! mode_gr = (fr_ps->header->version == 1) ? 2 : 1; +! + if ( frameData == NULL ) + { +! frameData = calloc( 1, sizeof(*frameData) ); + assert( frameData ); + } + if ( frameResults == NULL ) + { +! frameResults = calloc( 1, sizeof(*frameData) ); +! assert( frameData ); + } + if ( !PartHoldersInitialized ) + { +! headerPH = BF_newPartHolder( 12 ); + frameSIPH = BF_newPartHolder( 12 ); + + for ( ch = 0; ch < MAX_CHANNELS; ch++ ) +--- 57,86 ---- + */ + + void +! III_format_bitstream( lame_global_flags *gfp, +! int bitsPerFrame, + int l3_enc[2][2][576], + III_side_info_t *l3_side, +! III_scalefac_t scalefac[2][2], +! Bit_stream_struc *in_bs) + { +! int gr, ch; + bs = in_bs; +! + if ( frameData == NULL ) + { +! frameData = calloc( 1,sizeof *frameData); + assert( frameData ); + } + if ( frameResults == NULL ) + { +! frameResults = calloc( 1,sizeof *frameResults); +! assert( frameResults ); + } ++ + if ( !PartHoldersInitialized ) + { +! headerPH = BF_newPartHolder( 14 ); + frameSIPH = BF_newPartHolder( 12 ); + + for ( ch = 0; ch < MAX_CHANNELS; ch++ ) +*************** +*** 111,152 **** + PartHoldersInitialized = 1; + } + +! #if 1 +! for ( gr = 0; gr < mode_gr; gr++ ) +! for ( ch = 0; ch < stereo; ch++ ) +! { +! int *pi = &l3_enc[gr][ch][0]; +! double *pr = &xr[gr][ch][0]; +! for ( i = 0; i < 576; i++, pr++, pi++ ) +! { +! if ( (*pr < 0) && (*pi > 0) ) +! *pi *= -1; +! } +! } +! #endif + +- encodeSideInfo( l3_side ); +- encodeMainData( l3_enc, l3_side, scalefac ); +- write_ancillary_data( ancillary, ancillary_bits ); + +! if ( l3_side->resvDrain ) +! drain_into_ancillary_data( l3_side->resvDrain ); + /* + Put frameData together for the call + to BitstreamFrame() + */ +- frameData->putbits = putMyBits; + frameData->frameLength = bitsPerFrame; +! frameData->nGranules = mode_gr; +! frameData->nChannels = stereo; + frameData->header = headerPH->part; + frameData->frameSI = frameSIPH->part; + +! for ( ch = 0; ch < stereo; ch++ ) + frameData->channelSI[ch] = channelSIPH[ch]->part; + +! for ( gr = 0; gr < mode_gr; gr++ ) +! for ( ch = 0; ch < stereo; ch++ ) + { + frameData->spectrumSI[gr][ch] = spectrumSIPH[gr][ch]->part; + frameData->scaleFactors[gr][ch] = scaleFactorsPH[gr][ch]->part; +--- 98,124 ---- + PartHoldersInitialized = 1; + } + +! encodeSideInfo( gfp,l3_side ); +! encodeMainData( gfp,l3_enc, l3_side, scalefac ); + + +! +! drain_into_ancillary_data( l3_side->resvDrain ); + /* + Put frameData together for the call + to BitstreamFrame() + */ + frameData->frameLength = bitsPerFrame; +! frameData->nGranules = gfp->mode_gr; +! frameData->nChannels = gfp->stereo; + frameData->header = headerPH->part; + frameData->frameSI = frameSIPH->part; + +! for ( ch = 0; ch < gfp->stereo; ch++ ) + frameData->channelSI[ch] = channelSIPH[ch]->part; + +! for ( gr = 0; gr < gfp->mode_gr; gr++ ) +! for ( ch = 0; ch < gfp->stereo; ch++ ) + { + frameData->spectrumSI[gr][ch] = spectrumSIPH[gr][ch]->part; + frameData->scaleFactors[gr][ch] = scaleFactorsPH[gr][ch]->part; +*************** +*** 159,202 **** + + /* we set this here -- it will be tested in the next loops iteration */ + l3_side->main_data_begin = frameResults->nextBackPtr; + } + + void +! III_FlushBitstream() + { +! assert( PartHoldersInitialized ); +! BF_FlushBitstream( frameData, frameResults ); + } + + static unsigned slen1_tab[16] = { 0, 0, 0, 0, 3, 1, 1, 1, 2, 2, 2, 3, 3, 3, 4, 4 }; + static unsigned slen2_tab[16] = { 0, 1, 2, 3, 0, 1, 2, 3, 1, 2, 3, 1, 2, 3, 2, 3 }; + + static void +! encodeMainData( int l3_enc[2][2][576], + III_side_info_t *si, +! III_scalefac_t *scalefac ) + { +! int i, gr, ch, sfb, window, mode_gr; +! layer *info = fr_ps->header; + +- if ( info->version == 1 ) +- mode_gr = 2; +- else +- mode_gr = 1; + +! for ( gr = 0; gr < mode_gr; gr++ ) +! for ( ch = 0; ch < stereo; ch++ ) + scaleFactorsPH[gr][ch]->part->nrEntries = 0; + +! for ( gr = 0; gr < mode_gr; gr++ ) +! for ( ch = 0; ch < stereo; ch++ ) + codedDataPH[gr][ch]->part->nrEntries = 0; + +! if ( info->version == 1 ) + { /* MPEG 1 */ + for ( gr = 0; gr < 2; gr++ ) + { +! for ( ch = 0; ch < stereo; ch++ ) + { + BF_PartHolder **pph = &scaleFactorsPH[gr][ch]; + gr_info *gi = &(si->gr[gr].ch[ch].tt); +--- 131,171 ---- + + /* we set this here -- it will be tested in the next loops iteration */ + l3_side->main_data_begin = frameResults->nextBackPtr; ++ + } + + void +! III_FlushBitstream(void) + { +! if (PartHoldersInitialized!=0) +! BF_FlushBitstream( frameData, frameResults ); + } + + static unsigned slen1_tab[16] = { 0, 0, 0, 0, 3, 1, 1, 1, 2, 2, 2, 3, 3, 3, 4, 4 }; + static unsigned slen2_tab[16] = { 0, 1, 2, 3, 0, 1, 2, 3, 1, 2, 3, 1, 2, 3, 2, 3 }; + + static void +! encodeMainData( lame_global_flags *gfp, +! int l3_enc[2][2][576], + III_side_info_t *si, +! III_scalefac_t scalefac[2][2] ) + { +! int i, gr, ch, sfb, window; + + +! for ( gr = 0; gr < gfp->mode_gr; gr++ ) +! for ( ch = 0; ch < gfp->stereo; ch++ ) + scaleFactorsPH[gr][ch]->part->nrEntries = 0; + +! for ( gr = 0; gr < gfp->mode_gr; gr++ ) +! for ( ch = 0; ch < gfp->stereo; ch++ ) + codedDataPH[gr][ch]->part->nrEntries = 0; + +! if ( gfp->version == 1 ) + { /* MPEG 1 */ + for ( gr = 0; gr < 2; gr++ ) + { +! for ( ch = 0; ch < gfp->stereo; ch++ ) + { + BF_PartHolder **pph = &scaleFactorsPH[gr][ch]; + gr_info *gi = &(si->gr[gr].ch[ch].tt); +*************** +*** 204,253 **** + unsigned slen2 = slen2_tab[ gi->scalefac_compress ]; + int *ix = &l3_enc[gr][ch][0]; + +! if ( (gi->window_switching_flag == 1) && (gi->block_type == 2) ) + { + if ( gi->mixed_block_flag ) + { + for ( sfb = 0; sfb < 8; sfb++ ) +! *pph = BF_addEntry( *pph, scalefac->l[gr][ch][sfb], slen1 ); + + for ( sfb = 3; sfb < 6; sfb++ ) + for ( window = 0; window < 3; window++ ) +! *pph = BF_addEntry( *pph, scalefac->s[gr][ch][sfb][window], slen1 ); + + for ( sfb = 6; sfb < 12; sfb++ ) + for ( window = 0; window < 3; window++ ) +! *pph = BF_addEntry( *pph, scalefac->s[gr][ch][sfb][window], slen2 ); + + } + else + { + for ( sfb = 0; sfb < 6; sfb++ ) + for ( window = 0; window < 3; window++ ) +! *pph = BF_addEntry( *pph, scalefac->s[gr][ch][sfb][window], slen1 ); + + for ( sfb = 6; sfb < 12; sfb++ ) + for ( window = 0; window < 3; window++ ) +! *pph = BF_addEntry( *pph, scalefac->s[gr][ch][sfb][window], slen2 ); + } + } + else + { + if ( (gr == 0) || (si->scfsi[ch][0] == 0) ) + for ( sfb = 0; sfb < 6; sfb++ ) +! *pph = BF_addEntry( *pph, scalefac->l[gr][ch][sfb], slen1 ); + + if ( (gr == 0) || (si->scfsi[ch][1] == 0) ) + for ( sfb = 6; sfb < 11; sfb++ ) +! *pph = BF_addEntry( *pph, scalefac->l[gr][ch][sfb], slen1 ); + + if ( (gr == 0) || (si->scfsi[ch][2] == 0) ) + for ( sfb = 11; sfb < 16; sfb++ ) +! *pph = BF_addEntry( *pph, scalefac->l[gr][ch][sfb], slen2 ); + + if ( (gr == 0) || (si->scfsi[ch][3] == 0) ) + for ( sfb = 16; sfb < 21; sfb++ ) +! *pph = BF_addEntry( *pph, scalefac->l[gr][ch][sfb], slen2 ); + } + Huffmancodebits( &codedDataPH[gr][ch], ix, gi ); + } /* for ch */ +--- 173,224 ---- + unsigned slen2 = slen2_tab[ gi->scalefac_compress ]; + int *ix = &l3_enc[gr][ch][0]; + +! if (gi->block_type == SHORT_TYPE) + { ++ #ifdef ALLOW_MIXED + if ( gi->mixed_block_flag ) + { + for ( sfb = 0; sfb < 8; sfb++ ) +! *pph = BF_addEntry( *pph, scalefac[gr][ch].l[sfb], slen1 ); + + for ( sfb = 3; sfb < 6; sfb++ ) + for ( window = 0; window < 3; window++ ) +! *pph = BF_addEntry( *pph, scalefac[gr][ch].s[sfb][window], slen1 ); + + for ( sfb = 6; sfb < 12; sfb++ ) + for ( window = 0; window < 3; window++ ) +! *pph = BF_addEntry( *pph, scalefac[gr][ch].s[sfb][window], slen2 ); + + } + else ++ #endif + { + for ( sfb = 0; sfb < 6; sfb++ ) + for ( window = 0; window < 3; window++ ) +! *pph = BF_addEntry( *pph, scalefac[gr][ch].s[sfb][window], slen1 ); + + for ( sfb = 6; sfb < 12; sfb++ ) + for ( window = 0; window < 3; window++ ) +! *pph = BF_addEntry( *pph, scalefac[gr][ch].s[sfb][window], slen2 ); + } + } + else + { + if ( (gr == 0) || (si->scfsi[ch][0] == 0) ) + for ( sfb = 0; sfb < 6; sfb++ ) +! *pph = BF_addEntry( *pph, scalefac[gr][ch].l[sfb], slen1 ); + + if ( (gr == 0) || (si->scfsi[ch][1] == 0) ) + for ( sfb = 6; sfb < 11; sfb++ ) +! *pph = BF_addEntry( *pph, scalefac[gr][ch].l[sfb], slen1 ); + + if ( (gr == 0) || (si->scfsi[ch][2] == 0) ) + for ( sfb = 11; sfb < 16; sfb++ ) +! *pph = BF_addEntry( *pph, scalefac[gr][ch].l[sfb], slen2 ); + + if ( (gr == 0) || (si->scfsi[ch][3] == 0) ) + for ( sfb = 16; sfb < 21; sfb++ ) +! *pph = BF_addEntry( *pph, scalefac[gr][ch].l[sfb], slen2 ); + } + Huffmancodebits( &codedDataPH[gr][ch], ix, gi ); + } /* for ch */ +*************** +*** 256,277 **** + else + { /* MPEG 2 */ + gr = 0; +! for ( ch = 0; ch < stereo; ch++ ) + { + BF_PartHolder **pph = &scaleFactorsPH[gr][ch]; + gr_info *gi = &(si->gr[gr].ch[ch].tt); + int *ix = &l3_enc[gr][ch][0]; + int sfb_partition; +- + assert( gi->sfb_partition_table ); + +! if ( (gi->window_switching_flag == 1) && (gi->block_type == 2) ) + { + if ( gi->mixed_block_flag ) + { + sfb_partition = 0; + for ( sfb = 0; sfb < 8; sfb++ ) +! *pph = BF_addEntry( *pph, scalefac->l[gr][ch][sfb], gi->slen[sfb_partition] ); + + for ( sfb = 3, sfb_partition = 1; sfb_partition < 4; sfb_partition++ ) + { +--- 227,248 ---- + else + { /* MPEG 2 */ + gr = 0; +! for ( ch = 0; ch < gfp->stereo; ch++ ) + { + BF_PartHolder **pph = &scaleFactorsPH[gr][ch]; + gr_info *gi = &(si->gr[gr].ch[ch].tt); + int *ix = &l3_enc[gr][ch][0]; + int sfb_partition; + assert( gi->sfb_partition_table ); + +! if (gi->block_type == SHORT_TYPE) + { ++ #ifdef ALLOW_MIXED + if ( gi->mixed_block_flag ) + { + sfb_partition = 0; + for ( sfb = 0; sfb < 8; sfb++ ) +! *pph = BF_addEntry( *pph, scalefac[gr][ch].l[sfb], gi->slen[sfb_partition] ); + + for ( sfb = 3, sfb_partition = 1; sfb_partition < 4; sfb_partition++ ) + { +*************** +*** 279,288 **** + int slen = gi->slen[ sfb_partition ]; + for ( i = 0; i < sfbs; i++, sfb++ ) + for ( window = 0; window < 3; window++ ) +! *pph = BF_addEntry( *pph, scalefac->s[gr][ch][sfb][window], slen ); + } + } + else + { + for ( sfb = 0, sfb_partition = 0; sfb_partition < 4; sfb_partition++ ) + { +--- 250,260 ---- + int slen = gi->slen[ sfb_partition ]; + for ( i = 0; i < sfbs; i++, sfb++ ) + for ( window = 0; window < 3; window++ ) +! *pph = BF_addEntry( *pph, scalefac[gr][ch].s[sfb][window], slen ); + } + } + else ++ #endif + { + for ( sfb = 0, sfb_partition = 0; sfb_partition < 4; sfb_partition++ ) + { +*************** +*** 290,296 **** + int slen = gi->slen[ sfb_partition ]; + for ( i = 0; i < sfbs; i++, sfb++ ) + for ( window = 0; window < 3; window++ ) +! *pph = BF_addEntry( *pph, scalefac->s[gr][ch][sfb][window], slen ); + } + } + } +--- 262,268 ---- + int slen = gi->slen[ sfb_partition ]; + for ( i = 0; i < sfbs; i++, sfb++ ) + for ( window = 0; window < 3; window++ ) +! *pph = BF_addEntry( *pph, scalefac[gr][ch].s[sfb][window], slen ); + } + } + } +*************** +*** 301,486 **** + int sfbs = gi->sfb_partition_table[ sfb_partition ]; + int slen = gi->slen[ sfb_partition ]; + for ( i = 0; i < sfbs; i++, sfb++ ) +! *pph = BF_addEntry( *pph, scalefac->l[gr][ch][sfb], slen ); + } + } + Huffmancodebits( &codedDataPH[gr][ch], ix, gi ); + } /* for ch */ + } + } /* main_data */ + +! static unsigned int crc = 0; + +! static int encodeSideInfo( III_side_info_t *si ) + { +! int gr, ch, scfsi_band, region, window, bits_sent, mode_gr; +! layer *info = fr_ps->header; + +! mode_gr = (info->version == 1) ? 2 : 1; + + headerPH->part->nrEntries = 0; + headerPH = BF_addEntry( headerPH, 0xfff, 12 ); +! headerPH = BF_addEntry( headerPH, info->version, 1 ); +! headerPH = BF_addEntry( headerPH, 4 - info->lay, 2 ); +! headerPH = BF_addEntry( headerPH, !info->error_protection, 1 ); +! headerPH = BF_addEntry( headerPH, info->bitrate_index, 4 ); +! headerPH = BF_addEntry( headerPH, info->sampling_frequency, 2 ); +! headerPH = BF_addEntry( headerPH, info->padding, 1 ); +! headerPH = BF_addEntry( headerPH, info->extension, 1 ); +! headerPH = BF_addEntry( headerPH, info->mode, 2 ); +! headerPH = BF_addEntry( headerPH, info->mode_ext, 2 ); +! headerPH = BF_addEntry( headerPH, info->copyright, 1 ); +! headerPH = BF_addEntry( headerPH, info->original, 1 ); +! headerPH = BF_addEntry( headerPH, info->emphasis, 2 ); + + bits_sent = 32; +! +! if ( fr_ps->header->error_protection ) +! { +! headerPH = BF_addEntry( headerPH, crc, 16 ); +! bits_sent += 16; +! } + + frameSIPH->part->nrEntries = 0; + +! for (ch = 0; ch < stereo; ch++ ) + channelSIPH[ch]->part->nrEntries = 0; + +! for ( gr = 0; gr < mode_gr; gr++ ) +! for ( ch = 0; ch < stereo; ch++ ) + spectrumSIPH[gr][ch]->part->nrEntries = 0; + +! if ( info->version == 1 ) + { /* MPEG1 */ +! frameSIPH = BF_addEntry( frameSIPH, si->main_data_begin, 9 ); + +! if ( stereo == 2 ) +! frameSIPH = BF_addEntry( frameSIPH, si->private_bits, 3 ); + else +! frameSIPH = BF_addEntry( frameSIPH, si->private_bits, 5 ); + +! for ( ch = 0; ch < stereo; ch++ ) + for ( scfsi_band = 0; scfsi_band < 4; scfsi_band++ ) + { + BF_PartHolder **pph = &channelSIPH[ch]; +! *pph = BF_addEntry( *pph, si->scfsi[ch][scfsi_band], 1 ); + } + + for ( gr = 0; gr < 2; gr++ ) +! for ( ch = 0; ch < stereo; ch++ ) + { + BF_PartHolder **pph = &spectrumSIPH[gr][ch]; + gr_info *gi = &(si->gr[gr].ch[ch].tt); +! *pph = BF_addEntry( *pph, gi->part2_3_length, 12 ); +! *pph = BF_addEntry( *pph, gi->big_values, 9 ); +! *pph = BF_addEntry( *pph, gi->global_gain, 8 ); +! *pph = BF_addEntry( *pph, gi->scalefac_compress, 4 ); +! *pph = BF_addEntry( *pph, gi->window_switching_flag, 1 ); + + if ( gi->window_switching_flag ) + { +! *pph = BF_addEntry( *pph, gi->block_type, 2 ); +! *pph = BF_addEntry( *pph, gi->mixed_block_flag, 1 ); + + for ( region = 0; region < 2; region++ ) +! *pph = BF_addEntry( *pph, gi->table_select[region], 5 ); + for ( window = 0; window < 3; window++ ) +! *pph = BF_addEntry( *pph, gi->subblock_gain[window], 3 ); + } + else + { +! assert( gi->block_type == 0 ); + for ( region = 0; region < 3; region++ ) +! *pph = BF_addEntry( *pph, gi->table_select[region], 5 ); + +! *pph = BF_addEntry( *pph, gi->region0_count, 4 ); +! *pph = BF_addEntry( *pph, gi->region1_count, 3 ); + } + +! *pph = BF_addEntry( *pph, gi->preflag, 1 ); +! *pph = BF_addEntry( *pph, gi->scalefac_scale, 1 ); +! *pph = BF_addEntry( *pph, gi->count1table_select, 1 ); + } + +! if ( stereo == 2 ) + bits_sent += 256; + else + bits_sent += 136; + } + else + { /* MPEG2 */ +! frameSIPH = BF_addEntry( frameSIPH, si->main_data_begin, 8 ); + +! if ( stereo == 2 ) +! frameSIPH = BF_addEntry( frameSIPH, si->private_bits, 2 ); + else +! frameSIPH = BF_addEntry( frameSIPH, si->private_bits, 1 ); + + gr = 0; +! for ( ch = 0; ch < stereo; ch++ ) + { + BF_PartHolder **pph = &spectrumSIPH[gr][ch]; + gr_info *gi = &(si->gr[gr].ch[ch].tt); +! *pph = BF_addEntry( *pph, gi->part2_3_length, 12 ); +! *pph = BF_addEntry( *pph, gi->big_values, 9 ); +! *pph = BF_addEntry( *pph, gi->global_gain, 8 ); +! *pph = BF_addEntry( *pph, gi->scalefac_compress, 9 ); +! *pph = BF_addEntry( *pph, gi->window_switching_flag, 1 ); + + if ( gi->window_switching_flag ) + { +! *pph = BF_addEntry( *pph, gi->block_type, 2 ); +! *pph = BF_addEntry( *pph, gi->mixed_block_flag, 1 ); + + for ( region = 0; region < 2; region++ ) +! *pph = BF_addEntry( *pph, gi->table_select[region], 5 ); + for ( window = 0; window < 3; window++ ) +! *pph = BF_addEntry( *pph, gi->subblock_gain[window], 3 ); + } + else + { + for ( region = 0; region < 3; region++ ) +! *pph = BF_addEntry( *pph, gi->table_select[region], 5 ); + +! *pph = BF_addEntry( *pph, gi->region0_count, 4 ); +! *pph = BF_addEntry( *pph, gi->region1_count, 3 ); + } + +! *pph = BF_addEntry( *pph, gi->scalefac_scale, 1 ); +! *pph = BF_addEntry( *pph, gi->count1table_select, 1 ); + } +! if ( stereo == 2 ) + bits_sent += 136; + else + bits_sent += 72; + } +- return bits_sent; +- } + +! static void +! write_ancillary_data( char *theData, int lengthInBits ) +! { +! /* +! */ +! int bytesToSend = lengthInBits / 8; +! int remainingBits = lengthInBits % 8; +! unsigned wrd; +! int i; +! +! userFrameDataPH->part->nrEntries = 0; +! +! for ( i = 0; i < bytesToSend; i++ ) +! { +! wrd = theData[i]; +! userFrameDataPH = BF_addEntry( userFrameDataPH, wrd, 8 ); +! } +! if ( remainingBits ) +! { +! /* right-justify remaining bits */ +! wrd = theData[bytesToSend] >> (8 - remainingBits); +! userFrameDataPH = BF_addEntry( userFrameDataPH, wrd, remainingBits ); + } +! + } + + /* +--- 273,455 ---- + int sfbs = gi->sfb_partition_table[ sfb_partition ]; + int slen = gi->slen[ sfb_partition ]; + for ( i = 0; i < sfbs; i++, sfb++ ) +! *pph = BF_addEntry( *pph, scalefac[gr][ch].l[sfb], slen ); + } + } ++ ++ ++ + Huffmancodebits( &codedDataPH[gr][ch], ix, gi ); + } /* for ch */ + } + } /* main_data */ + +! static unsigned int crc = 0; /* (jo) current crc */ +! +! /* (jo) this wrapper function for BF_addEntry() updates also the crc */ +! static BF_PartHolder *CRC_BF_addEntry( BF_PartHolder *thePH, u_int value, u_int length ) +! { +! u_int bit = 1 << length; +! +! while((bit >>= 1)){ +! crc <<= 1; +! if (!(crc & 0x10000) ^ !(value & bit)) +! crc ^= CRC16_POLYNOMIAL; +! } +! crc &= 0xffff; +! return BF_addEntry(thePH, value, length); +! } +! +! +! + +! static int encodeSideInfo( lame_global_flags *gfp,III_side_info_t *si ) + { +! int gr, ch, scfsi_band, region, window, bits_sent; + +! crc = 0xffff; /* (jo) init crc16 for error_protection */ + + headerPH->part->nrEntries = 0; + headerPH = BF_addEntry( headerPH, 0xfff, 12 ); +! headerPH = BF_addEntry( headerPH, gfp->version, 1 ); +! headerPH = BF_addEntry( headerPH, 1, 2 ); +! headerPH = BF_addEntry( headerPH, !gfp->error_protection, 1 ); +! /* (jo) from now on call the CRC_BF_addEntry() wrapper to update crc */ +! headerPH = CRC_BF_addEntry( headerPH, gfp->bitrate_index, 4 ); +! headerPH = CRC_BF_addEntry( headerPH, gfp->samplerate_index, 2 ); +! headerPH = CRC_BF_addEntry( headerPH, gfp->padding, 1 ); +! headerPH = CRC_BF_addEntry( headerPH, gfp->extension, 1 ); +! headerPH = CRC_BF_addEntry( headerPH, gfp->mode, 2 ); +! headerPH = CRC_BF_addEntry( headerPH, gfp->mode_ext, 2 ); +! headerPH = CRC_BF_addEntry( headerPH, gfp->copyright, 1 ); +! headerPH = CRC_BF_addEntry( headerPH, gfp->original, 1 ); +! headerPH = CRC_BF_addEntry( headerPH, gfp->emphasis, 2 ); + + bits_sent = 32; +! +! /* (jo) see below for BF_addEntry( headerPH, crc, 16 ); */ + + frameSIPH->part->nrEntries = 0; + +! for (ch = 0; ch < gfp->stereo; ch++ ) + channelSIPH[ch]->part->nrEntries = 0; + +! for ( gr = 0; gr < gfp->mode_gr; gr++ ) +! for ( ch = 0; ch < gfp->stereo; ch++ ) + spectrumSIPH[gr][ch]->part->nrEntries = 0; + +! if ( gfp->version == 1 ) + { /* MPEG1 */ +! frameSIPH = CRC_BF_addEntry( frameSIPH, si->main_data_begin, 9 ); + +! if ( gfp->stereo == 2 ) +! frameSIPH = CRC_BF_addEntry( frameSIPH, si->private_bits, 3 ); + else +! frameSIPH = CRC_BF_addEntry( frameSIPH, si->private_bits, 5 ); + +! for ( ch = 0; ch < gfp->stereo; ch++ ) + for ( scfsi_band = 0; scfsi_band < 4; scfsi_band++ ) + { + BF_PartHolder **pph = &channelSIPH[ch]; +! *pph = CRC_BF_addEntry( *pph, si->scfsi[ch][scfsi_band], 1 ); + } + + for ( gr = 0; gr < 2; gr++ ) +! for ( ch = 0; ch < gfp->stereo; ch++ ) + { + BF_PartHolder **pph = &spectrumSIPH[gr][ch]; + gr_info *gi = &(si->gr[gr].ch[ch].tt); +! *pph = CRC_BF_addEntry( *pph, gi->part2_3_length, 12 ); +! *pph = CRC_BF_addEntry( *pph, gi->big_values, 9 ); +! *pph = CRC_BF_addEntry( *pph, gi->global_gain, 8 ); +! *pph = CRC_BF_addEntry( *pph, gi->scalefac_compress, 4 ); +! *pph = CRC_BF_addEntry( *pph, gi->window_switching_flag, 1 ); + + if ( gi->window_switching_flag ) + { +! *pph = CRC_BF_addEntry( *pph, gi->block_type, 2 ); +! *pph = CRC_BF_addEntry( *pph, gi->mixed_block_flag, 1 ); + + for ( region = 0; region < 2; region++ ) +! *pph = CRC_BF_addEntry( *pph, gi->table_select[region], 5 ); + for ( window = 0; window < 3; window++ ) +! *pph = CRC_BF_addEntry( *pph, gi->subblock_gain[window], 3 ); + } + else + { +! assert( gi->block_type == NORM_TYPE ); + for ( region = 0; region < 3; region++ ) +! *pph = CRC_BF_addEntry( *pph, gi->table_select[region], 5 ); + +! *pph = CRC_BF_addEntry( *pph, gi->region0_count, 4 ); +! *pph = CRC_BF_addEntry( *pph, gi->region1_count, 3 ); + } + +! *pph = CRC_BF_addEntry( *pph, gi->preflag, 1 ); +! *pph = CRC_BF_addEntry( *pph, gi->scalefac_scale, 1 ); +! *pph = CRC_BF_addEntry( *pph, gi->count1table_select, 1 ); + } + +! if ( gfp->stereo == 2 ) + bits_sent += 256; + else + bits_sent += 136; + } + else + { /* MPEG2 */ +! frameSIPH = CRC_BF_addEntry( frameSIPH, si->main_data_begin, 8 ); + +! if ( gfp->stereo == 2 ) +! frameSIPH = CRC_BF_addEntry( frameSIPH, si->private_bits, 2 ); + else +! frameSIPH = CRC_BF_addEntry( frameSIPH, si->private_bits, 1 ); + + gr = 0; +! for ( ch = 0; ch < gfp->stereo; ch++ ) + { + BF_PartHolder **pph = &spectrumSIPH[gr][ch]; + gr_info *gi = &(si->gr[gr].ch[ch].tt); +! *pph = CRC_BF_addEntry( *pph, gi->part2_3_length, 12 ); +! *pph = CRC_BF_addEntry( *pph, gi->big_values, 9 ); +! *pph = CRC_BF_addEntry( *pph, gi->global_gain, 8 ); +! *pph = CRC_BF_addEntry( *pph, gi->scalefac_compress, 9 ); +! *pph = CRC_BF_addEntry( *pph, gi->window_switching_flag, 1 ); + + if ( gi->window_switching_flag ) + { +! *pph = CRC_BF_addEntry( *pph, gi->block_type, 2 ); +! *pph = CRC_BF_addEntry( *pph, gi->mixed_block_flag, 1 ); + + for ( region = 0; region < 2; region++ ) +! *pph = CRC_BF_addEntry( *pph, gi->table_select[region], 5 ); + for ( window = 0; window < 3; window++ ) +! *pph = CRC_BF_addEntry( *pph, gi->subblock_gain[window], 3 ); + } + else + { + for ( region = 0; region < 3; region++ ) +! *pph = CRC_BF_addEntry( *pph, gi->table_select[region], 5 ); + +! *pph = CRC_BF_addEntry( *pph, gi->region0_count, 4 ); +! *pph = CRC_BF_addEntry( *pph, gi->region1_count, 3 ); + } + +! *pph = CRC_BF_addEntry( *pph, gi->scalefac_scale, 1 ); +! *pph = CRC_BF_addEntry( *pph, gi->count1table_select, 1 ); + } +! if ( gfp->stereo == 2 ) + bits_sent += 136; + else + bits_sent += 72; + } + +! if ( gfp->error_protection ) +! { /* (jo) error_protection: add crc16 information to header */ +! headerPH = BF_addEntry( headerPH, crc, 16 ); +! bits_sent += 16; + } +! +! return bits_sent; + } + + /* +*************** +*** 501,511 **** + /* + userFrameDataPH->part->nrEntries set by call to write_ancillary_data() + */ +! + for ( i = 0; i < wordsToSend; i++ ) + userFrameDataPH = BF_addEntry( userFrameDataPH, 0, 32 ); + if ( remainingBits ) +! userFrameDataPH = BF_addEntry( userFrameDataPH, 0, remainingBits ); + } + + /* +--- 470,481 ---- + /* + userFrameDataPH->part->nrEntries set by call to write_ancillary_data() + */ +! +! userFrameDataPH->part->nrEntries = 0; + for ( i = 0; i < wordsToSend; i++ ) + userFrameDataPH = BF_addEntry( userFrameDataPH, 0, 32 ); + if ( remainingBits ) +! userFrameDataPH = BF_addEntry( userFrameDataPH, 0, remainingBits ); + } + + /* +*************** +*** 517,541 **** + Huffmancodebits( BF_PartHolder **pph, int *ix, gr_info *gi ) + { + int L3_huffman_coder_count1( BF_PartHolder **pph, struct huffcodetab *h, int v, int w, int x, int y ); +- int bigv_bitcount( int ix[576], gr_info *cod_info ); + + int region1Start; + int region2Start; + int i, bigvalues, count1End; + int v, w, x, y, bits, cbits, xbits, stuffingBits; + unsigned int code, ext; +! struct huffcodetab *h; +! int bvbits, c1bits, tablezeros, r0, r1, r2, rt, *pr; + int bitsWritten = 0; +! int idx = 0; +! tablezeros = 0; +! r0 = r1 = r2 = 0; + + /* 1: Write the bigvalues */ + bigvalues = gi->big_values * 2; + if ( bigvalues ) + { +! if ( !(gi->mixed_block_flag) && gi->window_switching_flag && (gi->block_type == 2) ) + { /* Three short blocks */ + /* + Within each scalefactor band, data is given for successive +--- 487,509 ---- + Huffmancodebits( BF_PartHolder **pph, int *ix, gr_info *gi ) + { + int L3_huffman_coder_count1( BF_PartHolder **pph, struct huffcodetab *h, int v, int w, int x, int y ); + + int region1Start; + int region2Start; + int i, bigvalues, count1End; + int v, w, x, y, bits, cbits, xbits, stuffingBits; + unsigned int code, ext; +! #ifdef DEBUG +! int bvbits, c1bits; +! #endif + int bitsWritten = 0; +! + + /* 1: Write the bigvalues */ + bigvalues = gi->big_values * 2; + if ( bigvalues ) + { +! if ( !(gi->mixed_block_flag) && (gi->block_type == SHORT_TYPE) ) + { /* Three short blocks */ + /* + Within each scalefactor band, data is given for successive +*************** +*** 546,552 **** + int sfb, window, line, start, end; + + I192_3 *ix_s; +- int *scalefac = &sfBandIndex[fr_ps->header->sampling_frequency + (fr_ps->header->version * 3)].s[0]; + + ix_s = (I192_3 *) ix; + region1Start = 12; +--- 514,519 ---- +*************** +*** 555,562 **** + for ( sfb = 0; sfb < 13; sfb++ ) + { + unsigned tableindex = 100; +! start = scalefac[ sfb ]; +! end = scalefac[ sfb+1 ]; + + if ( start < region1Start ) + tableindex = gi->table_select[ 0 ]; +--- 522,529 ---- + for ( sfb = 0; sfb < 13; sfb++ ) + { + unsigned tableindex = 100; +! start = scalefac_band.s[ sfb ]; +! end = scalefac_band.s[ sfb+1 ]; + + if ( start < region1Start ) + tableindex = gi->table_select[ 0 ]; +*************** +*** 569,576 **** + { + x = (*ix_s)[line][window]; + y = (*ix_s)[line + 1][window]; +- assert( idx < 576 ); +- assert( idx >= 0 ); + bits = HuffmanCode( tableindex, x, y, &code, &ext, &cbits, &xbits ); + *pph = BF_addEntry( *pph, code, cbits ); + *pph = BF_addEntry( *pph, ext, xbits ); +--- 536,541 ---- +*************** +*** 580,591 **** + } + } + else +! if ( gi->mixed_block_flag && gi->block_type == 2 ) + { /* Mixed blocks long, short */ + int sfb, window, line, start, end; + unsigned tableindex; + I192_3 *ix_s; +- int *scalefac = &sfBandIndex[fr_ps->header->sampling_frequency + (fr_ps->header->version * 3)].s[0]; + + ix_s = (I192_3 *) ix; + +--- 545,556 ---- + } + } + else +! #ifdef ALLOW_MIXED +! if ( gi->mixed_block_flag && gi->block_type == SHORT_TYPE ) + { /* Mixed blocks long, short */ + int sfb, window, line, start, end; + unsigned tableindex; + I192_3 *ix_s; + + ix_s = (I192_3 *) ix; + +*************** +*** 608,615 **** + + for ( sfb = 3; sfb < 13; sfb++ ) + { +! start = scalefac[ sfb ]; +! end = scalefac[ sfb+1 ]; + + for ( window = 0; window < 3; window++ ) + for ( line = start; line < end; line += 2 ) +--- 573,580 ---- + + for ( sfb = 3; sfb < 13; sfb++ ) + { +! start = scalefac_band.s[ sfb ]; +! end = scalefac_band.s[ sfb+1 ]; + + for ( window = 0; window < 3; window++ ) + for ( line = start; line < end; line += 2 ) +*************** +*** 625,632 **** + + } + else + { /* Long blocks */ +- int *scalefac = &sfBandIndex[fr_ps->header->sampling_frequency + (fr_ps->header->version * 3)].l[0]; + unsigned scalefac_index = 100; + + if ( gi->mixed_block_flag ) +--- 590,597 ---- + + } + else ++ #endif + { /* Long blocks */ + unsigned scalefac_index = 100; + + if ( gi->mixed_block_flag ) +*************** +*** 638,649 **** + { + scalefac_index = gi->region0_count + 1; + assert( scalefac_index < 23 ); +! region1Start = scalefac[ scalefac_index ]; + scalefac_index += gi->region1_count + 1; + assert( scalefac_index < 23 ); +! region2Start = scalefac[ scalefac_index ]; +! assert( region1Start == gi->address1 ); + } + for ( i = 0; i < bigvalues; i += 2 ) + { + unsigned tableindex = 100; +--- 603,614 ---- + { + scalefac_index = gi->region0_count + 1; + assert( scalefac_index < 23 ); +! region1Start = scalefac_band.l[ scalefac_index ]; + scalefac_index += gi->region1_count + 1; + assert( scalefac_index < 23 ); +! region2Start = scalefac_band.l[ scalefac_index ]; + } ++ + for ( i = 0; i < bigvalues; i += 2 ) + { + unsigned tableindex = 100; +*************** +*** 651,725 **** + if ( i < region1Start ) + { + tableindex = gi->table_select[0]; +- pr = &r0; + } + else + if ( i < region2Start ) + { + tableindex = gi->table_select[1]; +- pr = &r1; + } + else + { + tableindex = gi->table_select[2]; +- pr = &r2; + } + assert( tableindex < 32 ); +- h = &ht[ tableindex ]; + /* get huffman code */ + x = ix[i]; + y = ix[i + 1]; + if ( tableindex ) + { + bits = HuffmanCode( tableindex, x, y, &code, &ext, &cbits, &xbits ); + *pph = BF_addEntry( *pph, code, cbits ); + *pph = BF_addEntry( *pph, ext, xbits ); +! bitsWritten += rt = bits; +! *pr += rt; +! } +! else +! { +! tablezeros += 1; +! *pr = 0; + } + } + } + } + bvbits = bitsWritten; + + /* 2: Write count1 area */ + assert( (gi->count1table_select < 2) ); +- h = &ht[gi->count1table_select + 32]; + count1End = bigvalues + (gi->count1 * 4); + assert( count1End <= 576 ); + for ( i = bigvalues; i < count1End; i += 4 ) + { + v = ix[i]; + w = ix[i+1]; + x = ix[i+2]; + y = ix[i+3]; +! bitsWritten += L3_huffman_coder_count1( pph, h, v, w, x, y ); + } + c1bits = bitsWritten - bvbits; + if ( (stuffingBits = gi->part2_3_length - gi->part2_length - bitsWritten) ) + { + int stuffingWords = stuffingBits / 32; + int remainingBits = stuffingBits % 32; +! assert( stuffingBits > 0 ); + + /* + Due to the nature of the Huffman code + tables, we will pad with ones + */ + while ( stuffingWords-- ) +! *pph = BF_addEntry( *pph, ~0, 32 ); + if ( remainingBits ) +! *pph = BF_addEntry( *pph, ~0, remainingBits ); + bitsWritten += stuffingBits; + } +! assert( bitsWritten == gi->part2_3_length - gi->part2_length ); + #ifdef DEBUG +! printf( "#### %d Huffman bits written (%02d + %02d), part2_length = %d, part2_3_length = %d, %d stuffing ####\n", + bitsWritten, bvbits, c1bits, gi->part2_length, gi->part2_3_length, stuffingBits ); + #endif + } +--- 616,688 ---- + if ( i < region1Start ) + { + tableindex = gi->table_select[0]; + } + else + if ( i < region2Start ) + { + tableindex = gi->table_select[1]; + } + else + { + tableindex = gi->table_select[2]; + } + assert( tableindex < 32 ); + /* get huffman code */ + x = ix[i]; + y = ix[i + 1]; ++ + if ( tableindex ) + { + bits = HuffmanCode( tableindex, x, y, &code, &ext, &cbits, &xbits ); + *pph = BF_addEntry( *pph, code, cbits ); + *pph = BF_addEntry( *pph, ext, xbits ); +! bitsWritten += bits; + } + } + } + } ++ #ifdef DEBUG + bvbits = bitsWritten; ++ #endif + + /* 2: Write count1 area */ + assert( (gi->count1table_select < 2) ); + count1End = bigvalues + (gi->count1 * 4); ++ + assert( count1End <= 576 ); ++ + for ( i = bigvalues; i < count1End; i += 4 ) + { + v = ix[i]; + w = ix[i+1]; + x = ix[i+2]; + y = ix[i+3]; +! bitsWritten += L3_huffman_coder_count1( pph, &ht[gi->count1table_select + 32], v, w, x, y ); + } ++ #ifdef DEBUG + c1bits = bitsWritten - bvbits; ++ #endif + if ( (stuffingBits = gi->part2_3_length - gi->part2_length - bitsWritten) ) + { + int stuffingWords = stuffingBits / 32; + int remainingBits = stuffingBits % 32; +! +! fprintf(stderr,"opps - adding stuffing bits = %i.\n",stuffingBits); +! fprintf(stderr,"this should not happen...\n"); + + /* + Due to the nature of the Huffman code + tables, we will pad with ones + */ + while ( stuffingWords-- ) +! *pph = BF_addEntry( *pph, ~(u_int)0, 32 ); + if ( remainingBits ) +! *pph = BF_addEntry( *pph, ~(u_int)0, remainingBits ); + bitsWritten += stuffingBits; + } +! assert( bitsWritten == (int)(gi->part2_3_length - gi->part2_length) ); + #ifdef DEBUG +! fprintf(stderr, "## %d Huffman bits written (%02d + %02d), part2_length = %d, part2_3_length = %d, %d stuffing ##\n", + bitsWritten, bvbits, c1bits, gi->part2_length, gi->part2_3_length, stuffingBits ); + #endif + } +*************** +*** 746,756 **** + signx = abs_and_sign( &x ); + signy = abs_and_sign( &y ); + +! p = v + (w << 1) + (x << 2) + (y << 3); + huffbits = h->table[p]; + len = h->hlen[ p ]; +! *pph = BF_addEntry( *pph, huffbits, len ); +! totalBits += len; + if ( v ) + { + *pph = BF_addEntry( *pph, signv, 1 ); +--- 709,722 ---- + signx = abs_and_sign( &x ); + signy = abs_and_sign( &y ); + +! /* bug fix from Leonid A. Kulakov 9/1999:*/ +! p = (v << 3) + (w << 2) + (x << 1) + y; +! + huffbits = h->table[p]; + len = h->hlen[ p ]; +! *pph = BF_addEntry(*pph, huffbits, len); +! totalBits= 0; +! #if 0 + if ( v ) + { + *pph = BF_addEntry( *pph, signv, 1 ); +*************** +*** 772,778 **** + *pph = BF_addEntry( *pph, signy, 1 ); + totalBits += 1; + } +! return totalBits; + } + + /* +--- 738,769 ---- + *pph = BF_addEntry( *pph, signy, 1 ); + totalBits += 1; + } +! #endif +! +! p=0; +! if ( v ) { +! p = signv; +! ++totalBits; +! } +! +! if ( w ){ +! p = 2*p + signw; +! ++totalBits; +! } +! +! if ( x ) { +! p = 2*p + signx; +! ++totalBits; +! } +! +! if ( y ) { +! p = 2*p + signy; +! ++totalBits; +! } +! +! *pph = BF_addEntry(*pph, p, totalBits); +! +! return totalBits+len; + } + + /* +*************** +*** 781,787 **** + int + HuffmanCode( int table_select, int x, int y, unsigned int *code, unsigned int *ext, int *cbits, int *xbits ) + { +! unsigned signx, signy, linbitsx, linbitsy, linbits, xlen, ylen, idx; + struct huffcodetab *h; + + *cbits = 0; +--- 772,778 ---- + int + HuffmanCode( int table_select, int x, int y, unsigned int *code, unsigned int *ext, int *cbits, int *xbits ) + { +! unsigned signx, signy, linbitsx, linbitsy, linbits, idx; + struct huffcodetab *h; + + *cbits = 0; +*************** +*** 795,807 **** + signx = abs_and_sign( &x ); + signy = abs_and_sign( &y ); + h = &(ht[table_select]); +- xlen = h->xlen; +- ylen = h->ylen; +- linbits = h->linbits; +- linbitsx = linbitsy = 0; + + if ( table_select > 15 ) + { /* ESC-table is used */ + if ( x > 14 ) + { + linbitsx = x - 15; +--- 786,796 ---- + signx = abs_and_sign( &x ); + signy = abs_and_sign( &y ); + h = &(ht[table_select]); + + if ( table_select > 15 ) + { /* ESC-table is used */ ++ linbits = h->xlen; ++ linbitsx = linbitsy = 0; + if ( x > 14 ) + { + linbitsx = x - 15; +*************** +*** 814,822 **** + assert( linbitsy <= h->linmax ); + y = 15; + } +! idx = (x * ylen) + y; + *code = h->table[idx]; +! *cbits = h->hlen[ idx ]; + if ( x > 14 ) + { + *ext |= linbitsx; +--- 803,811 ---- + assert( linbitsy <= h->linmax ); + y = 15; + } +! idx = x * 16 + y; + *code = h->table[idx]; +! *cbits = h->hlen[ idx ]; + if ( x > 14 ) + { + *ext |= linbitsx; +*************** +*** 843,849 **** + } + else + { /* No ESC-words */ +! idx = (x * ylen) + y; + *code = h->table[idx]; + *cbits += h->hlen[ idx ]; + if ( x != 0 ) +--- 832,838 ---- + } + else + { /* No ESC-words */ +! idx = x * 16 + y; + *code = h->table[idx]; + *cbits += h->hlen[ idx ]; + if ( x != 0 ) +*************** +*** 856,862 **** + { + *code <<= 1; + *code |= signy; +! *cbits += 1; + } + } + assert( *cbits <= 32 ); +--- 845,851 ---- + { + *code <<= 1; + *code |= signy; +! *cbits += 1; + } + } + assert( *cbits <= 32 ); +diff -r -c -N encoder/l3bitstream.h lame3.70/l3bitstream.h +*** encoder/l3bitstream.h Wed Jan 22 02:43:15 1997 +--- lame3.70/l3bitstream.h Thu Apr 6 12:50:49 2000 +*************** +*** 2,10 **** + * ISO MPEG Audio Subgroup Software Simulation Group (1996) + * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension + * +! * $Id: l3bitstream.h,v 1.1 1996/02/14 04:04:23 rowlands Exp $ + * + * $Log: l3bitstream.h,v $ + * Revision 1.1 1996/02/14 04:04:23 rowlands + * Initial revision + * +--- 2,26 ---- + * ISO MPEG Audio Subgroup Software Simulation Group (1996) + * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension + * +! * $Id: l3bitstream.h,v 1.5 2000/03/21 23:02:17 markt Exp $ + * + * $Log: l3bitstream.h,v $ ++ * Revision 1.5 2000/03/21 23:02:17 markt ++ * replaced all "gf." by gfp-> ++ * ++ * Revision 1.4 2000/03/14 20:45:04 markt ++ * removed "info" sturct. Removing fr_ps struct ++ * ++ * Revision 1.3 2000/02/01 11:26:32 takehiro ++ * scalefactor's structure changed ++ * ++ * Revision 1.2 1999/12/03 09:45:30 takehiro ++ * little bit cleanup ++ * ++ * Revision 1.1.1.1 1999/11/24 08:43:09 markt ++ * initial checkin of LAME ++ * Starting with LAME 3.57beta with some modifications ++ * + * Revision 1.1 1996/02/14 04:04:23 rowlands + * Initial revision + * +*************** +*** 14,34 **** + #ifndef L3_BITSTREAM_H + #define L3_BITSTREAM_H + +! #include "common.h" +! #include "encoder.h" + +! void III_format_bitstream( int bitsPerFrame, +! frame_params *in_fr_ps, + int l3_enc[2][2][576], + III_side_info_t *l3_side, +! III_scalefac_t *scalefac, +! Bit_stream_struc *in_bs, +! double (*xr)[2][576], +! char *ancillary, +! int anc_bits ); + + int HuffmanCode( int table_select, int x, int y, unsigned *code, unsigned int *extword, int *codebits, int *extbits ); +! void III_FlushBitstream(); + + int abs_and_sign( int *x ); /* returns signx and changes *x to abs(*x) */ + +--- 30,46 ---- + #ifndef L3_BITSTREAM_H + #define L3_BITSTREAM_H + +! #include "util.h" + +! void III_format_bitstream( lame_global_flags *gfp, +! int bitsPerFrame, + int l3_enc[2][2][576], + III_side_info_t *l3_side, +! III_scalefac_t scalefac[2][2], +! Bit_stream_struc *in_bs); + + int HuffmanCode( int table_select, int x, int y, unsigned *code, unsigned int *extword, int *codebits, int *extbits ); +! void III_FlushBitstream(void); + + int abs_and_sign( int *x ); /* returns signx and changes *x to abs(*x) */ + +diff -r -c -N encoder/l3psy.c lame3.70/l3psy.c +*** encoder/l3psy.c Wed Jan 22 02:43:15 1997 +--- lame3.70/l3psy.c Wed Dec 31 17:00:00 1969 +*************** +*** 1,1033 **** +- /********************************************************************** +- * ISO MPEG Audio Subgroup Software Simulation Group (1996) +- * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- * +- * $Id: l3psy.c,v 1.2 1997/01/19 22:28:29 rowlands Exp $ +- * +- * $Log: l3psy.c,v $ +- * Revision 1.2 1997/01/19 22:28:29 rowlands +- * Layer 3 bug fixes from Seymour Shlien +- * +- * Revision 1.1 1996/02/14 04:04:23 rowlands +- * Initial revision +- * +- * Received from Mike Coleman +- **********************************************************************/ +- /********************************************************************** +- * date programmers comment * +- * 2/25/91 Davis Pan start of version 1.0 records * +- * 5/10/91 W. Joseph Carter Ported to Macintosh and Unix. * +- * 7/10/91 Earle Jennings Ported to MsDos. * +- * replace of floats with FLOAT * +- * 2/11/92 W. Joseph Carter Fixed mem_alloc() arg for "absthr". * +- * 3/16/92 Masahiro Iwadare Modification for Layer III * +- * 17/4/93 Masahiro Iwadare Updated for IS Modification * +- **********************************************************************/ +- +- #include "common.h" +- #include "encoder.h" +- #include "l3psy.h" +- #include "l3side.h" +- #include <assert.h> +- +- #define maximum(x,y) ( (x>y) ? x : y ) +- #define minimum(x,y) ( (x<y) ? x : y ) +- +- void L3para_read( double sfreq, int numlines[CBANDS], int partition_l[HBLKSIZE], +- double minval[CBANDS], double qthr_l[CBANDS], double norm_l[CBANDS], +- double s3_l[CBANDS][CBANDS], int partition_s[HBLKSIZE_s], double qthr_s[CBANDS_s], +- double norm_s[CBANDS_s], double SNR_s[CBANDS_s], +- int cbw_l[SBMAX_l], int bu_l[SBMAX_l], int bo_l[SBMAX_l], +- double w1_l[SBMAX_l], double w2_l[SBMAX_l], +- int cbw_s[SBMAX_s], int bu_s[SBMAX_s], int bo_s[SBMAX_s], +- double w1_s[SBMAX_s], double w2_s[SBMAX_s] ); +- +- void L3psycho_anal( short int *buffer, short int savebuf[1344], int chn, int lay, FLOAT snr32[32], +- double sfreq, double ratio_d[21], double ratio_ds[12][3], +- double *pe, gr_info *cod_info ) +- { +- static double ratio[2][21]; +- static double ratio_s[2][12][3]; +- int blocktype; +- unsigned int b, i, j, k; +- double r_prime, phi_prime; /* not FLOAT */ +- FLOAT freq_mult, bval_lo, min_thres, sum_energy; +- double tb, temp1,temp2,temp3; +- +- /* nint(); Layer III */ +- double thr[CBANDS]; +- +- /* The static variables "r", "phi_sav", "new", "old" and "oldest" have */ +- /* to be remembered for the unpredictability measure. For "r" and */ +- /* "phi_sav", the first index from the left is the channel select and */ +- /* the second index is the "age" of the data. */ +- +- +- static FLOAT window_s[BLKSIZE_s] ; +- static int new = 0, old = 1, oldest = 0; +- static int init = 0, flush, sync_flush, syncsize, sfreq_idx; +- static double cw[HBLKSIZE], eb[CBANDS]; +- static double ctb[CBANDS]; +- static double SNR_l[CBANDS], SNR_s[CBANDS_s]; +- static int init_L3; +- static double minval[CBANDS],qthr_l[CBANDS],norm_l[CBANDS]; +- static double qthr_s[CBANDS_s],norm_s[CBANDS_s]; +- static double nb_1[2][CBANDS], nb_2[2][CBANDS]; +- static double s3_l[CBANDS][CBANDS]; /* s3_s[CBANDS_s][CBANDS_s]; */ +- +- /* Scale Factor Bands */ +- static int cbw_l[SBMAX_l],bu_l[SBMAX_l],bo_l[SBMAX_l] ; +- static int cbw_s[SBMAX_s],bu_s[SBMAX_s],bo_s[SBMAX_s] ; +- static double w1_l[SBMAX_l], w2_l[SBMAX_l]; +- static double w1_s[SBMAX_s], w2_s[SBMAX_s]; +- static double en[SBMAX_l], thm[SBMAX_l] ; +- static int blocktype_old[2] ; +- int sb,sblock; +- static int partition_l[HBLKSIZE],partition_s[HBLKSIZE_s]; +- +- +- /* The following static variables are constants. */ +- +- static double nmt = 5.5; +- +- static FLOAT crit_band[27] = {0, 100, 200, 300, 400, 510, 630, 770, +- 920, 1080, 1270,1480,1720,2000,2320, 2700, +- 3150, 3700, 4400,5300,6400,7700,9500,12000, +- 15500,25000,30000}; +- +- static FLOAT bmax[27] = {20.0, 20.0, 20.0, 20.0, 20.0, 17.0, 15.0, +- 10.0, 7.0, 4.4, 4.5, 4.5, 4.5, 4.5, +- 4.5, 4.5, 4.5, 4.5, 4.5, 4.5, 4.5, +- 4.5, 4.5, 4.5, 3.5, 3.5, 3.5}; +- +- /* The following pointer variables point to large areas of memory */ +- /* dynamically allocated by the mem_alloc() function. Dynamic memory */ +- /* allocation is used in order to avoid stack frame or data area */ +- /* overflow errors that otherwise would have occurred at compile time */ +- /* on the Macintosh computer. */ +- +- FLOAT *grouped_c, *grouped_e, *nb, *cb, *ecb, *bc; +- FLOAT *wsamp_r, *wsamp_i, *phi, *energy; +- static FLOAT energy_s[3][256]; +- static FLOAT phi_s[3][256] ; /* 256 samples not 129 */ +- FLOAT *c, *fthr; +- F32 *snrtmp; +- +- static int *numlines ; +- static int *partition; +- static FLOAT *cbval, *rnorm; +- static FLOAT *window; +- static FLOAT *absthr; +- static double *tmn; +- static FCB *s; +- static FHBLK *lthr; +- static F2HBLK *r, *phi_sav; +- +- /* These dynamic memory allocations simulate "automatic" variables */ +- /* placed on the stack. For each mem_alloc() call here, there must be */ +- /* a corresponding mem_free() call at the end of this function. */ +- +- grouped_c = (FLOAT *) mem_alloc(sizeof(FCB), "grouped_c"); +- grouped_e = (FLOAT *) mem_alloc(sizeof(FCB), "grouped_e"); +- nb = (FLOAT *) mem_alloc(sizeof(FCB), "nb"); +- cb = (FLOAT *) mem_alloc(sizeof(FCB), "cb"); +- ecb = (FLOAT *) mem_alloc(sizeof(FCB), "ecb"); +- bc = (FLOAT *) mem_alloc(sizeof(FCB), "bc"); +- wsamp_r = (FLOAT *) mem_alloc(sizeof(FBLK), "wsamp_r"); +- wsamp_i = (FLOAT *) mem_alloc(sizeof(FBLK), "wsamp_i"); +- phi = (FLOAT *) mem_alloc(sizeof(FBLK), "phi"); +- energy = (FLOAT *) mem_alloc(sizeof(FBLK), "energy"); +- c = (FLOAT *) mem_alloc(sizeof(FHBLK), "c"); +- fthr = (FLOAT *) mem_alloc(sizeof(FHBLK), "fthr"); +- snrtmp = (F32 *) mem_alloc(sizeof(F2_32), "snrtmp"); +- +- assert( lay == 3 ); +- if(init==0){ +- +- /* These dynamic memory allocations simulate "static" variables placed */ +- /* in the data space. Each mem_alloc() call here occurs only once at */ +- /* initialization time. The mem_free() function must not be called. */ +- numlines = (int *) mem_alloc(sizeof(ICB), "numlines"); +- partition = (int *) mem_alloc(sizeof(IHBLK), "partition"); +- cbval = (FLOAT *) mem_alloc(sizeof(FCB), "cbval"); +- rnorm = (FLOAT *) mem_alloc(sizeof(FCB), "rnorm"); +- window = (FLOAT *) mem_alloc(sizeof(FBLK), "window"); +- absthr = (FLOAT *) mem_alloc(sizeof(FHBLK), "absthr"); +- tmn = (double *) mem_alloc(sizeof(DCB), "tmn"); +- s = (FCB *) mem_alloc(sizeof(FCBCB), "s"); +- lthr = (FHBLK *) mem_alloc(sizeof(F2HBLK), "lthr"); +- r = (F2HBLK *) mem_alloc(sizeof(F22HBLK), "r"); +- phi_sav = (F2HBLK *) mem_alloc(sizeof(F22HBLK), "phi_sav"); +- +- /*#if 0 */ +- i = sfreq + 0.5; +- switch(i){ +- case 32000: sfreq_idx = 0; break; +- case 44100: sfreq_idx = 1; break; +- case 48000: sfreq_idx = 2; break; +- default: printf("error, invalid sampling frequency: %d Hz\n",i); +- exit(-1); +- } +- printf("absthr[][] sampling frequency index: %d\n",sfreq_idx); +- read_absthr(absthr, sfreq_idx); +- switch(lay){ +- case 1: sync_flush=576; flush=384; syncsize=1024; break; +- case 2: sync_flush=480; flush=576; syncsize=1056; break; +- case 3: sync_flush=768; flush=576; syncsize=1344; break; +- default: printf("Bad lay value:(%d)",lay); exit(-1); break; +- } +- /* #endif */ +- +- /* calculate HANN window coefficients */ +- /* for(i=0;i<BLKSIZE;i++) window[i] =0.5*(1-cos(2.0*PI*i/(BLKSIZE-1.0)));*/ +- for(i=0;i<BLKSIZE;i++) window[i] =0.5*(1-cos(2.0*PI*(i-0.5)/BLKSIZE)); +- for(i=0;i<BLKSIZE_s;i++)window_s[i]=0.5*(1-cos(2.0*PI*(i-0.5)/BLKSIZE_s)); +- /* reset states used in unpredictability measure */ +- for(i=0;i<HBLKSIZE;i++){ +- r[0][0][i]=r[1][0][i]=r[0][1][i]=r[1][1][i]=0; +- phi_sav[0][0][i]=phi_sav[1][0][i]=0; +- phi_sav[0][1][i]=phi_sav[1][1][i]=0; +- lthr[0][i] = 60802371420160.0; +- lthr[1][i] = 60802371420160.0; +- } +- /***************************************************************************** +- * Initialization: Compute the following constants for use later * +- * partition[HBLKSIZE] = the partition number associated with each * +- * frequency line * +- * cbval[CBANDS] = the center (average) bark value of each * +- * partition * +- * numlines[CBANDS] = the number of frequency lines in each partition * +- * tmn[CBANDS] = tone masking noise * +- *****************************************************************************/ +- /* compute fft frequency multiplicand */ +- freq_mult = sfreq/BLKSIZE; +- +- /* calculate fft frequency, then bval of each line (use fthr[] as tmp storage)*/ +- for(i=0;i<HBLKSIZE;i++){ +- temp1 = i*freq_mult; +- j = 1; +- while(temp1>crit_band[j])j++; +- fthr[i]=j-1+(temp1-crit_band[j-1])/(crit_band[j]-crit_band[j-1]); +- } +- partition[0] = 0; +- /* temp2 is the counter of the number of frequency lines in each partition */ +- temp2 = 1; +- cbval[0]=fthr[0]; +- bval_lo=fthr[0]; +- for(i=1;i<HBLKSIZE;i++){ +- if((fthr[i]-bval_lo)>0.33){ +- partition[i]=partition[i-1]+1; +- cbval[partition[i-1]] = cbval[partition[i-1]]/temp2; +- cbval[partition[i]] = fthr[i]; +- bval_lo = fthr[i]; +- numlines[partition[i-1]] = temp2; +- temp2 = 1; +- } +- else { +- partition[i]=partition[i-1]; +- cbval[partition[i]] += fthr[i]; +- temp2++; +- } +- } +- numlines[partition[i-1]] = temp2; +- cbval[partition[i-1]] = cbval[partition[i-1]]/temp2; +- +- /************************************************************************ +- * Now compute the spreading function, s[j][i], the value of the spread-* +- * ing function, centered at band j, for band i, store for later use * +- ************************************************************************/ +- for(j=0;j<CBANDS;j++){ +- for(i=0;i<CBANDS;i++){ +- temp1 = (cbval[i] - cbval[j])*1.05; +- if(temp1>=0.5 && temp1<=2.5){ +- temp2 = temp1 - 0.5; +- temp2 = 8.0 * (temp2*temp2 - 2.0 * temp2); +- } +- else temp2 = 0; +- temp1 += 0.474; +- temp3 = 15.811389+7.5*temp1-17.5*sqrt((double) (1.0+temp1*temp1)); +- if(temp3 <= -100) s[i][j] = 0; +- else { +- temp3 = (temp2 + temp3)*LN_TO_LOG10; +- s[i][j] = exp(temp3); +- } +- } +- } +- +- /* Calculate Tone Masking Noise values */ +- for(j=0;j<CBANDS;j++){ +- temp1 = 15.5 + cbval[j]; +- tmn[j] = (temp1>24.5) ? temp1 : 24.5; +- /* Calculate normalization factors for the net spreading functions */ +- rnorm[j] = 0; +- for(i=0;i<CBANDS;i++){ +- rnorm[j] += s[j][i]; +- } +- } +- init++; +- } +- +- /************************* End of Initialization *****************************/ +- switch(lay) { +- case 1: +- case 2: +- for ( i=0; i<lay; i++) +- { +- /***************************************************************************** +- * Net offset is 480 samples (1056-576) for layer 2; this is because one must* +- * stagger input data by 256 samples to synchronize psychoacoustic model with* +- * filter bank outputs, then stagger so that center of 1024 FFT window lines * +- * up with center of 576 "new" audio samples. * +- * * +- * For layer 1, the input data still needs to be staggered by 256 samples, * +- * then it must be staggered again so that the 384 "new" samples are centered* +- * in the 1024 FFT window. The net offset is then 576 and you need 448 "new"* +- * samples for each iteration to keep the 384 samples of interest centered * +- *****************************************************************************/ +- for (j=0; j<syncsize; j++) +- { +- if (j < (sync_flush) ) +- savebuf[j] = savebuf[j+flush]; +- else +- savebuf[j] = *buffer++; +- +- /**window data with HANN window***********************************************/ +- if (j<BLKSIZE) +- { +- wsamp_r[j] = window[j]*((FLOAT) savebuf[j]); +- wsamp_i[j] = 0; +- } +- } +- /**Compute FFT****************************************************************/ +- fft(wsamp_r,wsamp_i,energy,phi,1024); +- /***************************************************************************** +- * calculate the unpredictability measure, given energy[f] and phi[f] * +- *****************************************************************************/ +- for(j=0; j<HBLKSIZE; j++){ +- r_prime = 2.0 * r[chn][old][j] - r[chn][oldest][j]; +- phi_prime = 2.0 * phi_sav[chn][old][j] - phi_sav[chn][oldest][j]; +- r[chn][new][j] = sqrt((double) energy[j]); +- phi_sav[chn][new][j] = phi[j]; +- temp1 = r[chn][new][j] * cos((double) phi[j]) +- - r_prime * cos(phi_prime); +- temp2=r[chn][new][j] * sin((double) phi[j]) +- - r_prime * sin(phi_prime); +- temp3=r[chn][new][j] + fabs(r_prime); +- if(temp3 != 0)c[j]=sqrt(temp1*temp1+temp2*temp2)/temp3; +- else c[j] = 0; +- } +- /*only update data "age" pointers after you are done with the second channel */ +- /*for layer 1 computations, for the layer 2 double computations, the pointers*/ +- /*are reset automatically on the second pass */ +- if(lay==2 || chn==1){ +- if(new==0){new = 1; oldest = 1;} +- else {new = 0; oldest = 0;} +- if(old==0)old = 1; else old = 0; +- } +- /***************************************************************************** +- * Calculate the grouped, energy-weighted, unpredictability measure, * +- * grouped_c[], and the grouped energy. grouped_e[] * +- *****************************************************************************/ +- for(j=1;j<CBANDS;j++){ +- grouped_e[j] = 0; +- grouped_c[j] = 0; +- } +- grouped_e[0] = energy[0]; +- grouped_c[0] = energy[0]*c[0]; +- for(j=1;j<HBLKSIZE;j++){ +- grouped_e[partition[j]] += energy[j]; +- grouped_c[partition[j]] += energy[j]*c[j]; +- } +- /***************************************************************************** +- * convolve the grouped energy-weighted unpredictability measure * +- * and the grouped energy with the spreading function, s[j][k] * +- *****************************************************************************/ +- for(j=0;j<CBANDS;j++){ +- ecb[j] = 0; +- cb[j] = 0; +- for(k=0;k<CBANDS;k++){ +- if(s[j][k] != 0.0){ +- ecb[j] += s[j][k]*grouped_e[k]; +- cb[j] += s[j][k]*grouped_c[k]; +- } +- } +- if(ecb[j] !=0)cb[j] = cb[j]/ecb[j]; +- else cb[j] = 0; +- } +- /***************************************************************************** +- * Calculate the required SNR for each of the frequency partitions * +- * this whole section can be accomplished by a table lookup * +- *****************************************************************************/ +- for(j=0;j<CBANDS;j++){ +- if(cb[j]<.05)cb[j]=0.05; +- else if(cb[j]>.5)cb[j]=0.5; +- /* tb = -0.434294482*log((double) cb[j])-0.301029996; */ +- tb = -0.43 *log((double) cb[j]) - 0.29 ; +- if(tb<0.0) tb=0.0; else if(tb>1.0) tb=1.0; +- bc[j] = tmn[j]*tb + nmt*(1.0-tb); +- k = cbval[j] + 0.5; +- bc[j] = (bc[j] > bmax[k]) ? bc[j] : bmax[k]; +- bc[j] = exp((double) -bc[j]*LN_TO_LOG10); +- } +- /***************************************************************************** +- * Calculate the permissible noise energy level in each of the frequency * +- * partitions. Include absolute threshold and pre-echo controls * +- * this whole section can be accomplished by a table lookup * +- *****************************************************************************/ +- for(j=0;j<CBANDS;j++) +- if(rnorm[j] && numlines[j]) +- nb[j] = ecb[j]*bc[j]/(rnorm[j]*numlines[j]); +- else nb[j] = 0; +- for(j=0;j<HBLKSIZE;j++){ +- temp1=nb[partition[j]]; /* preliminary threshold */ +- temp1=(temp1>absthr[j])?temp1:absthr[j]; +- /*do not use pre-echo control for layer 2 because it may do bad things to the*/ +- /* MUSICAM bit allocation algorithm */ +- if(lay==1){ +- fthr[j] = (temp1 < lthr[chn][j]) ? temp1 : lthr[chn][j]; +- temp2 = temp1 * 0.00316; +- fthr[j] = (temp2 > fthr[j]) ? temp2 : fthr[j]; +- } +- else fthr[j] = temp1; +- lthr[chn][j] = LXMIN*temp1; +- } +- /***************************************************************************** +- * Translate the 512 threshold values to the 32 filter bands of the coder * +- *****************************************************************************/ +- for(j=0;j<193;j += 16){ +- min_thres = fthr[j]; +- sum_energy = energy[j]; +- for(k=1;k<17;k++){ +- if(min_thres>fthr[j+k]) min_thres = fthr[j+k]; +- sum_energy += energy[j+k]; +- } +- snrtmp[i][j/16] = sum_energy/(min_thres * 17.0); +- snrtmp[i][j/16] = 4.342944819 * log((double)snrtmp[i][j/16]); +- } +- for(j=208;j<(HBLKSIZE-1);j += 16){ +- min_thres = 0.0; +- sum_energy = 0.0; +- for(k=0;k<17;k++){ +- min_thres += fthr[j+k]; +- sum_energy += energy[j+k]; +- } +- snrtmp[i][j/16] = sum_energy/min_thres; +- snrtmp[i][j/16] = 4.342944819 * log((double)snrtmp[i][j/16]); +- } +- /***************************************************************************** +- * End of Psychoacuostic calculation loop * +- *****************************************************************************/ +- } +- for(i=0; i<32; i++){ +- if(lay==2) /* if(lay==2 && chn==2) MI */ +- snr32[i]=(snrtmp[0][i]>snrtmp[1][i])?snrtmp[0][i]:snrtmp[1][i]; +- else snr32[i]=snrtmp[0][i]; +- } +- break; +- +- /*************************************************************************/ +- /** LAYER 3 */ +- /*************************************************************************/ +- +- case 3: +- if ( init_L3 == 0 ) +- { +- L3para_read( sfreq,numlines,partition_l,minval,qthr_l,norm_l,s3_l, +- partition_s,qthr_s,norm_s,SNR_s, +- cbw_l,bu_l,bo_l,w1_l,w2_l, cbw_s,bu_s,bo_s,w1_s,w2_s ); +- init_L3 ++ ; +- } +- +- for ( j = 0; j < 21; j++ ) +- ratio_d[j] = ratio[chn][j]; +- for ( j = 0; j < 12; j++ ) +- for ( i = 0; i < 3; i++ ) +- ratio_ds[j][i] = ratio_s[chn][j][i]; +- +- if ( chn == 0 ) +- if ( new == 0 ) +- { +- new = 1; +- old = 0; +- oldest = 1; +- } +- else +- { +- new = 0; +- old = 1; +- oldest = 0; +- } +- +- +- /********************************************************************** +- * Delay signal by sync_flush=768 samples * +- **********************************************************************/ +- +- for ( j = 0; j < sync_flush; j++ ) /* for long window samples */ +- savebuf[j] = savebuf[j+flush]; +- +- for ( j = sync_flush; j < syncsize; j++ ) +- savebuf[j] = *buffer++; +- +- for ( j = 0; j < BLKSIZE; j++ ) +- { /**window data with HANN window**/ +- wsamp_r[j] = window[j] * savebuf[j]; +- wsamp_i[j] = 0.0; +- } +- +- +- /********************************************************************** +- * compute unpredicatability of first six spectral lines * +- **********************************************************************/ +- +- fft( wsamp_r, wsamp_i, energy, phi, 1024 ); /**long FFT**/ +- for ( j = 0; j < 6; j++ ) +- { /* calculate unpredictability measure cw */ +- r_prime = 2.0 * r[chn][old][j] - r[chn][oldest][j]; +- phi_prime = 2.0 * phi_sav[chn][old][j]-phi_sav[chn][oldest][j]; +- r[chn][new][j] = sqrt((double) energy[j]); +- phi_sav[chn][new][j] = phi[j]; +- temp1 = r[chn][new][j] * cos((double) phi[j]) +- - r_prime * cos(phi_prime); +- temp2 = r[chn][new][j] * sin((double) phi[j]) +- - r_prime * sin(phi_prime); +- temp3=r[chn][new][j] + fabs(r_prime); +- +- if ( temp3 != 0.0 ) +- cw[j] = sqrt( temp1*temp1+temp2*temp2 ) / temp3; +- else +- cw[j] = 0; +- } +- +- +- /********************************************************************** +- * compute unpredicatibility of next 200 spectral lines * +- **********************************************************************/ +- for ( sblock = 0; sblock < 3; sblock++ ) +- { /**window data with HANN window**/ +- for ( j = 0, k = 128 * (2 + sblock); j < 256; j++, k++ ) +- { +- wsamp_r[j] = window_s[j]* savebuf[k]; +- wsamp_i[j] = 0.0; +- } /* short FFT*/ +- +- fft( wsamp_r, wsamp_i, &energy_s[sblock][0], &phi_s[sblock][0], 256 ); +- } +- +- sblock = 1; +- +- for ( j = 6; j < 206; j += 4 ) +- {/* calculate unpredictability measure cw */ +- double r2, phi2, temp1, temp2, temp3; +- +- k = (j+2) / 4; +- r_prime = 2.0 * sqrt((double) energy_s[0][k]) +- - sqrt((double) energy_s[2][k]); +- phi_prime = 2.0 * phi_s[0][k] - phi_s[2][k]; +- r2 = sqrt((double) energy_s[1][k]); +- phi2 = phi_s[1][k]; +- temp1 = r2 * cos( phi2 ) - r_prime * cos( phi_prime ); +- temp2 = r2 * sin( phi2 ) - r_prime * sin( phi_prime ); +- temp3 = r2 + fabs( r_prime ); +- if ( temp3 != 0.0 ) +- cw[j] = sqrt( temp1 * temp1 + temp2 * temp2 ) / temp3; +- else +- cw[j] = 0.0; +- cw[j+1] = cw[j+2] = cw[j+3] = cw[j]; +- } +- +- +- /********************************************************************** +- * Set unpredicatiblility of remaining spectral lines to 0.4 * +- **********************************************************************/ +- for ( j = 206; j < HBLKSIZE; j++ ) +- cw[j] = 0.4; +- +- +- +- /********************************************************************** +- * Calculate the energy and the unpredictability in the threshold * +- * calculation partitions * +- **********************************************************************/ +- +- for ( b = 0; b < CBANDS; b++ ) +- { +- eb[b] = 0.0; +- cb[b] = 0.0; +- } +- for ( j = 0; j < HBLKSIZE; j++ ) +- { +- int tp = partition_l[j]; +- if ( tp >= 0 ) +- { +- eb[tp] += energy[j]; +- cb[tp] += cw[j] * energy[j]; +- } +- } +- +- +- /********************************************************************** +- * convolve the partitioned energy and unpredictability * +- * with the spreading function, s3_l[b][k] * +- ******************************************************************** */ +- +- for ( b = 0; b < CBANDS; b++ ) +- { +- ecb[b] = 0.0; +- ctb[b] = 0.0; +- } +- for ( b = 0;b < CBANDS; b++ ) +- { +- for ( k = 0; k < CBANDS; k++ ) +- { +- ecb[b] += s3_l[b][k] * eb[k]; /* sprdngf for Layer III */ +- ctb[b] += s3_l[b][k] * cb[k]; +- } +- } +- +- /* calculate the tonality of each threshold calculation partition */ +- /* calculate the SNR in each threshhold calculation partition */ +- +- for ( b = 0; b < CBANDS; b++ ) +- { +- double cbb,tbb; +- if (ecb[b] != 0.0 ) +- { +- cbb = ctb[b]/ecb[b]; +- if (cbb <0.01) cbb = 0.01; +- cbb = log( cbb); +- } +- else +- cbb = 0.0 ; +- tbb = -0.299 - 0.43*cbb; /* conv1=-0.299, conv2=-0.43 */ +- tbb = minimum( 1.0, maximum( 0.0, tbb) ) ; /* 0<tbb<1 */ +- SNR_l[b] = maximum( minval[b], 29.0*tbb+6.0*(1.0-tbb) ); +- } /* TMN=29.0,NMT=6.0 for all calculation partitions */ +- +- for ( b = 0; b < CBANDS; b++ ) /* calculate the threshold for each partition */ +- nb[b] = ecb[b] * norm_l[b] * exp( -SNR_l[b] * LN_TO_LOG10 ); +- +- for ( b = 0; b < CBANDS; b++ ) +- { /* pre-echo control */ +- double temp_1; /* BUG of IS */ +- temp_1 = minimum( nb[b], minimum(2.0*nb_1[chn][b],16.0*nb_2[chn][b]) ); +- thr[b] = maximum( qthr_l[b], temp_1 );/* rpelev=2.0, rpelev2=16.0 */ +- nb_2[chn][b] = nb_1[chn][b]; +- nb_1[chn][b] = nb[b]; +- } +- +- +- *pe = 0.0; /* calculate percetual entropy */ +- for ( b = 0; b < CBANDS; b++ ) +- { +- double tp ; +- tp = minimum( 0.0, log((thr[b]+1.0) / (eb[b]+1.0) ) ) ; /*not log*/ +- *pe -= numlines[b] * tp ; +- } /* thr[b] -> thr[b]+1.0 : for non sound portition */ +- +- #define switch_pe 1800 +- blocktype = NORM_TYPE; +- +- +- if ( *pe < switch_pe ) +- { /* no attack : use long blocks */ +- switch( blocktype_old[chn] ) +- { +- case NORM_TYPE: +- case STOP_TYPE: +- blocktype = NORM_TYPE; +- break; +- +- case SHORT_TYPE: +- blocktype = STOP_TYPE; +- break; +- +- case START_TYPE: +- fprintf( stderr, "Error in block selecting\n" ); +- abort(); +- break; /* problem */ +- } +- +- /* threshold calculation (part 2) */ +- for ( sb = 0; sb < SBMAX_l; sb++ ) +- { +- en[sb] = w1_l[sb] * eb[bu_l[sb]] + w2_l[sb] * eb[bo_l[sb]]; +- thm[sb] = w1_l[sb] *thr[bu_l[sb]] + w2_l[sb] * thr[bo_l[sb]]; +- for ( b = bu_l[sb]+1; b < bo_l[sb]; b++ ) +- { +- en[sb] += eb[b]; +- thm[sb] += thr[b]; +- } +- if ( en[sb] != 0.0 ) +- ratio[chn][sb] = thm[sb]/en[sb]; +- else +- ratio[chn][sb] = 0.0; +- } +- } +- else +- { +- /* attack : use short blocks */ +- blocktype = SHORT_TYPE; +- +- if ( blocktype_old[chn] == NORM_TYPE ) +- blocktype_old[chn] = START_TYPE; +- if ( blocktype_old[chn] == STOP_TYPE ) +- blocktype_old[chn] = SHORT_TYPE ; +- +- /* threshold calculation for short blocks */ +- +- for ( sblock = 0; sblock < 3; sblock++ ) +- { +- for ( b = 0; b < CBANDS_s; b++ ) +- { +- eb[b] = 0.0; +- ecb[b] = 0.0; +- } +- for ( j = 0; j < HBLKSIZE_s; j++ ) +- eb[partition_s[j]] += energy_s[sblock][j]; +- for ( b = 0; b < CBANDS_s; b++ ) +- for ( k = 0; k < CBANDS_s; k++ ) +- ecb[b] += s3_l[b][k] * eb[k]; +- for ( b = 0; b < CBANDS_s; b++ ) +- { +- nb[b] = ecb[b] * norm_l[b] * exp( (double) SNR_s[b] * LN_TO_LOG10 ); +- thr[b] = maximum (qthr_s[b],nb[b]); +- } +- for ( sb = 0; sb < SBMAX_s; sb++ ) +- { +- en[sb] = w1_s[sb] * eb[bu_s[sb]] + w2_s[sb] * eb[bo_s[sb]]; +- thm[sb] = w1_s[sb] *thr[bu_s[sb]] + w2_s[sb] * thr[bo_s[sb]]; +- for ( b = bu_s[sb]+1; b < bo_s[sb]; b++ ) +- { +- en[sb] += eb[b]; +- thm[sb] += thr[b]; +- } +- if ( en[sb] != 0.0 ) +- ratio_s[chn][sb][sblock] = thm[sb]/en[sb]; +- else +- ratio_s[chn][sb][sblock] = 0.0; +- } +- } +- } +- +- cod_info->block_type = blocktype_old[chn]; +- blocktype_old[chn] = blocktype; +- +- if ( cod_info->block_type == NORM_TYPE ) +- cod_info->window_switching_flag = 0; +- else +- cod_info->window_switching_flag = 1; +- cod_info->mixed_block_flag = 0; +- break; +- +- +- default: +- printf("error, invalid MPEG/audio coding layer: %d\n",lay); +- } +- +- /* These mem_free() calls must correspond with the mem_alloc() calls */ +- /* used at the beginning of this function to simulate "automatic" */ +- /* variables placed on the stack. */ +- +- mem_free((void **) &grouped_c); +- mem_free((void **) &grouped_e); +- mem_free((void **) &nb); +- mem_free((void **) &cb); +- mem_free((void **) &ecb); +- mem_free((void **) &bc); +- mem_free((void **) &wsamp_r); +- mem_free((void **) &wsamp_i); +- mem_free((void **) &phi); +- mem_free((void **) &energy); +- mem_free((void **) &c); +- mem_free((void **) &fthr); +- mem_free((void **) &snrtmp); +- } +- #ifdef DEBUG +- #undef DEBUG +- #endif +- +- +- void L3para_read(double sfreq, int *numlines, int *partition_l, double *minval, double *qthr_l, double *norm_l, double (*s3_l)[63], int *partition_s, double *qthr_s, double *norm_s, double *SNR, int *cbw_l, int *bu_l, int *bo_l, double *w1_l, double *w2_l, int *cbw_s, int *bu_s, int *bo_s, double *w1_s, double *w2_s) +- { +- double freq_tp; +- static double bval_l[CBANDS], bval_s[CBANDS]; +- int cbmax, cbmax_tp; +- static double s3_s[CBANDS][CBANDS]; +- +- FILE *fin; +- char tp[256]; +- int sbmax ; +- int i,j,k,k2,loop, part_max ; +- +- fin = OpenTableFile( "psy_data" ); +- if (fin == NULL) +- exit( 1 ); +- +- /* Read long block data */ +- +- for(loop=0;loop<6;loop++) +- { +- fscanf(fin,"freq = %lf partition = %d\n",&freq_tp,&cbmax_tp); +- cbmax_tp++; +- #ifdef DEBUG +- printf("freq = %f partition = %d\n",freq_tp,cbmax); +- #endif +- +- if (sfreq == freq_tp ) +- { +- cbmax = cbmax_tp; +- for(i=0,k2=0;i<cbmax_tp;i++) +- { +- fscanf(fin, +- "No=%d #lines=%d minval=%lf qthr=%lf norm=%lf bval=%lf\n", +- &j,&numlines[i],&minval[i],&qthr_l[i],&norm_l[i],&bval_l[i]); +- if (j!=i) +- { printf("please check \"psy_data\""); +- exit(-1); +- } +- for(k=0;k<numlines[i];k++) +- partition_l[k2++] = i ; +- #ifdef DEBUG +- printf("No=%2d #lines=%2d minval=%4.1f qthr=%8.3f norm=%5.3f bval=%8.3f\n", +- i,numlines[i],minval[i],qthr_l[i],norm_l[i],bval_l[i] ); +- #endif +- } +- } +- else +- { +- for(j=0;j<cbmax_tp;j++) +- { +- fgets(tp,255,fin); +- sscanf(tp,"No=%d %s\n",&i,tp); +- if (j!=i) +- { printf("please check \"psy_data.\"\n"); +- exit(-1); +- } +- } +- } +- } +- +- /************************************************************************ +- * Now compute the spreading function, s[j][i], the value of the spread-* +- * ing function, centered at band j, for band i, store for later use * +- ************************************************************************/ +- #ifdef DEBUG +- printf("freq = %f\n",sfreq); +- #endif +- part_max = cbmax ; +- for(i=0;i<part_max;i++) +- { +- double tempx,x,tempy,temp; +- for(j=0;j<part_max;j++) +- { +- tempx = (bval_l[i] - bval_l[j])*1.05; +- if (j>=i) tempx = (bval_l[i] - bval_l[j])*3.0; +- else tempx = (bval_l[i] - bval_l[j])*1.5; +- /* if (j>=i) tempx = (bval_l[j] - bval_l[i])*3.0; +- else tempx = (bval_l[j] - bval_l[i])*1.5; */ +- if(tempx>=0.5 && tempx<=2.5) +- { +- temp = tempx - 0.5; +- x = 8.0 * (temp*temp - 2.0 * temp); +- } +- else x = 0.0; +- tempx += 0.474; +- tempy = 15.811389 + 7.5*tempx - 17.5*sqrt(1.0+tempx*tempx); +- if (tempy <= -60.0) s3_l[i][j] = 0.0; +- else s3_l[i][j] = exp( (x + tempy)*LN_TO_LOG10 ); +- #ifdef DEBUG_S3 +- printf("s[%d][%d]=%f\n",i,j,s3_l[i][j]); +- #endif +- +- #ifdef DEBUGP +- printf("j=%d i=%d tempy=%f s[i][j]=%f \n",i,j,tempy,s[i][j]); +- minval[i] = bmax[j-1]; +- printf("minval[%d] = %f, j-1=%d %f\n",i, minval[i] , j,fthr[i]) ; +- #endif +- } +- } +- +- +- /* Read short block data */ +- +- for(loop=0;loop<6;loop++) +- { +- fscanf(fin,"freq = %lf partition = %d\n",&freq_tp,&cbmax_tp); +- cbmax_tp++; +- #ifdef DEBUG +- printf("freq = %f partition = %d\n",freq_tp,cbmax); +- #endif +- +- if (sfreq == freq_tp ) +- { +- cbmax = cbmax_tp; +- for(i=0,k2=0;i<cbmax_tp;i++) +- { +- fscanf(fin, +- "No=%d #lines=%d qthr=%lf norm=%lf SNR=%lf bval=%lf\n", +- &j,&numlines[i],&qthr_s[i],&norm_s[i],&SNR[i],&bval_s[i]); +- if (j!=i) +- { printf("please check \"psy_data\""); +- exit(-1); +- } +- for(k=0;k<numlines[i];k++) +- partition_s[k2++] = i ; +- #ifdef DEBUG +- printf("No=%2d #lines=%2d qthr=%8.3f norm=%5.3f SNR=%6.3f bval=%8.3f\n", +- i,numlines[i],qthr_s[i],norm_s[i],SNR[i],bval_s[i] ); +- #endif +- } +- } +- else +- { +- for(j=0;j<cbmax_tp;j++) +- { +- fgets(tp,255,fin); +- sscanf(tp,"No=%d %s\n",&i,tp); +- if (j!=i) +- { printf("please check \"psy_data.\"\n"); +- exit(-1); +- } +- } +- } +- } +- +- /************************************************************************ +- * Now compute the spreading function, s[j][i], the value of the spread-* +- * ing function, centered at band j, for band i, store for later use * +- ************************************************************************/ +- #ifdef DEBUG_S3 +- fpp=fopen("s3_s","w"); +- #endif +- part_max = cbmax ; +- for(i=0;i<part_max;i++) +- { +- double tempx,x,tempy,temp; +- for(j=0;j<part_max;j++) +- { +- tempx = (bval_s[i] - bval_s[j])*1.05; +- if (j>=i) tempx = (bval_s[i] - bval_s[j])*3.0; +- else tempx = (bval_s[i] - bval_s[j])*1.5; +- if(tempx>=0.5 && tempx<=2.5) +- { +- temp = tempx - 0.5; +- x = 8.0 * (temp*temp - 2.0 * temp); +- } +- else x = 0.0; +- tempx += 0.474; +- tempy = 15.811389 + 7.5*tempx - 17.5*sqrt(1.0+tempx*tempx); +- if (tempy <= -60.0) s3_s[i][j] = 0.0; +- else s3_s[i][j] = exp( (x + tempy)*LN_TO_LOG10 ); +- #ifdef DEBUG_S3 +- fprintf(fpp,"s3_s[%d][%d]=%f\n",i,j,s3_s[i][j]); +- #endif +- #ifdef DEBUGP +- printf("j=%d i=%d tempy=%f s[i][j]=%f \n",i,j,tempy,s[i][j]); +- minval[i] = bmax[j-1]; +- printf("minval[%d] = %f, j-1=%d %f\n",i, minval[i] , j,fthr[i]) ; +- #endif +- } +- } +- #ifdef DEBUG_S3 +- fclose(fpp); +- #endif +- /* Read long block data for converting threshold calculation +- partitions to scale factor bands */ +- +- for(loop=0;loop<6;loop++) +- { +- fscanf(fin,"freq=%lf sb=%d\n",&freq_tp,&sbmax); +- sbmax++; +- #ifdef DEBUG +- printf("freq = %f sb = %d\n",freq_tp,sbmax); +- #endif +- +- if (sfreq == freq_tp) +- { +- for(i=0;i<sbmax;i++) +- { +- fscanf(fin, +- "sb=%d cbw=%d bu=%d bo=%d w1=%lf w2=%lf\n", +- &j,&cbw_l[i],&bu_l[i],&bo_l[i],&w1_l[i],&w2_l[i]); +- if (j!=i) +- { printf("30:please check \"psy_data\"\n"); +- exit(-1); +- } +- #ifdef DEBUG +- printf( +- "sb=%2d cbw=%1d bu=%2d bo=%2d w1=%5.3f w2=%5.3f\n", +- j,cbw_l[i],bu_l[i],bo_l[i],w1_l[i],w2_l[i]); +- #endif +- if (i!=0) +- if ( (bo_l[i] != (bu_l[i]+cbw_l[i])) || +- (fabs(1.0-w1_l[i]-w2_l[i-1]) > 0.01 ) ) +- { printf("31:please check \"psy_data.\"\n"); +- exit(-1); +- } +- } +- } +- else +- { +- for(j=0;j<sbmax;j++) +- { +- fgets(tp,255,fin); +- sscanf(tp,"sb=%d %s\n",&i,tp); +- if (j!=i) +- { printf("please check \"psy_data.\"\n"); +- exit(-1); +- } +- } +- } +- } +- +- /* Read short block data for converting threshold calculation +- partitions to scale factor bands */ +- +- for(loop=0;loop<6;loop++) +- { +- fscanf(fin,"freq=%lf sb=%d\n",&freq_tp,&sbmax); +- sbmax++; +- #ifdef DEBUG +- printf("freq = %f sb = %d\n",freq_tp,sbmax); +- #endif +- +- if (sfreq == freq_tp) +- { +- for(i=0;i<sbmax;i++) +- { +- fscanf(fin, +- "sb=%d cbw=%d bu=%d bo=%d w1=%lf w2=%lf\n", +- &j,&cbw_s[i],&bu_s[i],&bo_s[i],&w1_s[i],&w2_s[i]); +- if (j!=i) +- { printf("30:please check \"psy_data\"\n"); +- exit(-1); +- } +- #ifdef DEBUG +- printf( +- "sb=%2d cbw=%1d bu=%2d bo=%2d w1=%5.3f w2=%5.3f\n", +- j,cbw_s[i],bu_s[i],bo_s[i],w1_s[i],w2_s[i]); +- #endif +- if (i!=0) +- if ( (bo_s[i] != (bu_s[i]+cbw_s[i])) || +- (fabs(1.0-w1_s[i]-w2_s[i-1]) > 0.01 ) ) +- { printf("31:please check \"psy_data.\"\n"); +- exit(-1); +- } +- } +- } +- else +- { +- for(j=0;j<sbmax;j++) +- { +- fgets(tp,255,fin); +- sscanf(tp,"sb=%d %s\n",&i,tp); +- if (j!=i) +- { printf("please check \"psy_data.\"\n"); +- exit(-1); +- } +- } +- } +- } +- +- } +--- 0 ---- +diff -r -c -N encoder/l3psy.h lame3.70/l3psy.h +*** encoder/l3psy.h Wed Jan 22 02:43:16 1997 +--- lame3.70/l3psy.h Wed Dec 31 17:00:00 1969 +*************** +*** 1,37 **** +- /********************************************************************** +- * ISO MPEG Audio Subgroup Software Simulation Group (1996) +- * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- * +- * $Id: l3psy.h,v 1.1 1996/02/14 04:04:23 rowlands Exp $ +- * +- * $Log: l3psy.h,v $ +- * Revision 1.1 1996/02/14 04:04:23 rowlands +- * Initial revision +- * +- * Received from Mike Coleman +- **********************************************************************/ +- +- #ifndef L3PSY_DOT_H_ +- #define L3PSY_DOT_H_ +- /* #define CBANDS 63 */ +- #define CBANDS_s 42 +- #define BLKSIZE_s 256 +- #define HBLKSIZE_s 129 +- #define TCBMAX_l 63 +- #define TCBMAX_s 42 +- #define SBMAX_l 21 +- #define SBMAX_s 12 +- +- /* #define switch_pe 1800 */ +- #define NORM_TYPE 0 +- #define START_TYPE 1 +- #define SHORT_TYPE 2 +- #define STOP_TYPE 3 +- +- /* l3psy.c */ +- #include "l3side.h" +- void L3psycho_anal( short int *buffer, short int savebuf[1344], int chn, int lay, FLOAT snr32[32], +- double sfreq, double ratio_d[21], double ratio_ds[12][3], +- double *pe, gr_info *cod_info ); +- /* void read_absthr(float *absthr, int table); */ +- #endif +--- 0 ---- +diff -r -c -N encoder/l3side.h lame3.70/l3side.h +*** encoder/l3side.h Wed Jan 22 02:43:16 1997 +--- lame3.70/l3side.h Thu Apr 6 12:50:49 2000 +*************** +*** 1,16 **** + /********************************************************************** +- * ISO MPEG Audio Subgroup Software Simulation Group (1996) +- * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- * +- * $Id: l3side.h,v 1.1 1996/02/14 04:04:23 rowlands Exp $ +- * +- * $Log: l3side.h,v $ +- * Revision 1.1 1996/02/14 04:04:23 rowlands +- * Initial revision +- * +- * Received from Mike Coleman +- **********************************************************************/ +- /********************************************************************** + * date programmers comment * + * 25. 6.92 Toshiyuki Ishino Ver 1.0 * + * 29.10.92 Masahiro Iwadare Ver 2.0 * +--- 1,4 ---- +*************** +*** 20,61 **** + + #ifndef L3_SIDE_H + #define L3_SIDE_H + + /* Layer III side information. */ + +! /* #define CBLIMIT_SHORT 12 */ +! +! void l3deco(); +! +! typedef double D576[576]; + typedef int I576[576]; +! typedef double D192_3[192][3]; + typedef int I192_3[192][3]; + +- typedef struct { +- unsigned cbl_max; +- unsigned cbs_start; +- } III_cb_info; +- + + typedef struct { +! double l[2][2][21]; +! double s[2][2][12][3]; +! } III_psy_ratio; +! +! typedef struct { +! double l[2][2][21]; +! double s[2][2][12][3]; + } III_psy_xmin; + + typedef struct { +! double xr[576]; +! double xr_s[3][192]; +! double xmin[21]; +! double xmin_s[3][192]; +! int ix[576]; +! int ix_s[3][192]; +! } III_input3; /* ch */ + + typedef struct { + unsigned part2_3_length; +--- 8,33 ---- + + #ifndef L3_SIDE_H + #define L3_SIDE_H ++ #include "encoder.h" ++ #include "machine.h" + + /* Layer III side information. */ + +! typedef FLOAT8 D576[576]; + typedef int I576[576]; +! typedef FLOAT8 D192_3[192][3]; + typedef int I192_3[192][3]; + + + typedef struct { +! FLOAT8 l[SBPSY_l + 1]; +! FLOAT8 s[SBPSY_s + 1][3]; + } III_psy_xmin; + + typedef struct { +! III_psy_xmin thm; +! III_psy_xmin en; +! } III_psy_ratio; + + typedef struct { + unsigned part2_3_length; +*************** +*** 77,86 **** + unsigned part2_length; + unsigned sfb_lmax; + unsigned sfb_smax; +! unsigned address1; +! unsigned address2; +! unsigned address3; +! double quantizerStepSize; + /* added for LSF */ + unsigned *sfb_partition_table; + unsigned slen[4]; +--- 49,55 ---- + unsigned part2_length; + unsigned sfb_lmax; + unsigned sfb_smax; +! unsigned count1bits; + /* added for LSF */ + unsigned *sfb_partition_table; + unsigned slen[4]; +*************** +*** 92,108 **** + int resvDrain; + unsigned scfsi[2][4]; + struct { +! struct gr_info_s { + gr_info tt; + } ch[2]; + } gr[2]; + } III_side_info_t; + + /* Layer III scale factors. */ +! + typedef struct { +! int l[2][2][22]; /* [cb] */ +! int s[2][2][13][3]; /* [window][cb] */ +! } III_scalefac_t; /* [gr][ch] */ + + #endif +--- 61,78 ---- + int resvDrain; + unsigned scfsi[2][4]; + struct { +! struct gr_info_ss { + gr_info tt; + } ch[2]; + } gr[2]; + } III_side_info_t; + + /* Layer III scale factors. */ +! /* note: there are only SBPSY_l=(SBMAX_l-1) and SBPSY_s=(SBMAX_s-1) scalefactors. +! * Dont know why these would be dimensioned SBMAX_l and SBMAX-s */ + typedef struct { +! int l[SBMAX_l]; /* [cb] */ +! int s[SBMAX_s][3]; /* [window][cb] */ +! } III_scalefac_t; /* [gr][ch] */ + + #endif +diff -r -c -N encoder/lame.c lame3.70/lame.c +*** encoder/lame.c Wed Dec 31 17:00:00 1969 +--- lame3.70/lame.c Thu Apr 6 12:50:51 2000 +*************** +*** 0 **** +--- 1,1428 ---- ++ /* ++ * LAME MP3 encoding engine ++ * ++ * Copyright (c) 1999 Mark Taylor ++ * ++ * This library is free software; you can redistribute it and/or ++ * modify it under the terms of the GNU Library General Public ++ * License as published by the Free Software Foundation; either ++ * version 2 of the License, or (at your option) any later version. ++ * ++ * This library is distributed in the hope that it will be useful, ++ * but WITHOUT ANY WARRANTY; without even the implied warranty of ++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU ++ * Library General Public License for more details. ++ * ++ * You should have received a copy of the GNU Library General Public ++ * License along with this library; if not, write to the ++ * Free Software Foundation, Inc., 59 Temple Place - Suite 330, ++ * Boston, MA 02111-1307, USA. ++ */ ++ ++ ++ #include <assert.h> ++ ++ #ifdef HAVEGTK ++ #include "gtkanal.h" ++ #include <gtk/gtk.h> ++ #endif ++ #include "lame.h" ++ #include "util.h" ++ #include "timestatus.h" ++ #include "psymodel.h" ++ #include "newmdct.h" ++ #include "quantize.h" ++ #include "quantize-pvt.h" ++ #include "l3bitstream.h" ++ #include "formatBitstream.h" ++ #include "version.h" ++ #include "VbrTag.h" ++ #include "id3tag.h" ++ #include "tables.h" ++ #include "brhist.h" ++ #include "get_audio.h" ++ ++ #ifdef __riscos__ ++ #include "asmstuff.h" ++ #endif ++ ++ ++ /* Global variable definitions for lame.c */ ++ static Bit_stream_struc bs; ++ static III_side_info_t l3_side; ++ #define MFSIZE (1152+1152+ENCDELAY-MDCTDELAY) ++ static short int mfbuf[2][MFSIZE]; ++ static int mf_size; ++ static int mf_samples_to_encode; ++ ++ ++ ++ /******************************************************************** ++ * initialize internal params based on data in gf ++ * (globalflags struct filled in by calling program) ++ * ++ ********************************************************************/ ++ void lame_init_params(lame_global_flags *gfp) ++ { ++ int i; ++ FLOAT compression_ratio; ++ ++ ++ memset(&bs, 0, sizeof(Bit_stream_struc)); ++ memset(&l3_side,0x00,sizeof(III_side_info_t)); ++ ++ ++ gfp->frameNum=0; ++ InitFormatBitStream(); ++ if (gfp->num_channels==1) { ++ gfp->mode = MPG_MD_MONO; ++ } ++ gfp->stereo=2; ++ if (gfp->mode == MPG_MD_MONO) gfp->stereo=1; ++ ++ #ifdef BRHIST ++ if (gfp->silent) { ++ disp_brhist=0; /* turn of VBR historgram */ ++ } ++ if (!gfp->VBR) { ++ disp_brhist=0; /* turn of VBR historgram */ ++ } ++ #endif ++ ++ /* set the output sampling rate, and resample options if necessary ++ samplerate = input sample rate ++ resamplerate = ouput sample rate ++ */ ++ if (gfp->out_samplerate==0) { ++ /* user did not specify output sample rate */ ++ gfp->out_samplerate=gfp->in_samplerate; /* default */ ++ ++ ++ /* if resamplerate is not valid, find a valid value */ ++ if (gfp->out_samplerate>=48000) gfp->out_samplerate=48000; ++ else if (gfp->out_samplerate>=44100) gfp->out_samplerate=44100; ++ else if (gfp->out_samplerate>=32000) gfp->out_samplerate=32000; ++ else if (gfp->out_samplerate>=24000) gfp->out_samplerate=24000; ++ else if (gfp->out_samplerate>=22050) gfp->out_samplerate=22050; ++ else gfp->out_samplerate=16000; ++ ++ ++ if (gfp->brate>0) { ++ /* check if user specified bitrate requires downsampling */ ++ compression_ratio = gfp->out_samplerate*16*gfp->stereo/(1000.0*gfp->brate); ++ if (!gfp->VBR && compression_ratio > 13 ) { ++ /* automatic downsample, if possible */ ++ gfp->out_samplerate = (10*1000.0*gfp->brate)/(16*gfp->stereo); ++ if (gfp->out_samplerate<=16000) gfp->out_samplerate=16000; ++ else if (gfp->out_samplerate<=22050) gfp->out_samplerate=22050; ++ else if (gfp->out_samplerate<=24000) gfp->out_samplerate=24000; ++ else if (gfp->out_samplerate<=32000) gfp->out_samplerate=32000; ++ else if (gfp->out_samplerate<=44100) gfp->out_samplerate=44100; ++ else gfp->out_samplerate=48000; ++ } ++ } ++ } ++ ++ gfp->mode_gr = (gfp->out_samplerate <= 24000) ? 1 : 2; /* mode_gr = 2 */ ++ gfp->encoder_delay = ENCDELAY; ++ gfp->framesize = gfp->mode_gr*576; ++ ++ if (gfp->brate==0) { /* user didn't specify a bitrate, use default */ ++ gfp->brate=128; ++ if (gfp->mode_gr==1) gfp->brate=64; ++ } ++ ++ ++ gfp->resample_ratio=1; ++ if (gfp->out_samplerate != gfp->in_samplerate) gfp->resample_ratio = (FLOAT)gfp->in_samplerate/(FLOAT)gfp->out_samplerate; ++ ++ /* estimate total frames. must be done after setting sampling rate so ++ * we know the framesize. */ ++ gfp->totalframes=0; ++ gfp->totalframes = 2+ gfp->num_samples/(gfp->resample_ratio*gfp->framesize); ++ ++ ++ ++ /* 44.1kHz at 56kbs/channel: compression factor of 12.6 ++ 44.1kHz at 64kbs/channel: compression factor of 11.025 ++ 44.1kHz at 80kbs/channel: compression factor of 8.82 ++ 22.05kHz at 24kbs: 14.7 ++ 22.05kHz at 32kbs: 11.025 ++ 22.05kHz at 40kbs: 8.82 ++ 16kHz at 16kbs: 16.0 ++ 16kHz at 24kbs: 10.7 ++ ++ compression_ratio ++ 11 .70? ++ 12 sox resample .66 ++ 14.7 sox resample .45 ++ ++ */ ++ if (gfp->brate >= 320) gfp->VBR=0; /* dont bother with VBR at 320kbs */ ++ compression_ratio = gfp->out_samplerate*16*gfp->stereo/(1000.0*gfp->brate); ++ ++ ++ /* for VBR, take a guess at the compression_ratio */ ++ /* VBR_q compression like ++ 0 4.4 320kbs ++ 1 5.4 256kbs ++ 3 7.4 192kbs ++ 4 8.8 160kbs ++ 6 10.4 128kbs ++ */ ++ if (gfp->VBR && compression_ratio>11) { ++ compression_ratio = 4.4 + gfp->VBR_q; ++ } ++ ++ ++ /* At higher quality (lower compression) use STEREO instead of JSTEREO. ++ * (unless the user explicitly specified a mode ) */ ++ if ( (!gfp->mode_fixed) && (gfp->mode !=MPG_MD_MONO)) { ++ if (compression_ratio < 9 ) { ++ gfp->mode = MPG_MD_STEREO; ++ } ++ } ++ ++ ++ ++ /****************************************************************/ ++ /* if a filter has not been enabled, see if we should add one: */ ++ /****************************************************************/ ++ if (gfp->lowpassfreq == 0) { ++ /* If the user has not selected their own filter, add a lowpass ++ * filter based on the compression ratio. Formula based on ++ 44.1 /160 4.4x ++ 44.1 /128 5.5x keep all bands ++ 44.1 /96kbs 7.3x keep band 28 ++ 44.1 /80kbs 8.8x keep band 25 ++ 44.1khz/64kbs 11x keep band 21 22? ++ ++ 16khz/24kbs 10.7x keep band 21 ++ 22kHz/32kbs 11x keep band ? ++ 22kHz/24kbs 14.7x keep band 16 ++ 16 16 16x keep band 14 ++ */ ++ ++ ++ /* Should we use some lowpass filters? */ ++ int band = 1+floor(.5 + 14-18*log(compression_ratio/16.0)); ++ if (band < 31) { ++ gfp->lowpass1 = band/31.0; ++ gfp->lowpass2 = band/31.0; ++ } ++ } ++ ++ /****************************************************************/ ++ /* apply user driven filters*/ ++ /****************************************************************/ ++ if ( gfp->highpassfreq > 0 ) { ++ gfp->highpass1 = 2.0*gfp->highpassfreq/gfp->out_samplerate; /* will always be >=0 */ ++ if ( gfp->highpasswidth >= 0 ) { ++ gfp->highpass2 = 2.0*(gfp->highpassfreq+gfp->highpasswidth)/gfp->out_samplerate; ++ } else { ++ /* 15% above on default */ ++ /* gfp->highpass2 = 1.15*2.0*gfp->highpassfreq/gfp->out_samplerate; */ ++ gfp->highpass2 = 1.00*2.0*gfp->highpassfreq/gfp->out_samplerate; ++ } ++ gfp->highpass1 = Min( 1, gfp->highpass1 ); ++ gfp->highpass2 = Min( 1, gfp->highpass2 ); ++ } ++ ++ if ( gfp->lowpassfreq > 0 ) { ++ gfp->lowpass2 = 2.0*gfp->lowpassfreq/gfp->out_samplerate; /* will always be >=0 */ ++ if ( gfp->lowpasswidth >= 0 ) { ++ gfp->lowpass1 = 2.0*(gfp->lowpassfreq-gfp->lowpasswidth)/gfp->out_samplerate; ++ if ( gfp->lowpass1 < 0 ) { /* has to be >= 0 */ ++ gfp->lowpass1 = 0; ++ } ++ } else { ++ /* 15% below on default */ ++ /* gfp->lowpass1 = 0.85*2.0*gfp->lowpassfreq/gfp->out_samplerate; */ ++ gfp->lowpass1 = 1.00*2.0*gfp->lowpassfreq/gfp->out_samplerate; ++ } ++ gfp->lowpass1 = Min( 1, gfp->lowpass1 ); ++ gfp->lowpass2 = Min( 1, gfp->lowpass2 ); ++ } ++ ++ ++ /***************************************************************/ ++ /* compute info needed for polyphase filter */ ++ /***************************************************************/ ++ if (gfp->filter_type==0) { ++ int band,maxband,minband; ++ FLOAT8 amp,freq; ++ if (gfp->lowpass1 > 0) { ++ minband=999; ++ maxband=-1; ++ for (band=0; band <=31 ; ++band) { ++ freq = band/31.0; ++ amp = 1; ++ /* this band and above will be zeroed: */ ++ if (freq >= gfp->lowpass2) { ++ gfp->lowpass_band= Min(gfp->lowpass_band,band); ++ amp=0; ++ } ++ if (gfp->lowpass1 < freq && freq < gfp->lowpass2) { ++ minband = Min(minband,band); ++ maxband = Max(maxband,band); ++ amp = cos((PI/2)*(gfp->lowpass1-freq)/(gfp->lowpass2-gfp->lowpass1)); ++ } ++ /* printf("lowpass band=%i amp=%f \n",band,amp);*/ ++ } ++ /* compute the *actual* transition band implemented by the polyphase filter */ ++ if (minband==999) gfp->lowpass1 = (gfp->lowpass_band-.75)/31.0; ++ else gfp->lowpass1 = (minband-.75)/31.0; ++ gfp->lowpass2 = gfp->lowpass_band/31.0; ++ } ++ ++ /* make sure highpass filter is within 90% of whan the effective highpass ++ * frequency will be */ ++ if (gfp->highpass2 > 0) ++ if (gfp->highpass2 < .9*(.75/31.0) ) { ++ gfp->highpass1=0; gfp->highpass2=0; ++ fprintf(stderr,"Warning: highpass filter disabled. highpass frequency to small\n"); ++ } ++ ++ ++ if (gfp->highpass2 > 0) { ++ minband=999; ++ maxband=-1; ++ for (band=0; band <=31; ++band) { ++ freq = band/31.0; ++ amp = 1; ++ /* this band and below will be zereod */ ++ if (freq <= gfp->highpass1) { ++ gfp->highpass_band = Max(gfp->highpass_band,band); ++ amp=0; ++ } ++ if (gfp->highpass1 < freq && freq < gfp->highpass2) { ++ minband = Min(minband,band); ++ maxband = Max(maxband,band); ++ amp = cos((PI/2)*(gfp->highpass2-freq)/(gfp->highpass2-gfp->highpass1)); ++ } ++ /* printf("highpass band=%i amp=%f \n",band,amp);*/ ++ } ++ /* compute the *actual* transition band implemented by the polyphase filter */ ++ gfp->highpass1 = gfp->highpass_band/31.0; ++ if (maxband==-1) gfp->highpass2 = (gfp->highpass_band+.75)/31.0; ++ else gfp->highpass2 = (maxband+.75)/31.0; ++ } ++ /* ++ printf("lowpass band with amp=0: %i \n",gfp->lowpass_band); ++ printf("highpass band with amp=0: %i \n",gfp->highpass_band); ++ */ ++ } ++ ++ ++ ++ /***************************************************************/ ++ /* compute info needed for FIR filter */ ++ /***************************************************************/ ++ if (gfp->filter_type==1) { ++ } ++ ++ ++ ++ ++ gfp->mode_ext=MPG_MD_LR_LR; ++ gfp->stereo = (gfp->mode == MPG_MD_MONO) ? 1 : 2; ++ ++ ++ gfp->samplerate_index = SmpFrqIndex((long)gfp->out_samplerate, &gfp->version); ++ if( gfp->samplerate_index < 0) { ++ display_bitrates(stderr); ++ exit(1); ++ } ++ if( (gfp->bitrate_index = BitrateIndex(gfp->brate, gfp->version,gfp->out_samplerate)) < 0) { ++ display_bitrates(stderr); ++ exit(1); ++ } ++ ++ ++ /* choose a min/max bitrate for VBR */ ++ if (gfp->VBR) { ++ /* if the user didn't specify VBR_max_bitrate: */ ++ if (0==gfp->VBR_max_bitrate_kbps) { ++ /* default max bitrate is 256kbs */ ++ /* we do not normally allow 320bps frams with VBR, unless: */ ++ gfp->VBR_max_bitrate=13; /* default: allow 256kbs */ ++ if (gfp->VBR_min_bitrate_kbps>=256) gfp->VBR_max_bitrate=14; ++ if (gfp->VBR_q == 0) gfp->VBR_max_bitrate=14; /* allow 320kbs */ ++ if (gfp->VBR_q >= 4) gfp->VBR_max_bitrate=12; /* max = 224kbs */ ++ if (gfp->VBR_q >= 8) gfp->VBR_max_bitrate=9; /* low quality, max = 128kbs */ ++ }else{ ++ if( (gfp->VBR_max_bitrate = BitrateIndex(gfp->VBR_max_bitrate_kbps, gfp->version,gfp->out_samplerate)) < 0) { ++ display_bitrates(stderr); ++ exit(1); ++ } ++ } ++ if (0==gfp->VBR_min_bitrate_kbps) { ++ gfp->VBR_min_bitrate=1; /* 32 kbps */ ++ }else{ ++ if( (gfp->VBR_min_bitrate = BitrateIndex(gfp->VBR_min_bitrate_kbps, gfp->version,gfp->out_samplerate)) < 0) { ++ display_bitrates(stderr); ++ exit(1); ++ } ++ } ++ ++ } ++ ++ ++ if (gfp->VBR) gfp->quality=Min(gfp->quality,2); /* always use quality <=2 with VBR */ ++ /* dont allow forced mid/side stereo for mono output */ ++ if (gfp->mode == MPG_MD_MONO) gfp->force_ms=0; ++ ++ ++ /* Do not write VBR tag if VBR flag is not specified */ ++ if (gfp->VBR==0) gfp->bWriteVbrTag=0; ++ ++ /* some file options not allowed if output is: not specified or stdout */ ++ ++ if (gfp->outPath!=NULL && gfp->outPath[0]=='-' ) { ++ gfp->bWriteVbrTag=0; /* turn off VBR tag */ ++ } ++ ++ if (gfp->outPath==NULL || gfp->outPath[0]=='-' ) { ++ id3tag.used=0; /* turn of id3 tagging */ ++ } ++ ++ ++ ++ if (gfp->gtkflag) { ++ gfp->bWriteVbrTag=0; /* disable Xing VBR tag */ ++ } ++ ++ init_bit_stream_w(&bs); ++ ++ ++ ++ /* set internal feature flags. USER should not access these since ++ * some combinations will produce strange results */ ++ ++ /* no psymodel, no noise shaping */ ++ if (gfp->quality==9) { ++ gfp->filter_type=0; ++ gfp->psymodel=0; ++ gfp->quantization=0; ++ gfp->noise_shaping=0; ++ gfp->noise_shaping_stop=0; ++ gfp->use_best_huffman=0; ++ } ++ ++ if (gfp->quality==8) gfp->quality=7; ++ ++ /* use psymodel (for short block and m/s switching), but no noise shapping */ ++ if (gfp->quality==7) { ++ gfp->filter_type=0; ++ gfp->psymodel=1; ++ gfp->quantization=0; ++ gfp->noise_shaping=0; ++ gfp->noise_shaping_stop=0; ++ gfp->use_best_huffman=0; ++ } ++ ++ if (gfp->quality==6) gfp->quality=5; ++ ++ if (gfp->quality==5) { ++ /* the default */ ++ gfp->filter_type=0; ++ gfp->psymodel=1; ++ gfp->quantization=0; ++ gfp->noise_shaping=1; ++ gfp->noise_shaping_stop=0; ++ gfp->use_best_huffman=0; ++ } ++ ++ if (gfp->quality==4) gfp->quality=2; ++ if (gfp->quality==3) gfp->quality=2; ++ ++ if (gfp->quality==2) { ++ gfp->filter_type=0; ++ gfp->psymodel=1; ++ gfp->quantization=1; ++ gfp->noise_shaping=1; ++ gfp->noise_shaping_stop=0; ++ gfp->use_best_huffman=1; ++ } ++ ++ if (gfp->quality==1) { ++ gfp->filter_type=0; ++ gfp->psymodel=1; ++ gfp->quantization=1; ++ gfp->noise_shaping=1; ++ gfp->noise_shaping_stop=1; ++ gfp->use_best_huffman=1; ++ } ++ ++ if (gfp->quality==0) { ++ /* 0..1 quality */ ++ gfp->filter_type=1; /* not yet coded */ ++ gfp->psymodel=1; ++ gfp->quantization=1; ++ gfp->noise_shaping=3; /* not yet coded */ ++ gfp->noise_shaping_stop=2; /* not yet coded */ ++ gfp->use_best_huffman=2; /* not yet coded */ ++ exit(-99); ++ } ++ ++ ++ for (i = 0; i < SBMAX_l + 1; i++) { ++ scalefac_band.l[i] = ++ sfBandIndex[gfp->samplerate_index + (gfp->version * 3)].l[i]; ++ } ++ for (i = 0; i < SBMAX_s + 1; i++) { ++ scalefac_band.s[i] = ++ sfBandIndex[gfp->samplerate_index + (gfp->version * 3)].s[i]; ++ } ++ ++ ++ ++ if (gfp->bWriteVbrTag) ++ { ++ /* Write initial VBR Header to bitstream */ ++ InitVbrTag(&bs,1-gfp->version,gfp->mode,gfp->samplerate_index); ++ } ++ ++ #ifdef HAVEGTK ++ gtkflag=gfp->gtkflag; ++ #endif ++ ++ #ifdef BRHIST ++ if (gfp->VBR) { ++ if (disp_brhist) ++ brhist_init(gfp,1, 14); ++ } else ++ disp_brhist = 0; ++ #endif ++ return; ++ } ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ /************************************************************************ ++ * ++ * print_config ++ * ++ * PURPOSE: Prints the encoding parameters used ++ * ++ ************************************************************************/ ++ void lame_print_config(lame_global_flags *gfp) ++ { ++ static const char *mode_names[4] = { "stereo", "j-stereo", "dual-ch", "single-ch" }; ++ FLOAT out_samplerate=gfp->out_samplerate/1000.0; ++ FLOAT in_samplerate = gfp->resample_ratio*out_samplerate; ++ FLOAT compression= ++ (FLOAT)(gfp->stereo*16*out_samplerate)/(FLOAT)(gfp->brate); ++ ++ lame_print_version(stderr); ++ if (gfp->num_channels==2 && gfp->stereo==1) { ++ fprintf(stderr, "Autoconverting from stereo to mono. Setting encoding to mono mode.\n"); ++ } ++ if (gfp->resample_ratio!=1) { ++ fprintf(stderr,"Resampling: input=%ikHz output=%ikHz\n", ++ (int)in_samplerate,(int)out_samplerate); ++ } ++ if (gfp->highpass2>0.0) ++ fprintf(stderr, "Using polyphase highpass filter, transition band: %.0f Hz - %.0f Hz\n", ++ gfp->highpass1*out_samplerate*500, ++ gfp->highpass2*out_samplerate*500); ++ if (gfp->lowpass1>0.0) ++ fprintf(stderr, "Using polyphase lowpass filter, transition band: %.0f Hz - %.0f Hz\n", ++ gfp->lowpass1*out_samplerate*500, ++ gfp->lowpass2*out_samplerate*500); ++ ++ if (gfp->gtkflag) { ++ fprintf(stderr, "Analyzing %s \n",gfp->inPath); ++ } ++ else { ++ fprintf(stderr, "Encoding %s to %s\n", ++ (strcmp(gfp->inPath, "-")? gfp->inPath : "stdin"), ++ (strcmp(gfp->outPath, "-")? gfp->outPath : "stdout")); ++ if (gfp->VBR) ++ fprintf(stderr, "Encoding as %.1fkHz VBR(q=%i) %s MPEG%i LayerIII qval=%i\n", ++ gfp->out_samplerate/1000.0, ++ gfp->VBR_q,mode_names[gfp->mode],2-gfp->version,gfp->quality); ++ else ++ fprintf(stderr, "Encoding as %.1f kHz %d kbps %s MPEG%i LayerIII (%4.1fx) qval=%i\n", ++ gfp->out_samplerate/1000.0,gfp->brate, ++ mode_names[gfp->mode],2-gfp->version,compression,gfp->quality); ++ } ++ fflush(stderr); ++ } ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ /************************************************************************ ++ * ++ * encodeframe() Layer 3 ++ * ++ * encode a single frame ++ * ++ ************************************************************************ ++ lame_encode_frame() ++ ++ ++ gr 0 gr 1 ++ inbuf: |--------------|---------------|-------------| ++ MDCT output: |--------------|---------------|-------------| ++ ++ FFT's <---------1024----------> ++ <---------1024--------> ++ ++ ++ ++ inbuf = buffer of PCM data size=MP3 framesize ++ encoder acts on inbuf[ch][0], but output is delayed by MDCTDELAY ++ so the MDCT coefficints are from inbuf[ch][-MDCTDELAY] ++ ++ psy-model FFT has a 1 granule day, so we feed it data for the next granule. ++ FFT is centered over granule: 224+576+224 ++ So FFT starts at: 576-224-MDCTDELAY ++ ++ MPEG2: FFT ends at: BLKSIZE+576-224-MDCTDELAY ++ MPEG1: FFT ends at: BLKSIZE+2*576-224-MDCTDELAY (1904) ++ ++ FFT starts at 576-224-MDCTDELAY (304) = 576-FFTOFFSET ++ ++ */ ++ int lame_encode_frame(lame_global_flags *gfp, ++ short int inbuf_l[],short int inbuf_r[], ++ int mf_size,char *mp3buf, int mp3buf_size) ++ { ++ static unsigned long frameBits; ++ static unsigned long bitsPerSlot; ++ static FLOAT8 frac_SpF; ++ static FLOAT8 slot_lag; ++ static unsigned long sentBits = 0; ++ FLOAT8 xr[2][2][576]; ++ int l3_enc[2][2][576]; ++ int mp3count; ++ III_psy_ratio masking_ratio[2][2]; /*LR ratios */ ++ III_psy_ratio masking_MS_ratio[2][2]; /*MS ratios */ ++ III_psy_ratio (*masking)[2][2]; /*LR ratios and MS ratios*/ ++ III_scalefac_t scalefac[2][2]; ++ short int *inbuf[2]; ++ ++ typedef FLOAT8 pedata[2][2]; ++ pedata pe,pe_MS; ++ pedata *pe_use; ++ ++ int ch,gr,mean_bits; ++ int bitsPerFrame; ++ ++ int check_ms_stereo; ++ static FLOAT8 ms_ratio[2]={0,0}; ++ FLOAT8 ms_ratio_next=0; ++ FLOAT8 ms_ratio_prev=0; ++ static FLOAT8 ms_ener_ratio[2]={0,0}; ++ ++ memset((char *) masking_ratio, 0, sizeof(masking_ratio)); ++ memset((char *) masking_MS_ratio, 0, sizeof(masking_MS_ratio)); ++ memset((char *) scalefac, 0, sizeof(scalefac)); ++ inbuf[0]=inbuf_l; ++ inbuf[1]=inbuf_r; ++ ++ gfp->mode_ext = MPG_MD_LR_LR; ++ ++ if (gfp->frameNum==0 ) { ++ /* Figure average number of 'slots' per frame. */ ++ FLOAT8 avg_slots_per_frame; ++ FLOAT8 sampfreq = gfp->out_samplerate/1000.0; ++ int bit_rate = gfp->brate; ++ sentBits = 0; ++ bitsPerSlot = 8; ++ avg_slots_per_frame = (bit_rate*gfp->framesize) / ++ (sampfreq* bitsPerSlot); ++ /* -f fast-math option causes some strange rounding here, be carefull: */ ++ frac_SpF = avg_slots_per_frame - floor(avg_slots_per_frame + 1e-9); ++ if (fabs(frac_SpF) < 1e-9) frac_SpF = 0; ++ ++ slot_lag = -frac_SpF; ++ gfp->padding = 1; ++ if (frac_SpF==0) gfp->padding = 0; ++ /* check FFT will not use a negative starting offset */ ++ assert(576>=FFTOFFSET); ++ /* check if we have enough data for FFT */ ++ assert(mf_size>=(BLKSIZE+gfp->framesize-FFTOFFSET)); ++ } ++ ++ ++ /********************** padding *****************************/ ++ switch (gfp->padding_type) { ++ case 0: ++ gfp->padding=0; ++ break; ++ case 1: ++ gfp->padding=1; ++ break; ++ case 2: ++ default: ++ if (gfp->VBR) { ++ gfp->padding=0; ++ } else { ++ if (gfp->disable_reservoir) { ++ gfp->padding = 0; ++ /* if the user specified --nores, dont very gfp->padding either */ ++ /* tiny changes in frac_SpF rounding will cause file differences */ ++ }else{ ++ if (frac_SpF != 0) { ++ if (slot_lag > (frac_SpF-1.0) ) { ++ slot_lag -= frac_SpF; ++ gfp->padding = 0; ++ } ++ else { ++ gfp->padding = 1; ++ slot_lag += (1-frac_SpF); ++ } ++ } ++ } ++ } ++ } ++ ++ ++ /********************** status display *****************************/ ++ if (!gfp->gtkflag && !gfp->silent) { ++ int mod = gfp->version == 0 ? 200 : 50; ++ if (gfp->frameNum%mod==0) { ++ timestatus(gfp->out_samplerate,gfp->frameNum,gfp->totalframes,gfp->framesize); ++ #ifdef BRHIST ++ if (disp_brhist) ++ { ++ brhist_add_count(); ++ brhist_disp(); ++ } ++ #endif ++ } ++ } ++ ++ ++ if (gfp->psymodel) { ++ /* psychoacoustic model ++ * psy model has a 1 granule (576) delay that we must compensate for ++ * (mt 6/99). ++ */ ++ short int *bufp[2]; /* address of beginning of left & right granule */ ++ int blocktype[2]; ++ ++ ms_ratio_prev=ms_ratio[gfp->mode_gr-1]; ++ for (gr=0; gr < gfp->mode_gr ; gr++) { ++ ++ for ( ch = 0; ch < gfp->stereo; ch++ ) ++ bufp[ch] = &inbuf[ch][576 + gr*576-FFTOFFSET]; ++ ++ L3psycho_anal( gfp,bufp, gr, ++ &ms_ratio[gr],&ms_ratio_next,&ms_ener_ratio[gr], ++ masking_ratio, masking_MS_ratio, ++ pe[gr],pe_MS[gr],blocktype); ++ ++ for ( ch = 0; ch < gfp->stereo; ch++ ) ++ l3_side.gr[gr].ch[ch].tt.block_type=blocktype[ch]; ++ ++ } ++ }else{ ++ for (gr=0; gr < gfp->mode_gr ; gr++) ++ for ( ch = 0; ch < gfp->stereo; ch++ ) { ++ l3_side.gr[gr].ch[ch].tt.block_type=NORM_TYPE; ++ pe[gr][ch]=700; ++ } ++ } ++ ++ ++ /* block type flags */ ++ for( gr = 0; gr < gfp->mode_gr; gr++ ) { ++ for ( ch = 0; ch < gfp->stereo; ch++ ) { ++ gr_info *cod_info = &l3_side.gr[gr].ch[ch].tt; ++ cod_info->mixed_block_flag = 0; /* never used by this model */ ++ if (cod_info->block_type == NORM_TYPE ) ++ cod_info->window_switching_flag = 0; ++ else ++ cod_info->window_switching_flag = 1; ++ } ++ } ++ ++ /* polyphase filtering / mdct */ ++ mdct_sub48(gfp,inbuf[0], inbuf[1], xr, &l3_side); ++ ++ /* use m/s gfp->stereo? */ ++ check_ms_stereo = (gfp->mode == MPG_MD_JOINT_STEREO); ++ if (check_ms_stereo) { ++ /* make sure block type is the same in each channel */ ++ check_ms_stereo = ++ (l3_side.gr[0].ch[0].tt.block_type==l3_side.gr[0].ch[1].tt.block_type) && ++ (l3_side.gr[1].ch[0].tt.block_type==l3_side.gr[1].ch[1].tt.block_type); ++ } ++ if (check_ms_stereo) { ++ /* ms_ratio = is like the ratio of side_energy/total_energy */ ++ FLOAT8 ms_ratio_ave,ms_ener_ratio_ave; ++ /* ms_ratio_ave = .5*(ms_ratio[0] + ms_ratio[1]);*/ ++ ms_ratio_ave = .25*(ms_ratio[0] + ms_ratio[1]+ ++ ms_ratio_prev + ms_ratio_next); ++ ms_ener_ratio_ave = .5*(ms_ener_ratio[0]+ms_ener_ratio[1]); ++ if ( ms_ratio_ave <.35 /*&& ms_ener_ratio_ave<.75*/ ) gfp->mode_ext = MPG_MD_MS_LR; ++ } ++ if (gfp->force_ms) gfp->mode_ext = MPG_MD_MS_LR; ++ ++ ++ #ifdef HAVEGTK ++ if (gfp->gtkflag) { ++ int j; ++ for ( gr = 0; gr < gfp->mode_gr; gr++ ) { ++ for ( ch = 0; ch < gfp->stereo; ch++ ) { ++ pinfo->ms_ratio[gr]=ms_ratio[gr]; ++ pinfo->ms_ener_ratio[gr]=ms_ener_ratio[gr]; ++ pinfo->blocktype[gr][ch]= ++ l3_side.gr[gr].ch[ch].tt.block_type; ++ for ( j = 0; j < 576; j++ ) pinfo->xr[gr][ch][j]=xr[gr][ch][j]; ++ /* if MS stereo, switch to MS psy data */ ++ if (gfp->mode_ext==MPG_MD_MS_LR) { ++ pinfo->pe[gr][ch]=pinfo->pe[gr][ch+2]; ++ pinfo->ers[gr][ch]=pinfo->ers[gr][ch+2]; ++ memcpy(pinfo->energy[gr][ch],pinfo->energy[gr][ch+2], ++ sizeof(pinfo->energy[gr][ch])); ++ } ++ } ++ } ++ } ++ #endif ++ ++ ++ ++ ++ /* bit and noise allocation */ ++ if (MPG_MD_MS_LR == gfp->mode_ext) { ++ masking = &masking_MS_ratio; /* use MS masking */ ++ pe_use=&pe_MS; ++ } else { ++ masking = &masking_ratio; /* use LR masking */ ++ pe_use=&pe; ++ } ++ ++ ++ /* ++ VBR_iteration_loop_new( gfp,*pe_use, ms_ratio, xr, masking, &l3_side, l3_enc, ++ &scalefac); ++ */ ++ ++ ++ if (gfp->VBR) { ++ VBR_iteration_loop( gfp,*pe_use, ms_ratio, xr, *masking, &l3_side, l3_enc, ++ scalefac); ++ }else{ ++ iteration_loop( gfp,*pe_use, ms_ratio, xr, *masking, &l3_side, l3_enc, ++ scalefac); ++ } ++ ++ ++ ++ ++ #ifdef BRHIST ++ brhist_temp[gfp->bitrate_index]++; ++ #endif ++ ++ ++ /* write the frame to the bitstream */ ++ getframebits(gfp,&bitsPerFrame,&mean_bits); ++ III_format_bitstream( gfp,bitsPerFrame, l3_enc, &l3_side, ++ scalefac, &bs); ++ ++ ++ frameBits = bs.totbit - sentBits; ++ ++ ++ if ( frameBits % bitsPerSlot ) /* a program failure */ ++ fprintf( stderr, "Sent %ld bits = %ld slots plus %ld\n", ++ frameBits, frameBits/bitsPerSlot, ++ frameBits%bitsPerSlot ); ++ sentBits += frameBits; ++ ++ /* copy mp3 bit buffer into array */ ++ mp3count = copy_buffer(mp3buf,mp3buf_size,&bs); ++ ++ if (gfp->bWriteVbrTag) AddVbrFrame((int)(sentBits/8)); ++ ++ #ifdef HAVEGTK ++ if (gfp->gtkflag) { ++ int j; ++ for ( ch = 0; ch < gfp->stereo; ch++ ) { ++ for ( j = 0; j < FFTOFFSET; j++ ) ++ pinfo->pcmdata[ch][j] = pinfo->pcmdata[ch][j+gfp->framesize]; ++ for ( j = FFTOFFSET; j < 1600; j++ ) { ++ pinfo->pcmdata[ch][j] = inbuf[ch][j-FFTOFFSET]; ++ } ++ } ++ } ++ #endif ++ gfp->frameNum++; ++ ++ return mp3count; ++ } ++ ++ ++ ++ int fill_buffer_resample(lame_global_flags *gfp,short int *outbuf,int desired_len, ++ short int *inbuf,int len,int *num_used,int ch) { ++ ++ static FLOAT8 itime[2]; ++ #define OLDBUFSIZE 5 ++ static short int inbuf_old[2][OLDBUFSIZE]; ++ static int init[2]={0,0}; ++ int i,j=0,k,linear,value; ++ ++ if (gfp->frameNum==0 && !init[ch]) { ++ init[ch]=1; ++ itime[ch]=0; ++ memset((char *) inbuf_old[ch], 0, sizeof(short int)*OLDBUFSIZE); ++ } ++ if (gfp->frameNum!=0) init[ch]=0; /* reset, for next time framenum=0 */ ++ ++ ++ /* if downsampling by an integer multiple, use linear resampling, ++ * otherwise use quadratic */ ++ linear = ( fabs(gfp->resample_ratio - floor(.5+gfp->resample_ratio)) < .0001 ); ++ ++ /* time of j'th element in inbuf = itime + j/ifreq; */ ++ /* time of k'th element in outbuf = j/ofreq */ ++ for (k=0;k<desired_len;k++) { ++ int y0,y1,y2,y3; ++ FLOAT8 x0,x1,x2,x3; ++ FLOAT8 time0; ++ ++ time0 = k*gfp->resample_ratio; /* time of k'th output sample */ ++ j = floor( time0 -itime[ch] ); ++ /* itime[ch] + j; */ /* time of j'th input sample */ ++ if (j+2 >= len) break; /* not enough data in input buffer */ ++ ++ x1 = time0-(itime[ch]+j); ++ x2 = x1-1; ++ y1 = (j<0) ? inbuf_old[ch][OLDBUFSIZE+j] : inbuf[j]; ++ y2 = ((1+j)<0) ? inbuf_old[ch][OLDBUFSIZE+1+j] : inbuf[1+j]; ++ ++ /* linear resample */ ++ if (linear) { ++ outbuf[k] = floor(.5 + (y2*x1-y1*x2) ); ++ } else { ++ /* quadratic */ ++ x0 = x1+1; ++ x3 = x1-2; ++ y0 = ((j-1)<0) ? inbuf_old[ch][OLDBUFSIZE+(j-1)] : inbuf[j-1]; ++ y3 = ((j+2)<0) ? inbuf_old[ch][OLDBUFSIZE+(j+2)] : inbuf[j+2]; ++ value = floor(.5 + ++ -y0*x1*x2*x3/6 + y1*x0*x2*x3/2 - y2*x0*x1*x3/2 +y3*x0*x1*x2/6 ++ ); ++ if (value > 32767) outbuf[k]=32767; ++ else if (value < -32767) outbuf[k]=-32767; ++ else outbuf[k]=value; ++ ++ /* ++ printf("k=%i new=%i [ %i %i %i %i ]\n",k,outbuf[k], ++ y0,y1,y2,y3); ++ */ ++ } ++ } ++ ++ ++ /* k = number of samples added to outbuf */ ++ /* last k sample used data from j,j+1, or j+1 overflowed buffer */ ++ /* remove num_used samples from inbuf: */ ++ *num_used = Min(len,j+2); ++ itime[ch] += *num_used - k*gfp->resample_ratio; ++ for (i=0;i<OLDBUFSIZE;i++) ++ inbuf_old[ch][i]=inbuf[*num_used + i -OLDBUFSIZE]; ++ return k; ++ } ++ ++ ++ ++ ++ int fill_buffer(lame_global_flags *gfp,short int *outbuf,int desired_len,short int *inbuf,int len) { ++ int j; ++ j=Min(desired_len,len); ++ memcpy( (char *) outbuf,(char *)inbuf,sizeof(short int)*j); ++ return j; ++ } ++ ++ ++ ++ ++ /* ++ * THE MAIN LAME ENCODING INTERFACE ++ * mt 3/00 ++ * ++ * input pcm data, output (maybe) mp3 frames. ++ * This routine handles all buffering, resampling and filtering for you. ++ * The required mp3buffer_size can be computed from num_samples, ++ * samplerate and encoding rate, but here is a worst case estimate: ++ * ++ * mp3buffer_size in bytes = 1.25*num_samples + 7200 ++ * ++ * return code = number of bytes output in mp3buffer. can be 0 ++ */ ++ int lame_encode_buffer(lame_global_flags *gfp, ++ short int buffer_l[], short int buffer_r[],int nsamples, ++ char *mp3buf, int mp3buf_size) ++ { ++ static int frame_buffered=0; ++ int mp3size=0,ret,i,ch,mf_needed; ++ ++ short int *in_buffer[2]; ++ in_buffer[0] = buffer_l; ++ in_buffer[1] = buffer_r; ++ ++ /* some sanity checks */ ++ assert(ENCDELAY>=MDCTDELAY); ++ assert(BLKSIZE-FFTOFFSET >= 0); ++ mf_needed = BLKSIZE+gfp->framesize-FFTOFFSET; ++ assert(MFSIZE>=mf_needed); ++ ++ /* The reason for ++ * int mf_samples_to_encode = ENCDELAY + 288; ++ * ENCDELAY = internal encoder delay. And then we have to add 288 ++ * because of the 50% MDCT overlap. A 576 MDCT granule decodes to ++ * 1152 samples. To synthesize the 576 samples centered under this granule ++ * we need the previous granule for the first 288 samples (no problem), and ++ * the next granule for the next 288 samples (not possible if this is last ++ * granule). So we need to pad with 288 samples to make sure we can ++ * encode the 576 samples we are interested in. ++ */ ++ if (gfp->frameNum==0 && !frame_buffered) { ++ memset((char *) mfbuf, 0, sizeof(mfbuf)); ++ frame_buffered=1; ++ mf_samples_to_encode = ENCDELAY+288; ++ mf_size=ENCDELAY-MDCTDELAY; /* we pad input with this many 0's */ ++ } ++ if (gfp->frameNum==1) { ++ /* reset, for the next time frameNum==0 */ ++ frame_buffered=0; ++ } ++ ++ if (gfp->num_channels==2 && gfp->stereo==1) { ++ /* downsample to mono */ ++ for (i=0; i<nsamples; ++i) { ++ in_buffer[0][i]=((int)in_buffer[0][i]+(int)in_buffer[1][i])/2; ++ in_buffer[1][i]=0; ++ } ++ } ++ ++ ++ while (nsamples > 0) { ++ int n_in=0; ++ int n_out=0; ++ /* copy in new samples */ ++ for (ch=0; ch<gfp->stereo; ch++) { ++ if (gfp->resample_ratio!=1) { ++ n_out=fill_buffer_resample(gfp,&mfbuf[ch][mf_size],gfp->framesize, ++ in_buffer[ch],nsamples,&n_in,ch); ++ } else { ++ n_out=fill_buffer(gfp,&mfbuf[ch][mf_size],gfp->framesize,in_buffer[ch],nsamples); ++ n_in = n_out; ++ } ++ in_buffer[ch] += n_in; ++ } ++ ++ ++ nsamples -= n_in; ++ mf_size += n_out; ++ assert(mf_size<=MFSIZE); ++ mf_samples_to_encode += n_out; ++ ++ if (mf_size >= mf_needed) { ++ /* encode the frame */ ++ ret = lame_encode_frame(gfp,mfbuf[0],mfbuf[1],mf_size,mp3buf,mp3buf_size); ++ if (ret == -1) { ++ /* fatel error: mp3buffer was too small */ ++ return -1; ++ } ++ mp3buf += ret; ++ mp3size += ret; ++ ++ /* shift out old samples */ ++ mf_size -= gfp->framesize; ++ mf_samples_to_encode -= gfp->framesize; ++ for (ch=0; ch<gfp->stereo; ch++) ++ for (i=0; i<mf_size; i++) ++ mfbuf[ch][i]=mfbuf[ch][i+gfp->framesize]; ++ } ++ } ++ assert(nsamples==0); ++ return mp3size; ++ } ++ ++ ++ ++ ++ int lame_encode_buffer_interleaved(lame_global_flags *gfp, ++ short int buffer[], int nsamples, char *mp3buf, int mp3buf_size) ++ { ++ static int frame_buffered=0; ++ int mp3size=0,ret,i,ch,mf_needed; ++ ++ /* some sanity checks */ ++ assert(ENCDELAY>=MDCTDELAY); ++ assert(BLKSIZE-FFTOFFSET >= 0); ++ mf_needed = BLKSIZE+gfp->framesize-FFTOFFSET; ++ assert(MFSIZE>=mf_needed); ++ ++ if (gfp->num_channels == 1) { ++ return lame_encode_buffer(gfp,buffer, NULL ,nsamples,mp3buf,mp3buf_size); ++ } ++ ++ if (gfp->resample_ratio!=1) { ++ short int *buffer_l; ++ short int *buffer_r; ++ buffer_l=malloc(sizeof(short int)*nsamples); ++ buffer_r=malloc(sizeof(short int)*nsamples); ++ if (buffer_l == NULL || buffer_r == NULL) { ++ return -1; ++ } ++ for (i=0; i<nsamples; i++) { ++ buffer_l[i]=buffer[2*i]; ++ buffer_r[i]=buffer[2*i+1]; ++ } ++ ret = lame_encode_buffer(gfp,buffer_l,buffer_r,nsamples,mp3buf,mp3buf_size); ++ free(buffer_l); ++ free(buffer_r); ++ return ret; ++ } ++ ++ ++ if (gfp->frameNum==0 && !frame_buffered) { ++ memset((char *) mfbuf, 0, sizeof(mfbuf)); ++ frame_buffered=1; ++ mf_samples_to_encode = ENCDELAY+288; ++ mf_size=ENCDELAY-MDCTDELAY; /* we pad input with this many 0's */ ++ } ++ if (gfp->frameNum==1) { ++ /* reset, for the next time frameNum==0 */ ++ frame_buffered=0; ++ } ++ ++ if (gfp->num_channels==2 && gfp->stereo==1) { ++ /* downsample to mono */ ++ for (i=0; i<nsamples; ++i) { ++ buffer[2*i]=((int)buffer[2*i]+(int)buffer[2*i+1])/2; ++ buffer[2*i+1]=0; ++ } ++ } ++ ++ ++ while (nsamples > 0) { ++ int n_out; ++ /* copy in new samples */ ++ n_out = Min(gfp->framesize,nsamples); ++ for (i=0; i<n_out; ++i) { ++ mfbuf[0][mf_size+i]=buffer[2*i]; ++ mfbuf[1][mf_size+i]=buffer[2*i+1]; ++ } ++ buffer += 2*n_out; ++ ++ nsamples -= n_out; ++ mf_size += n_out; ++ assert(mf_size<=MFSIZE); ++ mf_samples_to_encode += n_out; ++ ++ if (mf_size >= mf_needed) { ++ /* encode the frame */ ++ ret = lame_encode_frame(gfp,mfbuf[0],mfbuf[1],mf_size,mp3buf,mp3buf_size); ++ if (ret == -1) { ++ /* fatel error: mp3buffer was too small */ ++ return -1; ++ } ++ mp3buf += ret; ++ mp3size += ret; ++ ++ /* shift out old samples */ ++ mf_size -= gfp->framesize; ++ mf_samples_to_encode -= gfp->framesize; ++ for (ch=0; ch<gfp->stereo; ch++) ++ for (i=0; i<mf_size; i++) ++ mfbuf[ch][i]=mfbuf[ch][i+gfp->framesize]; ++ } ++ } ++ assert(nsamples==0); ++ return mp3size; ++ } ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ /* old LAME interface */ ++ /* With this interface, it is the users responsibilty to keep track of the ++ * buffered, unencoded samples. Thus mf_samples_to_encode is not incremented. ++ * ++ * lame_encode() is also used to flush the PCM input buffer by ++ * lame_encode_finish() ++ */ ++ int lame_encode(lame_global_flags *gfp, short int in_buffer[2][1152],char *mp3buf,int size){ ++ int imp3,save; ++ save = mf_samples_to_encode; ++ imp3= lame_encode_buffer(gfp,in_buffer[0],in_buffer[1],576*gfp->mode_gr, ++ mp3buf,size); ++ mf_samples_to_encode = save; ++ return imp3; ++ } ++ ++ ++ ++ ++ /* initialize mp3 encoder */ ++ void lame_init(lame_global_flags *gfp) ++ { ++ ++ /* ++ * Disable floating point exepctions ++ */ ++ #ifdef __FreeBSD__ ++ # include <floatingpoint.h> ++ { ++ /* seet floating point mask to the Linux default */ ++ fp_except_t mask; ++ mask=fpgetmask(); ++ /* if bit is set, we get SIGFPE on that error! */ ++ fpsetmask(mask & ~(FP_X_INV|FP_X_DZ)); ++ /* fprintf(stderr,"FreeBSD mask is 0x%x\n",mask); */ ++ } ++ #endif ++ #if defined(__riscos__) && !defined(ABORTFP) ++ /* Disable FPE's under RISC OS */ ++ /* if bit is set, we disable trapping that error! */ ++ /* _FPE_IVO : invalid operation */ ++ /* _FPE_DVZ : divide by zero */ ++ /* _FPE_OFL : overflow */ ++ /* _FPE_UFL : underflow */ ++ /* _FPE_INX : inexact */ ++ DisableFPETraps( _FPE_IVO | _FPE_DVZ | _FPE_OFL ); ++ #endif ++ ++ ++ /* ++ * Debugging stuff ++ * The default is to ignore FPE's, unless compiled with -DABORTFP ++ * so add code below to ENABLE FPE's. ++ */ ++ ++ #if defined(ABORTFP) && !defined(__riscos__) ++ #if defined(_MSC_VER) ++ { ++ #include <float.h> ++ unsigned int mask; ++ mask=_controlfp( 0, 0 ); ++ mask&=~(_EM_OVERFLOW|_EM_UNDERFLOW|_EM_ZERODIVIDE|_EM_INVALID); ++ mask=_controlfp( mask, _MCW_EM ); ++ } ++ #elif defined(__CYGWIN__) ++ # define _FPU_GETCW(cw) __asm__ ("fnstcw %0" : "=m" (*&cw)) ++ # define _FPU_SETCW(cw) __asm__ ("fldcw %0" : : "m" (*&cw)) ++ ++ # define _EM_INEXACT 0x00000001 /* inexact (precision) */ ++ # define _EM_UNDERFLOW 0x00000002 /* underflow */ ++ # define _EM_OVERFLOW 0x00000004 /* overflow */ ++ # define _EM_ZERODIVIDE 0x00000008 /* zero divide */ ++ # define _EM_INVALID 0x00000010 /* invalid */ ++ { ++ unsigned int mask; ++ _FPU_GETCW(mask); ++ /* Set the FPU control word to abort on most FPEs */ ++ mask &= ~(_EM_UNDERFLOW | _EM_OVERFLOW | _EM_ZERODIVIDE | _EM_INVALID); ++ _FPU_SETCW(mask); ++ } ++ # else ++ { ++ # include <fpu_control.h> ++ #ifndef _FPU_GETCW ++ #define _FPU_GETCW(cw) __asm__ ("fnstcw %0" : "=m" (*&cw)) ++ #endif ++ #ifndef _FPU_SETCW ++ #define _FPU_SETCW(cw) __asm__ ("fldcw %0" : : "m" (*&cw)) ++ #endif ++ unsigned int mask; ++ _FPU_GETCW(mask); ++ /* Set the Linux mask to abort on most FPE's */ ++ /* if bit is set, we _mask_ SIGFPE on that error! */ ++ /* mask &= ~( _FPU_MASK_IM | _FPU_MASK_ZM | _FPU_MASK_OM | _FPU_MASK_UM );*/ ++ mask &= ~( _FPU_MASK_IM | _FPU_MASK_ZM | _FPU_MASK_OM ); ++ _FPU_SETCW(mask); ++ } ++ #endif ++ #endif /* ABORTFP && !__riscos__ */ ++ ++ ++ ++ /* Global flags. set defaults here */ ++ gfp->allow_diff_short=0; ++ gfp->ATHonly=0; ++ gfp->noATH=0; ++ gfp->bWriteVbrTag=1; ++ gfp->cwlimit=0; ++ gfp->disable_reservoir=0; ++ gfp->experimentalX = 0; ++ gfp->experimentalY = 0; ++ gfp->experimentalZ = 0; ++ gfp->frameNum=0; ++ gfp->gtkflag=0; ++ gfp->quality=5; ++ gfp->input_format=sf_unknown; ++ ++ gfp->filter_type=0; ++ gfp->lowpassfreq=0; ++ gfp->highpassfreq=0; ++ gfp->lowpasswidth=-1; ++ gfp->highpasswidth=-1; ++ gfp->lowpass1=0; ++ gfp->lowpass2=0; ++ gfp->highpass1=0; ++ gfp->highpass2=0; ++ gfp->lowpass_band=32; ++ gfp->highpass_band=-1; ++ ++ gfp->no_short_blocks=0; ++ gfp->resample_ratio=1; ++ gfp->padding_type=2; ++ gfp->padding=0; ++ gfp->swapbytes=0; ++ gfp->silent=0; ++ gfp->totalframes=0; ++ gfp->VBR=0; ++ gfp->VBR_q=4; ++ gfp->VBR_min_bitrate_kbps=0; ++ gfp->VBR_max_bitrate_kbps=0; ++ gfp->VBR_min_bitrate=1; ++ gfp->VBR_max_bitrate=13; ++ ++ ++ gfp->version = 1; /* =1 Default: MPEG-1 */ ++ gfp->mode = MPG_MD_JOINT_STEREO; ++ gfp->mode_fixed=0; ++ gfp->force_ms=0; ++ gfp->brate=0; ++ gfp->copyright=0; ++ gfp->original=1; ++ gfp->extension=0; ++ gfp->error_protection=0; ++ gfp->emphasis=0; ++ gfp->in_samplerate=1000*44.1; ++ gfp->out_samplerate=0; ++ gfp->num_channels=2; ++ gfp->num_samples=MAX_U_32_NUM; ++ ++ gfp->inPath=NULL; ++ gfp->outPath=NULL; ++ id3tag.used=0; ++ ++ } ++ ++ ++ ++ /*****************************************************************/ ++ /* flush internal mp3 buffers, */ ++ /*****************************************************************/ ++ int lame_encode_finish(lame_global_flags *gfp,char *mp3buffer, int mp3buffer_size) ++ { ++ int imp3,mp3count,mp3buffer_size_remaining; ++ short int buffer[2][1152]; ++ memset((char *)buffer,0,sizeof(buffer)); ++ mp3count = 0; ++ ++ while (mf_samples_to_encode > 0) { ++ ++ mp3buffer_size_remaining = mp3buffer_size - mp3count; ++ /* if user specifed buffer size = 0, dont check size */ ++ if (mp3buffer_size == 0) mp3buffer_size_remaining=0; ++ imp3=lame_encode(gfp,buffer,mp3buffer,mp3buffer_size_remaining); ++ ++ if (imp3 == -1) { ++ /* fatel error: mp3buffer too small */ ++ desalloc_buffer(&bs); /* Deallocate all buffers */ ++ return -1; ++ } ++ mp3buffer += imp3; ++ mp3count += imp3; ++ mf_samples_to_encode -= gfp->framesize; ++ } ++ ++ ++ gfp->frameNum--; ++ if (!gfp->gtkflag && !gfp->silent) { ++ timestatus(gfp->out_samplerate,gfp->frameNum,gfp->totalframes,gfp->framesize); ++ #ifdef BRHIST ++ if (disp_brhist) ++ { ++ brhist_add_count(); ++ brhist_disp(); ++ brhist_disp_total(gfp); ++ } ++ #endif ++ fprintf(stderr,"\n"); ++ fflush(stderr); ++ } ++ ++ ++ III_FlushBitstream(); ++ mp3buffer_size_remaining = mp3buffer_size - mp3count; ++ /* if user specifed buffer size = 0, dont check size */ ++ if (mp3buffer_size == 0) mp3buffer_size_remaining=0; ++ ++ imp3= copy_buffer(mp3buffer,mp3buffer_size_remaining,&bs); ++ if (imp3 == -1) { ++ /* fatel error: mp3buffer too small */ ++ desalloc_buffer(&bs); /* Deallocate all buffers */ ++ return -1; ++ } ++ ++ mp3count += imp3; ++ desalloc_buffer(&bs); /* Deallocate all buffers */ ++ return mp3count; ++ } ++ ++ ++ /*****************************************************************/ ++ /* write VBR Xing header, and ID3 tag, if asked for */ ++ /*****************************************************************/ ++ void lame_mp3_tags(lame_global_flags *gfp) ++ { ++ if (gfp->bWriteVbrTag) ++ { ++ /* Calculate relative quality of VBR stream ++ * 0=best, 100=worst */ ++ int nQuality=gfp->VBR_q*100/9; ++ /* Write Xing header again */ ++ PutVbrTag(gfp->outPath,nQuality,1-gfp->version); ++ } ++ ++ ++ /* write an ID3 tag */ ++ if(id3tag.used) { ++ id3_buildtag(&id3tag); ++ id3_writetag(gfp->outPath, &id3tag); ++ } ++ } ++ ++ ++ void lame_version(lame_global_flags *gfp,char *ostring) { ++ strncpy(ostring,get_lame_version(),20); ++ } ++ +diff -r -c -N encoder/lame.dsp lame3.70/lame.dsp +*** encoder/lame.dsp Wed Dec 31 17:00:00 1969 +--- lame3.70/lame.dsp Thu Apr 6 12:50:51 2000 +*************** +*** 0 **** +--- 1,336 ---- ++ # Microsoft Developer Studio Project File - Name="lame" - Package Owner=<4> ++ # Microsoft Developer Studio Generated Build File, Format Version 5.00 ++ # ** DO NOT EDIT ** ++ ++ # TARGTYPE "Win32 (x86) Console Application" 0x0103 ++ ++ CFG=lame - Win32 Debug GTK ++ !MESSAGE This is not a valid makefile. To build this project using NMAKE, ++ !MESSAGE use the Export Makefile command and run ++ !MESSAGE ++ !MESSAGE NMAKE /f "lame.mak". ++ !MESSAGE ++ !MESSAGE You can specify a configuration when running NMAKE ++ !MESSAGE by defining the macro CFG on the command line. For example: ++ !MESSAGE ++ !MESSAGE NMAKE /f "lame.mak" CFG="lame - Win32 Debug GTK" ++ !MESSAGE ++ !MESSAGE Possible choices for configuration are: ++ !MESSAGE ++ !MESSAGE "lame - Win32 Release" (based on "Win32 (x86) Console Application") ++ !MESSAGE "lame - Win32 Debug" (based on "Win32 (x86) Console Application") ++ !MESSAGE "lame - Win32 Debug GTK" (based on "Win32 (x86) Console Application") ++ !MESSAGE "lame - Win32 Release GTK" (based on\ ++ "Win32 (x86) Console Application") ++ !MESSAGE ++ ++ # Begin Project ++ # PROP Scc_ProjName "" ++ # PROP Scc_LocalPath "" ++ CPP=xicl.exe ++ RSC=rc.exe ++ ++ !IF "$(CFG)" == "lame - Win32 Release" ++ ++ # PROP BASE Use_MFC 0 ++ # PROP BASE Use_Debug_Libraries 0 ++ # PROP BASE Output_Dir "Release" ++ # PROP BASE Intermediate_Dir "Release" ++ # PROP BASE Target_Dir "" ++ # PROP Use_MFC 0 ++ # PROP Use_Debug_Libraries 0 ++ # PROP Output_Dir "Release" ++ # PROP Intermediate_Dir "Release" ++ # PROP Ignore_Export_Lib 0 ++ # PROP Target_Dir "" ++ # ADD BASE CPP /nologo /W3 /GX /O2 /D "WIN32" /D "NDEBUG" /D "_CONSOLE" /D "_MBCS" /YX /FD /c ++ # ADD CPP /nologo /W3 /GX /Ox /Ot /Og /I "./WinGTK/gtk-plus" /I "./WinGTK/glib-1.2" /D "NDEBUG" /D "WIN32" /D "_CONSOLE" /D "_MBCS" /D "LAMEPARSE" /D "HAVEMPGLIB" /D "LAMESNDFILE" /YX /FD /c ++ # ADD BASE RSC /l 0x409 /d "NDEBUG" ++ # ADD RSC /l 0x409 /d "NDEBUG" ++ BSC32=bscmake.exe ++ # ADD BASE BSC32 /nologo ++ # ADD BSC32 /nologo ++ LINK32=xilink.exe ++ # ADD BASE LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:console /machine:I386 ++ # ADD LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:console /profile /map /machine:I386 ++ ++ !ELSEIF "$(CFG)" == "lame - Win32 Debug" ++ ++ # PROP BASE Use_MFC 0 ++ # PROP BASE Use_Debug_Libraries 1 ++ # PROP BASE Output_Dir "Debug" ++ # PROP BASE Intermediate_Dir "Debug" ++ # PROP BASE Target_Dir "" ++ # PROP Use_MFC 0 ++ # PROP Use_Debug_Libraries 1 ++ # PROP Output_Dir "Debug" ++ # PROP Intermediate_Dir "Debug" ++ # PROP Ignore_Export_Lib 0 ++ # PROP Target_Dir "" ++ # ADD BASE CPP /nologo /W3 /Gm /GX /Zi /Od /D "WIN32" /D "_DEBUG" /D "_CONSOLE" /D "_MBCS" /YX /FD /c ++ # ADD CPP /nologo /W3 /Gm /GX /Zi /Od /D "_DEBUG" /D "WIN32" /D "_CONSOLE" /D "_MBCS" /D BS_FORMAT=BINARY /YX /FD /c ++ # ADD BASE RSC /l 0x409 /d "_DEBUG" ++ # ADD RSC /l 0x409 /d "_DEBUG" ++ BSC32=bscmake.exe ++ # ADD BASE BSC32 /nologo ++ # ADD BSC32 /nologo ++ LINK32=xilink.exe ++ # ADD BASE LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:console /debug /machine:I386 /pdbtype:sept ++ # ADD LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:console /debug /machine:I386 /pdbtype:sept ++ ++ !ELSEIF "$(CFG)" == "lame - Win32 Debug GTK" ++ ++ # PROP BASE Use_MFC 0 ++ # PROP BASE Use_Debug_Libraries 1 ++ # PROP BASE Output_Dir "lame___W" ++ # PROP BASE Intermediate_Dir "lame___W" ++ # PROP BASE Ignore_Export_Lib 0 ++ # PROP BASE Target_Dir "" ++ # PROP Use_MFC 0 ++ # PROP Use_Debug_Libraries 1 ++ # PROP Output_Dir "DebugGTK" ++ # PROP Intermediate_Dir "DebugGTK" ++ # PROP Ignore_Export_Lib 0 ++ # PROP Target_Dir "" ++ # ADD BASE CPP /nologo /Zp4 /W3 /Gm /GX /Zi /Od /I "../WinGTK/gtk-plus" /I "../WinGTK/glib-1.2" /D "_DEBUG" /D "WIN32" /D "_CONSOLE" /D "_MBCS" /D "HAVEGTK" /YX /FD /c ++ # ADD CPP /nologo /W3 /Gm /GX /Zi /Od /I "./WinGTK/gtk-plus" /I "./WinGTK/glib-1.2" /D "_DEBUG" /D "WIN32" /D "_CONSOLE" /D "_MBCS" /D "LAMESNDFILE" /D "HAVEMPGLIB" /D "HAVEGTK" /YX /FD /c ++ # ADD BASE RSC /l 0x409 /d "_DEBUG" ++ # ADD RSC /l 0x409 /d "_DEBUG" ++ BSC32=bscmake.exe ++ # ADD BASE BSC32 /nologo ++ # ADD BSC32 /nologo ++ LINK32=xilink.exe ++ # ADD BASE LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:console /debug /machine:I386 /pdbtype:sept ++ # ADD LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib glib-1.2.lib gdk-1.3.lib gtk-1.3.lib /nologo /subsystem:console /debug /machine:I386 /pdbtype:sept /libpath:"./WinGTK/glib-1.2" /libpath:"./WinGTK/gtk-plus/gdk" /libpath:"./WinGTK/gtk-plus/gtk" ++ ++ !ELSEIF "$(CFG)" == "lame - Win32 Release GTK" ++ ++ # PROP BASE Use_MFC 0 ++ # PROP BASE Use_Debug_Libraries 0 ++ # PROP BASE Output_Dir "lame___0" ++ # PROP BASE Intermediate_Dir "lame___0" ++ # PROP BASE Ignore_Export_Lib 0 ++ # PROP BASE Target_Dir "" ++ # PROP Use_MFC 0 ++ # PROP Use_Debug_Libraries 0 ++ # PROP Output_Dir "ReleaseGTK" ++ # PROP Intermediate_Dir "ReleaseGTK" ++ # PROP Ignore_Export_Lib 0 ++ # PROP Target_Dir "" ++ # ADD BASE CPP /nologo /W3 /GX /O2 /I "../WinGTK/gtk-plus" /I "../WinGTK/glib-1.2" /D "NDEBUG" /D "WIN32" /D "_CONSOLE" /D "_MBCS" /D "HAVEGTK" /YX /FD /c ++ # ADD CPP /nologo /W3 /GX /O2 /I "./WinGTK/gtk-plus" /I "./WinGTK/glib-1.2" /D "NDEBUG" /D "WIN32" /D "_CONSOLE" /D "_MBCS" /D "LAMEPARSE" /D "HAVEMPGLIB" /D "LAMESNDFILE" /D "HAVEGTK" /YX /FD /c ++ # ADD BASE RSC /l 0x409 /d "NDEBUG" ++ # ADD RSC /l 0x409 /d "NDEBUG" ++ BSC32=bscmake.exe ++ # ADD BASE BSC32 /nologo ++ # ADD BSC32 /nologo ++ LINK32=xilink.exe ++ # ADD BASE LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:console /profile /map /machine:I386 ++ # ADD LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib glib-1.2.lib gdk-1.3.lib gtk-1.3.lib /nologo /subsystem:console /profile /map /machine:I386 /libpath:"./WinGTK/glib-1.2" /libpath:"./WinGTK/gtk-plus/gdk" /libpath:"./WinGTK/gtk-plus/gtk" ++ ++ !ENDIF ++ ++ # Begin Target ++ ++ # Name "lame - Win32 Release" ++ # Name "lame - Win32 Debug" ++ # Name "lame - Win32 Debug GTK" ++ # Name "lame - Win32 Release GTK" ++ # Begin Source File ++ ++ SOURCE=.\encoder.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\fft.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\fft.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\formatBitstream.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\formatBitstream.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\get_audio.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\get_audio.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\gpkplotting.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\gpkplotting.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\gtkanal.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\gtkanal.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\id3tag.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\id3tag.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\ieeefloat.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\ieeefloat.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=".\l3bitstream-pvt.h" ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\l3bitstream.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\l3bitstream.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\l3side.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\lame.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\lame.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\machine.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\main.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\newmdct.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\newmdct.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\parse.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\portableio.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\portableio.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\psymodel.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\psymodel.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=".\quantize-pvt.c" ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=".\quantize-pvt.h" ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\quantize.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\quantize.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\reservoir.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\reservoir.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\tables.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\tables.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\takehiro.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\timestatus.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\timestatus.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\util.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\util.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\vbrquantize.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\VbrTag.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\VbrTag.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\version.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\version.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\WinGTK.txt ++ # End Source File ++ # End Target ++ # End Project +diff -r -c -N encoder/lame.dsw lame3.70/lame.dsw +*** encoder/lame.dsw Wed Dec 31 17:00:00 1969 +--- lame3.70/lame.dsw Wed Nov 24 01:43:20 1999 +*************** +*** 0 **** +--- 1,44 ---- ++ Microsoft Developer Studio Workspace File, Format Version 5.00 ++ # WARNING: DO NOT EDIT OR DELETE THIS WORKSPACE FILE! ++ ++ ############################################################################### ++ ++ Project: "lame"=.\lame.dsp - Package Owner=<4> ++ ++ Package=<5> ++ {{{ ++ }}} ++ ++ Package=<4> ++ {{{ ++ Begin Project Dependency ++ Project_Dep_Name mpglib ++ End Project Dependency ++ }}} ++ ++ ############################################################################### ++ ++ Project: "mpglib"=.\mpglib\mpglib.dsp - Package Owner=<4> ++ ++ Package=<5> ++ {{{ ++ }}} ++ ++ Package=<4> ++ {{{ ++ }}} ++ ++ ############################################################################### ++ ++ Global: ++ ++ Package=<5> ++ {{{ ++ }}} ++ ++ Package=<3> ++ {{{ ++ }}} ++ ++ ############################################################################### ++ +diff -r -c -N encoder/lame.h lame3.70/lame.h +*** encoder/lame.h Wed Dec 31 17:00:00 1969 +--- lame3.70/lame.h Thu Apr 6 12:50:51 2000 +*************** +*** 0 **** +--- 1,336 ---- ++ /* ++ * Interface to MP3 LAME encoding engine ++ * ++ * Copyright (c) 1999 Mark Taylor ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License as published by ++ * the Free Software Foundation; either version 2, or (at your option) ++ * any later version. ++ * ++ * This program is distributed in the hope that it will be useful, ++ * but WITHOUT ANY WARRANTY; without even the implied warranty of ++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the ++ * GNU General Public License for more details. ++ * ++ * You should have received a copy of the GNU General Public License ++ * along with this program; see the file COPYING. If not, write to ++ * the Free Software Foundation, 675 Mass Ave, Cambridge, MA 02139, USA. ++ */ ++ #ifndef LAME_H_INCLUDE ++ #define LAME_H_INCLUDE ++ #include <stdio.h> ++ ++ /* maximum size of mp3buffer needed if you encode at most 1152 samples for ++ each call to lame_encode_buffer. see lame_encode_buffer() below */ ++ #define LAME_MAXMP3BUFFER 16384 ++ ++ ++ typedef enum sound_file_format_e { ++ sf_unknown, sf_wave, sf_aiff, sf_mp3, sf_raw ++ } sound_file_format; ++ ++ ++ /*********************************************************************** ++ * ++ * Global Variables. ++ * ++ * substantiated in lame.c ++ * ++ * Initilized and default values set by gf=lame_init() ++ * gf is a pointer to this struct, which the user may use to ++ * override any of the default values ++ * ++ * a call to lame_set_params() is also needed ++ * ++ ***********************************************************************/ ++ typedef struct { ++ /* input file description */ ++ unsigned long num_samples; /* number of samples. default=2^32-1 */ ++ int num_channels; /* input number of channels. default=2 */ ++ int in_samplerate; /* input_samp_rate. default=44.1kHz */ ++ int out_samplerate; /* output_samp_rate. (usually determined automatically) */ ++ ++ ++ /* general control params */ ++ int gtkflag; /* run frame analyzer? */ ++ int bWriteVbrTag; /* add Xing VBR tag? */ ++ int quality; /* quality setting 0=best, 9=worst */ ++ int silent; /* disable some status output */ ++ int mode; /* 0,1,2,3 stereo,jstereo,dual channel,mono */ ++ int mode_fixed; /* use specified the mode, do not use lame's opinion of the best mode */ ++ int force_ms; /* force M/S mode. requires mode=1 */ ++ int brate; /* bitrate */ ++ ++ /* frame params */ ++ int copyright; /* mark as copyright. default=0 */ ++ int original; /* mark as original. default=1 */ ++ int error_protection; /* use 2 bytes per frame for a CRC checksum. default=0*/ ++ int padding_type; /* 0=no padding, 1=always pad, 2=adjust padding */ ++ int extension; /* the MP3 'private extension' bit. meaningless */ ++ ++ /* quantization/noise shaping */ ++ int disable_reservoir; /* use bit reservoir? */ ++ int experimentalX; ++ int experimentalY; ++ int experimentalZ; ++ ++ /* VBR control */ ++ int VBR; ++ int VBR_q; ++ int VBR_min_bitrate_kbps; ++ int VBR_max_bitrate_kbps; ++ ++ ++ /* resampling and filtering */ ++ int lowpassfreq; /* freq in Hz. 0=lame choses. -1=no filter */ ++ int highpassfreq; /* freq in Hz. 0=lame choses. -1=no filter */ ++ int lowpasswidth; /* freq width of filter, in Hz (default=15%)*/ ++ int highpasswidth; /* freq width of filter, in Hz (default=15%)*/ ++ ++ ++ /* input file reading - not used if calling program does the i/o */ ++ sound_file_format input_format; ++ int swapbytes; /* force byte swapping default=0*/ ++ char *inPath; /* name of input file */ ++ char *outPath; /* name of output file. */ ++ /* Note: outPath must be set if you want Xing VBR or id3 tags ++ * written */ ++ ++ ++ /* psycho acoustics and other aguments which you should not change ++ * unless you know what you are doing */ ++ int ATHonly; /* only use ATH */ ++ int noATH; /* disable ATH */ ++ float cwlimit; /* predictability limit */ ++ int allow_diff_short; /* allow blocktypes to differ between channels ? */ ++ int no_short_blocks; /* disable short blocks */ ++ int emphasis; /* obsolete */ ++ ++ ++ ++ /********************************************************************/ ++ /* internal variables NOT set by calling program, and should not be */ ++ /* modified by the calling program */ ++ /********************************************************************/ ++ long int frameNum; /* frame counter */ ++ long totalframes; /* frames: 0..totalframes-1 (estimate)*/ ++ int encoder_delay; ++ int framesize; ++ int version; /* 0=MPEG2 1=MPEG1 */ ++ int padding; /* padding for the current frame? */ ++ int mode_gr; /* granules per frame */ ++ int stereo; /* number of channels */ ++ int VBR_min_bitrate; /* min bitrate index */ ++ int VBR_max_bitrate; /* max bitrate index */ ++ float resample_ratio; /* input_samp_rate/output_samp_rate */ ++ int bitrate_index; ++ int samplerate_index; ++ int mode_ext; ++ ++ /* lowpass and highpass filter control */ ++ float lowpass1,lowpass2; /* normalized frequency bounds of passband */ ++ float highpass1,highpass2; /* normalized frequency bounds of passband */ ++ ++ /* polyphase filter (filter_type=0) */ ++ int lowpass_band; /* zero bands >= lowpass_band in the polyphase filterbank */ ++ int highpass_band; /* zero bands <= highpass_band */ ++ ++ ++ ++ int filter_type; /* 0=polyphase filter, 1= FIR filter 2=MDCT filter(bad)*/ ++ int quantization; /* 0 = ISO formual, 1=best amplitude */ ++ int noise_shaping; /* 0 = none ++ 1 = ISO AAC model ++ 2 = allow scalefac_select=1 ++ */ ++ ++ int noise_shaping_stop; /* 0 = stop at over=0, all scalefacs amplified or ++ a scalefac has reached max value ++ 1 = stop when all scalefacs amplified or ++ a scalefac has reached max value ++ 2 = stop when all scalefacs amplified ++ */ ++ ++ int psymodel; /* 0 = none 1=gpsycho */ ++ int use_best_huffman; /* 0 = no. 1=outside loop 2=inside loop(slow) */ ++ ++ ++ } lame_global_flags; ++ ++ ++ ++ ++ ++ ++ /* ++ ++ The LAME API ++ ++ */ ++ ++ ++ /* REQUIRED: initialize the encoder. sets default for all encoder paramters, ++ * returns pointer to encoder parameters listed above ++ */ ++ void lame_init(lame_global_flags *); ++ ++ ++ ++ ++ /********************************************************************* ++ * command line argument parsing & option setting. Only supported ++ * if libmp3lame compiled with LAMEPARSE defined ++ *********************************************************************/ ++ /* OPTIONAL: call this to print an error with a brief command line usage guide and quit ++ * only supported if libmp3lame compiled with LAMEPARSE defined. ++ */ ++ void lame_usage(lame_global_flags *, char *); ++ ++ /* OPTIONAL: call this to print a command line interface usage guide and quit */ ++ void lame_help(lame_global_flags *, char *); ++ ++ /* OPTIONAL: get the version number, in a string. of the form: "3.63 (beta)" or ++ just "3.63". Max allows length is 20 characters */ ++ void lame_version(lame_global_flags *, char *); ++ ++ ++ /* OPTIONAL: set internal options via command line argument parsing ++ * You can skip this call if you like the default values, or if ++ * set the encoder parameters your self ++ */ ++ void lame_parse_args(lame_global_flags *, int argc, char **argv); ++ ++ ++ ++ ++ ++ /* REQUIRED: sets more internal configuration based on data provided ++ * above ++ */ ++ void lame_init_params(lame_global_flags *); ++ ++ ++ /* OPTONAL: print internal lame configuration on stderr*/ ++ void lame_print_config(lame_global_flags *); ++ ++ ++ ++ ++ /* input pcm data, output (maybe) mp3 frames. ++ * This routine handles all buffering, resampling and filtering for you. ++ * ++ * leftpcm[] array of 16bit pcm data, left channel ++ * rightpcm[] array of 16bit pcm data, right channel ++ * num_samples number of samples in leftpcm[] and rightpcm[] (if stereo) ++ * mp3buffer pointer to buffer where mp3 output is written ++ * mp3buffer_size size of mp3buffer, in bytes ++ * return code number of bytes output in mp3buffer. can be 0 ++ * if return code = -1: mp3buffer was too small ++ * ++ * The required mp3buffer_size can be computed from num_samples, ++ * samplerate and encoding rate, but here is a worst case estimate: ++ * ++ * mp3buffer_size in bytes = 1.25*num_samples + 7200 ++ * ++ * I think a tighter bound could be: (mt, March 2000) ++ * MPEG1: ++ * num_samples*(bitrate/8)/samplerate + 4*1152*(bitrate/8)/samplerate + 512 ++ * MPEG2: ++ * num_samples*(bitrate/8)/samplerate + 4*576*(bitrate/8)/samplerate + 256 ++ * ++ * but test first if you use that! ++ * ++ * set mp3buffer_size = 0 and LAME will not check if mp3buffer_size is ++ * large enough. ++ * ++ * NOTE: if gfp->num_channels=2, but gfp->mode = 3 (mono), the L & R channels ++ * will be averaged into the L channel before encoding only the L channel ++ * This will overwrite the data in leftpcm[] and rightpcm[]. ++ * ++ */ ++ int lame_encode_buffer(lame_global_flags *,short int leftpcm[], short int rightpcm[],int num_samples, ++ char *mp3buffer,int mp3buffer_size); ++ ++ /* as above, but input has L & R channel data interleaved. Note: ++ * num_samples = number of samples in the L (or R) ++ * channel, not the total number of samples in pcm[] ++ */ ++ int lame_encode_buffer_interleaved(lame_global_flags *,short int pcm[], ++ int num_samples, char *mp3buffer,int mp3buffer_size); ++ ++ ++ ++ /* input 1 pcm frame, output (maybe) 1 mp3 frame. ++ * return code = number of bytes output in mp3buffer. can be 0 ++ * NOTE: this interface is outdated, please use lame_encode_buffer() instead ++ * declair mp3buffer with: char mp3buffer[LAME_MAXMP3BUFFER] ++ * if return code = -1: mp3buffer was too small ++ */ ++ int lame_encode(lame_global_flags *,short int Buffer[2][1152],char *mp3buffer,int mp3buffer_size); ++ ++ ++ ++ /* REQUIRED: lame_encode_finish will flush the buffers and may return a ++ * final few mp3 frames. mp3buffer should be at least 7200 bytes. ++ * ++ * return code = number of bytes output to mp3buffer. can be 0 ++ */ ++ int lame_encode_finish(lame_global_flags *,char *mp3buffer, int size); ++ ++ ++ /* OPTIONAL: lame_mp3_tags will append id3 and Xing VBR tags to ++ the mp3 file with name given by gf->outPath. These calls open the file, ++ write tags, and close the file, so make sure the the encoding is finished ++ before calling these routines. ++ Note: if VBR and id3 tags are turned off by the user, or turned off ++ by LAME because the output is not a regular file, this call does nothing ++ */ ++ void lame_mp3_tags(lame_global_flags *); ++ ++ ++ ++ ++ /********************************************************************* ++ * lame file i/o. Only supported ++ * if libmp3lame compiled with LAMESNDFILE or LIBSNDFILE ++ *********************************************************************/ ++ /* OPTIONAL: open the input file, and parse headers if possible ++ * you can skip this call if you will do your own PCM input ++ */ ++ void lame_init_infile(lame_global_flags *); ++ ++ /* OPTIONAL: read one frame of PCM data from audio input file opened by ++ * lame_init_infile. Input file can be wav, aiff, raw pcm, anything ++ * supported by libsndfile, or an mp3 file ++ */ ++ int lame_readframe(lame_global_flags *,short int Buffer[2][1152]); ++ ++ /* OPTIONAL: close the sound input file if lame_init_infile() was used */ ++ void lame_close_infile(lame_global_flags *); ++ ++ ++ ++ ++ ++ /********************************************************************* ++ * a simple interface to mpglib, part of mpg123, is also included if ++ * libmp3lame is compiled with HAVEMPGLIB ++ * input 1 mp3 frame, output (maybe) 1 pcm frame. ++ * lame_decode return code: -1: error. 0: need more data. n>0: size of pcm output ++ *********************************************************************/ ++ int lame_decode_init(void); ++ int lame_decode(char *mp3buf,int len,short pcm_l[],short pcm_r[]); ++ /* read mp3 file until mpglib returns one frame of PCM data */ ++ #ifdef AMIGA_MPEGA ++ int lame_decode_initfile(const char *fullname,int *stereo,int *samp,int *bitrate, unsigned long *nsamp); ++ int lame_decode_fromfile(FILE *fd,short int pcm_l[], short int pcm_r[]); ++ #else ++ int lame_decode_initfile(FILE *fd,int *stereo,int *samp,int *bitrate, unsigned long *nsamp); ++ int lame_decode_fromfile(FILE *fd,short int pcm_l[],short int pcm_r[]); ++ #endif ++ ++ ++ ++ ++ #endif +diff -r -c -N encoder/libmp3lame/CVS/Entries lame3.70/libmp3lame/CVS/Entries +*** encoder/libmp3lame/CVS/Entries Wed Dec 31 17:00:00 1969 +--- lame3.70/libmp3lame/CVS/Entries Thu Apr 6 12:50:37 2000 +*************** +*** 0 **** +--- 1 ---- ++ D +diff -r -c -N encoder/libmp3lame/CVS/Repository lame3.70/libmp3lame/CVS/Repository +*** encoder/libmp3lame/CVS/Repository Wed Dec 31 17:00:00 1969 +--- lame3.70/libmp3lame/CVS/Repository Thu Apr 6 12:50:37 2000 +*************** +*** 0 **** +--- 1 ---- ++ lame/libmp3lame +diff -r -c -N encoder/libmp3lame/CVS/Root lame3.70/libmp3lame/CVS/Root +*** encoder/libmp3lame/CVS/Root Wed Dec 31 17:00:00 1969 +--- lame3.70/libmp3lame/CVS/Root Thu Apr 6 12:50:37 2000 +*************** +*** 0 **** +--- 1 ---- ++ markt@cvs.lame.sourceforge.net:/cvsroot/lame +diff -r -c -N encoder/libmp3lame/CVS/Tag lame3.70/libmp3lame/CVS/Tag +*** encoder/libmp3lame/CVS/Tag Wed Dec 31 17:00:00 1969 +--- lame3.70/libmp3lame/CVS/Tag Thu Apr 6 12:50:53 2000 +*************** +*** 0 **** +--- 1 ---- ++ Tlame3_70 +diff -r -c -N encoder/loop-pvt.h lame3.70/loop-pvt.h +*** encoder/loop-pvt.h Wed Jan 22 02:43:16 1997 +--- lame3.70/loop-pvt.h Wed Dec 31 17:00:00 1969 +*************** +*** 1,119 **** +- /********************************************************************** +- * ISO MPEG Audio Subgroup Software Simulation Group (1996) +- * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- * +- * $Id: loop-pvt.h,v 1.1 1996/02/14 04:04:23 rowlands Exp $ +- * +- * Private interface declarations for loop.c +- * +- * $Log: loop-pvt.h,v $ +- * Revision 1.1 1996/02/14 04:04:23 rowlands +- * Initial revision +- * +- * Received from Mike Coleman +- **********************************************************************/ +- +- #ifndef LOOP_PVT_H +- #define LOOP_PVT_H +- +- /* +- Revision History: +- +- Date Programmer Comment +- ========== ========================= =============================== +- 1995/10/01 mc@fivebats.com created +- +- */ +- +- int outer_loop( double xr[2][2][576], /*vector of the magnitudees of the spectral values */ +- int max_bits, +- III_psy_xmin *l3_xmin, /* the allowed distortion of the scalefactor */ +- int l3_enc[2][2][576], /* vector of quantized values ix(0..575) */ +- frame_params *fr_ps, +- III_scalefac_t *scalefac, /* scalefactors */ +- int gr, +- int ch, +- III_side_info_t *l3_side ); +- +- int part2_length( III_scalefac_t *scalefac, +- frame_params *fr_ps, +- int gr, +- int ch, +- III_side_info_t *si ); +- +- int quantanf_init( double xr[576] ); +- +- int inner_loop( double xr[2][2][576], +- int l3_enc[2][2][576], +- int max_bits, +- gr_info *cod_info, +- int gr, +- int ch ); +- void calc_xmin( double xr[2][2][576], +- III_psy_ratio *ratio, +- gr_info *cod_info, +- III_psy_xmin *l3_xmin, +- int gr, +- int ch ); +- double xr_max( double xr[576], +- unsigned int begin, +- unsigned int end ); +- +- void calc_scfsi( double xr[576], +- III_side_info_t *l3_side, +- III_psy_xmin *l3_xmin, +- int ch, +- int gr ); +- +- void gr_deco( gr_info *cod_info ); +- +- +- int count_bit( int ix[576], unsigned int start, unsigned int end, unsigned int table); +- int bigv_bitcount( int ix[576], gr_info *cod_info ); +- int choose_table( int max); +- void bigv_tab_select( int ix[576], gr_info *cod_info ); +- void subdivide( gr_info *cod_info ); +- int count1_bitcount( int ix[576], gr_info *cod_info ); +- void calc_runlen( int ix[576], +- gr_info *cod_info ); +- int scale_bitcount( III_scalefac_t *scalefac, +- gr_info *cod_info, +- int gr, +- int ch ); +- void calc_noise( double xr[576], +- int ix[576], +- gr_info *cod_info, +- double xfsf[4][CBLIMIT] ); +- +- +- int loop_break( III_scalefac_t *scalefac, +- gr_info *cod_info, +- int gr, +- int ch ); +- void preemphasis( double xr[576], +- double xfsf[4][CBLIMIT], +- III_psy_xmin *l3_xmin, +- int gr, +- int ch, +- III_side_info_t *l3_side ); +- int amp_scalefac_bands( double xr[576], +- double xfsf[4][CBLIMIT], +- III_psy_xmin *l3_xmin, +- III_side_info_t *l3_side, +- III_scalefac_t *scalefac, +- int gr, +- int ch, +- int iteration ); +- void quantize( double xr[576], +- int ix[576], +- gr_info *cod_info ); +- int ix_max( int ix[576], +- unsigned int begin, +- unsigned int end ); +- +- +- int +- new_choose_table( int ix[576], +- unsigned int begin, +- unsigned int end ); +- #endif +--- 0 ---- +diff -r -c -N encoder/loop.c lame3.70/loop.c +*** encoder/loop.c Wed Jan 22 02:43:16 1997 +--- lame3.70/loop.c Wed Dec 31 17:00:00 1969 +*************** +*** 1,2220 **** +- /********************************************************************** +- * ISO MPEG Audio Subgroup Software Simulation Group (1996) +- * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- * +- * $Id: loop.c,v 1.2 1997/01/19 22:28:29 rowlands Exp $ +- * +- * $Log: loop.c,v $ +- * Revision 1.2 1997/01/19 22:28:29 rowlands +- * Layer 3 bug fixes from Seymour Shlien +- * +- * Revision 1.1 1996/02/14 04:04:23 rowlands +- * Initial revision +- * +- * Received from Mike Coleman +- **********************************************************************/ +- /********************************************************************** +- * date programmers comment * +- *25. 6.92 Toshiyuki Ishino Ver 1.0 * +- *29.10.92 Masahiro Iwadare Ver 2.0 * +- *17. 4.93 Masahiro Iwadare Updated for IS Modification * +- *04.11.93 Seymour Shlien Speed up inner loop * +- *09.09.95 mc@fivebats.com Several changes for updated IS,* +- * and some MPEG2-LSF support * +- *20.12.96 seymour.shlien@crc.doc.ca Fixed some bugs and improved * +- * the appearance * +- *********************************************************************/ +- +- #include <stdio.h> +- #include <stdlib.h> +- #include <math.h> +- #include <assert.h> +- #include "l3side.h" +- #include "loop.h" +- #include "huffman.h" +- #include "l3bitstream.h" +- #include "reservoir.h" +- #include "loop-pvt.h" +- +- /* #define DEBUG */ +- /* #define DEBUGSC */ +- /* #define PERFORM 3 */ +- /* If PERFORM is defined to some number, then a file encode.log +- is preduced showing the intermediate results produced by the +- outer_loop code for the frame number = PERFORM. +- */ +- #define BIN_SEARCH +- /* for speeding up the iteration_loop algorithm */ +- +- #ifdef PERFORM +- FILE *log_output; +- extern int frameNum; +- float worst_xfsf_to_xmin_ratio(III_psy_xmin *l3_xmin, double xfsf[4][CBLIMIT] +- ,int block_type,int gr,int ch); +- #endif +- +- /* New SS 20-12-96 */ +- #ifdef BIN_SEARCH +- int bin_search_StepSize(int desired_rate, double start, int ix[576], +- double xrs[576], gr_info * cod_info); +- int count_bits(); +- float worst_xfsf_to_xmin_ratio(); +- #endif +- +- +- /* +- Here are MPEG1 Table B.8 and MPEG2 Table B.1 +- -- Layer III scalefactor bands. +- Index into this using a method such as: +- idx = fr_ps->header->sampling_frequency +- + (fr_ps->header->version * 3) +- */ +- +- struct scalefac_struct sfBandIndex[6] = +- { +- +- { /* Table B.2.b: 22.05 kHz */ +- {0,6,12,18,24,30,36,44,54,66,80,96,116,140,168,200,238,284,336,396,464,522,576}, +- {0,4,8,12,18,24,32,42,56,74,100,132,174,192} +- }, +- { /* Table B.2.c: 24 kHz */ +- {0,6,12,18,24,30,36,44,54,66,80,96,114,136,162,194,232,278,330,394,464,540,576}, +- {0,4,8,12,18,26,36,48,62,80,104,136,180,192} +- }, +- { /* Table B.2.a: 16 kHz */ +- {0,6,12,18,24,30,36,44,45,66,80,96,116,140,168,200,238,248,336,396,464,522,576}, +- {0,4,8,12,18,26,36,48,62,80,104,134,174,192} +- }, +- { /* Table B.8.b: 44.1 kHz */ +- {0,4,8,12,16,20,24,30,36,44,52,62,74,90,110,134,162,196,238,288,342,418,576}, +- {0,4,8,12,16,22,30,40,52,66,84,106,136,192} +- }, +- { /* Table B.8.c: 48 kHz */ +- {0,4,8,12,16,20,24,30,36,42,50,60,72,88,106,128,156,190,230,276,330,384,576}, +- {0,4,8,12,16,22,28,38,50,64,80,100,126,192} +- }, +- { /* Table B.8.a: 32 kHz */ +- {0,4,8,12,16,20,24,30,36,44,54,66,82,102,126,156,194,240,296,364,448,550,576}, +- {0,4,8,12,16,22,30,42,58,78,104,138,180,192} +- } +- }; +- +- /* +- The following table is used to implement the scalefactor +- partitioning for MPEG2 as described in section +- 2.4.3.2 of the IS. The indexing corresponds to the +- way the tables are presented in the IS: +- +- [table_number][row_in_table][column of nr_of_sfb] +- */ +- static unsigned nr_of_sfb_block[6][3][4] = +- { +- { +- {6, 5, 5, 5}, +- {9, 9, 9, 9}, +- {6, 9, 9, 9} +- }, +- { +- {6, 5, 7, 3}, +- {9, 9, 12, 6}, +- {6, 9, 12, 6} +- }, +- { +- {11, 10, 0, 0}, +- {18, 18, 0, 0}, +- {15,18,0,0} +- }, +- { +- {7, 7, 7, 0}, +- {12, 12, 12, 0}, +- {6, 15, 12, 0} +- }, +- { +- {6, 6, 6, 3}, +- {12, 9, 9, 6}, +- {6, 12, 9, 6} +- }, +- { +- {8, 8, 5, 0}, +- {15,12,9,0}, +- {6,18,9,0} +- } +- }; +- +- /* +- table of largest scalefactors for MPEG2 +- */ +- static unsigned max_sfac_tab[6][4] = +- { +- {4, 4, 3, 3}, +- {4, 4, 3, 0}, +- {3, 2, 0, 0}, +- {4, 5, 5, 0}, +- {3, 3, 3, 0}, +- {2, 2, 0, 0} +- }; +- +- /* Table B.6: layer3 preemphasis */ +- int pretab[21] = +- { +- 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, +- 1, 1, 1, 1, 2, 2, 3, 3, 3, 2 +- }; +- +- /* This is the scfsi_band table from 2.4.2.7 of the IS */ +- int scfsi_band_long[5] = { 0, 6, 11, 16, 21 }; +- +- int *scalefac_band_long = &sfBandIndex[3].l[0]; +- int *scalefac_band_short = &sfBandIndex[3].s[0]; +- +- +- +- /************************************************************************/ +- /* iteration_loop() */ +- /************************************************************************/ +- void +- iteration_loop( double pe[][2], double xr_org[2][2][576], III_psy_ratio *ratio, +- III_side_info_t *l3_side, int l3_enc[2][2][576], +- int mean_bits, int stereo, double xr_dec[2][2][576], +- III_scalefac_t *scalefac, frame_params *fr_ps, +- int ancillary_pad, int bitsPerFrame ) +- { +- static int firstcall = 1; +- III_psy_xmin l3_xmin; +- gr_info *cod_info; +- layer *info; +- int *main_data_begin; +- +- int max_bits; +- int ch, gr, sfb, i, mode_gr; +- static int huffman_read_flag = 0; +- +- double xr[2][2][576]; +- I576 *ix; +- +- main_data_begin = &l3_side->main_data_begin; +- l3_side->resvDrain = 0; +- +- if ( firstcall ) +- { +- *main_data_begin = 0; +- firstcall = 0; +- #ifdef PERFORM +- log_output = fopen("encode.log","w"); +- #endif +- } +- +- info = fr_ps->header; +- mode_gr = (info->version == 1) ? 2 : 1; +- +- scalefac_band_long = &sfBandIndex[info->sampling_frequency + (info->version * 3)].l[0]; +- scalefac_band_short = &sfBandIndex[info->sampling_frequency + (info->version * 3)].s[0]; +- +- /* reading huffman code table */ +- if (huffman_read_flag == 0) { +- FILE *fi = OpenTableFile( "huffcode" ); +- if ( fi == NULL ) +- exit( EXIT_FAILURE ); +- read_huffcodetab( fi ); +- huffman_read_flag++; +- fclose( fi ); +- } +- +- +- for ( gr = 0; gr < mode_gr; gr++ ) +- { +- for ( ch = 0; ch < stereo; ch++ ) +- { +- for ( i = 0; i < 576; i++ ) +- xr[gr][ch][i] = xr_org[gr][ch][i]; +- } +- } +- +- ResvFrameBegin( fr_ps, l3_side, mean_bits, bitsPerFrame ); +- +- for ( gr = 0; gr < mode_gr; gr++ ) +- { +- for ( ch = 0; ch < stereo; ch++ ) +- { +- ix = (I576 *) l3_enc[gr][ch]; +- cod_info = (gr_info *) &(l3_side->gr[gr].ch[ch]); +- gr_deco(cod_info); +- calc_xmin( xr, ratio, cod_info, &l3_xmin, gr, ch ); +- +- #ifdef DEBUG +- printf( "----- start gr[%d] ch[%1d] : block_type=%1d, window_switching_flag=%1d (loop)\n", +- gr,ch, cod_info->block_type, cod_info->window_switching_flag ); +- #endif +- +- if ( info->version == 1 ) +- calc_scfsi( xr[gr][ch], l3_side, &l3_xmin, ch, gr ); +- +- /* calculation of number of available bit( per granule ) */ +- max_bits = ResvMaxBits( fr_ps, l3_side, &pe[gr][ch], mean_bits ); +- #ifdef DEBUG +- printf( " max_bits = %d, mean_bits = %d (iteration_loop)\n", max_bits, mean_bits ); +- #endif +- +- /* reset of iteration variables */ +- +- for ( sfb = 0; sfb < 21; sfb++ ) +- scalefac->l[gr][ch][sfb] = 0; +- for ( sfb = 0; sfb < 13; sfb++ ) +- for ( i = 0; i < 3; i++ ) +- scalefac->s[gr][ch][sfb][i] = 0; +- +- for ( i = 0; i < 4; i++ ) +- cod_info->slen[i] = 0; +- cod_info->sfb_partition_table = &nr_of_sfb_block[0][0][0]; +- +- cod_info->part2_3_length = 0; +- cod_info->big_values = 0; +- cod_info->count1 = 0; +- cod_info->scalefac_compress = 0; +- cod_info->table_select[0] = 0; +- cod_info->table_select[1] = 0; +- cod_info->table_select[2] = 0; +- cod_info->subblock_gain[0] = 0; +- cod_info->subblock_gain[1] = 0; +- cod_info->subblock_gain[2] = 0; +- cod_info->region0_count = 0; +- cod_info->region1_count = 0; +- cod_info->part2_length = 0; +- cod_info->preflag = 0; +- cod_info->scalefac_scale = 0; +- cod_info->quantizerStepSize = 0.0; +- cod_info->count1table_select= 0; +- +- /* all spectral values zero ? */ +- if ( fabs(xr_max(xr[gr][ch], 0, 576)) != 0.0 ) +- { +- cod_info->quantizerStepSize = +- (double) quantanf_init( xr[gr][ch] ); +- cod_info->part2_3_length = outer_loop( xr, max_bits, &l3_xmin, +- l3_enc, fr_ps, scalefac, +- gr, ch, l3_side ); +- } +- ResvAdjust( fr_ps, cod_info, l3_side, mean_bits ); +- +- cod_info->global_gain = nint( cod_info->quantizerStepSize + 210.0 ); +- assert( cod_info->global_gain < 256 ); +- } /* for ch */ +- } /* for gr */ +- ResvFrameEnd( fr_ps, l3_side, mean_bits ); +- } +- +- +- +- /************************************************************************/ +- /* quantanf_init */ +- /************************************************************************/ +- int quantanf_init( double xr[576] ) +- /* Function: Calculate the first quantization step quantanf. */ +- { +- int i, tp = 0; +- double system_const, minlimit; +- double sfm = 0.0, sum1 = 0.0, sum2 = 0.0; +- +- system_const = 8.0; +- minlimit = -100.0; +- +- for ( i = 0; i < 576; i++ ) +- { +- if ( xr[i] != 0 ) +- { +- double tpd = xr[i] * xr[i]; +- sum1 += log( tpd ); +- sum2 += tpd; +- } +- } +- if ( sum2 != 0.0 ) +- { +- sfm = exp( sum1 / 576.0 ) / (sum2 / 576.0); +- tp = nint( system_const * log(sfm) ); +- if ( tp < minlimit ) +- tp = minlimit; +- #ifdef DEBUG +- printf(" quantanf = %d (quantanf_init)\n",tp ); +- #endif +- } +- return(tp-70.0); /* SS 19-12-96. Starting value of +- global_gain or quantizerStepSize +- has to be reduced for iteration_loop +- */ +- } +- +- +- +- +- +- /************************************************************************/ +- /* outer_loop */ +- /************************************************************************/ +- /* Function: The outer iteration loop controls the masking conditions */ +- /* of all scalefactorbands. It computes the best scalefac and */ +- /* global gain. This module calls the inner iteration loop */ +- /************************************************************************/ +- int outer_loop( +- double xr[2][2][576], /* magnitudes of the spectral values */ +- int max_bits, +- III_psy_xmin *l3_xmin, /* the allowed distortion of the scalefactor */ +- int l3_enc[2][2][576], /* vector of quantized values ix(0..575) */ +- frame_params *fr_ps, +- III_scalefac_t *scalefac, /* scalefactors */ +- int gr, int ch, III_side_info_t *l3_side ) +- { +- int status ; +- int scalesave_l[CBLIMIT], scalesave_s[CBLIMIT][3]; +- int sfb, bits, huff_bits, save_preflag, save_compress; +- double xfsf[4][CBLIMIT]; +- int i, over, iteration; +- float max_ratio; +- +- +- /* D576 *xrs; */ /* to eliminate warning messages from gcc compiler */ +- /* I576 *ix; */ /* replace this code with below. S. Shlien 15-1-97 */ +- +- double *xrs; +- int *ix; +- gr_info *cod_info = &l3_side->gr[gr].ch[ch].tt; +- +- /* xrs = (D576 *) &xr[gr][ch][0]; */ +- /* ix = (I576 *) l3_enc[gr][ch]; */ +- +- +- xrs = (double *) &(xr[gr][ch][0]); +- ix = (int *) &(l3_enc[gr][ch][0]); +- +- +- iteration = 0; +- #ifdef PERFORM +- if(frameNum == PERFORM) +- fprintf(log_output,"\n\nframe = %d ch = %d gr= %d\n",frameNum,ch,gr); +- #endif +- do +- { +- iteration += 1; +- cod_info->part2_length = part2_length( scalefac, fr_ps, gr, ch, l3_side ); +- huff_bits = max_bits - cod_info->part2_length; +- +- #ifdef BIN_SEARCH +- if(iteration == 1) +- { +- bin_search_StepSize(max_bits,cod_info->quantizerStepSize, +- ix,xrs,cod_info); /* speeds things up a bit */ +- } +- #endif +- #ifdef PERFORM +- if(frameNum==PERFORM) +- fprintf(log_output,"\n Interim Results %d\n\n",iteration); +- bits = test_inner_loop( xr, l3_enc, huff_bits, cod_info, gr, ch, +- xfsf,l3_xmin); +- #else +- bits = inner_loop( xr, l3_enc, huff_bits, cod_info, gr, ch ); +- #endif +- +- calc_noise( &xr[gr][ch][0], &l3_enc[gr][ch][0], cod_info, xfsf ); /* distortion calculation */ +- +- for ( sfb = 0; sfb < CBLIMIT; sfb++ ) /* save scaling factors */ +- scalesave_l[sfb] = scalefac->l[gr][ch][sfb]; +- +- for ( sfb = 0; sfb < SFB_SMAX; sfb++ ) +- for ( i = 0; i < 3; i++ ) +- scalesave_s[sfb][i] = scalefac->s[gr][ch][sfb][i]; +- +- save_preflag = cod_info->preflag; +- save_compress = cod_info->scalefac_compress; +- +- preemphasis( &xr[gr][ch][0], xfsf, l3_xmin, gr, ch, l3_side ); +- +- +- +- #ifdef PERFORM +- if(frameNum == PERFORM) +- { +- fprintf(log_output,"\nbits = %d huff_bits= %d ", bits,huff_bits); +- fprintf(log_output," max_bits = %d\n",max_bits); +- fprintf(log_output,"Stepsize = %f ",cod_info->quantizerStepSize); +- fprintf(log_output," scale_bits = %d \n", cod_info->part2_length ); +- print_scalefacs(log_output,scalefac,cod_info->block_type,gr,ch); +- /*if (gr==0 && ch==0) +- print_quantized_values(log_output,l3_enc[gr][ch] ,cod_info);*/ +- max_ratio = worst_xfsf_to_xmin_ratio(l3_xmin,xfsf,cod_info->block_type,gr,ch); +- fprintf(log_output,"max_ratio = %6.2f\n",max_ratio ); +- print_ratios(log_output,l3_xmin,xfsf,cod_info->block_type,gr,ch); +- fprintf(log_output,"\n\n"); +- fflush(log_output); +- } +- #endif +- over = amp_scalefac_bands( &xr[gr][ch][0], xfsf, l3_xmin, +- l3_side, scalefac, gr, ch, iteration ); +- +- #if 1 +- if ( (status = loop_break(scalefac, cod_info, gr, ch)) == 0 ) +- { +- if ( fr_ps->header->version == 1 ) +- status = scale_bitcount( scalefac, cod_info, gr, ch ); +- else +- status = scale_bitcount_lsf( scalefac, cod_info, gr, ch ); +- } +- #else +- status = loop_break( scalefac, cod_info, gr, ch ); +- if ( fr_ps->header->version == 1 ) +- status += scale_bitcount( scalefac, cod_info, gr, ch ); +- else +- status += scale_bitcount_lsf( scalefac, cod_info, gr, ch ); +- #endif +- +- } +- while ( (status == 0) && (over > 0) ); +- +- cod_info->preflag = save_preflag; +- cod_info->scalefac_compress = save_compress; +- +- for ( sfb = 0; sfb < 21; sfb++ ) +- scalefac->l[gr][ch][sfb] = scalesave_l[sfb]; +- +- for ( i = 0; i < 3; i++ ) +- for ( sfb = 0; sfb < 12; sfb++ ) +- scalefac->s[gr][ch][sfb][i] = scalesave_s[sfb][i]; +- +- cod_info->part2_length = part2_length( scalefac, fr_ps, gr, ch, l3_side ); +- cod_info->part2_3_length = cod_info->part2_length + bits; +- +- #ifdef PERFORM +- if(frameNum == PERFORM) +- { +- fprintf(log_output,"\n Final Results\n"); +- fprintf(log_output,"bits = %d huff_bits= %d", bits,huff_bits); +- fprintf(log_output," max_bits = %d\n",max_bits); +- fprintf(log_output," Stepsize = %f ",cod_info->quantizerStepSize); +- fprintf(log_output, " scale_bits = %d \n", cod_info->part2_length ); +- max_ratio = worst_xfsf_to_xmin_ratio(l3_xmin,xfsf,cod_info->block_type,gr,ch); +- print_scalefacs(log_output,scalefac,cod_info->block_type,gr,ch); +- fprintf(log_output,"max_ratio = %6.2f\n",max_ratio ); +- print_ratios(log_output,l3_xmin,xfsf,cod_info->block_type,gr,ch); +- fflush(log_output); +- } +- #endif +- return cod_info->part2_3_length; +- } +- +- +- +- +- /***************************************************************************/ +- /* inner_loop */ +- /***************************************************************************/ +- /* The code selects the best quantizerStepSize for a particular set +- /* of scalefacs */ +- +- int +- inner_loop( double xr[2][2][576], int l3_enc[2][2][576], int max_bits, +- gr_info *cod_info, int gr, int ch ) +- { +- int bits, c1bits, bvbits; +- double *xrs; /* D576 *xr; */ +- int *ix; /* I576 *ix; */ +- xrs = &xr[gr][ch][0]; +- ix = l3_enc[gr][ch]; +- +- assert( max_bits >= 0 ); +- cod_info->quantizerStepSize -= 1.0;; +- do +- { +- do +- { +- cod_info->quantizerStepSize += 1.0; +- quantize( xrs, ix, cod_info ); +- } +- while ( ix_max(ix, 0, 576) > 8191 + 14 ); /* within table range? */ +- +- calc_runlen( ix, cod_info ); /*rzero,count1,big_values*/ +- bits = c1bits = count1_bitcount( ix, cod_info ); /*count1_table selection*/ +- subdivide( cod_info ); /* bigvalues sfb division */ +- bigv_tab_select( ix, cod_info ); /* codebook selection*/ +- bits += bvbits = bigv_bitcount( ix, cod_info ); /* bit count */ +- +- #ifdef PERFORM +- if(frameNum == PERFORM) +- fprintf(log_output,"StepSize=%f bits = %d huff_bits = %d\n", +- cod_info->quantizerStepSize,bits,max_bits); +- #endif +- +- } +- while ( bits > max_bits ); +- +- return bits; +- } +- +- +- +- /***************************************************************************/ +- /* calc_scfsi */ +- /***************************************************************************/ +- /* calculation of the scalefactor select information ( scfsi ) */ +- +- void calc_scfsi( double xr[576], III_side_info_t *l3_side, +- III_psy_xmin *l3_xmin, int ch, int gr ) +- { +- static int en_tot[2][2]; /* ch,gr */ +- static int en[2][2][21]; +- static int xm[2][2][21]; +- static int xrmax[2][2]; +- +- int en_tot_krit = 10; +- int en_dif_krit = 100; +- int en_scfsi_band_krit = 10; +- int xm_scfsi_band_krit = 10; +- +- int scfsi_band; +- unsigned scfsi_set; +- +- int sfb, start, end, i; +- int condition = 0; +- double temp, log2 = log( 2.0 ); +- gr_info *cod_info = &l3_side->gr[gr].ch[ch].tt; +- +- xrmax[gr][ch] = xr_max( xr, 0, 576 ); +- scfsi_set = 0; +- +- /* the total energy of the granule */ +- for ( temp = 0.0, i = 0; i < 576; i++ ) +- temp += xr[i] * xr[i]; +- if ( temp == 0.0 ) +- en_tot[gr][ch] = 0.0; +- else +- en_tot[gr][ch] = log( temp ) / log2 ; +- +- /* the energy of each scalefactor band, en */ +- /* the allowed distortion of each scalefactor band, xm */ +- +- if ( cod_info->window_switching_flag == 0 || +- cod_info->block_type != 2 ) +- for ( sfb = 0; sfb < 21; sfb++ ) +- { +- start = scalefac_band_long[ sfb ]; +- end = scalefac_band_long[ sfb+1 ]; +- +- for ( temp = 0.0, i = start; i < end; i++ ) +- temp += xr[i] * xr[i]; +- if ( temp == 0.0 ) +- en[gr][ch][sfb] = 0.0; +- else +- en[gr][ch][sfb] = log( temp )/ log2; +- +- if ( l3_xmin->l[gr][ch][sfb] == 0.0 ) +- xm[gr][ch][sfb] = 0.0; +- else +- xm[gr][ch][sfb] = log( l3_xmin->l[gr][ch][sfb] ) / log2; +- } +- if ( gr == 1 ) +- { +- int gr2, tp; +- +- for ( gr2 = 0; gr2 < 2; gr2++ ) +- { +- /* The spectral values are not all zero */ +- if ( xrmax[ch][gr2] != 0.0 ) +- condition++; +- /* None of the granules contains short blocks */ +- if ( (cod_info->window_switching_flag == 0) || +- (cod_info->block_type != 2) ) +- condition++; +- } +- if ( abs(en_tot[0] - en_tot[1]) < en_tot_krit ) +- condition++; +- for ( tp = 0, sfb = 0; sfb < 21; sfb++ ) +- tp += abs( en[ch][0][sfb] - en[ch][1][sfb] ); +- if ( tp < en_dif_krit ) +- condition++; +- +- if ( condition == 6 ) +- { +- for ( scfsi_band = 0; scfsi_band < 4; scfsi_band++ ) +- { +- int sum0 = 0, sum1 = 0; +- l3_side->scfsi[ch][scfsi_band] = 0; +- start = scfsi_band_long[scfsi_band]; +- end = scfsi_band_long[scfsi_band+1]; +- for ( sfb = start; sfb < end; sfb++ ) +- { +- sum0 += abs( en[ch][0][sfb] - en[ch][1][sfb] ); +- sum1 += abs( xm[ch][0][sfb] - xm[ch][1][sfb] ); +- } +- +- if ( sum0 < en_scfsi_band_krit && sum1 < xm_scfsi_band_krit ) +- { +- l3_side->scfsi[ch][scfsi_band] = 1; +- scfsi_set |= (1 << scfsi_band); +- } +- else +- l3_side->scfsi[ch][scfsi_band] = 0; +- } /* for scfsi_band */ +- #ifdef DEBUG +- fprintf( stderr, "calc_scfsi: scfsi_set = 0x%02x\n", scfsi_set ); +- #endif +- } /* if condition == 6 */ +- else +- for ( scfsi_band = 0; scfsi_band < 4; scfsi_band++ ) +- l3_side->scfsi[ch][scfsi_band] = 0; +- } /* if gr == 1 */ +- } +- +- +- +- /***************************************************************************/ +- /* part2_length */ +- /***************************************************************************/ +- +- /* calculates the number of bits needed to encode the scalefacs in the */ +- /* main data block */ +- +- int part2_length( III_scalefac_t *scalefac, frame_params *fr_ps, +- int gr, int ch, III_side_info_t *si ) +- { +- int slen1, slen2, slen3, slen4, bits, sfb, window, partition; +- gr_info *gi = &si->gr[gr].ch[ch].tt; +- +- bits = 0; +- if ( fr_ps->header->version == 1 ) +- { +- static int slen1_tab[16] = { 0, 0, 0, 0, 3, 1, 1, 1, 2, 2, 2, 3, 3, 3, 4, 4 }; +- static int slen2_tab[16] = { 0, 1, 2, 3, 0, 1, 2, 3, 1, 2, 3, 1, 2, 3, 2, 3 }; +- +- slen1 = slen1_tab[ gi->scalefac_compress ]; +- slen2 = slen2_tab[ gi->scalefac_compress ]; +- +- if ( (gi->window_switching_flag == 1) && (gi->block_type == 2) ) +- { +- if ( gi->mixed_block_flag ) +- { +- bits += (8 * slen1) + (9 * slen1) + (18 * slen2); +- } +- else +- { +- bits += (18 * slen1) + (18 * slen2); +- } +- } +- else +- { +- if ( (gr == 0) || (si->scfsi[ch][0] == 0) ) +- bits += (6 * slen1); +- +- if ( (gr == 0) || (si->scfsi[ch][1] == 0) ) +- /* bits += (6 * slen1); This is wrong SS 19-12-96 */ +- bits += (5 * slen1); +- +- if ( (gr == 0) || (si->scfsi[ch][2] == 0) ) +- /* bits += (6 * slen1); This is wrong SS 19-12-96 */ +- bits += (5 * slen2); +- +- if ( (gr == 0) || (si->scfsi[ch][3] == 0) ) +- /* bits += (6 * slen1); This is wrong SS 19-12-96 */ +- bits += (5 * slen2); +- } +- } +- else +- { /* MPEG 2 */ +- assert( gi->sfb_partition_table ); +- for ( partition = 0; partition < 4; partition++ ) +- bits += gi->slen[partition] * gi->sfb_partition_table[partition]; +- } +- return bits; +- } +- +- +- +- /*************************************************************************/ +- /* scale_bitcount */ +- /*************************************************************************/ +- +- /* Also calculates the number of bits necessary to code the scalefactors. */ +- +- int scale_bitcount( III_scalefac_t *scalefac, gr_info *cod_info, +- int gr, int ch ) +- { +- int i, k, sfb, max_slen1 = 0, max_slen2 = 0, /*a, b, */ ep = 2; +- +- static int slen1[16] = { 0, 0, 0, 0, 3, 1, 1, 1, 2, 2, 2, 3, 3, 3, 4, 4 }; +- static int slen2[16] = { 0, 1, 2, 3, 0, 1, 2, 3, 1, 2, 3, 1, 2, 3, 2, 3 }; +- static int pow2[5] = { 1, 2, 4, 8, 16 }; +- +- if ( cod_info->window_switching_flag != 0 && cod_info->block_type == 2 ) +- { +- if ( cod_info->mixed_block_flag == 0 ) +- { +- /* a = 18; b = 18; */ +- for ( i = 0; i < 3; i++ ) +- { +- for ( sfb = 0; sfb < 6; sfb++ ) +- if ( scalefac->s[gr][ch][sfb][i] > max_slen1 ) +- max_slen1 = scalefac->s[gr][ch][sfb][i]; +- for (sfb = 6; sfb < 12; sfb++ ) +- if ( scalefac->s[gr][ch][sfb][i] > max_slen2 ) +- max_slen2 = scalefac->s[gr][ch][sfb][i]; +- } +- } +- else +- {/* mixed_block_flag = 1 */ +- /* a = 17; b = 18; */ +- for ( sfb = 0; sfb < 8; sfb++ ) +- if ( scalefac->l[gr][ch][sfb] > max_slen1 ) +- max_slen1 = scalefac->l[gr][ch][sfb]; +- for ( i = 0; i < 3; i++ ) +- { +- for ( sfb = 3; sfb < 6; sfb++ ) +- if ( scalefac->s[gr][ch][sfb][i] > max_slen1 ) +- max_slen1 = scalefac->s[gr][ch][sfb][i]; +- for ( sfb = 6; sfb < 12; sfb++ ) +- if ( scalefac->s[gr][ch][sfb][i] > max_slen2 ) +- max_slen2 = scalefac->s[gr][ch][sfb][i]; +- } +- } +- } +- else +- { /* block_type == 1,2,or 3 */ +- /* a = 11; b = 10; */ +- for ( sfb = 0; sfb < 11; sfb++ ) +- if ( scalefac->l[gr][ch][sfb] > max_slen1 ) +- max_slen1 = scalefac->l[gr][ch][sfb]; +- for ( sfb = 11; sfb < 21; sfb++ ) +- if ( scalefac->l[gr][ch][sfb] > max_slen2 ) +- max_slen2 = scalefac->l[gr][ch][sfb]; +- } +- +- for ( k = 0; k < 16; k++ ) +- { +- if ( (max_slen1 < pow2[slen1[k]]) && (max_slen2 < pow2[slen2[k]]) ) +- { +- ep = 0; +- break; +- } +- } +- +- if ( ep == 0 ) +- cod_info->scalefac_compress = k; +- #ifdef DEBUG +- if ( ep != 0 ) +- printf( "---WARNING !! Amplification of some bands over limits\n" ); +- #endif +- return ep; +- } +- +- +- +- +- /*************************************************************************/ +- /* scale_bitcount_lsf */ +- /*************************************************************************/ +- +- /* Also counts the number of bits to encode the scalefacs but for MPEG 2 */ +- /* Lower sampling frequencies (24, 22.05 and 16 kHz.) */ +- +- /* This is reverse-engineered from section 2.4.3.2 of the MPEG2 IS, */ +- /* "Audio Decoding Layer III" */ +- +- int scale_bitcount_lsf( III_scalefac_t *scalefac, gr_info *cod_info, +- int gr, int ch ) +- { +- int table_number, row_in_table, partition, nr_sfb, window, over; +- int i, k, sfb, max_sfac[ 4 ]; +- unsigned *partition_table; +- +- /* +- Set partition table. Note that should try to use table one, +- but do not yet... +- */ +- if ( cod_info->preflag ) +- table_number = 2; +- else +- table_number = 0; +- +- for ( i = 0; i < 4; i++ ) +- max_sfac[i] = 0; +- +- if ( cod_info->window_switching_flag != 0 && cod_info->block_type == 2 ) +- { +- if ( cod_info->mixed_block_flag == 0 ) +- { +- row_in_table = 1; +- partition_table = &nr_of_sfb_block[table_number][row_in_table][0]; +- for ( sfb = 0, partition = 0; partition < 4; partition++ ) +- { +- nr_sfb = partition_table[ partition ] / 3; +- for ( i = 0; i < nr_sfb; i++, sfb++ ) +- for ( window = 0; window < 3; window++ ) +- if ( scalefac->s[gr][ch][sfb][window] > max_sfac[partition] ) +- max_sfac[partition] = scalefac->s[gr][ch][sfb][window]; +- } +- } +- else +- {/* mixed_block_flag = 1 */ +- row_in_table = 2; +- partition_table = &nr_of_sfb_block[table_number][row_in_table][0]; +- partition = 0; +- nr_sfb = partition_table[ partition ]; +- for ( i = 0; i < nr_sfb; i++, sfb++ ) +- if ( scalefac->l[gr][ch][sfb] > max_sfac[partition] ) +- max_sfac[partition] = scalefac->l[gr][ch][sfb]; +- +- for ( sfb = 0, partition = 1; partition < 4; partition++ ) +- { +- nr_sfb = partition_table[ partition ] / 3; +- for ( i = 0; i < nr_sfb; i++, sfb++ ) +- for ( window = 0; window < 3; window++ ) +- if ( scalefac->s[gr][ch][sfb][window] > max_sfac[partition] ) +- max_sfac[partition] = scalefac->s[gr][ch][sfb][window]; +- } +- } +- } +- else +- { +- row_in_table = 0; +- partition_table = &nr_of_sfb_block[table_number][row_in_table][0]; +- partition = 0; +- for ( sfb = 0, partition = 0; partition < 4; partition++ ) +- { +- nr_sfb = partition_table[ partition ]; +- for ( i = 0; i < nr_sfb; i++, sfb++ ) +- if ( scalefac->l[gr][ch][sfb] > max_sfac[partition] ) +- max_sfac[partition] = scalefac->l[gr][ch][sfb]; +- } +- } +- +- for ( over = 0, partition = 0; partition < 4; partition++ ) +- { +- if ( max_sfac[partition] > max_sfac_tab[table_number][partition] ) +- over++; +- } +- if ( !over ) +- { +- /* +- Since no bands have been over-amplified, we can set scalefac_compress +- and slen[] for the formatter +- */ +- static int log2tab[] = { 0, 1, 2, 2, 3, 3, 3, 3, 4, 4, 4, 4, 4, 4, 4, 4 }; +- +- unsigned slen1, slen2, slen3, slen4; +- +- cod_info->sfb_partition_table = &nr_of_sfb_block[table_number][row_in_table][0]; +- for ( partition = 0; partition < 4; partition++ ) +- cod_info->slen[partition] = log2tab[max_sfac[partition]]; +- +- /* set scalefac_compress */ +- slen1 = cod_info->slen[ 0 ]; +- slen2 = cod_info->slen[ 1 ]; +- slen3 = cod_info->slen[ 2 ]; +- slen4 = cod_info->slen[ 3 ]; +- +- switch ( table_number ) +- { +- case 0: +- cod_info->scalefac_compress = (((slen1 * 5) + slen2) << 4) +- + (slen3 << 2) +- + slen4; +- break; +- +- case 1: +- cod_info->scalefac_compress = 400 +- + (((slen1 * 5) + slen2) << 2) +- + slen3; +- break; +- +- case 2: +- cod_info->scalefac_compress = 500 + (slen1 * 3) + slen2; +- break; +- +- default: +- fprintf( stderr, "intensity stereo not implemented yet\n" ); +- exit( EXIT_FAILURE ); +- break; +- } +- } +- #ifdef DEBUG +- if ( over ) +- printf( "---WARNING !! Amplification of some bands over limits\n" ); +- #endif +- return over; +- } +- +- +- /*************************************************************************/ +- /* calc_noise */ +- /*************************************************************************/ +- +- /* Function: Calculate the distortion introduced by the quantization */ +- /* in each scale factor band. */ +- +- void calc_noise( double xr[576], int ix[576], gr_info *cod_info, +- double xfsf[4][CBLIMIT] ) +- { +- int start, end, sfb, l, i; +- double sum,step,bw; +- +- D192_3 *xr_s; +- I192_3 *ix_s; +- +- xr_s = (D192_3 *) xr; +- ix_s = (I192_3 *) ix; +- +- step = pow( 2.0, (cod_info->quantizerStepSize) * 0.25 ); +- for ( sfb = 0; sfb < cod_info->sfb_lmax; sfb++ ) +- { +- start = scalefac_band_long[ sfb ]; +- end = scalefac_band_long[ sfb+1 ]; +- bw = end - start; +- +- for ( sum = 0.0, l = start; l < end; l++ ) +- { +- double temp; +- temp = fabs( xr[l] ) - pow( (double) ix[l], 4.0 / 3.0 ) * step; +- sum += temp * temp; +- } +- xfsf[0][sfb] = sum / bw; +- } +- +- for ( i = 0; i < 3; i++ ) +- { +- step = pow( 2.0, (cod_info->quantizerStepSize) * 0.25 ); /* subblock_gain ? */ +- for ( sfb = cod_info->sfb_smax; sfb < 12; sfb++ ) +- { +- start = scalefac_band_short[ sfb ]; +- end = scalefac_band_short[ sfb+1 ]; +- bw = end - start; +- +- for ( sum = 0.0, l = start; l < end; l++ ) +- { +- double temp; +- temp = fabs( (*xr_s)[l][i] ) - pow( (double) (*ix_s)[l][i], 4.0 / 3.0 ) * step; +- sum += temp * temp; +- } +- xfsf[i+1][sfb] = sum / bw; +- } +- } +- } +- +- +- +- +- /*************************************************************************/ +- /* calc_xmin */ +- /*************************************************************************/ +- +- /* +- Calculate the allowed distortion for each scalefactor band, +- as determined by the psychoacoustic model. +- xmin(sb) = ratio(sb) * en(sb) / bw(sb) +- */ +- +- void calc_xmin( double xr[2][2][576], III_psy_ratio *ratio, +- gr_info *cod_info, III_psy_xmin *l3_xmin, +- int gr, int ch ) +- { +- int start, end, sfb, l, b; +- double en, bw; +- +- D192_3 *xr_s; +- +- xr_s = (D192_3 *) xr[gr][ch] ; +- +- for ( sfb = cod_info->sfb_smax; sfb < SFB_SMAX - 1; sfb++ ) +- { +- start = scalefac_band_short[ sfb ]; +- end = scalefac_band_short[ sfb + 1 ]; +- bw = end - start; +- for ( b = 0; b < 3; b++ ) +- { +- for ( en = 0.0, l = start; l < end; l++ ) +- en += (*xr_s)[l][b] * (*xr_s)[l][b]; +- l3_xmin->s[gr][ch][sfb][b] = ratio->s[gr][ch][sfb][b] * en / bw; +- } +- } +- +- for ( sfb = 0; sfb < cod_info->sfb_lmax; sfb++ ) +- { +- start = scalefac_band_long[ sfb ]; +- end = scalefac_band_long[ sfb+1 ]; +- bw = end - start; +- +- for ( en = 0.0, l = start; l < end; l++ ) +- en += xr[gr][ch][l] * xr[gr][ch][l]; +- l3_xmin->l[gr][ch][sfb] = ratio->l[gr][ch][sfb] * en / bw; +- } +- } +- +- +- +- /*************************************************************************/ +- /* loop_break */ +- /*************************************************************************/ +- +- /* Function: Returns zero if there is a scalefac which has not been +- amplified. Otherwise it returns one. +- */ +- +- int loop_break( III_scalefac_t *scalefac, gr_info *cod_info, +- int gr, int ch ) +- { +- int i, sfb, temp = 1; +- +- for ( sfb = 0; sfb < cod_info->sfb_lmax; sfb++ ) +- if ( scalefac->l[gr][ch][sfb] == 0 ) +- temp = 0; +- +- for ( sfb = cod_info->sfb_smax; sfb < 12; sfb++ ) +- for ( i = 0; i < 3; i++ ) +- if ( scalefac->s[gr][ch][sfb][i] == 0 ) +- temp = 0; +- #ifdef DEBUG +- if ( temp != 0 ) +- printf( "---WARNING !! All scalefactor bands amplified\n" ); +- #endif +- return temp; +- } +- +- +- +- /*************************************************************************/ +- /* preemphasis */ +- /*************************************************************************/ +- +- /* +- See ISO 11172-3 section C.1.5.4.3.4 +- */ +- +- void preemphasis( double xr[576], double xfsf[4][CBLIMIT], +- III_psy_xmin *l3_xmin, +- int gr, int ch, III_side_info_t *l3_side ) +- { +- int i, sfb, start, end, scfsi_band, over; +- double ifqstep; +- gr_info *cod_info = &l3_side->gr[gr].ch[ch].tt; +- +- if ( gr == 1 ) +- { +- /* +- If the second granule is being coded and scfsi is active in +- at least one scfsi_band, the preemphasis in the second granule +- is set equal to the setting in the first granule +- */ +- for ( scfsi_band = 0; scfsi_band < 4; scfsi_band++ ) +- if ( l3_side->scfsi[ch][scfsi_band] ) +- { +- cod_info->preflag = l3_side->gr[0].ch[ch].tt.preflag; +- return; +- } +- +- } +- +- /* +- Preemphasis is switched on if in all the upper four scalefactor +- bands the actual distortion exceeds the threshold after the +- first call of the inner loop +- */ +- if ( cod_info->block_type != 2 && cod_info->preflag == 0 ) +- { +- over = 0; +- for ( sfb = 17; sfb < 21; sfb++ ) +- if ( xfsf[0][sfb] > l3_xmin->l[gr][ch][sfb] ) +- over++; +- +- if (over == 4 ) +- { +- cod_info->preflag = 1; +- ifqstep = ( cod_info->scalefac_scale == 0 ) ? sqrt(2.) +- : pow( 2.0, (0.5 * (1.0 + (double) cod_info->scalefac_scale)) ); +- +- for ( sfb = 0; sfb < cod_info->sfb_lmax; sfb++ ) +- { +- l3_xmin->l[gr][ch][sfb] *= pow( ifqstep, 2.0 * (double) pretab[sfb] ); +- start = scalefac_band_long[ sfb ]; +- end = scalefac_band_long[ sfb+1 ]; +- for( i = start; i < end; i++ ) +- xr[i] *= pow( ifqstep, (double) pretab[sfb] ); +- } +- } +- } +- } +- +- +- /*************************************************************************/ +- /* amp_scalefac_bands */ +- /*************************************************************************/ +- +- /* +- Amplify the scalefactor bands that violate the masking threshold. +- See ISO 11172-3 Section C.1.5.4.3.5 +- */ +- +- int amp_scalefac_bands( double xr[576], double xfsf[4][CBLIMIT], +- III_psy_xmin *l3_xmin, III_side_info_t *l3_side, +- III_scalefac_t *scalefac, +- int gr, int ch, int iteration ) +- { +- int start, end, l, sfb, i, scfsi_band, over = 0; +- double ifqstep, ifqstep2; +- D192_3 *xr_s; +- gr_info *cod_info, *gr0; +- int copySF, preventSF; +- cod_info = &l3_side->gr[gr].ch[ch].tt; +- gr0 = &l3_side->gr[0].ch[ch].tt; +- +- xr_s = (D192_3 *) xr; +- copySF = 0; +- preventSF = 0; +- +- if ( cod_info->scalefac_scale == 0 ) +- ifqstep = sqrt( 2.0 ); +- else +- ifqstep = pow( 2.0, 0.5 * (1.0 + (double) cod_info->scalefac_scale) ); +- +- if ( gr == 1 ) +- { +- /* +- If the second granule is being coded and scfsi is active in at +- least one scfsi_band... +- */ +- for ( scfsi_band = 0; scfsi_band < 4; scfsi_band++ ) +- if ( l3_side->scfsi[ch][scfsi_band] ) +- { +- /* +- a) ifqstep has to be set similar to the +- first granule... +- */ +- if ( gr0->scalefac_scale == 0 ) +- ifqstep = sqrt( 2.0 ); +- else +- ifqstep = pow( 2.0, 0.5 * (1.0 + (double) gr0->scalefac_scale) ); +- +- if ( iteration == 1 ) +- { +- /* +- b) If it is the first iteration, the scalefactors +- of scalefactor bands in which scfsi is enabled +- must be taken from the first granule +- */ +- copySF = 1; +- } +- else +- { +- /* +- c) If it is not the first iteration, the amplification +- must be prevented for scalefactor bands in which +- scfsi is enabled +- */ +- preventSF = 1; +- } +- break; +- } +- +- } +- +- ifqstep2 = ifqstep * ifqstep; +- scfsi_band = 0; +- +- for ( sfb = 0; sfb < cod_info->sfb_lmax; sfb++ ) +- { +- if ( copySF || preventSF ) +- { +- if ( sfb == scfsi_band_long[scfsi_band + 1] ) +- scfsi_band += 1; +- if ( l3_side->scfsi[ch][scfsi_band] ) +- { +- if ( copySF ) +- scalefac->l[gr][ch][sfb] = scalefac->l[0][ch][sfb]; +- continue; +- } +- } +- if ( xfsf[0][sfb] > l3_xmin->l[gr][ch][sfb] ) +- { +- over++; +- l3_xmin->l[gr][ch][sfb] *= ifqstep2; +- scalefac->l[gr][ch][sfb]++; +- start = scalefac_band_long[sfb]; +- end = scalefac_band_long[sfb+1]; +- for ( l = start; l < end; l++ ) +- xr[l] *= ifqstep; +- } +- } +- +- /* +- Note that scfsi is not enabled for frames containing +- short blocks +- */ +- for ( i = 0; i < 3; i++ ) +- for ( sfb = cod_info->sfb_smax; sfb < 12; sfb++ ) +- if ( xfsf[i+1][sfb] > l3_xmin->s[gr][ch][sfb][i] ) +- { +- over++; +- l3_xmin->s[gr][ch][sfb][i] *= ifqstep2; +- scalefac->s[gr][ch][sfb][i]++; +- #ifdef DEBUGSC +- printf( "cod_info->scalefac[%d][%d] = %d (amp_scale)\n", +- i,sfb,scalefac->s[gr][ch][sfb][i] ); +- #endif +- start = scalefac_band_short[sfb]; +- end = scalefac_band_short[sfb+1]; +- for ( l = start; l < end; l++ ) +- (*xr_s)[l][i] *= ifqstep; +- } +- return over; +- } +- +- +- +- +- /*************************************************************************/ +- /* quantize */ +- /*************************************************************************/ +- +- /* +- Function: Quantization of the vector xr ( -> ix) +- */ +- +- void quantize( double xr[576], int ix[576], gr_info *cod_info ) +- { +- int i, b, l_end, s_start; +- double step, quantizerStepSize; +- +- D192_3 *xr_s; +- I192_3 *ix_s; +- +- xr_s = (D192_3 *) xr; +- ix_s = (I192_3 *) ix; +- +- quantizerStepSize = (double) cod_info->quantizerStepSize; +- +- for ( i = 0; i < 576; i++ ) +- ix[i] = 0; +- +- if ( cod_info->quantizerStepSize == 0.0 ) +- step = 1.0; +- else +- step = pow ( 2.0, quantizerStepSize * 0.25 ); +- +- if ( cod_info->window_switching_flag != 0 && cod_info->block_type == 2 ) +- if ( cod_info->mixed_block_flag == 0 ) +- { +- l_end = 0; +- s_start = 0; +- } +- else +- { +- l_end = 18 * 2; +- s_start = 6 * 2; +- } +- else +- { +- l_end = 576; +- s_start = 192; +- } +- +- for ( i = 0; i < l_end; i++ ) +- ix[i] = nint( pow(fabs(xr[i]) / step, 0.75) - 0.0946 ); +- +- if ( s_start < 192 ) +- for ( b = 0; b < 3; b++ ) +- { +- step = pow( 2.0, (quantizerStepSize + 8.0 * (double) cod_info->subblock_gain[b]) * 0.25 ); +- for ( i = s_start; i < 192; i++ ) +- (*ix_s)[i][b] = nint( pow(fabs((*xr_s)[i][b]) / step, 0.75) - 0.0946 ); +- } +- } +- +- +- +- +- /*************************************************************************/ +- /* ix_max */ +- /*************************************************************************/ +- +- /* +- Function: Calculate the maximum of ix from 0 to 575 +- */ +- +- int ix_max( int ix[576], unsigned int begin, unsigned int end ) +- { +- int i, max = 0; +- +- for ( i = begin; i < end; i++ ) +- { +- int x = abs( ix[i] ); +- if ( x > max ) +- max = x; +- } +- return max; +- } +- +- +- /*************************************************************************/ +- /* xr_max */ +- /*************************************************************************/ +- +- /* +- Function: Calculate the maximum of xr[576] from 0 to 575 +- */ +- +- double xr_max( double xr[576], unsigned int begin, unsigned int end ) +- { +- int i; +- double max = 0.0, temp; +- +- for ( i = begin; i < end; i++ ) +- if( (temp = fabs(xr[i])) > max ) +- max = temp; +- return max; +- } +- +- +- +- /* Noiseless coding -- Huffman coding */ +- +- +- /*************************************************************************/ +- /* calc_runlen */ +- /*************************************************************************/ +- +- /* +- Function: Calculation of rzero, count1, big_values +- (Partitions ix into big values, quadruples and zeros). +- */ +- +- void calc_runlen( int ix[576], gr_info *cod_info ) +- { +- int i; +- int rzero = 0; +- +- if ( cod_info->window_switching_flag && (cod_info->block_type == 2) ) +- { /* short blocks */ +- cod_info->count1 = 0; +- cod_info->big_values = 288; +- } +- else +- { +- for ( i = 576; i > 1; i -= 2 ) +- if ( ix[i-1] == 0 && ix[i-2] == 0 ) +- rzero++; +- else +- break; +- +- cod_info->count1 = 0 ; +- for ( ; i > 3; i -= 4 ) +- if ( abs(ix[i-1]) <= 1 +- && abs(ix[i-2]) <= 1 +- && abs(ix[i-3]) <= 1 +- && abs(ix[i-4]) <= 1 ) +- cod_info->count1++; +- else +- break; +- +- cod_info->big_values = i/2; +- } +- assert( (2 * rzero + 4 * cod_info->count1 + 2 * cod_info->big_values) == 576 ); +- } +- +- +- +- /*************************************************************************/ +- /* count1_bitcount */ +- /*************************************************************************/ +- +- /* +- Determines the number of bits to encode the quadruples. +- */ +- +- int count1_bitcount( int ix[ 576 ], gr_info *cod_info ) +- { +- int abs_and_sign( int *x ); +- +- int p, i, k, bitsum_count1; +- int v, w, x, y, signbits; +- int sum0 = 0, sum1 = 0; +- +- for ( i = cod_info->big_values * 2, k = 0; k < cod_info->count1; i += 4, k++ ) +- { +- v = ix[ i ]; +- w = ix[ i + 1 ]; +- x = ix[ i + 2 ]; +- y = ix[ i + 3 ]; +- +- abs_and_sign( &v ); +- abs_and_sign( &w ); +- abs_and_sign( &x ); +- abs_and_sign( &y ); +- +- p = v + (w << 1) + (x << 2) + (y << 3); +- +- signbits = 0; +- +- if ( v != 0 ) +- signbits += 1; +- if ( w != 0 ) +- signbits += 1; +- if ( x != 0 ) +- signbits += 1; +- if ( y != 0 ) +- signbits += 1; +- +- sum0 += signbits; +- sum1 += signbits; +- +- sum0 += ht[ 32 ].hlen[ p ]; +- sum1 += ht[ 33 ].hlen[ p ]; +- } +- +- if ( sum0 < sum1 ) +- { +- bitsum_count1 = sum0; +- cod_info->count1table_select = 0; +- } +- else +- { +- bitsum_count1 = sum1; +- cod_info->count1table_select = 1; +- } +- return( bitsum_count1 ); +- } +- +- +- +- +- +- struct +- { +- unsigned region0_count; +- unsigned region1_count; +- } subdv_table[ 23 ] = +- { +- {0, 0}, /* 0 bands */ +- {0, 0}, /* 1 bands */ +- {0, 0}, /* 2 bands */ +- {0, 0}, /* 3 bands */ +- {0, 0}, /* 4 bands */ +- {0, 1}, /* 5 bands */ +- {1, 1}, /* 6 bands */ +- {1, 1}, /* 7 bands */ +- {1, 2}, /* 8 bands */ +- {2, 2}, /* 9 bands */ +- {2, 3}, /* 10 bands */ +- {2, 3}, /* 11 bands */ +- {3, 4}, /* 12 bands */ +- {3, 4}, /* 13 bands */ +- {3, 4}, /* 14 bands */ +- {4, 5}, /* 15 bands */ +- {4, 5}, /* 16 bands */ +- {4, 6}, /* 17 bands */ +- {5, 6}, /* 18 bands */ +- {5, 6}, /* 19 bands */ +- {5, 7}, /* 20 bands */ +- {6, 7}, /* 21 bands */ +- {6, 7}, /* 22 bands */ +- }; +- +- +- +- +- /*************************************************************************/ +- /* subdivide */ +- /*************************************************************************/ +- +- /* presumable subdivides the bigvalue region which will +- use separate Huffman tables. +- */ +- +- void subdivide( gr_info *cod_info ) +- { +- int scfb_anz = 0; +- int bigvalues_region; +- +- if ( cod_info->big_values == 0 ) +- { /* no big_values region */ +- cod_info->region0_count = 0; +- cod_info->region1_count = 0; +- } +- else +- { +- bigvalues_region = 2 * cod_info->big_values; +- +- if ( (cod_info->window_switching_flag == 0) ) +- { /* long blocks */ +- int thiscount, index; +- /* Calculate scfb_anz */ +- while ( scalefac_band_long[scfb_anz] < bigvalues_region ) +- scfb_anz++; +- assert( scfb_anz < 23 ); +- +- cod_info->region0_count = subdv_table[scfb_anz].region0_count; +- thiscount = cod_info->region0_count; +- index = thiscount + 1; +- while ( thiscount && (scalefac_band_long[index] > bigvalues_region) ) +- { +- thiscount -= 1; +- index -= 1; +- } +- cod_info->region0_count = thiscount; +- +- cod_info->region1_count = subdv_table[scfb_anz].region1_count; +- index = cod_info->region0_count + cod_info->region1_count + 2; +- thiscount = cod_info->region1_count; +- while ( thiscount && (scalefac_band_long[index] > bigvalues_region) ) +- { +- thiscount -= 1; +- index -= 1; +- } +- cod_info->region1_count = thiscount; +- cod_info->address1 = scalefac_band_long[cod_info->region0_count+1]; +- cod_info->address2 = scalefac_band_long[cod_info->region0_count +- + cod_info->region1_count + 2 ]; +- cod_info->address3 = bigvalues_region; +- } +- else +- { +- if ( (cod_info->block_type == 2) && (cod_info->mixed_block_flag == 0) ) +- { +- cod_info->region0_count = 8; +- cod_info->region1_count = 36; +- cod_info->address1 = 36; +- cod_info->address2 = bigvalues_region; +- cod_info->address3 = 0; +- } +- else +- { +- cod_info->region0_count = 7; +- cod_info->region1_count = 13; +- cod_info->address1 = scalefac_band_long[cod_info->region0_count+1]; +- cod_info->address2 = bigvalues_region; +- cod_info->address3 = 0; +- } +- } +- } +- } +- +- +- +- +- /*************************************************************************/ +- /* bigv_tab_select */ +- /*************************************************************************/ +- +- /* +- /* Function: Select huffman code tables for bigvalues regions +- */ +- +- void bigv_tab_select( int ix[576], gr_info *cod_info ) +- { +- /* int max; */ +- +- cod_info->table_select[0] = 0; +- cod_info->table_select[1] = 0; +- cod_info->table_select[2] = 0; +- +- if ( cod_info->window_switching_flag && (cod_info->block_type == 2) ) +- { +- /* +- Within each scalefactor band, data is given for successive +- time windows, beginning with window 0 and ending with window 2. +- Within each window, the quantized values are then arranged in +- order of increasing frequency... +- */ +- int sfb, window, line, start, end, max1, max2, x, y; +- int region1Start; +- int *pmax; +- +- region1Start = 12; +- max1 = max2 = 0; +- +- for ( sfb = 0; sfb < 13; sfb++ ) +- { +- start = scalefac_band_short[ sfb ]; +- end = scalefac_band_short[ sfb+1 ]; +- +- if ( start < region1Start ) +- pmax = &max1; +- else +- pmax = &max2; +- +- for ( window = 0; window < 3; window++ ) +- for ( line = start; line < end; line += 2 ) +- { +- assert( line >= 0 ); +- assert( line < 576 ); +- x = abs( ix[ (line * 3) + window ] ); +- y = abs( ix[ ((line + 1) * 3) + window ]); +- *pmax = *pmax > x ? *pmax : x; +- *pmax = *pmax > y ? *pmax : y; +- } +- } +- cod_info->table_select[0] = choose_table( max1 ); +- cod_info->table_select[1] = choose_table( max2 ); +- } +- else +- { +- if ( cod_info->address1 > 0 ) +- cod_info->table_select[0] = new_choose_table( ix, 0, cod_info->address1 ); +- +- if ( cod_info->address2 > cod_info->address1 ) +- cod_info->table_select[1] = new_choose_table( ix, cod_info->address1, cod_info->address2 ); +- +- if ( cod_info->big_values * 2 > cod_info->address2 ) +- cod_info->table_select[2] = new_choose_table( ix, cod_info->address2, cod_info->big_values * 2 ); +- } +- } +- +- +- +- +- /*************************************************************************/ +- /* new_choose table */ +- /*************************************************************************/ +- +- /* +- Choose the Huffman table that will encode ix[begin..end] with +- the fewest bits. +- +- Note: This code contains knowledge about the sizes and characteristics +- of the Huffman tables as defined in the IS (Table B.7), and will not work +- with any arbitrary tables. +- */ +- +- int new_choose_table( int ix[576], unsigned int begin, unsigned int end ) +- { +- int i, max; +- int choice[ 2 ]; +- int sum[ 2 ]; +- +- max = ix_max( ix, begin, end ); +- +- if ( max == 0 ) +- return 0; +- +- max = abs( max ); +- +- choice[ 0 ] = 0; +- choice[ 1 ] = 0; +- +- if ( max < 15 ) +- { +- /* try tables with no linbits */ +- for ( i = 0; i < 14; i++ ) +- if ( ht[i].xlen > max ) +- { +- choice[ 0 ] = i; +- break; +- } +- assert( choice[0] ); +- +- sum[ 0 ] = count_bit( ix, begin, end, choice[0] ); +- +- switch ( choice[0] ) +- { +- case 2: +- sum[ 1 ] = count_bit( ix, begin, end, 3 ); +- if ( sum[1] <= sum[0] ) +- choice[ 0 ] = 3; +- break; +- +- case 5: +- sum[ 1 ] = count_bit( ix, begin, end, 6 ); +- if ( sum[1] <= sum[0] ) +- choice[ 0 ] = 6; +- break; +- +- case 7: +- sum[ 1 ] = count_bit( ix, begin, end, 8 ); +- if ( sum[1] <= sum[0] ) +- { +- choice[ 0 ] = 8; +- sum[ 0 ] = sum[ 1 ]; +- } +- sum[ 1 ] = count_bit( ix, begin, end, 9 ); +- if ( sum[1] <= sum[0] ) +- choice[ 0 ] = 9; +- break; +- +- case 10: +- sum[ 1 ] = count_bit( ix, begin, end, 11 ); +- if ( sum[1] <= sum[0] ) +- { +- choice[ 0 ] = 11; +- sum[ 0 ] = sum[ 1 ]; +- } +- sum[ 1 ] = count_bit( ix, begin, end, 12 ); +- if ( sum[1] <= sum[0] ) +- choice[ 0 ] = 12; +- break; +- +- case 13: +- sum[ 1 ] = count_bit( ix, begin, end, 15 ); +- if ( sum[1] <= sum[0] ) +- choice[ 0 ] = 15; +- break; +- +- default: +- break; +- } +- } +- else +- { +- /* try tables with linbits */ +- max -= 15; +- +- for ( i = 15; i < 24; i++ ) +- { +- if ( ht[i].linmax >= max ) +- { +- choice[ 0 ] = i; +- break; +- } +- } +- for ( i = 24; i < 32; i++ ) +- { +- if ( ht[i].linmax >= max ) +- { +- choice[ 1 ] = i; +- break; +- } +- } +- assert( choice[0] ); +- assert( choice[1] ); +- +- sum[ 0 ] = count_bit( ix, begin, end, choice[0] ); +- sum[ 1 ] = count_bit( ix, begin, end, choice[1] ); +- if ( sum[1] < sum[0] ) +- choice[ 0 ] = choice[ 1 ]; +- } +- return choice[ 0 ]; +- } +- +- +- +- /*************************************************************************/ +- /* choose table */ +- /*************************************************************************/ +- +- int choose_table( int max ) +- { +- int i, choice; +- +- if ( max == 0 ) +- return 0; +- +- max = abs( max ); +- choice = 0; +- +- if ( max < 15 ) +- { +- for ( i = 0; i < 15; i++ ) +- { +- if ( ht[i].xlen > max ) +- { +- choice = i; +- break; +- } +- } +- } +- else +- { +- max -= 15; +- for (i = 15; i < 32; i++ ) +- { +- if ( ht[i].linmax >= max ) +- { +- choice = i; +- break; +- } +- } +- } +- assert( choice ); +- return choice; +- } +- +- +- /*************************************************************************/ +- /* bigv_bitcount */ +- /*************************************************************************/ +- +- /* +- Function: Count the number of bits necessary to code the bigvalues region. +- */ +- +- int bigv_bitcount( int ix[576], gr_info *gi ) +- { +- int bits = 0; +- +- if ( gi->window_switching_flag && gi->block_type == 2 ) +- { +- /* +- Within each scalefactor band, data is given for successive +- time windows, beginning with window 0 and ending with window 2. +- Within each window, the quantized values are then arranged in +- order of increasing frequency... +- */ +- int sfb, window, line, start, end; +- I192_3 *ix_s; +- +- if ( gi->mixed_block_flag ) +- { +- unsigned int table; +- +- if ( (table = gi->table_select[0]) != 0 ) +- bits += count_bit( ix, 0, gi->address1, table ); +- sfb = 2; +- } +- else +- sfb = 0; +- +- ix_s = (I192_3 *) &ix[0]; +- +- for ( ; sfb < 13; sfb++ ) +- { +- unsigned tableindex = 100; +- +- start = scalefac_band_short[ sfb ]; +- end = scalefac_band_short[ sfb+1 ]; +- +- if ( start < 12 ) +- tableindex = gi->table_select[ 0 ]; +- else +- tableindex = gi->table_select[ 1 ]; +- assert( tableindex < 32 ); +- +- for ( window = 0; window < 3; window++ ) +- for ( line = start; line < end; line += 2 ) +- { +- unsigned int code, ext; +- int cbits, xbits; +- int x = (*ix_s)[line][window]; +- int y = (*ix_s)[line + 1][window]; +- bits += HuffmanCode( tableindex, x, y, &code, &ext, &cbits, &xbits ); +- } +- } +- } +- else +- { +- unsigned int table; +- +- if( (table=gi->table_select[0]) != 0 ) /* region0 */ +- bits += count_bit(ix, 0, gi->address1, table ); +- if( (table=gi->table_select[1]) != 0 ) /* region1 */ +- bits += count_bit(ix, gi->address1, gi->address2, table ); +- if( (table=gi->table_select[2]) != 0 ) /* region2 */ +- bits += count_bit(ix, gi->address2, gi->address3, table ); +- } +- return bits; +- } +- +- +- +- /*************************************************************************/ +- /* count_bit */ +- /*************************************************************************/ +- +- /* +- Function: Count the number of bits necessary to code the subregion. +- */ +- +- int count_bit( int ix[576], unsigned int start, unsigned int end, unsigned int table ) +- { +- int i, sum; +- +- sum = 0; +- for ( i = start; i < end; i += 2 ) +- { +- unsigned int code, ext; +- int cbits, xbits; +- sum += HuffmanCode( table, ix[i], ix[i+1], &code, &ext, &cbits, &xbits ); +- } +- return sum; +- } +- +- +- +- #ifndef HAVE_NINT +- int +- nint( double in ) +- { +- int temp; +- +- if( in < 0 ) temp = (int)(in - 0.5); +- else temp = (int)(in + 0.5); +- +- return(temp); +- } +- +- double +- aint(double in) { +- return((long) in); +- } +- #endif +- +- +- +- /* +- Seymour's comment: Jan 8 1995 +- When mixed_block_flag is set, the low subbands 0-1 undergo the long +- window transform and are each split into 18 frequency lines, while +- the remaining 30 subbands undergo the short window transform and are +- each split into 6 frequency lines. A problem now arises, as neither +- the short or long scale factor bands apply to this mixed spectrum. +- The standard resolves this situation by using the first 8 long scale +- factor bands for the low spectrum and the short scale factor bands +- in the range of 3 to 11 (inclusive) for the remaining frequency lines. +- These scale factor bands do not match exactly to the 0-1 subbands +- for all sampling frequencies (32,44.1 and 48 kHz); however they +- were designed so that there would not be a frequency gap or overlap +- at the switch over point. (Note multiply short frequency lines by 3 +- to account for wider frequency line.) +- */ +- +- +- +- /*************************************************************************/ +- /* gr_deco */ +- /*************************************************************************/ +- +- void gr_deco( gr_info *cod_info ) +- { +- if ( cod_info->window_switching_flag != 0 && cod_info->block_type == 2 ) +- if ( cod_info->mixed_block_flag == 0 ) +- { +- cod_info->sfb_lmax = 0; /* No sb*/ +- cod_info->sfb_smax = 0; +- } +- else +- { +- cod_info->sfb_lmax = 8; +- cod_info->sfb_smax = 3; +- } +- else +- { +- cod_info->sfb_lmax = SFB_LMAX - 1; +- cod_info->sfb_smax = SFB_SMAX - 1; /* No sb */ +- } +- } +- +- +- +- +- +- +- +- +- /* The following optional code written by Seymour Shlien +- will speed up the outer_loop code which is called +- by iteration_loop. When BIN_SEARCH is defined, the +- outer_loop function precedes the call to the function inner_loop +- with a call to bin_search gain defined below, which +- returns a good starting quantizerStepSize. +- */ +- +- #if defined(BIN_SEARCH) || defined(PERFORM) +- int count_bits(ix,cod_info) +- int *ix; /* I576 *ix; */ +- gr_info *cod_info; +- { +- int bits,max; +- calc_runlen(ix,cod_info); /*rzero,count1,big_values*/ +- max = ix_max( ix, 0,576); +- if(max > 8192) return 100000; /* report unsuitable quantizer */ +- bits = count1_bitcount(ix, cod_info); /*count1_table selection*/ +- subdivide(cod_info); /* bigvalues sfb division */ +- bigv_tab_select(ix,cod_info); /* codebook selection*/ +- bits += bigv_bitcount(ix,cod_info); /* bit count */ +- /* printf("\nglobal_gain = %f bits= %d ",cod_info->quantizerStepSize,bits);*/ +- return bits; +- } +- #endif +- +- +- #ifdef BIN_SEARCH +- +- int bin_search_StepSize(int desired_rate, double start, int *ix, +- double xrs[576], gr_info * cod_info) +- { +- double top,bot,next,last; +- int bit; +- top = start; +- bot = 200; +- next = start; +- do +- { +- last = next; +- next = aint((top+bot)/2.0); +- cod_info->quantizerStepSize = next; +- quantize(xrs,ix,cod_info); +- bit = count_bits(ix,cod_info); +- if (bit>desired_rate) top = next; +- else bot = next; +- /* printf("\n%f %f %f %d %d",next, top,bot,bit,desired_rate);*/ +- } +- while ((bit != desired_rate) && fabs(last - next) > 1.0); +- return next; +- } +- +- #endif +- +- +- +- +- #ifdef PERFORM +- +- /* The following code is used for exposing some problems with +- the outer_loop code. PERFORM should be defined to the +- frame number you wish to have additional output recorded +- in the file encode.log - Seymour Shlien 14-Jan-97 +- */ +- +- /* +- float worst_xfsf_to_xmin_ratio(l3_xmin,xfsf,block_type,gr,ch) +- double xfsf[4][CBLIMIT]; +- III_psy_xmin *l3_xmin; +- int block_type,gr,ch; +- */ +- float worst_xfsf_to_xmin_ratio(III_psy_xmin *l3_xmin, double xfsf[4][CBLIMIT] +- ,int block_type,int gr,int ch) +- +- { +- float ratio,maxratio; +- int i,j; +- maxratio =-100.0; +- if (block_type != 2) +- for(i=0;i<21;i++) +- { +- ratio = 10.*log10(xfsf[0][i] /l3_xmin->l[gr][ch][i]); +- if (ratio > maxratio) maxratio = ratio; +- } +- else +- { +- for(j=0;j<3;j++) +- /* for(i = cod_info->sfb_smax; i <SFB_SMAX; i++) */ +- for(i = 0; i <11; i++) +- { +- ratio = 10.*log10(xfsf[j+1][i] /l3_xmin->s[gr][ch][i][j]); +- if (ratio > maxratio) maxratio = ratio; +- } +- } +- return maxratio; +- } +- +- +- print_ratios(handle_out,l3_xmin,xfsf,block_type,gr,ch) +- FILE *handle_out; +- double xfsf[4][CBLIMIT]; +- III_psy_xmin *l3_xmin; +- int gr,ch; +- int block_type; +- { +- float ratio; +- int i,j; +- if(block_type !=2) +- for (i=0;i<21;i++) +- { +- ratio = 100.0; /* signals undefined value in output */ +- if(l3_xmin->l[gr][ch][i] >1.0e-20) +- ratio = 10.*log10(xfsf[0][i] /l3_xmin->l[gr][ch][i]); +- fprintf(handle_out,"%6.2f ",ratio); +- if(i%5==4) fprintf(handle_out,"\n"); +- } +- else +- +- for(j=0;j<3;j++) +- { fprintf(handle_out,"\n block %d\n",j); +- for(i = 0; i <11; i++) +- { +- ratio = 10.*log10(xfsf[j+1][i] /l3_xmin->s[gr][ch][i][j]); +- fprintf(handle_out,"%6.2f ",ratio); +- if(i%5==4) fprintf(handle_out,"\n"); +- } +- } +- fprintf(handle_out,"\n"); +- } +- +- +- print_scalefacs(handle_out,scalefac,block_type,gr,ch) +- FILE *handle_out; +- III_scalefac_t *scalefac; +- int gr,ch; +- int block_type; +- { +- int sfb,j; +- +- if(block_type !=2) +- for ( sfb = 0; sfb < 21; sfb++ ) +- { +- fprintf(handle_out,"%6d ", scalefac->l[gr][ch][sfb]); +- if(sfb%5==4) fprintf(handle_out,"\n"); +- } +- else +- for (j=0;j<3;j++) +- { +- fprintf(handle_out,"\n block %d\n",j); +- for (sfb=0;sfb<11;sfb++) +- { +- fprintf(handle_out,"%6d ",scalefac->s[gr][ch][sfb][j]); +- if(sfb%5==4) fprintf(handle_out,"\n"); +- } +- } +- fprintf(handle_out,"\n"); +- } +- +- +- print_quantized_values(FILE *handle, int ix[576], gr_info *cod_info) +- { +- int sfb,start,end,i,bw; +- for (sfb=0;sfb<cod_info->sfb_lmax;sfb++) +- { +- start = scalefac_band_long[sfb]; +- end = scalefac_band_long[sfb+1]; +- bw = end - start; +- fprintf(handle,"scalefac band %d from %d to %d\n",sfb,start,end); +- for (i=0;i<bw;i++) +- { +- fprintf(handle,"%8d",ix[start+i]); +- if(i%5==4) fprintf(handle,"\n"); +- } +- fprintf(handle,"\n"); +- } +- } +- +- +- test_inner_loop(double xr[2][2][576], int l3_enc[2][2][576], int max_bits, +- gr_info *cod_info, int gr, int ch, double xfsf[4][CBLIMIT], +- III_psy_xmin *l3_xmin) +- { +- int bits, c1bits, bvbits; +- double *xrs; /* D576 *xr; */ +- int *ix; /* I576 *ix; */ +- xrs = &xr[gr][ch][0]; +- ix = l3_enc[gr][ch]; +- +- assert( max_bits >= 0 ); +- cod_info->quantizerStepSize -= 1.0;; +- do +- { +- do +- { +- cod_info->quantizerStepSize += 1.0; +- quantize( xrs, ix, cod_info ); +- } +- while ( ix_max(ix, 0, 576) > 8191 + 14 ); /* within table range? */ +- +- bits = count_bits(ix,cod_info); +- +- if(frameNum == PERFORM) +- { +- fprintf(log_output,"StepSize=%f bits = %d huff_bits = %d\n", +- cod_info->quantizerStepSize,bits,max_bits); +- calc_noise( &xr[gr][ch][0], &l3_enc[gr][ch][0], cod_info, xfsf ); +- /* distortion calculation */ +- print_ratios(log_output,l3_xmin,xfsf,cod_info->block_type,gr,ch); +- fprintf(log_output,"\n\n"); +- } +- +- } +- while ( bits > max_bits ); +- +- return bits; +- } +- +- #endif PERFORM +--- 0 ---- +diff -r -c -N encoder/loop.h lame3.70/loop.h +*** encoder/loop.h Wed Jan 22 02:43:16 1997 +--- lame3.70/loop.h Wed Dec 31 17:00:00 1969 +*************** +*** 1,90 **** +- /********************************************************************** +- * ISO MPEG Audio Subgroup Software Simulation Group (1996) +- * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- * +- * $Id: loop.h,v 1.1 1996/02/14 04:04:23 rowlands Exp $ +- * +- * $Log: loop.h,v $ +- * Revision 1.1 1996/02/14 04:04:23 rowlands +- * Initial revision +- * +- * Received from Mike Coleman +- **********************************************************************/ +- +- #ifndef LOOP_DOT_H +- #define LOOP_DOT_H +- #include "common.h" +- #include "l3side.h" +- +- /********************************************************************** +- * date programmers comment * +- * 25. 6.92 Toshiyuki Ishino Ver 1.0 * +- * 29.10.92 Masahiro Iwadare Ver 2.0 * +- * 17. 4.93 Masahiro Iwadare Updated for IS Modification * +- * * +- *********************************************************************/ +- +- extern int cont_flag; +- +- #define e 2.71828182845 +- +- /*#define SBLIMIT 32*/ +- #define CBLIMIT 21 +- +- #define SFB_LMAX 22 +- #define SFB_SMAX 13 +- +- extern int pretab[]; +- +- struct scalefac_struct +- { +- int l[23]; +- int s[14]; +- }; +- +- extern struct scalefac_struct sfBandIndex[]; /* Table B.8 -- in loop.c */ +- +- +- void iteration_loop( double pe[][2], double xr_org[2][2][576], III_psy_ratio *ratio, +- III_side_info_t *l3_side, int l3_enc[2][2][576], int mean_bits, +- int stereo, double xr_dec[2][2][576], +- III_scalefac_t *scalefac, frame_params *fr_ps, +- int ancillary_pad, int bitsPerFrame ); +- +- +- int nint( double in ); +- +- /* #define PI 3.1415926535 */ +- #define maximum(A,B) ( (A) > (B) ? (A) : (B) ) +- #define minimum(A,B) ( (A) < (B) ? (A) : (B) ) +- #define signum( A ) ( (A) > 0 ? 1 : -1 ) +- +- /* GLOBALE VARIABLE */ +- +- /*extern FILE *debp; +- extern FILE *huffcp,*huffdp; +- extern FILE *musicin,*cod_music;*/ +- +- /* Beachte:Partitonen nur fuer 48 kHz */ +- /* andere ev. fehlerhaft */ +- +- /* static int scalefac_band_long[22]; +- static int scalefac_band_short[13]; +- */ +- /*static int huffman_tab_quad[2][16][2]; +- extern int bigv_cod_tab[17][2][16][16]; +- extern int bigv_dec_tab[17][512][5]; +- extern int bigv_root_node[17]; +- extern int count1_root_node[2];*/ +- /* extern int cod_tab_info[2][31]; */ +- +- +- +- +- /* in max spalte war um 1 zuviel */ +- /* [0][i] : maximalwert der Tabelle mit index i +- [1][i] : anzahl der linbits der TAb. i */ +- +- extern int bit_buffer[50000]; +- +- #endif +--- 0 ---- +diff -r -c -N encoder/machine.h lame3.70/machine.h +*** encoder/machine.h Wed Dec 31 17:00:00 1969 +--- lame3.70/machine.h Thu Apr 6 12:50:51 2000 +*************** +*** 0 **** +--- 1,112 ---- ++ /* ++ * Machine dependent defines/includes for LAME. ++ * ++ * Copyright (c) 1999 A.L. Faber ++ * ++ * This library is free software; you can redistribute it and/or ++ * modify it under the terms of the GNU Library General Public ++ * License as published by the Free Software Foundation; either ++ * version 2 of the License, or (at your option) any later version. ++ * ++ * This library is distributed in the hope that it will be useful, ++ * but WITHOUT ANY WARRANTY; without even the implied warranty of ++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU ++ * Library General Public License for more details. ++ * ++ * You should have received a copy of the GNU Library General Public ++ * License along with this library; if not, write to the ++ * Free Software Foundation, Inc., 59 Temple Place - Suite 330, ++ * Boston, MA 02111-1307, USA. ++ */ ++ ++ ++ #ifndef MACHINE_H_INCLUDED ++ #define MACHINE_H_INCLUDED ++ ++ #include <stdio.h> ++ #include <string.h> ++ #include <math.h> ++ #include <stdlib.h> ++ #include <ctype.h> ++ #include <signal.h> ++ #include <sys/types.h> ++ #include <sys/stat.h> ++ #include <fcntl.h> ++ #include <errno.h> ++ ++ ++ /* three different types of pow() functions: ++ 1. table lookup ++ 2. pow() ++ 3. exp() on some machines this is claimed to be faster than pow() ++ */ ++ ++ ++ #define POW20(x) pow20[x] ++ /* ++ #define POW20(x) pow(2.0,((double)(x)-210)*.25) ++ #define POW20(x) exp( ((double)(x)-210)*(.25*LOG2) ) ++ */ ++ ++ #define IPOW20(x) ipow20[x] ++ /* ++ #define IPOW20(x) exp( -((double)(x)-210)*.1875*LOG2 ) ++ #define IPOW20(x) pow(2.0,-((double)(x)-210)*.1875) ++ */ ++ ++ ++ ++ #if ( defined(_MSC_VER) && !defined(INLINE)) ++ # define INLINE _inline ++ #elif defined(__SASC) || defined(__GNUC__) ++ # define INLINE __inline ++ #else ++ # define INLINE ++ #endif ++ ++ #if ( defined(_MSC_VER)) ++ # pragma warning( disable : 4244 ) ++ # pragma warning( disable : 4305 ) ++ #endif ++ ++ #if ( defined(_MSC_VER) || defined(__BORLANDC__) ) ++ # define WIN32_LEAN_AND_MEAN ++ # include <windows.h> ++ #else ++ typedef float FLOAT; ++ #endif ++ ++ ++ /* MH: the x86 asm quantization routines in quantize-pvt.c ++ are designed to work only with 8-byte doubles or 4-byte ++ floats. if you use a different type (e.g. 10-byte extended ++ precision long doubles, as supported by ICL), you will need ++ to disable the ASM routines (or fix them :) */ ++ ++ #define FLOAT8_is_double ++ typedef double FLOAT8; ++ ++ /*#define FLOAT8_is_float*/ ++ /*typedef float FLOAT8; */ ++ ++ ++ ++ ++ #if defined _WIN32 && !defined __CYGWIN__ ++ typedef unsigned long u_long; ++ typedef unsigned int u_int; ++ typedef unsigned short u_short; ++ typedef unsigned char u_char; ++ #elif defined __DECALPHA__ ++ # do nothing ++ #elif !defined __GNUC__ || defined __STRICT_ANSI__ ++ typedef unsigned long u_long; ++ typedef unsigned int u_int; ++ typedef unsigned short u_short; ++ typedef unsigned char u_char; ++ #endif ++ ++ ++ ++ ++ #endif +diff -r -c -N encoder/main.c lame3.70/main.c +*** encoder/main.c Wed Dec 31 17:00:00 1969 +--- lame3.70/main.c Thu Apr 6 12:50:51 2000 +*************** +*** 0 **** +--- 1,142 ---- ++ #include <stdlib.h> ++ #include <string.h> ++ #include <stdio.h> ++ #include <fcntl.h> ++ #ifdef _WIN32 ++ /* needed to set stdout to binary */ ++ #include <io.h> ++ #endif ++ #include "lame.h" ++ ++ ++ #ifdef HAVEGTK ++ #include "gtkanal.h" ++ #include <gtk/gtk.h> ++ #endif ++ ++ #ifdef __riscos__ ++ #include "asmstuff.h" ++ #endif ++ ++ ++ ++ ++ /************************************************************************ ++ * ++ * main ++ * ++ * PURPOSE: MPEG-1,2 Layer III encoder with GPSYCHO ++ * psychoacoustic model. ++ * ++ ************************************************************************/ ++ ++ ++ int main(int argc, char **argv) ++ { ++ ++ char mp3buffer[LAME_MAXMP3BUFFER]; ++ short int Buffer[2][1152]; ++ int iread,imp3; ++ lame_global_flags gf; ++ FILE *outf; ++ #ifdef __riscos__ ++ int i; ++ #endif ++ ++ ++ lame_init(&gf); /* initialize libmp3lame */ ++ if(argc==1) lame_usage(&gf,argv[0]); /* no command-line args, print usage, exit */ ++ ++ /* parse the command line arguments, setting various flags in the ++ * struct 'gf'. If you want to parse your own arguments, ++ * or call libmp3lame from a program which uses a GUI to set arguments, ++ * skip this call and set the values of interest in the gf struct. ++ * (see lame.h for documentation about these parameters) ++ */ ++ lame_parse_args(&gf,argc, argv); ++ ++ if (!gf.gtkflag) { ++ /* open the MP3 output file */ ++ if (!strcmp(gf.outPath, "-")) { ++ #ifdef __EMX__ ++ _fsetmode(stdout,"b"); ++ #elif (defined __BORLANDC__) ++ setmode(_fileno(stdout), O_BINARY); ++ #elif (defined __CYGWIN__) ++ setmode(fileno(stdout), _O_BINARY); ++ #elif (defined _WIN32) ++ _setmode(_fileno(stdout), _O_BINARY); ++ #endif ++ outf = stdout; ++ } else { ++ if ((outf = fopen(gf.outPath, "wb")) == NULL) { ++ fprintf(stderr,"Could not create \"%s\".\n", gf.outPath); ++ exit(1); ++ } ++ } ++ #ifdef __riscos__ ++ /* Assign correct file type */ ++ for (i = 0; gf.outPath[i]; i++) ++ if (gf.outPath[i] == '.') gf.outPath[i] = '/'; ++ SetFiletype(gf.outPath, 0x1ad); ++ #endif ++ } ++ ++ ++ /* open the wav/aiff/raw pcm or mp3 input file. This call will ++ * open the file with name gf.inFile, try to parse the headers and ++ * set gf.samplerate, gf.num_channels, gf.num_samples. ++ * if you want to do your own file input, skip this call and set ++ * these values yourself. ++ */ ++ lame_init_infile(&gf); ++ ++ /* Now that all the options are set, lame needs to analyze them and ++ * set some more options ++ */ ++ lame_init_params(&gf); ++ lame_print_config(&gf); /* print usefull information about options being used */ ++ ++ ++ ++ ++ #ifdef HAVEGTK ++ if (gf.gtkflag) gtk_init (&argc, &argv); ++ if (gf.gtkflag) gtkcontrol(&gf); ++ else ++ #endif ++ { ++ ++ /* encode until we hit eof */ ++ do { ++ /* read in 'iread' samples */ ++ iread=lame_readframe(&gf,Buffer); ++ ++ ++ /* encode */ ++ imp3=lame_encode_buffer(&gf,Buffer[0],Buffer[1],iread, ++ mp3buffer,(int)sizeof(mp3buffer)); ++ ++ /* was our output buffer big enough? */ ++ if (imp3==-1) { ++ fprintf(stderr,"mp3 buffer is not big enough... \n"); ++ exit(1); ++ } ++ ++ if (fwrite(mp3buffer,1,imp3,outf) != imp3) { ++ fprintf(stderr,"Error writing mp3 output"); ++ exit(1); ++ } ++ } while (iread); ++ } ++ ++ imp3=lame_encode_finish(&gf,mp3buffer,(int)sizeof(mp3buffer)); /* may return one more mp3 frame */ ++ fwrite(mp3buffer,1,imp3,outf); ++ fclose(outf); ++ lame_close_infile(&gf); /* close the input file */ ++ lame_mp3_tags(&gf); /* add id3 or VBR tags to mp3 file */ ++ return 0; ++ } ++ ++ ++ +diff -r -c -N encoder/makefile.unix lame3.70/makefile.unix +*** encoder/makefile.unix Wed Jan 22 02:43:23 1997 +--- lame3.70/makefile.unix Wed Dec 31 17:00:00 1969 +*************** +*** 1,77 **** +- ############################################################################ +- ## ISO MPEG Audio Subgroup Software Simulation Group (1996) +- ## ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- ## +- ## $Id: makefile.unix,v 1.1 1996/02/14 05:28:25 rowlands Exp $ +- ## +- ## Makefile for encoder for vanilla UNIX platform using standard make. +- ## +- ## $Log: makefile.unix,v $ +- ## Revision 1.1 1996/02/14 05:28:25 rowlands +- ## Initial revision +- ## +- ############################################################################ +- +- # MODE= -O -DUNIX +- MODE= -g -DUNIX +- #MODE= -DMSC60 -AH -Zi -Gt +- OBJ_SUFFIX=.o +- EXE_SUFFIX= +- +- PROGRAM = encode$(EXE_SUFFIX) +- +- all: $(PROGRAM) +- +- $(PROGRAM) : \ +- common.c common.h encode.c encoder.h formatBitstream.c \ +- formatBitstream.h huffman.c huffman.h ieeefloat.c ieeefloat.h \ +- l3bitstream-pvt.h l3bitstream.c l3bitstream.h l3psy.c l3psy.h \ +- l3side.h loop-pvt.h loop.c loop.h mdct.c mdct.h musicin.c \ +- portableio.c portableio.h psy.c reservoir.c reservoir.h subs.c tonal.c +- +- HEDS = \ +- common.h encoder.h formatBitstream.h huffman.h ieeefloat.h \ +- l3bitstream-pvt.h l3bitstream.h l3psy.h l3side.h loop-pvt.h loop.h \ +- mdct.h portableio.h reservoir.h +- +- OBJS = \ +- common$(OBJ_SUFFIX) \ +- encode$(OBJ_SUFFIX) \ +- formatBitstream$(OBJ_SUFFIX) \ +- huffman$(OBJ_SUFFIX) \ +- ieeefloat$(OBJ_SUFFIX) \ +- l3bitstream$(OBJ_SUFFIX) \ +- l3psy$(OBJ_SUFFIX) \ +- loop$(OBJ_SUFFIX) \ +- mdct$(OBJ_SUFFIX) \ +- musicin$(OBJ_SUFFIX) \ +- portableio$(OBJ_SUFFIX) \ +- psy$(OBJ_SUFFIX) \ +- reservoir$(OBJ_SUFFIX) \ +- subs$(OBJ_SUFFIX) \ +- tonal$(OBJ_SUFFIX) +- +- LFLAG = +- +- CC = gcc +- LN = gcc +- LNEXTRA=-lm +- #CC = cl +- #LN = cl +- +- .c$(OBJ_SUFFIX): $(HEDS) +- $(CC) -c $(MODE) $*.c +- +- all: $(PROGRAM) +- +- default: all +- +- +- $(PROGRAM) : $(OBJS) $(HEDS) +- $(LN) -o $(PROGRAM) $(MODE) $(LFLAG) $(OBJS) $(LNEXTRA) +- +- clean: +- rm -f $(OBJS) +- +- veryclean: +- rm -f $(PROGRAM) $(OBJS) +--- 0 ---- +diff -r -c -N encoder/mdct.c lame3.70/mdct.c +*** encoder/mdct.c Wed Jan 22 02:43:17 1997 +--- lame3.70/mdct.c Wed Dec 31 17:00:00 1969 +*************** +*** 1,212 **** +- /********************************************************************** +- * ISO MPEG Audio Subgroup Software Simulation Group (1996) +- * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- * +- * $Id: mdct.c,v 1.1 1996/02/14 04:04:23 rowlands Exp $ +- * +- * $Log: mdct.c,v $ +- * Revision 1.1 1996/02/14 04:04:23 rowlands +- * Initial revision +- * +- * Received from Mike Coleman +- **********************************************************************/ +- +- #include "common.h" +- #include "l3side.h" +- #include "mdct.h" +- +- double ca[8], cs[8]; +- +- /* +- This is table B.9: coefficients for aliasing reduction +- */ +- static double c[8] = { -0.6,-0.535,-0.33,-0.185,-0.095,-0.041,-0.0142, -0.0037 }; +- +- void mdct_sub( L3SBS (*sb_sample), double (*mdct_freq)[2][576], int stereo, III_side_info_t *l3_side, int mode_gr ) +- { +- gr_info *cod_info; +- double mdct_in[36]; +- int ch,gr,band,k,j; +- double bu,bd; +- static int init = 0; +- int block_type; +- double (*mdct_enc)[2][32][18] = (double (*)[2][32][18]) mdct_freq; +- +- if ( init == 0 ) +- { +- /* prepare the aliasing reduction butterflies */ +- for ( k = 0; k < 8; k++ ) +- { +- double sq; +- sq = sqrt( 1.0 + c[k] * c[k] ); +- ca[k] = c[k] / sq; +- cs[k] = 1.0 / sq; +- } +- init++; +- } +- +- for ( gr = 0; gr < mode_gr; gr++ ) +- for ( ch = 0; ch < stereo; ch++ ) +- { +- cod_info = (gr_info *) &(l3_side->gr[gr].ch[ch]) ; +- block_type = cod_info->block_type; +- +- /* +- Compensate for inversion in the analysis filter +- */ +- for ( band = 0; band < 32; band++ ) +- for ( k = 0; k < 18; k++ ) +- if ( (band & 1) && (k & 1) ) +- (*sb_sample)[ch][gr+1][k][band] *= -1.0; +- +- /* +- Perform imdct of 18 previous subband samples +- + 18 current subband samples +- */ +- for ( band = 0; band < 32; band++ ) +- { +- for ( k = 0; k < 18; k++ ) +- { +- mdct_in[k] = (*sb_sample)[ch][ gr ][k][band]; +- mdct_in[k+18] = (*sb_sample)[ch][gr+1][k][band]; +- } +- if ( cod_info->mixed_block_flag && (band < 2) ) +- block_type = 0; +- +- mdct( mdct_in, &mdct_enc[gr][ch][band][0], block_type ); +- } +- +- /* +- Perform aliasing reduction butterfly +- on long blocks +- */ +- if ( block_type != 2 ) +- for ( band = 0; band < 31; band++ ) +- for ( k = 0; k < 8; k++ ) +- { +- bu = mdct_enc[gr][ch][band][17-k] * cs[k] + mdct_enc[gr][ch][band+1][k] * ca[k]; +- bd = mdct_enc[gr][ch][band+1][k] * cs[k] - mdct_enc[gr][ch][band][17-k] * ca[k]; +- mdct_enc[gr][ch][band][17-k] = bu; +- mdct_enc[gr][ch][band+1][k] = bd; +- } +- +- } +- +- /* +- Save latest granule's subband samples to be used in +- the next mdct call +- */ +- for ( ch = 0; ch < stereo; ch++ ) +- for ( j = 0; j < 18; j++ ) +- for ( band = 0; band < 32; band++ ) +- (*sb_sample)[ch][0][j][band] = (*sb_sample)[ch][mode_gr][j][band]; +- } +- +- void mdct( double *in, double *out, int block_type ) +- { +- /*-------------------------------------------------------------------*/ +- /* */ +- /* Function: Calculation of the MDCT */ +- /* In the case of long blocks ( block_type 0,1,3 ) there are */ +- /* 36 coefficents in the time domain and 18 in the frequency */ +- /* domain. */ +- /* In the case of short blocks (block_type 2 ) there are 3 */ +- /* transformations with short length. This leads to 12 coefficents */ +- /* in the time and 6 in the frequency domain. In this case the */ +- /* results are stored side by side in the vector out[]. */ +- /* */ +- /* New layer3 */ +- /* */ +- /*-------------------------------------------------------------------*/ +- +- int l,k,i,m,N; +- double sum; +- static double win[4][36]; +- static int init = 0; +- static double cos_s[6][12], cos_l[18][36]; +- +- +- if ( init == 0 ) +- { +- /* type 0 */ +- for ( i = 0; i < 36; i++ ) +- win[0][i] = sin( PI/36 * (i + 0.5) ); +- /* type 1*/ +- for ( i = 0; i < 18; i++ ) +- win[1][i] = sin( PI/36 * (i + 0.5) ); +- for ( i = 18; i < 24; i++ ) +- win[1][i] = 1.0; +- for ( i = 24; i < 30; i++ ) +- win[1][i] = sin( PI/12 * ( i + 0.5 - 18) ); +- for ( i = 30; i < 36; i++ ) +- win[1][i] = 0.0; +- /* type 3*/ +- for ( i = 0; i < 6; i++ ) +- win[3][i] = 0.0; +- for ( i = 6; i < 12; i++ ) +- win[3][i] = sin( PI/12 * (i + 0.5 - 6) ); +- for ( i = 12; i < 18; i++ ) +- win[3][i] = 1.0; +- for ( i = 18; i < 36; i++ ) +- win[3][i] = sin( PI/36 * (i + 0.5) ); +- /* type 2*/ +- for ( i = 0; i < 12; i++ ) +- win[2][i] = sin( PI/12 * (i + 0.5) ); +- for ( i = 12; i < 36; i++ ) +- win[2][i] = 0.0; +- +- N = 12; +- for ( m = 0; m < N / 2; m++ ) +- for ( k = 0; k < N; k++ ) +- cos_s[m][k] = cos( (PI /(2 * N)) * (2 * k + 1 + N / 2) * +- (2 * m + 1) ) / (N / 4); +- +- N = 36; +- for ( m = 0; m < N / 2; m++ ) +- for ( k = 0; k < N; k++ ) +- cos_l[m][k] = cos( (PI / (2 * N)) * (2 * k + 1 + N / 2) * +- (2 * m + 1) ) / (N / 4); +- +- init++; +- } +- +- if ( block_type == 2 ) +- { +- N = 12; +- for ( l = 0; l < 3; l++ ) +- { +- for ( m = 0; m < N / 2; m++ ) +- { +- for ( sum = 0.0, k = 0; k < N; k++ ) +- sum += win[block_type][k] * in[k + 6 * l + 6] * cos_s[m][k]; +- out[ 3 * m + l] = sum; +- } +- } +- } +- else +- { +- N = 36; +- for ( m = 0; m < N / 2; m++ ) +- { +- for ( sum = 0.0, k = 0; k < N; k++ ) +- sum += win[block_type][k] * in[k] * cos_l[m][k]; +- out[m] = sum; +- } +- } +- } +- +- void +- delay( double (*xr)[2][576], int stereo ) +- { +- static double xr_buff[2][576]; +- double xr_buff2[2][576]; +- unsigned int i,j; +- +- for (i=0;i<stereo;i++) +- { +- for (j=0;j<576;j++) xr_buff2[i][j] = xr_buff[i][j]; +- for (j=0;j<576;j++) xr_buff[i][j] = xr[1][i][j]; +- for (j=0;j<576;j++) xr[1][i][j] = xr[0][i][j]; +- for (j=0;j<576;j++) xr[0][i][j] = xr_buff2[i][j]; +- } +- } +--- 0 ---- +diff -r -c -N encoder/mdct.h lame3.70/mdct.h +*** encoder/mdct.h Wed Jan 22 02:43:17 1997 +--- lame3.70/mdct.h Wed Dec 31 17:00:00 1969 +*************** +*** 1,25 **** +- /********************************************************************** +- * ISO MPEG Audio Subgroup Software Simulation Group (1996) +- * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- * +- * $Id: mdct.h,v 1.1 1996/02/14 04:04:23 rowlands Exp $ +- * +- * $Log: mdct.h,v $ +- * Revision 1.1 1996/02/14 04:04:23 rowlands +- * Initial revision +- * +- * Received from Mike Coleman +- **********************************************************************/ +- +- #ifndef MDCT_DOT_H +- #define MDCT_DOT_H +- void mdct(double *in, double *out, int block_type); +- void inv_mdct(double *in, double *out, int block_type); +- +- typedef double D32_18[SBLIMIT][18]; +- typedef double L3SBS[2][3][18][SBLIMIT]; /* [gr][ch] */ +- +- void mdct_sub(L3SBS (*sb_sample), double (*mdct_freq)[2][576], int stereo, III_side_info_t *l3_side, int mode_gr ); +- void mdct_sub_dec(double (*mdct_freq)[2][576], double inv_mdct_dec[3][2][18][32], int stereo, III_side_info_t *l3_side); +- void delay(double (*xr)[2][576], int stereo); +- #endif +--- 0 ---- +diff -r -c -N encoder/mlame lame3.70/mlame +*** encoder/mlame Wed Dec 31 17:00:00 1969 +--- lame3.70/mlame Sun Jan 30 22:49:48 2000 +*************** +*** 0 **** +--- 1,81 ---- ++ #!/bin/bash ++ #!/usr/local/bin/bash ++ ############################################################################ ++ # ++ # Run the LAME encoder on multiple files, with option to delete .wav files ++ # after encoding. "mlame -h" will give instructions. ++ # ++ # Robert Hegemann <Robert.Hegemann@gmx.de> ++ # ++ ############################################################################ ++ ++ mp3coder="lame" ++ options="-h -d -m j -b 128" ++ rmsrc=false ++ ++ helptext="\ ++ \nThis script runs the LAME mp3 encoder on multiple files: \n\n\ ++ $0 [options] <file 1> ... <file n>\n\ ++ \n\ ++ options:\n\ ++ -h this help text\n\ ++ -r remove files after encoding\n\ ++ -o \"<lame options>\" overrides script default options \"${options}\"\n\ ++ \n\ ++ example:\n\ ++ $0 -r -o \"-v -V 0 -b 112\" a*.wav z*.aif\n\ ++ \n\ ++ " ++ ++ # process command-line options ++ # this could be extended to fake the ++ # commandline interface of the mp3encoder ++ ++ while getopts ":o:r" optn; do ++ case $optn in ++ o ) options=$OPTARG # replace default options ++ ;; ++ r ) rmsrc=true ++ ;; ++ \? ) printf "$helptext" ++ exit 1 ++ ;; ++ esac ++ done ++ shift $(($OPTIND - 1)) ++ ++ # process input-files ++ ++ for filename in "$@"; do ++ case $filename in ++ *[*?]* ) # means shell couldn�t extend *.wav, etc. ++ echo "warning: no $filename file(s) found" ++ ;; ++ *[.][wW][aA][vV] ) ++ name=${filename%[.][wW][aA][vV]} ++ if $mp3coder $options "$filename" "${name}.mp3" ++ then ++ if [ $rmsrc = true ]; then ++ rm -f "$filename" ++ fi ++ fi ++ ;; ++ *[.][aA][iI][fF] ) ++ name=${filename%[.][aA][iI][fF]} ++ if $mp3coder $options "$filename" "${name}.mp3" ++ then ++ if [ $rmsrc = true ]; then ++ rm -f "$filename" ++ fi ++ fi ++ ;; ++ * ) ++ if $mp3coder $options "$filename" "${filename}.mp3" ++ then ++ if [ $rmsrc = true ]; then ++ rm -f "$filename" ++ fi ++ fi ++ ;; ++ esac ++ done +diff -r -c -N encoder/mp3rtp.c lame3.70/mp3rtp.c +*** encoder/mp3rtp.c Wed Dec 31 17:00:00 1969 +--- lame3.70/mp3rtp.c Thu Mar 16 12:05:26 2000 +*************** +*** 0 **** +--- 1,171 ---- ++ #include <stdlib.h> ++ #include <string.h> ++ #include <time.h> ++ #include <unistd.h> ++ #include "lame.h" ++ #include "rtp.h" ++ ++ ++ /* ++ ++ encode (via LAME) to mp3 with RTP streaming of the output. ++ ++ Author: Felix von Leitner <leitner@vim.org> ++ ++ mp3rtp ip:port:ttl [lame encoding options] infile outfile ++ ++ example: ++ ++ arecord -b 16 -s 22050 -w | ./mp3rtp 224.17.23.42:5004:2 -b 56 - /dev/null ++ ++ ++ ++ */ ++ ++ ++ struct rtpheader RTPheader; ++ struct sockaddr_in rtpsi; ++ int rtpsocket; ++ ++ void rtp_output(char *mp3buffer,int mp3size) ++ { ++ sendrtp(rtpsocket,&rtpsi,&RTPheader,mp3buffer,mp3size); ++ RTPheader.timestamp+=5; ++ RTPheader.b.sequence++; ++ } ++ ++ void rtp_usage(void) { ++ fprintf(stderr,"usage: mp3rtp ip:port:ttl [encoder options] <infile> <outfile>\n"); ++ exit(1); ++ } ++ ++ ++ ++ char mp3buffer[LAME_MAXMP3BUFFER]; ++ ++ ++ /************************************************************************ ++ * ++ * main ++ * ++ * PURPOSE: MPEG-1,2 Layer III encoder with GPSYCHO ++ * psychoacoustic model. ++ * ++ ************************************************************************/ ++ ++ ++ int main(int argc, char **argv) ++ { ++ ++ int i,port,ttl; ++ char *tmp,*Arg; ++ lame_global_flags gf; ++ int iread,imp3; ++ FILE *outf; ++ short int Buffer[2][1152]; ++ ++ if(argc<=2) { ++ rtp_usage(); ++ exit(1); ++ } ++ ++ /* process args */ ++ Arg = argv[1]; ++ tmp=strchr(Arg,':'); ++ ++ if (!tmp) { ++ rtp_usage(); ++ exit(1); ++ } ++ *tmp++=0; ++ port=atoi(tmp); ++ if (port<=0) { ++ rtp_usage(); ++ exit(1); ++ } ++ tmp=strchr(tmp,':'); ++ if (!tmp) { ++ rtp_usage(); ++ exit(1); ++ } ++ *tmp++=0; ++ ttl=atoi(tmp); ++ if (tmp<=0) { ++ rtp_usage(); ++ exit(1); ++ } ++ rtpsocket=makesocket(Arg,port,ttl,&rtpsi); ++ srand(getpid() ^ time(0)); ++ initrtp(&RTPheader); ++ ++ ++ /* initialize encoder */ ++ lame_init(&gf); ++ ++ /* Remove the argumets that are rtp related, and then ++ * parse the command line arguments, setting various flags in the ++ * struct pointed to by 'gf'. If you want to parse your own arguments, ++ * or call libmp3lame from a program which uses a GUI to set arguments, ++ * skip this call and set the values of interest in the gf struct. ++ * (see lame.h for documentation about these parameters) ++ */ ++ for (i=1; i<argc-1; i++) /* remove first argument, it was for rtp */ ++ argv[i]=argv[i+1]; ++ lame_parse_args(&gf,argc-1, argv); ++ ++ /* open the output file. Filename parsed into gf.inPath */ ++ if (!strcmp(gf.outPath, "-")) { ++ #ifdef __EMX__ ++ _fsetmode(stdout,"b"); ++ #elif (defined __BORLANDC__) ++ setmode(_fileno(stdout), O_BINARY); ++ #elif (defined __CYGWIN__) ++ setmode(fileno(stdout), _O_BINARY); ++ #elif (defined _WIN32) ++ _setmode(_fileno(stdout), _O_BINARY); ++ #endif ++ outf = stdout; ++ } else { ++ if ((outf = fopen(gf.outPath, "wb")) == NULL) { ++ fprintf(stderr,"Could not create \"%s\".\n", gf.outPath); ++ exit(1); ++ } ++ } ++ ++ ++ /* open the wav/aiff/raw pcm or mp3 input file. This call will ++ * open the file with name gf.inFile, try to parse the headers and ++ * set gf.samplerate, gf.num_channels, gf.num_samples. ++ * if you want to do your own file input, skip this call and set ++ * these values yourself. ++ */ ++ lame_init_infile(&gf); ++ ++ ++ /* Now that all the options are set, lame needs to analyze them and ++ * set some more options ++ */ ++ lame_init_params(&gf); ++ lame_print_config(&gf); /* print usefull information about options being used */ ++ ++ /* encode until we hit eof */ ++ do { ++ /* read in 'iread' samples */ ++ iread=lame_readframe(&gf,Buffer); ++ /* encode the frame */ ++ imp3=lame_encode_buffer(&gf,Buffer[0],Buffer[1],iread, ++ mp3buffer,sizeof(mp3buffer)); ++ fwrite(mp3buffer,1,imp3,outf); /* write the MP3 output to file */ ++ rtp_output(mp3buffer,imp3); /* write MP3 output to RTP port */ ++ } while (iread); ++ ++ ++ imp3=lame_encode_finish(&gf,mp3buffer,sizeof(mp3buffer)); /* may return one or more mp3 frame */ ++ fwrite(mp3buffer,1,imp3,outf); ++ rtp_output(mp3buffer,imp3); ++ fclose(outf); ++ lame_close_infile(&gf); /* close the sound input file */ ++ lame_mp3_tags(&gf); /* add id3 or VBR tags to mp3 file */ ++ return 0; ++ } ++ +diff -r -c -N encoder/mp3x.c lame3.70/mp3x.c +*** encoder/mp3x.c Wed Dec 31 17:00:00 1969 +--- lame3.70/mp3x.c Thu Mar 16 12:05:26 2000 +*************** +*** 0 **** +--- 1,39 ---- ++ #include "lame.h" ++ ++ #include "gtkanal.h" ++ #include <gtk/gtk.h> ++ ++ ++ ++ ++ /************************************************************************ ++ * ++ * main ++ * ++ * PURPOSE: MPEG-1,2 Layer III encoder with GPSYCHO ++ * psychoacoustic model. ++ * ++ ************************************************************************/ ++ int main(int argc, char **argv) ++ { ++ char mp3buffer[LAME_MAXMP3BUFFER]; ++ lame_global_flags gf; ++ ++ lame_init(&gf); ++ if(argc==1) lame_usage(&gf,argv[0]); /* no command-line args */ ++ ++ lame_parse_args(&gf,argc, argv); ++ gf.gtkflag=1; ++ lame_init_infile(&gf); ++ lame_init_params(&gf); ++ lame_print_config(&gf); ++ ++ ++ gtk_init (&argc, &argv); ++ gtkcontrol(&gf); ++ ++ lame_encode_finish(&gf,mp3buffer,sizeof(mp3buffer)); ++ lame_close_infile(&gf); ++ return 0; ++ } ++ +diff -r -c -N encoder/mpglib/.cvsignore lame3.70/mpglib/.cvsignore +*** encoder/mpglib/.cvsignore Wed Dec 31 17:00:00 1969 +--- lame3.70/mpglib/.cvsignore Tue Nov 30 12:30:46 1999 +*************** +*** 0 **** +--- 1,9 ---- ++ *.d ++ Debug ++ DebugGTK ++ Release ++ ReleaseGTK ++ *.ncb ++ *.plg ++ *.opt ++ *.dll +diff -r -c -N encoder/mpglib/CVS/Entries lame3.70/mpglib/CVS/Entries +*** encoder/mpglib/CVS/Entries Wed Dec 31 17:00:00 1969 +--- lame3.70/mpglib/CVS/Entries Thu Apr 6 12:50:53 2000 +*************** +*** 0 **** +--- 1,16 ---- ++ /.cvsignore/1.1/Tue Nov 30 19:30:46 1999//Tlame3_70 ++ /Makefile/1.1.1.1/Wed Nov 24 08:44:31 1999//Tlame3_70 ++ /README/1.1.1.1/Wed Nov 24 08:44:33 1999//Tlame3_70 ++ /TODO/1.1.1.1/Wed Nov 24 08:44:33 1999//Tlame3_70 ++ /common.c/1.4/Thu Mar 23 22:12:26 2000//Tlame3_70 ++ /dct64_i386.c/1.2/Mon Mar 6 19:53:03 2000//Tlame3_70 ++ /decode_i386.c/1.2/Mon Mar 6 19:53:03 2000//Tlame3_70 ++ /huffman.h/1.1.1.1/Wed Nov 24 08:44:45 1999//Tlame3_70 ++ /interface.c/1.2/Mon Mar 6 19:53:03 2000//Tlame3_70 ++ /layer3.c/1.13/Thu Apr 6 18:50:53 2000//Tlame3_70 ++ /main.c/1.13/Fri Mar 17 00:05:38 2000//Tlame3_70 ++ /mpg123.h/1.3/Sat Feb 19 13:33:12 2000//Tlame3_70 ++ /mpglib.dsp/1.6/Fri Mar 17 00:05:38 2000//Tlame3_70 ++ /mpglib.h/1.2/Wed Jan 5 17:40:35 2000//Tlame3_70 ++ /tabinit.c/1.2/Mon Mar 6 19:53:03 2000//Tlame3_70 ++ D +diff -r -c -N encoder/mpglib/CVS/Repository lame3.70/mpglib/CVS/Repository +*** encoder/mpglib/CVS/Repository Wed Dec 31 17:00:00 1969 +--- lame3.70/mpglib/CVS/Repository Thu Apr 6 12:50:37 2000 +*************** +*** 0 **** +--- 1 ---- ++ lame/mpglib +diff -r -c -N encoder/mpglib/CVS/Root lame3.70/mpglib/CVS/Root +*** encoder/mpglib/CVS/Root Wed Dec 31 17:00:00 1969 +--- lame3.70/mpglib/CVS/Root Thu Apr 6 12:50:37 2000 +*************** +*** 0 **** +--- 1 ---- ++ markt@cvs.lame.sourceforge.net:/cvsroot/lame +diff -r -c -N encoder/mpglib/CVS/Tag lame3.70/mpglib/CVS/Tag +*** encoder/mpglib/CVS/Tag Wed Dec 31 17:00:00 1969 +--- lame3.70/mpglib/CVS/Tag Thu Apr 6 12:50:53 2000 +*************** +*** 0 **** +--- 1 ---- ++ Tlame3_70 +diff -r -c -N encoder/mpglib/Makefile lame3.70/mpglib/Makefile +*** encoder/mpglib/Makefile Wed Dec 31 17:00:00 1969 +--- lame3.70/mpglib/Makefile Wed Nov 24 01:44:31 1999 +*************** +*** 0 **** +--- 1,22 ---- ++ ++ CC=gcc ++ CFLAGS=-Wall -g ++ ++ all: mpglib ++ ++ ++ *.o: mpg123.h mpglib.h ++ ++ mpglib: common.o dct64_i386.o decode_i386.o layer3.o tabinit.o interface.o main.o ++ $(CC) -o mpglib common.o dct64_i386.o decode_i386.o layer3.o \ ++ tabinit.o interface.o main.o -lm ++ ++ lib: common.o dct64_i386.o decode_i386.o layer3.o tabinit.o interface.o main.o ++ ar -svru libmpg123.a main.o common.o dct64_i386.o decode_i386.o layer3.o \ ++ tabinit.o interface.o ++ ++ ++ clean: ++ rm *.o mpglib ++ ++ +diff -r -c -N encoder/mpglib/README lame3.70/mpglib/README +*** encoder/mpglib/README Wed Dec 31 17:00:00 1969 +--- lame3.70/mpglib/README Wed Nov 24 01:44:33 1999 +*************** +*** 0 **** +--- 1,33 ---- ++ MP3 library ++ ----------- ++ Version 0.2 ++ ++ This decoder is a 'light' version (thrown out all unnecessay parts) ++ from the mpg123 package. I made this for a company. ++ ++ Currently only Layer3 is enabled to save some space. Layer1,2 isn't ++ tested at all. The interface will not change significantly. ++ A backport to the mpg123 package is planed. ++ ++ comiled and tested only on Solaris 2.6 ++ main.c contains a simple demo application for library. ++ ++ COPYING: you may use this source under GPL terms! ++ ++ PLEASE NOTE: This software may contain patented alogrithm (at least ++ patented in some countries). It may be not allowed to sell/use products ++ based on this source code in these countries. Check this out first! ++ ++ COPYRIGHT of MP3 music: ++ Please note, that the duplicating of copyrighted music without explicit ++ permission violates the rights of the owner. ++ ++ SENDING PATCHES: ++ Maybe I change the copyright policy (ie some kind of more free BSD licencse). ++ Please consider this when sending patches/changes. ++ ++ FEEDBACK: ++ I'm interessted to here from you, when you use this package as part ++ of another project. ++ ++ +diff -r -c -N encoder/mpglib/TODO lame3.70/mpglib/TODO +*** encoder/mpglib/TODO Wed Dec 31 17:00:00 1969 +--- lame3.70/mpglib/TODO Wed Nov 24 01:44:33 1999 +*************** +*** 0 **** +--- 1,2 ---- ++ ++ apply 'VBR' bug +diff -r -c -N encoder/mpglib/common.c lame3.70/mpglib/common.c +*** encoder/mpglib/common.c Wed Dec 31 17:00:00 1969 +--- lame3.70/mpglib/common.c Thu Mar 23 15:12:26 2000 +*************** +*** 0 **** +--- 1,265 ---- ++ #ifdef HAVEMPGLIB ++ #include <ctype.h> ++ #include <stdlib.h> ++ #include <signal.h> ++ ++ #include <sys/types.h> ++ #include <sys/stat.h> ++ #include <fcntl.h> ++ ++ #include "mpg123.h" ++ ++ struct parameter param = { 1 , 1 , 0 , 0 }; ++ ++ int tabsel_123[2][3][16] = { ++ { {0,32,64,96,128,160,192,224,256,288,320,352,384,416,448,}, ++ {0,32,48,56, 64, 80, 96,112,128,160,192,224,256,320,384,}, ++ {0,32,40,48, 56, 64, 80, 96,112,128,160,192,224,256,320,} }, ++ ++ { {0,32,48,56,64,80,96,112,128,144,160,176,192,224,256,}, ++ {0,8,16,24,32,40,48,56,64,80,96,112,128,144,160,}, ++ {0,8,16,24,32,40,48,56,64,80,96,112,128,144,160,} } ++ }; ++ ++ long freqs[9] = { 44100, 48000, 32000, ++ 22050, 24000, 16000 , ++ 11025 , 12000 , 8000 }; ++ ++ int bitindex; ++ unsigned char *wordpointer; ++ unsigned char *pcm_sample; ++ int pcm_point = 0; ++ ++ ++ #if 0 ++ static void get_II_stuff(struct frame *fr) ++ { ++ static int translate[3][2][16] = ++ { { { 0,2,2,2,2,2,2,0,0,0,1,1,1,1,1,0 } , ++ { 0,2,2,0,0,0,1,1,1,1,1,1,1,1,1,0 } } , ++ { { 0,2,2,2,2,2,2,0,0,0,0,0,0,0,0,0 } , ++ { 0,2,2,0,0,0,0,0,0,0,0,0,0,0,0,0 } } , ++ { { 0,3,3,3,3,3,3,0,0,0,1,1,1,1,1,0 } , ++ { 0,3,3,0,0,0,1,1,1,1,1,1,1,1,1,0 } } }; ++ ++ int table,sblim; ++ static struct al_table *tables[5] = ++ { alloc_0, alloc_1, alloc_2, alloc_3 , alloc_4 }; ++ static int sblims[5] = { 27 , 30 , 8, 12 , 30 }; ++ ++ if(fr->lsf) ++ table = 4; ++ else ++ table = translate[fr->sampling_frequency][2-fr->stereo][fr->bitrate_index]; ++ sblim = sblims[table]; ++ ++ fr->alloc = tables[table]; ++ fr->II_sblimit = sblim; ++ } ++ #endif ++ ++ #define HDRCMPMASK 0xfffffd00 ++ ++ #if 0 ++ int head_check(unsigned long head) ++ { ++ if( (head & 0xffe00000) != 0xffe00000) ++ return FALSE; ++ if(!((head>>17)&3)) ++ return FALSE; ++ if( ((head>>12)&0xf) == 0xf) ++ return FALSE; ++ if( ((head>>10)&0x3) == 0x3 ) ++ return FALSE; ++ return TRUE; ++ } ++ #endif ++ ++ /* ++ * the code a header and write the information ++ * into the frame structure ++ */ ++ int decode_header(struct frame *fr,unsigned long newhead) ++ { ++ ++ ++ if( newhead & (1<<20) ) { ++ fr->lsf = (newhead & (1<<19)) ? 0x0 : 0x1; ++ fr->mpeg25 = 0; ++ } ++ else { ++ fr->lsf = 1; ++ fr->mpeg25 = 1; ++ } ++ ++ ++ fr->lay = 4-((newhead>>17)&3); ++ if( ((newhead>>10)&0x3) == 0x3) { ++ fprintf(stderr,"Stream error\n"); ++ exit(1); ++ } ++ if(fr->mpeg25) { ++ fr->sampling_frequency = 6 + ((newhead>>10)&0x3); ++ } ++ else ++ fr->sampling_frequency = ((newhead>>10)&0x3) + (fr->lsf*3); ++ fr->error_protection = ((newhead>>16)&0x1)^0x1; ++ ++ if(fr->mpeg25) /* allow Bitrate change for 2.5 ... */ ++ fr->bitrate_index = ((newhead>>12)&0xf); ++ ++ fr->bitrate_index = ((newhead>>12)&0xf); ++ fr->padding = ((newhead>>9)&0x1); ++ fr->extension = ((newhead>>8)&0x1); ++ fr->mode = ((newhead>>6)&0x3); ++ fr->mode_ext = ((newhead>>4)&0x3); ++ fr->copyright = ((newhead>>3)&0x1); ++ fr->original = ((newhead>>2)&0x1); ++ fr->emphasis = newhead & 0x3; ++ ++ fr->stereo = (fr->mode == MPG_MD_MONO) ? 1 : 2; ++ ++ if(!fr->bitrate_index) ++ { ++ fprintf(stderr,"Free format not supported.\n"); ++ return (0); ++ } ++ ++ switch(fr->lay) ++ { ++ case 1: ++ #if 0 ++ fr->do_layer = do_layer1; ++ fr->jsbound = (fr->mode == MPG_MD_JOINT_STEREO) ? ++ (fr->mode_ext<<2)+4 : 32; ++ fr->framesize = (long) tabsel_123[fr->lsf][0][fr->bitrate_index] * 12000; ++ fr->framesize /= freqs[fr->sampling_frequency]; ++ fr->framesize = ((fr->framesize+fr->padding)<<2)-4; ++ #else ++ fprintf(stderr,"layer=1 Not supported!\n"); ++ #endif ++ break; ++ case 2: ++ #if 0 ++ fr->do_layer = do_layer2; ++ get_II_stuff(fr); ++ fr->jsbound = (fr->mode == MPG_MD_JOINT_STEREO) ? ++ (fr->mode_ext<<2)+4 : fr->II_sblimit; ++ fr->framesize = (long) tabsel_123[fr->lsf][1][fr->bitrate_index] * 144000; ++ fr->framesize /= freqs[fr->sampling_frequency]; ++ fr->framesize += fr->padding - 4; ++ #else ++ fprintf(stderr,"layer=2 Not supported!\n"); ++ #endif ++ break; ++ case 3: ++ #if 0 ++ fr->do_layer = do_layer3; ++ if(fr->lsf) ++ ssize = (fr->stereo == 1) ? 9 : 17; ++ else ++ ssize = (fr->stereo == 1) ? 17 : 32; ++ #endif ++ ++ #if 0 ++ if(fr->error_protection) ++ ssize += 2; ++ #endif ++ fr->framesize = (long) tabsel_123[fr->lsf][2][fr->bitrate_index] * 144000; ++ fr->framesize /= freqs[fr->sampling_frequency]<<(fr->lsf); ++ fr->framesize = fr->framesize + fr->padding - 4; ++ break; ++ default: ++ fprintf(stderr,"Sorry, unknown layer type.\n"); ++ return (0); ++ } ++ ++ /* print_header(fr); */ ++ ++ return 1; ++ } ++ ++ ++ #if 1 ++ void print_header(struct frame *fr) ++ { ++ static char *modes[4] = { "Stereo", "Joint-Stereo", "Dual-Channel", "Single-Channel" }; ++ static char *layers[4] = { "Unknown" , "I", "II", "III" }; ++ ++ fprintf(stderr,"MPEG %s, Layer: %s, Freq: %ld, mode: %s, modext: %d, BPF : %d\n", ++ fr->mpeg25 ? "2.5" : (fr->lsf ? "2.0" : "1.0"), ++ layers[fr->lay],freqs[fr->sampling_frequency], ++ modes[fr->mode],fr->mode_ext,fr->framesize+4); ++ fprintf(stderr,"Channels: %d, copyright: %s, original: %s, CRC: %s, emphasis: %d.\n", ++ fr->stereo,fr->copyright?"Yes":"No", ++ fr->original?"Yes":"No",fr->error_protection?"Yes":"No", ++ fr->emphasis); ++ fprintf(stderr,"Bitrate: %d Kbits/s, Extension value: %d\n", ++ tabsel_123[fr->lsf][fr->lay-1][fr->bitrate_index],fr->extension); ++ } ++ ++ void print_header_compact(struct frame *fr) ++ { ++ static char *modes[4] = { "stereo", "joint-stereo", "dual-channel", "mono" }; ++ static char *layers[4] = { "Unknown" , "I", "II", "III" }; ++ ++ fprintf(stderr,"MPEG %s layer %s, %d kbit/s, %ld Hz %s\n", ++ fr->mpeg25 ? "2.5" : (fr->lsf ? "2.0" : "1.0"), ++ layers[fr->lay], ++ tabsel_123[fr->lsf][fr->lay-1][fr->bitrate_index], ++ freqs[fr->sampling_frequency], modes[fr->mode]); ++ } ++ ++ #endif ++ ++ unsigned int getbits(int number_of_bits) ++ { ++ unsigned long rval; ++ ++ if(!number_of_bits) ++ return 0; ++ ++ { ++ rval = wordpointer[0]; ++ rval <<= 8; ++ rval |= wordpointer[1]; ++ rval <<= 8; ++ rval |= wordpointer[2]; ++ rval <<= bitindex; ++ rval &= 0xffffff; ++ ++ bitindex += number_of_bits; ++ ++ rval >>= (24-number_of_bits); ++ ++ wordpointer += (bitindex>>3); ++ bitindex &= 7; ++ } ++ return rval; ++ } ++ ++ unsigned int getbits_fast(int number_of_bits) ++ { ++ unsigned long rval; ++ ++ { ++ rval = wordpointer[0]; ++ rval <<= 8; ++ rval |= wordpointer[1]; ++ rval <<= bitindex; ++ rval &= 0xffff; ++ bitindex += number_of_bits; ++ ++ rval >>= (16-number_of_bits); ++ ++ wordpointer += (bitindex>>3); ++ bitindex &= 7; ++ } ++ return rval; ++ } ++ ++ ++ ++ ++ ++ #endif +diff -r -c -N encoder/mpglib/dct64_i386.c lame3.70/mpglib/dct64_i386.c +*** encoder/mpglib/dct64_i386.c Wed Dec 31 17:00:00 1969 +--- lame3.70/mpglib/dct64_i386.c Mon Mar 6 12:53:03 2000 +*************** +*** 0 **** +--- 1,317 ---- ++ #ifdef HAVEMPGLIB ++ /* ++ * Discrete Cosine Tansform (DCT) for subband synthesis ++ * optimized for machines with no auto-increment. ++ * The performance is highly compiler dependend. Maybe ++ * the dct64.c version for 'normal' processor may be faster ++ * even for Intel processors. ++ */ ++ ++ #include "mpg123.h" ++ ++ static void dct64_1(real *out0,real *out1,real *b1,real *b2,real *samples) ++ { ++ ++ { ++ register real *costab = pnts[0]; ++ ++ b1[0x00] = samples[0x00] + samples[0x1F]; ++ b1[0x1F] = (samples[0x00] - samples[0x1F]) * costab[0x0]; ++ ++ b1[0x01] = samples[0x01] + samples[0x1E]; ++ b1[0x1E] = (samples[0x01] - samples[0x1E]) * costab[0x1]; ++ ++ b1[0x02] = samples[0x02] + samples[0x1D]; ++ b1[0x1D] = (samples[0x02] - samples[0x1D]) * costab[0x2]; ++ ++ b1[0x03] = samples[0x03] + samples[0x1C]; ++ b1[0x1C] = (samples[0x03] - samples[0x1C]) * costab[0x3]; ++ ++ b1[0x04] = samples[0x04] + samples[0x1B]; ++ b1[0x1B] = (samples[0x04] - samples[0x1B]) * costab[0x4]; ++ ++ b1[0x05] = samples[0x05] + samples[0x1A]; ++ b1[0x1A] = (samples[0x05] - samples[0x1A]) * costab[0x5]; ++ ++ b1[0x06] = samples[0x06] + samples[0x19]; ++ b1[0x19] = (samples[0x06] - samples[0x19]) * costab[0x6]; ++ ++ b1[0x07] = samples[0x07] + samples[0x18]; ++ b1[0x18] = (samples[0x07] - samples[0x18]) * costab[0x7]; ++ ++ b1[0x08] = samples[0x08] + samples[0x17]; ++ b1[0x17] = (samples[0x08] - samples[0x17]) * costab[0x8]; ++ ++ b1[0x09] = samples[0x09] + samples[0x16]; ++ b1[0x16] = (samples[0x09] - samples[0x16]) * costab[0x9]; ++ ++ b1[0x0A] = samples[0x0A] + samples[0x15]; ++ b1[0x15] = (samples[0x0A] - samples[0x15]) * costab[0xA]; ++ ++ b1[0x0B] = samples[0x0B] + samples[0x14]; ++ b1[0x14] = (samples[0x0B] - samples[0x14]) * costab[0xB]; ++ ++ b1[0x0C] = samples[0x0C] + samples[0x13]; ++ b1[0x13] = (samples[0x0C] - samples[0x13]) * costab[0xC]; ++ ++ b1[0x0D] = samples[0x0D] + samples[0x12]; ++ b1[0x12] = (samples[0x0D] - samples[0x12]) * costab[0xD]; ++ ++ b1[0x0E] = samples[0x0E] + samples[0x11]; ++ b1[0x11] = (samples[0x0E] - samples[0x11]) * costab[0xE]; ++ ++ b1[0x0F] = samples[0x0F] + samples[0x10]; ++ b1[0x10] = (samples[0x0F] - samples[0x10]) * costab[0xF]; ++ } ++ ++ ++ { ++ register real *costab = pnts[1]; ++ ++ b2[0x00] = b1[0x00] + b1[0x0F]; ++ b2[0x0F] = (b1[0x00] - b1[0x0F]) * costab[0]; ++ b2[0x01] = b1[0x01] + b1[0x0E]; ++ b2[0x0E] = (b1[0x01] - b1[0x0E]) * costab[1]; ++ b2[0x02] = b1[0x02] + b1[0x0D]; ++ b2[0x0D] = (b1[0x02] - b1[0x0D]) * costab[2]; ++ b2[0x03] = b1[0x03] + b1[0x0C]; ++ b2[0x0C] = (b1[0x03] - b1[0x0C]) * costab[3]; ++ b2[0x04] = b1[0x04] + b1[0x0B]; ++ b2[0x0B] = (b1[0x04] - b1[0x0B]) * costab[4]; ++ b2[0x05] = b1[0x05] + b1[0x0A]; ++ b2[0x0A] = (b1[0x05] - b1[0x0A]) * costab[5]; ++ b2[0x06] = b1[0x06] + b1[0x09]; ++ b2[0x09] = (b1[0x06] - b1[0x09]) * costab[6]; ++ b2[0x07] = b1[0x07] + b1[0x08]; ++ b2[0x08] = (b1[0x07] - b1[0x08]) * costab[7]; ++ ++ b2[0x10] = b1[0x10] + b1[0x1F]; ++ b2[0x1F] = (b1[0x1F] - b1[0x10]) * costab[0]; ++ b2[0x11] = b1[0x11] + b1[0x1E]; ++ b2[0x1E] = (b1[0x1E] - b1[0x11]) * costab[1]; ++ b2[0x12] = b1[0x12] + b1[0x1D]; ++ b2[0x1D] = (b1[0x1D] - b1[0x12]) * costab[2]; ++ b2[0x13] = b1[0x13] + b1[0x1C]; ++ b2[0x1C] = (b1[0x1C] - b1[0x13]) * costab[3]; ++ b2[0x14] = b1[0x14] + b1[0x1B]; ++ b2[0x1B] = (b1[0x1B] - b1[0x14]) * costab[4]; ++ b2[0x15] = b1[0x15] + b1[0x1A]; ++ b2[0x1A] = (b1[0x1A] - b1[0x15]) * costab[5]; ++ b2[0x16] = b1[0x16] + b1[0x19]; ++ b2[0x19] = (b1[0x19] - b1[0x16]) * costab[6]; ++ b2[0x17] = b1[0x17] + b1[0x18]; ++ b2[0x18] = (b1[0x18] - b1[0x17]) * costab[7]; ++ } ++ ++ { ++ register real *costab = pnts[2]; ++ ++ b1[0x00] = b2[0x00] + b2[0x07]; ++ b1[0x07] = (b2[0x00] - b2[0x07]) * costab[0]; ++ b1[0x01] = b2[0x01] + b2[0x06]; ++ b1[0x06] = (b2[0x01] - b2[0x06]) * costab[1]; ++ b1[0x02] = b2[0x02] + b2[0x05]; ++ b1[0x05] = (b2[0x02] - b2[0x05]) * costab[2]; ++ b1[0x03] = b2[0x03] + b2[0x04]; ++ b1[0x04] = (b2[0x03] - b2[0x04]) * costab[3]; ++ ++ b1[0x08] = b2[0x08] + b2[0x0F]; ++ b1[0x0F] = (b2[0x0F] - b2[0x08]) * costab[0]; ++ b1[0x09] = b2[0x09] + b2[0x0E]; ++ b1[0x0E] = (b2[0x0E] - b2[0x09]) * costab[1]; ++ b1[0x0A] = b2[0x0A] + b2[0x0D]; ++ b1[0x0D] = (b2[0x0D] - b2[0x0A]) * costab[2]; ++ b1[0x0B] = b2[0x0B] + b2[0x0C]; ++ b1[0x0C] = (b2[0x0C] - b2[0x0B]) * costab[3]; ++ ++ b1[0x10] = b2[0x10] + b2[0x17]; ++ b1[0x17] = (b2[0x10] - b2[0x17]) * costab[0]; ++ b1[0x11] = b2[0x11] + b2[0x16]; ++ b1[0x16] = (b2[0x11] - b2[0x16]) * costab[1]; ++ b1[0x12] = b2[0x12] + b2[0x15]; ++ b1[0x15] = (b2[0x12] - b2[0x15]) * costab[2]; ++ b1[0x13] = b2[0x13] + b2[0x14]; ++ b1[0x14] = (b2[0x13] - b2[0x14]) * costab[3]; ++ ++ b1[0x18] = b2[0x18] + b2[0x1F]; ++ b1[0x1F] = (b2[0x1F] - b2[0x18]) * costab[0]; ++ b1[0x19] = b2[0x19] + b2[0x1E]; ++ b1[0x1E] = (b2[0x1E] - b2[0x19]) * costab[1]; ++ b1[0x1A] = b2[0x1A] + b2[0x1D]; ++ b1[0x1D] = (b2[0x1D] - b2[0x1A]) * costab[2]; ++ b1[0x1B] = b2[0x1B] + b2[0x1C]; ++ b1[0x1C] = (b2[0x1C] - b2[0x1B]) * costab[3]; ++ } ++ ++ { ++ register real const cos0 = pnts[3][0]; ++ register real const cos1 = pnts[3][1]; ++ ++ b2[0x00] = b1[0x00] + b1[0x03]; ++ b2[0x03] = (b1[0x00] - b1[0x03]) * cos0; ++ b2[0x01] = b1[0x01] + b1[0x02]; ++ b2[0x02] = (b1[0x01] - b1[0x02]) * cos1; ++ ++ b2[0x04] = b1[0x04] + b1[0x07]; ++ b2[0x07] = (b1[0x07] - b1[0x04]) * cos0; ++ b2[0x05] = b1[0x05] + b1[0x06]; ++ b2[0x06] = (b1[0x06] - b1[0x05]) * cos1; ++ ++ b2[0x08] = b1[0x08] + b1[0x0B]; ++ b2[0x0B] = (b1[0x08] - b1[0x0B]) * cos0; ++ b2[0x09] = b1[0x09] + b1[0x0A]; ++ b2[0x0A] = (b1[0x09] - b1[0x0A]) * cos1; ++ ++ b2[0x0C] = b1[0x0C] + b1[0x0F]; ++ b2[0x0F] = (b1[0x0F] - b1[0x0C]) * cos0; ++ b2[0x0D] = b1[0x0D] + b1[0x0E]; ++ b2[0x0E] = (b1[0x0E] - b1[0x0D]) * cos1; ++ ++ b2[0x10] = b1[0x10] + b1[0x13]; ++ b2[0x13] = (b1[0x10] - b1[0x13]) * cos0; ++ b2[0x11] = b1[0x11] + b1[0x12]; ++ b2[0x12] = (b1[0x11] - b1[0x12]) * cos1; ++ ++ b2[0x14] = b1[0x14] + b1[0x17]; ++ b2[0x17] = (b1[0x17] - b1[0x14]) * cos0; ++ b2[0x15] = b1[0x15] + b1[0x16]; ++ b2[0x16] = (b1[0x16] - b1[0x15]) * cos1; ++ ++ b2[0x18] = b1[0x18] + b1[0x1B]; ++ b2[0x1B] = (b1[0x18] - b1[0x1B]) * cos0; ++ b2[0x19] = b1[0x19] + b1[0x1A]; ++ b2[0x1A] = (b1[0x19] - b1[0x1A]) * cos1; ++ ++ b2[0x1C] = b1[0x1C] + b1[0x1F]; ++ b2[0x1F] = (b1[0x1F] - b1[0x1C]) * cos0; ++ b2[0x1D] = b1[0x1D] + b1[0x1E]; ++ b2[0x1E] = (b1[0x1E] - b1[0x1D]) * cos1; ++ } ++ ++ { ++ register real const cos0 = pnts[4][0]; ++ ++ b1[0x00] = b2[0x00] + b2[0x01]; ++ b1[0x01] = (b2[0x00] - b2[0x01]) * cos0; ++ b1[0x02] = b2[0x02] + b2[0x03]; ++ b1[0x03] = (b2[0x03] - b2[0x02]) * cos0; ++ b1[0x02] += b1[0x03]; ++ ++ b1[0x04] = b2[0x04] + b2[0x05]; ++ b1[0x05] = (b2[0x04] - b2[0x05]) * cos0; ++ b1[0x06] = b2[0x06] + b2[0x07]; ++ b1[0x07] = (b2[0x07] - b2[0x06]) * cos0; ++ b1[0x06] += b1[0x07]; ++ b1[0x04] += b1[0x06]; ++ b1[0x06] += b1[0x05]; ++ b1[0x05] += b1[0x07]; ++ ++ b1[0x08] = b2[0x08] + b2[0x09]; ++ b1[0x09] = (b2[0x08] - b2[0x09]) * cos0; ++ b1[0x0A] = b2[0x0A] + b2[0x0B]; ++ b1[0x0B] = (b2[0x0B] - b2[0x0A]) * cos0; ++ b1[0x0A] += b1[0x0B]; ++ ++ b1[0x0C] = b2[0x0C] + b2[0x0D]; ++ b1[0x0D] = (b2[0x0C] - b2[0x0D]) * cos0; ++ b1[0x0E] = b2[0x0E] + b2[0x0F]; ++ b1[0x0F] = (b2[0x0F] - b2[0x0E]) * cos0; ++ b1[0x0E] += b1[0x0F]; ++ b1[0x0C] += b1[0x0E]; ++ b1[0x0E] += b1[0x0D]; ++ b1[0x0D] += b1[0x0F]; ++ ++ b1[0x10] = b2[0x10] + b2[0x11]; ++ b1[0x11] = (b2[0x10] - b2[0x11]) * cos0; ++ b1[0x12] = b2[0x12] + b2[0x13]; ++ b1[0x13] = (b2[0x13] - b2[0x12]) * cos0; ++ b1[0x12] += b1[0x13]; ++ ++ b1[0x14] = b2[0x14] + b2[0x15]; ++ b1[0x15] = (b2[0x14] - b2[0x15]) * cos0; ++ b1[0x16] = b2[0x16] + b2[0x17]; ++ b1[0x17] = (b2[0x17] - b2[0x16]) * cos0; ++ b1[0x16] += b1[0x17]; ++ b1[0x14] += b1[0x16]; ++ b1[0x16] += b1[0x15]; ++ b1[0x15] += b1[0x17]; ++ ++ b1[0x18] = b2[0x18] + b2[0x19]; ++ b1[0x19] = (b2[0x18] - b2[0x19]) * cos0; ++ b1[0x1A] = b2[0x1A] + b2[0x1B]; ++ b1[0x1B] = (b2[0x1B] - b2[0x1A]) * cos0; ++ b1[0x1A] += b1[0x1B]; ++ ++ b1[0x1C] = b2[0x1C] + b2[0x1D]; ++ b1[0x1D] = (b2[0x1C] - b2[0x1D]) * cos0; ++ b1[0x1E] = b2[0x1E] + b2[0x1F]; ++ b1[0x1F] = (b2[0x1F] - b2[0x1E]) * cos0; ++ b1[0x1E] += b1[0x1F]; ++ b1[0x1C] += b1[0x1E]; ++ b1[0x1E] += b1[0x1D]; ++ b1[0x1D] += b1[0x1F]; ++ } ++ ++ out0[0x10*16] = b1[0x00]; ++ out0[0x10*12] = b1[0x04]; ++ out0[0x10* 8] = b1[0x02]; ++ out0[0x10* 4] = b1[0x06]; ++ out0[0x10* 0] = b1[0x01]; ++ out1[0x10* 0] = b1[0x01]; ++ out1[0x10* 4] = b1[0x05]; ++ out1[0x10* 8] = b1[0x03]; ++ out1[0x10*12] = b1[0x07]; ++ ++ b1[0x08] += b1[0x0C]; ++ out0[0x10*14] = b1[0x08]; ++ b1[0x0C] += b1[0x0a]; ++ out0[0x10*10] = b1[0x0C]; ++ b1[0x0A] += b1[0x0E]; ++ out0[0x10* 6] = b1[0x0A]; ++ b1[0x0E] += b1[0x09]; ++ out0[0x10* 2] = b1[0x0E]; ++ b1[0x09] += b1[0x0D]; ++ out1[0x10* 2] = b1[0x09]; ++ b1[0x0D] += b1[0x0B]; ++ out1[0x10* 6] = b1[0x0D]; ++ b1[0x0B] += b1[0x0F]; ++ out1[0x10*10] = b1[0x0B]; ++ out1[0x10*14] = b1[0x0F]; ++ ++ b1[0x18] += b1[0x1C]; ++ out0[0x10*15] = b1[0x10] + b1[0x18]; ++ out0[0x10*13] = b1[0x18] + b1[0x14]; ++ b1[0x1C] += b1[0x1a]; ++ out0[0x10*11] = b1[0x14] + b1[0x1C]; ++ out0[0x10* 9] = b1[0x1C] + b1[0x12]; ++ b1[0x1A] += b1[0x1E]; ++ out0[0x10* 7] = b1[0x12] + b1[0x1A]; ++ out0[0x10* 5] = b1[0x1A] + b1[0x16]; ++ b1[0x1E] += b1[0x19]; ++ out0[0x10* 3] = b1[0x16] + b1[0x1E]; ++ out0[0x10* 1] = b1[0x1E] + b1[0x11]; ++ b1[0x19] += b1[0x1D]; ++ out1[0x10* 1] = b1[0x11] + b1[0x19]; ++ out1[0x10* 3] = b1[0x19] + b1[0x15]; ++ b1[0x1D] += b1[0x1B]; ++ out1[0x10* 5] = b1[0x15] + b1[0x1D]; ++ out1[0x10* 7] = b1[0x1D] + b1[0x13]; ++ b1[0x1B] += b1[0x1F]; ++ out1[0x10* 9] = b1[0x13] + b1[0x1B]; ++ out1[0x10*11] = b1[0x1B] + b1[0x17]; ++ out1[0x10*13] = b1[0x17] + b1[0x1F]; ++ out1[0x10*15] = b1[0x1F]; ++ } ++ ++ /* ++ * the call via dct64 is a trick to force GCC to use ++ * (new) registers for the b1,b2 pointer to the bufs[xx] field ++ */ ++ void dct64(real *a,real *b,real *c) ++ { ++ real bufs[0x40]; ++ dct64_1(a,b,bufs,bufs+0x20,c); ++ } ++ ++ ++ #endif +diff -r -c -N encoder/mpglib/decode_i386.c lame3.70/mpglib/decode_i386.c +*** encoder/mpglib/decode_i386.c Wed Dec 31 17:00:00 1969 +--- lame3.70/mpglib/decode_i386.c Mon Mar 6 12:53:03 2000 +*************** +*** 0 **** +--- 1,156 ---- ++ #ifdef HAVEMPGLIB ++ /* ++ * Mpeg Layer-1,2,3 audio decoder ++ * ------------------------------ ++ * copyright (c) 1995,1996,1997 by Michael Hipp, All rights reserved. ++ * See also 'README' ++ * ++ * slighlty optimized for machines without autoincrement/decrement. ++ * The performance is highly compiler dependend. Maybe ++ * the decode.c version for 'normal' processor may be faster ++ * even for Intel processors. ++ */ ++ ++ #include <stdlib.h> ++ #include <math.h> ++ #include <string.h> ++ ++ #include "mpg123.h" ++ #include "mpglib.h" ++ ++ extern struct mpstr *gmp; ++ ++ /* old WRITE_SAMPLE */ ++ #define WRITE_SAMPLE(samples,sum,clip) \ ++ if( (sum) > 32767.0) { *(samples) = 0x7fff; (clip)++; } \ ++ else if( (sum) < -32768.0) { *(samples) = -0x8000; (clip)++; } \ ++ else { *(samples) = sum; } ++ ++ int synth_1to1_mono(real *bandPtr,unsigned char *samples,int *pnt) ++ { ++ short samples_tmp[64]; ++ short *tmp1 = samples_tmp; ++ int i,ret; ++ int pnt1 = 0; ++ ++ ret = synth_1to1(bandPtr,0,(unsigned char *) samples_tmp,&pnt1); ++ samples += *pnt; ++ ++ for(i=0;i<32;i++) { ++ *( (short *) samples) = *tmp1; ++ samples += 2; ++ tmp1 += 2; ++ } ++ *pnt += 64; ++ ++ return ret; ++ } ++ ++ ++ int synth_1to1(real *bandPtr,int channel,unsigned char *out,int *pnt) ++ { ++ static const int step = 2; ++ int bo; ++ short *samples = (short *) (out + *pnt); ++ ++ real *b0,(*buf)[0x110]; ++ int clip = 0; ++ int bo1; ++ ++ bo = gmp->synth_bo; ++ ++ if(!channel) { ++ bo--; ++ bo &= 0xf; ++ buf = gmp->synth_buffs[0]; ++ } ++ else { ++ samples++; ++ buf = gmp->synth_buffs[1]; ++ } ++ ++ if(bo & 0x1) { ++ b0 = buf[0]; ++ bo1 = bo; ++ dct64(buf[1]+((bo+1)&0xf),buf[0]+bo,bandPtr); ++ } ++ else { ++ b0 = buf[1]; ++ bo1 = bo+1; ++ dct64(buf[0]+bo,buf[1]+bo+1,bandPtr); ++ } ++ ++ gmp->synth_bo = bo; ++ ++ { ++ register int j; ++ real *window = decwin + 16 - bo1; ++ ++ for (j=16;j;j--,b0+=0x10,window+=0x20,samples+=step) ++ { ++ real sum; ++ sum = window[0x0] * b0[0x0]; ++ sum -= window[0x1] * b0[0x1]; ++ sum += window[0x2] * b0[0x2]; ++ sum -= window[0x3] * b0[0x3]; ++ sum += window[0x4] * b0[0x4]; ++ sum -= window[0x5] * b0[0x5]; ++ sum += window[0x6] * b0[0x6]; ++ sum -= window[0x7] * b0[0x7]; ++ sum += window[0x8] * b0[0x8]; ++ sum -= window[0x9] * b0[0x9]; ++ sum += window[0xA] * b0[0xA]; ++ sum -= window[0xB] * b0[0xB]; ++ sum += window[0xC] * b0[0xC]; ++ sum -= window[0xD] * b0[0xD]; ++ sum += window[0xE] * b0[0xE]; ++ sum -= window[0xF] * b0[0xF]; ++ ++ WRITE_SAMPLE(samples,sum,clip); ++ } ++ ++ { ++ real sum; ++ sum = window[0x0] * b0[0x0]; ++ sum += window[0x2] * b0[0x2]; ++ sum += window[0x4] * b0[0x4]; ++ sum += window[0x6] * b0[0x6]; ++ sum += window[0x8] * b0[0x8]; ++ sum += window[0xA] * b0[0xA]; ++ sum += window[0xC] * b0[0xC]; ++ sum += window[0xE] * b0[0xE]; ++ WRITE_SAMPLE(samples,sum,clip); ++ b0-=0x10,window-=0x20,samples+=step; ++ } ++ window += bo1<<1; ++ ++ for (j=15;j;j--,b0-=0x10,window-=0x20,samples+=step) ++ { ++ real sum; ++ sum = -window[-0x1] * b0[0x0]; ++ sum -= window[-0x2] * b0[0x1]; ++ sum -= window[-0x3] * b0[0x2]; ++ sum -= window[-0x4] * b0[0x3]; ++ sum -= window[-0x5] * b0[0x4]; ++ sum -= window[-0x6] * b0[0x5]; ++ sum -= window[-0x7] * b0[0x6]; ++ sum -= window[-0x8] * b0[0x7]; ++ sum -= window[-0x9] * b0[0x8]; ++ sum -= window[-0xA] * b0[0x9]; ++ sum -= window[-0xB] * b0[0xA]; ++ sum -= window[-0xC] * b0[0xB]; ++ sum -= window[-0xD] * b0[0xC]; ++ sum -= window[-0xE] * b0[0xD]; ++ sum -= window[-0xF] * b0[0xE]; ++ sum -= window[-0x0] * b0[0xF]; ++ ++ WRITE_SAMPLE(samples,sum,clip); ++ } ++ } ++ *pnt += 128; ++ ++ return clip; ++ } ++ ++ ++ #endif +diff -r -c -N encoder/mpglib/huffman.h lame3.70/mpglib/huffman.h +*** encoder/mpglib/huffman.h Wed Dec 31 17:00:00 1969 +--- lame3.70/mpglib/huffman.h Wed Nov 24 01:44:45 1999 +*************** +*** 0 **** +--- 1,332 ---- ++ /* ++ * huffman tables ... recalcualted to work with my optimzed ++ * decoder scheme (MH) ++ * ++ * probably we could save a few bytes of memory, because the ++ * smaller tables are often the part of a bigger table ++ */ ++ ++ struct newhuff ++ { ++ unsigned int linbits; ++ short *table; ++ }; ++ ++ static short tab0[] = ++ { ++ 0 ++ }; ++ ++ static short tab1[] = ++ { ++ -5, -3, -1, 17, 1, 16, 0 ++ }; ++ ++ static short tab2[] = ++ { ++ -15, -11, -9, -5, -3, -1, 34, 2, 18, -1, 33, 32, 17, -1, 1, ++ 16, 0 ++ }; ++ ++ static short tab3[] = ++ { ++ -13, -11, -9, -5, -3, -1, 34, 2, 18, -1, 33, 32, 16, 17, -1, ++ 1, 0 ++ }; ++ ++ static short tab5[] = ++ { ++ -29, -25, -23, -15, -7, -5, -3, -1, 51, 35, 50, 49, -3, -1, 19, ++ 3, -1, 48, 34, -3, -1, 18, 33, -1, 2, 32, 17, -1, 1, 16, ++ 0 ++ }; ++ ++ static short tab6[] = ++ { ++ -25, -19, -13, -9, -5, -3, -1, 51, 3, 35, -1, 50, 48, -1, 19, ++ 49, -3, -1, 34, 2, 18, -3, -1, 33, 32, 1, -1, 17, -1, 16, ++ 0 ++ }; ++ ++ static short tab7[] = ++ { ++ -69, -65, -57, -39, -29, -17, -11, -7, -3, -1, 85, 69, -1, 84, 83, ++ -1, 53, 68, -3, -1, 37, 82, 21, -5, -1, 81, -1, 5, 52, -1, ++ 80, -1, 67, 51, -5, -3, -1, 36, 66, 20, -1, 65, 64, -11, -7, ++ -3, -1, 4, 35, -1, 50, 3, -1, 19, 49, -3, -1, 48, 34, 18, ++ -5, -1, 33, -1, 2, 32, 17, -1, 1, 16, 0 ++ }; ++ ++ static short tab8[] = ++ { ++ -65, -63, -59, -45, -31, -19, -13, -7, -5, -3, -1, 85, 84, 69, 83, ++ -3, -1, 53, 68, 37, -3, -1, 82, 5, 21, -5, -1, 81, -1, 52, ++ 67, -3, -1, 80, 51, 36, -5, -3, -1, 66, 20, 65, -3, -1, 4, ++ 64, -1, 35, 50, -9, -7, -3, -1, 19, 49, -1, 3, 48, 34, -1, ++ 2, 32, -1, 18, 33, 17, -3, -1, 1, 16, 0 ++ }; ++ ++ static short tab9[] = ++ { ++ -63, -53, -41, -29, -19, -11, -5, -3, -1, 85, 69, 53, -1, 83, -1, ++ 84, 5, -3, -1, 68, 37, -1, 82, 21, -3, -1, 81, 52, -1, 67, ++ -1, 80, 4, -7, -3, -1, 36, 66, -1, 51, 64, -1, 20, 65, -5, ++ -3, -1, 35, 50, 19, -1, 49, -1, 3, 48, -5, -3, -1, 34, 2, ++ 18, -1, 33, 32, -3, -1, 17, 1, -1, 16, 0 ++ }; ++ ++ static short tab10[] = ++ { ++ -125,-121,-111, -83, -55, -35, -21, -13, -7, -3, -1, 119, 103, -1, 118, ++ 87, -3, -1, 117, 102, 71, -3, -1, 116, 86, -1, 101, 55, -9, -3, ++ -1, 115, 70, -3, -1, 85, 84, 99, -1, 39, 114, -11, -5, -3, -1, ++ 100, 7, 112, -1, 98, -1, 69, 53, -5, -1, 6, -1, 83, 68, 23, ++ -17, -5, -1, 113, -1, 54, 38, -5, -3, -1, 37, 82, 21, -1, 81, ++ -1, 52, 67, -3, -1, 22, 97, -1, 96, -1, 5, 80, -19, -11, -7, ++ -3, -1, 36, 66, -1, 51, 4, -1, 20, 65, -3, -1, 64, 35, -1, ++ 50, 3, -3, -1, 19, 49, -1, 48, 34, -7, -3, -1, 18, 33, -1, ++ 2, 32, 17, -1, 1, 16, 0 ++ }; ++ ++ static short tab11[] = ++ { ++ -121,-113, -89, -59, -43, -27, -17, -7, -3, -1, 119, 103, -1, 118, 117, ++ -3, -1, 102, 71, -1, 116, -1, 87, 85, -5, -3, -1, 86, 101, 55, ++ -1, 115, 70, -9, -7, -3, -1, 69, 84, -1, 53, 83, 39, -1, 114, ++ -1, 100, 7, -5, -1, 113, -1, 23, 112, -3, -1, 54, 99, -1, 96, ++ -1, 68, 37, -13, -7, -5, -3, -1, 82, 5, 21, 98, -3, -1, 38, ++ 6, 22, -5, -1, 97, -1, 81, 52, -5, -1, 80, -1, 67, 51, -1, ++ 36, 66, -15, -11, -7, -3, -1, 20, 65, -1, 4, 64, -1, 35, 50, ++ -1, 19, 49, -5, -3, -1, 3, 48, 34, 33, -5, -1, 18, -1, 2, ++ 32, 17, -3, -1, 1, 16, 0 ++ }; ++ ++ static short tab12[] = ++ { ++ -115, -99, -73, -45, -27, -17, -9, -5, -3, -1, 119, 103, 118, -1, 87, ++ 117, -3, -1, 102, 71, -1, 116, 101, -3, -1, 86, 55, -3, -1, 115, ++ 85, 39, -7, -3, -1, 114, 70, -1, 100, 23, -5, -1, 113, -1, 7, ++ 112, -1, 54, 99, -13, -9, -3, -1, 69, 84, -1, 68, -1, 6, 5, ++ -1, 38, 98, -5, -1, 97, -1, 22, 96, -3, -1, 53, 83, -1, 37, ++ 82, -17, -7, -3, -1, 21, 81, -1, 52, 67, -5, -3, -1, 80, 4, ++ 36, -1, 66, 20, -3, -1, 51, 65, -1, 35, 50, -11, -7, -5, -3, ++ -1, 64, 3, 48, 19, -1, 49, 34, -1, 18, 33, -7, -5, -3, -1, ++ 2, 32, 0, 17, -1, 1, 16 ++ }; ++ ++ static short tab13[] = ++ { ++ -509,-503,-475,-405,-333,-265,-205,-153,-115, -83, -53, -35, -21, -13, -9, ++ -7, -5, -3, -1, 254, 252, 253, 237, 255, -1, 239, 223, -3, -1, 238, ++ 207, -1, 222, 191, -9, -3, -1, 251, 206, -1, 220, -1, 175, 233, -1, ++ 236, 221, -9, -5, -3, -1, 250, 205, 190, -1, 235, 159, -3, -1, 249, ++ 234, -1, 189, 219, -17, -9, -3, -1, 143, 248, -1, 204, -1, 174, 158, ++ -5, -1, 142, -1, 127, 126, 247, -5, -1, 218, -1, 173, 188, -3, -1, ++ 203, 246, 111, -15, -7, -3, -1, 232, 95, -1, 157, 217, -3, -1, 245, ++ 231, -1, 172, 187, -9, -3, -1, 79, 244, -3, -1, 202, 230, 243, -1, ++ 63, -1, 141, 216, -21, -9, -3, -1, 47, 242, -3, -1, 110, 156, 15, ++ -5, -3, -1, 201, 94, 171, -3, -1, 125, 215, 78, -11, -5, -3, -1, ++ 200, 214, 62, -1, 185, -1, 155, 170, -1, 31, 241, -23, -13, -5, -1, ++ 240, -1, 186, 229, -3, -1, 228, 140, -1, 109, 227, -5, -1, 226, -1, ++ 46, 14, -1, 30, 225, -15, -7, -3, -1, 224, 93, -1, 213, 124, -3, ++ -1, 199, 77, -1, 139, 184, -7, -3, -1, 212, 154, -1, 169, 108, -1, ++ 198, 61, -37, -21, -9, -5, -3, -1, 211, 123, 45, -1, 210, 29, -5, ++ -1, 183, -1, 92, 197, -3, -1, 153, 122, 195, -7, -5, -3, -1, 167, ++ 151, 75, 209, -3, -1, 13, 208, -1, 138, 168, -11, -7, -3, -1, 76, ++ 196, -1, 107, 182, -1, 60, 44, -3, -1, 194, 91, -3, -1, 181, 137, ++ 28, -43, -23, -11, -5, -1, 193, -1, 152, 12, -1, 192, -1, 180, 106, ++ -5, -3, -1, 166, 121, 59, -1, 179, -1, 136, 90, -11, -5, -1, 43, ++ -1, 165, 105, -1, 164, -1, 120, 135, -5, -1, 148, -1, 119, 118, 178, ++ -11, -3, -1, 27, 177, -3, -1, 11, 176, -1, 150, 74, -7, -3, -1, ++ 58, 163, -1, 89, 149, -1, 42, 162, -47, -23, -9, -3, -1, 26, 161, ++ -3, -1, 10, 104, 160, -5, -3, -1, 134, 73, 147, -3, -1, 57, 88, ++ -1, 133, 103, -9, -3, -1, 41, 146, -3, -1, 87, 117, 56, -5, -1, ++ 131, -1, 102, 71, -3, -1, 116, 86, -1, 101, 115, -11, -3, -1, 25, ++ 145, -3, -1, 9, 144, -1, 72, 132, -7, -5, -1, 114, -1, 70, 100, ++ 40, -1, 130, 24, -41, -27, -11, -5, -3, -1, 55, 39, 23, -1, 113, ++ -1, 85, 7, -7, -3, -1, 112, 54, -1, 99, 69, -3, -1, 84, 38, ++ -1, 98, 53, -5, -1, 129, -1, 8, 128, -3, -1, 22, 97, -1, 6, ++ 96, -13, -9, -5, -3, -1, 83, 68, 37, -1, 82, 5, -1, 21, 81, ++ -7, -3, -1, 52, 67, -1, 80, 36, -3, -1, 66, 51, 20, -19, -11, ++ -5, -1, 65, -1, 4, 64, -3, -1, 35, 50, 19, -3, -1, 49, 3, ++ -1, 48, 34, -3, -1, 18, 33, -1, 2, 32, -3, -1, 17, 1, 16, ++ 0 ++ }; ++ ++ static short tab15[] = ++ { ++ -495,-445,-355,-263,-183,-115, -77, -43, -27, -13, -7, -3, -1, 255, 239, ++ -1, 254, 223, -1, 238, -1, 253, 207, -7, -3, -1, 252, 222, -1, 237, ++ 191, -1, 251, -1, 206, 236, -7, -3, -1, 221, 175, -1, 250, 190, -3, ++ -1, 235, 205, -1, 220, 159, -15, -7, -3, -1, 249, 234, -1, 189, 219, ++ -3, -1, 143, 248, -1, 204, 158, -7, -3, -1, 233, 127, -1, 247, 173, ++ -3, -1, 218, 188, -1, 111, -1, 174, 15, -19, -11, -3, -1, 203, 246, ++ -3, -1, 142, 232, -1, 95, 157, -3, -1, 245, 126, -1, 231, 172, -9, ++ -3, -1, 202, 187, -3, -1, 217, 141, 79, -3, -1, 244, 63, -1, 243, ++ 216, -33, -17, -9, -3, -1, 230, 47, -1, 242, -1, 110, 240, -3, -1, ++ 31, 241, -1, 156, 201, -7, -3, -1, 94, 171, -1, 186, 229, -3, -1, ++ 125, 215, -1, 78, 228, -15, -7, -3, -1, 140, 200, -1, 62, 109, -3, ++ -1, 214, 227, -1, 155, 185, -7, -3, -1, 46, 170, -1, 226, 30, -5, ++ -1, 225, -1, 14, 224, -1, 93, 213, -45, -25, -13, -7, -3, -1, 124, ++ 199, -1, 77, 139, -1, 212, -1, 184, 154, -7, -3, -1, 169, 108, -1, ++ 198, 61, -1, 211, 210, -9, -5, -3, -1, 45, 13, 29, -1, 123, 183, ++ -5, -1, 209, -1, 92, 208, -1, 197, 138, -17, -7, -3, -1, 168, 76, ++ -1, 196, 107, -5, -1, 182, -1, 153, 12, -1, 60, 195, -9, -3, -1, ++ 122, 167, -1, 166, -1, 192, 11, -1, 194, -1, 44, 91, -55, -29, -15, ++ -7, -3, -1, 181, 28, -1, 137, 152, -3, -1, 193, 75, -1, 180, 106, ++ -5, -3, -1, 59, 121, 179, -3, -1, 151, 136, -1, 43, 90, -11, -5, ++ -1, 178, -1, 165, 27, -1, 177, -1, 176, 105, -7, -3, -1, 150, 74, ++ -1, 164, 120, -3, -1, 135, 58, 163, -17, -7, -3, -1, 89, 149, -1, ++ 42, 162, -3, -1, 26, 161, -3, -1, 10, 160, 104, -7, -3, -1, 134, ++ 73, -1, 148, 57, -5, -1, 147, -1, 119, 9, -1, 88, 133, -53, -29, ++ -13, -7, -3, -1, 41, 103, -1, 118, 146, -1, 145, -1, 25, 144, -7, ++ -3, -1, 72, 132, -1, 87, 117, -3, -1, 56, 131, -1, 102, 71, -7, ++ -3, -1, 40, 130, -1, 24, 129, -7, -3, -1, 116, 8, -1, 128, 86, ++ -3, -1, 101, 55, -1, 115, 70, -17, -7, -3, -1, 39, 114, -1, 100, ++ 23, -3, -1, 85, 113, -3, -1, 7, 112, 54, -7, -3, -1, 99, 69, ++ -1, 84, 38, -3, -1, 98, 22, -3, -1, 6, 96, 53, -33, -19, -9, ++ -5, -1, 97, -1, 83, 68, -1, 37, 82, -3, -1, 21, 81, -3, -1, ++ 5, 80, 52, -7, -3, -1, 67, 36, -1, 66, 51, -1, 65, -1, 20, ++ 4, -9, -3, -1, 35, 50, -3, -1, 64, 3, 19, -3, -1, 49, 48, ++ 34, -9, -7, -3, -1, 18, 33, -1, 2, 32, 17, -3, -1, 1, 16, ++ 0 ++ }; ++ ++ static short tab16[] = ++ { ++ -509,-503,-461,-323,-103, -37, -27, -15, -7, -3, -1, 239, 254, -1, 223, ++ 253, -3, -1, 207, 252, -1, 191, 251, -5, -1, 175, -1, 250, 159, -3, ++ -1, 249, 248, 143, -7, -3, -1, 127, 247, -1, 111, 246, 255, -9, -5, ++ -3, -1, 95, 245, 79, -1, 244, 243, -53, -1, 240, -1, 63, -29, -19, ++ -13, -7, -5, -1, 206, -1, 236, 221, 222, -1, 233, -1, 234, 217, -1, ++ 238, -1, 237, 235, -3, -1, 190, 205, -3, -1, 220, 219, 174, -11, -5, ++ -1, 204, -1, 173, 218, -3, -1, 126, 172, 202, -5, -3, -1, 201, 125, ++ 94, 189, 242, -93, -5, -3, -1, 47, 15, 31, -1, 241, -49, -25, -13, ++ -5, -1, 158, -1, 188, 203, -3, -1, 142, 232, -1, 157, 231, -7, -3, ++ -1, 187, 141, -1, 216, 110, -1, 230, 156, -13, -7, -3, -1, 171, 186, ++ -1, 229, 215, -1, 78, -1, 228, 140, -3, -1, 200, 62, -1, 109, -1, ++ 214, 155, -19, -11, -5, -3, -1, 185, 170, 225, -1, 212, -1, 184, 169, ++ -5, -1, 123, -1, 183, 208, 227, -7, -3, -1, 14, 224, -1, 93, 213, ++ -3, -1, 124, 199, -1, 77, 139, -75, -45, -27, -13, -7, -3, -1, 154, ++ 108, -1, 198, 61, -3, -1, 92, 197, 13, -7, -3, -1, 138, 168, -1, ++ 153, 76, -3, -1, 182, 122, 60, -11, -5, -3, -1, 91, 137, 28, -1, ++ 192, -1, 152, 121, -1, 226, -1, 46, 30, -15, -7, -3, -1, 211, 45, ++ -1, 210, 209, -5, -1, 59, -1, 151, 136, 29, -7, -3, -1, 196, 107, ++ -1, 195, 167, -1, 44, -1, 194, 181, -23, -13, -7, -3, -1, 193, 12, ++ -1, 75, 180, -3, -1, 106, 166, 179, -5, -3, -1, 90, 165, 43, -1, ++ 178, 27, -13, -5, -1, 177, -1, 11, 176, -3, -1, 105, 150, -1, 74, ++ 164, -5, -3, -1, 120, 135, 163, -3, -1, 58, 89, 42, -97, -57, -33, ++ -19, -11, -5, -3, -1, 149, 104, 161, -3, -1, 134, 119, 148, -5, -3, ++ -1, 73, 87, 103, 162, -5, -1, 26, -1, 10, 160, -3, -1, 57, 147, ++ -1, 88, 133, -9, -3, -1, 41, 146, -3, -1, 118, 9, 25, -5, -1, ++ 145, -1, 144, 72, -3, -1, 132, 117, -1, 56, 131, -21, -11, -5, -3, ++ -1, 102, 40, 130, -3, -1, 71, 116, 24, -3, -1, 129, 128, -3, -1, ++ 8, 86, 55, -9, -5, -1, 115, -1, 101, 70, -1, 39, 114, -5, -3, ++ -1, 100, 85, 7, 23, -23, -13, -5, -1, 113, -1, 112, 54, -3, -1, ++ 99, 69, -1, 84, 38, -3, -1, 98, 22, -1, 97, -1, 6, 96, -9, ++ -5, -1, 83, -1, 53, 68, -1, 37, 82, -1, 81, -1, 21, 5, -33, ++ -23, -13, -7, -3, -1, 52, 67, -1, 80, 36, -3, -1, 66, 51, 20, ++ -5, -1, 65, -1, 4, 64, -1, 35, 50, -3, -1, 19, 49, -3, -1, ++ 3, 48, 34, -3, -1, 18, 33, -1, 2, 32, -3, -1, 17, 1, 16, ++ 0 ++ }; ++ ++ static short tab24[] = ++ { ++ -451,-117, -43, -25, -15, -7, -3, -1, 239, 254, -1, 223, 253, -3, -1, ++ 207, 252, -1, 191, 251, -5, -1, 250, -1, 175, 159, -1, 249, 248, -9, ++ -5, -3, -1, 143, 127, 247, -1, 111, 246, -3, -1, 95, 245, -1, 79, ++ 244, -71, -7, -3, -1, 63, 243, -1, 47, 242, -5, -1, 241, -1, 31, ++ 240, -25, -9, -1, 15, -3, -1, 238, 222, -1, 237, 206, -7, -3, -1, ++ 236, 221, -1, 190, 235, -3, -1, 205, 220, -1, 174, 234, -15, -7, -3, ++ -1, 189, 219, -1, 204, 158, -3, -1, 233, 173, -1, 218, 188, -7, -3, ++ -1, 203, 142, -1, 232, 157, -3, -1, 217, 126, -1, 231, 172, 255,-235, ++ -143, -77, -45, -25, -15, -7, -3, -1, 202, 187, -1, 141, 216, -5, -3, ++ -1, 14, 224, 13, 230, -5, -3, -1, 110, 156, 201, -1, 94, 186, -9, ++ -5, -1, 229, -1, 171, 125, -1, 215, 228, -3, -1, 140, 200, -3, -1, ++ 78, 46, 62, -15, -7, -3, -1, 109, 214, -1, 227, 155, -3, -1, 185, ++ 170, -1, 226, 30, -7, -3, -1, 225, 93, -1, 213, 124, -3, -1, 199, ++ 77, -1, 139, 184, -31, -15, -7, -3, -1, 212, 154, -1, 169, 108, -3, ++ -1, 198, 61, -1, 211, 45, -7, -3, -1, 210, 29, -1, 123, 183, -3, ++ -1, 209, 92, -1, 197, 138, -17, -7, -3, -1, 168, 153, -1, 76, 196, ++ -3, -1, 107, 182, -3, -1, 208, 12, 60, -7, -3, -1, 195, 122, -1, ++ 167, 44, -3, -1, 194, 91, -1, 181, 28, -57, -35, -19, -7, -3, -1, ++ 137, 152, -1, 193, 75, -5, -3, -1, 192, 11, 59, -3, -1, 176, 10, ++ 26, -5, -1, 180, -1, 106, 166, -3, -1, 121, 151, -3, -1, 160, 9, ++ 144, -9, -3, -1, 179, 136, -3, -1, 43, 90, 178, -7, -3, -1, 165, ++ 27, -1, 177, 105, -1, 150, 164, -17, -9, -5, -3, -1, 74, 120, 135, ++ -1, 58, 163, -3, -1, 89, 149, -1, 42, 162, -7, -3, -1, 161, 104, ++ -1, 134, 119, -3, -1, 73, 148, -1, 57, 147, -63, -31, -15, -7, -3, ++ -1, 88, 133, -1, 41, 103, -3, -1, 118, 146, -1, 25, 145, -7, -3, ++ -1, 72, 132, -1, 87, 117, -3, -1, 56, 131, -1, 102, 40, -17, -7, ++ -3, -1, 130, 24, -1, 71, 116, -5, -1, 129, -1, 8, 128, -1, 86, ++ 101, -7, -5, -1, 23, -1, 7, 112, 115, -3, -1, 55, 39, 114, -15, ++ -7, -3, -1, 70, 100, -1, 85, 113, -3, -1, 54, 99, -1, 69, 84, ++ -7, -3, -1, 38, 98, -1, 22, 97, -5, -3, -1, 6, 96, 53, -1, ++ 83, 68, -51, -37, -23, -15, -9, -3, -1, 37, 82, -1, 21, -1, 5, ++ 80, -1, 81, -1, 52, 67, -3, -1, 36, 66, -1, 51, 20, -9, -5, ++ -1, 65, -1, 4, 64, -1, 35, 50, -1, 19, 49, -7, -5, -3, -1, ++ 3, 48, 34, 18, -1, 33, -1, 2, 32, -3, -1, 17, 1, -1, 16, ++ 0 ++ }; ++ ++ static short tab_c0[] = ++ { ++ -29, -21, -13, -7, -3, -1, 11, 15, -1, 13, 14, -3, -1, 7, 5, ++ 9, -3, -1, 6, 3, -1, 10, 12, -3, -1, 2, 1, -1, 4, 8, ++ 0 ++ }; ++ ++ static short tab_c1[] = ++ { ++ -15, -7, -3, -1, 15, 14, -1, 13, 12, -3, -1, 11, 10, -1, 9, ++ 8, -7, -3, -1, 7, 6, -1, 5, 4, -3, -1, 3, 2, -1, 1, ++ 0 ++ }; ++ ++ ++ ++ static struct newhuff ht[] = ++ { ++ { /* 0 */ 0 , tab0 } , ++ { /* 2 */ 0 , tab1 } , ++ { /* 3 */ 0 , tab2 } , ++ { /* 3 */ 0 , tab3 } , ++ { /* 0 */ 0 , tab0 } , ++ { /* 4 */ 0 , tab5 } , ++ { /* 4 */ 0 , tab6 } , ++ { /* 6 */ 0 , tab7 } , ++ { /* 6 */ 0 , tab8 } , ++ { /* 6 */ 0 , tab9 } , ++ { /* 8 */ 0 , tab10 } , ++ { /* 8 */ 0 , tab11 } , ++ { /* 8 */ 0 , tab12 } , ++ { /* 16 */ 0 , tab13 } , ++ { /* 0 */ 0 , tab0 } , ++ { /* 16 */ 0 , tab15 } , ++ ++ { /* 16 */ 1 , tab16 } , ++ { /* 16 */ 2 , tab16 } , ++ { /* 16 */ 3 , tab16 } , ++ { /* 16 */ 4 , tab16 } , ++ { /* 16 */ 6 , tab16 } , ++ { /* 16 */ 8 , tab16 } , ++ { /* 16 */ 10, tab16 } , ++ { /* 16 */ 13, tab16 } , ++ { /* 16 */ 4 , tab24 } , ++ { /* 16 */ 5 , tab24 } , ++ { /* 16 */ 6 , tab24 } , ++ { /* 16 */ 7 , tab24 } , ++ { /* 16 */ 8 , tab24 } , ++ { /* 16 */ 9 , tab24 } , ++ { /* 16 */ 11, tab24 } , ++ { /* 16 */ 13, tab24 } ++ }; ++ ++ static struct newhuff htc[] = ++ { ++ { /* 1 , 1 , */ 0 , tab_c0 } , ++ { /* 1 , 1 , */ 0 , tab_c1 } ++ }; ++ ++ +diff -r -c -N encoder/mpglib/interface.c lame3.70/mpglib/interface.c +*** encoder/mpglib/interface.c Wed Dec 31 17:00:00 1969 +--- lame3.70/mpglib/interface.c Mon Mar 6 12:53:03 2000 +*************** +*** 0 **** +--- 1,220 ---- ++ #ifdef HAVEMPGLIB ++ #include <stdlib.h> ++ #include <stdio.h> ++ ++ #include "mpg123.h" ++ #include "mpglib.h" ++ ++ ++ /* Global mp .. it's a hack */ ++ struct mpstr *gmp; ++ ++ ++ BOOL InitMP3(struct mpstr *mp) ++ { ++ memset(mp,0,sizeof(struct mpstr)); ++ ++ mp->framesize = 0; ++ mp->fsizeold = -1; ++ mp->bsize = 0; ++ mp->head = mp->tail = NULL; ++ mp->fr.single = -1; ++ mp->bsnum = 0; ++ mp->synth_bo = 1; ++ ++ make_decode_tables(32767); ++ init_layer3(SBLIMIT); ++ ++ return !0; ++ } ++ ++ void ExitMP3(struct mpstr *mp) ++ { ++ struct buf *b,*bn; ++ ++ b = mp->tail; ++ while(b) { ++ free(b->pnt); ++ bn = b->next; ++ free(b); ++ b = bn; ++ } ++ } ++ ++ static struct buf *addbuf(struct mpstr *mp,char *buf,int size) ++ { ++ struct buf *nbuf; ++ ++ nbuf = (struct buf*) malloc( sizeof(struct buf) ); ++ if(!nbuf) { ++ fprintf(stderr,"Out of memory!\n"); ++ return NULL; ++ } ++ nbuf->pnt = (unsigned char*) malloc(size); ++ if(!nbuf->pnt) { ++ free(nbuf); ++ return NULL; ++ } ++ nbuf->size = size; ++ memcpy(nbuf->pnt,buf,size); ++ nbuf->next = NULL; ++ nbuf->prev = mp->head; ++ nbuf->pos = 0; ++ ++ if(!mp->tail) { ++ mp->tail = nbuf; ++ } ++ else { ++ mp->head->next = nbuf; ++ } ++ ++ mp->head = nbuf; ++ mp->bsize += size; ++ ++ return nbuf; ++ } ++ ++ static void remove_buf(struct mpstr *mp) ++ { ++ struct buf *buf = mp->tail; ++ ++ mp->tail = buf->next; ++ if(mp->tail) ++ mp->tail->prev = NULL; ++ else { ++ mp->tail = mp->head = NULL; ++ } ++ ++ free(buf->pnt); ++ free(buf); ++ ++ } ++ ++ static int read_buf_byte(struct mpstr *mp) ++ { ++ unsigned int b; ++ ++ int pos; ++ ++ pos = mp->tail->pos; ++ while(pos >= mp->tail->size) { ++ remove_buf(mp); ++ pos = mp->tail->pos; ++ if(!mp->tail) { ++ fprintf(stderr,"Fatal error!\n"); ++ exit(1); ++ } ++ } ++ ++ b = mp->tail->pnt[pos]; ++ mp->bsize--; ++ mp->tail->pos++; ++ ++ ++ return b; ++ } ++ ++ static void read_head(struct mpstr *mp) ++ { ++ unsigned long head; ++ ++ head = read_buf_byte(mp); ++ head <<= 8; ++ head |= read_buf_byte(mp); ++ head <<= 8; ++ head |= read_buf_byte(mp); ++ head <<= 8; ++ head |= read_buf_byte(mp); ++ ++ mp->header = head; ++ } ++ ++ int decodeMP3(struct mpstr *mp,char *in,int isize,char *out, ++ int osize,int *done) ++ { ++ int len; ++ ++ gmp = mp; ++ ++ if(osize < 4608) { ++ fprintf(stderr,"To less out space\n"); ++ return MP3_ERR; ++ } ++ ++ if(in) { ++ if(addbuf(mp,in,isize) == NULL) { ++ return MP3_ERR; ++ } ++ } ++ ++ ++ /* First decode header */ ++ if(mp->framesize == 0) { ++ if(mp->bsize < 4) { ++ return MP3_NEED_MORE; ++ } ++ read_head(mp); ++ decode_header(&mp->fr,mp->header); ++ mp->framesize = mp->fr.framesize; ++ } ++ ++ /* printf(" fr.framesize = %i \n",mp->fr.framesize); ++ printf(" bsize = %i \n",mp->bsize); ++ */ ++ ++ if(mp->fr.framesize > mp->bsize) { ++ return MP3_NEED_MORE; ++ } ++ wordpointer = mp->bsspace[mp->bsnum] + 512; ++ mp->bsnum = (mp->bsnum + 1) & 0x1; ++ bitindex = 0; ++ ++ len = 0; ++ while(len < mp->framesize) { ++ int nlen; ++ int blen = mp->tail->size - mp->tail->pos; ++ if( (mp->framesize - len) <= blen) { ++ nlen = mp->framesize-len; ++ } ++ else { ++ nlen = blen; ++ } ++ memcpy(wordpointer+len,mp->tail->pnt+mp->tail->pos,nlen); ++ len += nlen; ++ mp->tail->pos += nlen; ++ mp->bsize -= nlen; ++ if(mp->tail->pos == mp->tail->size) { ++ remove_buf(mp); ++ } ++ } ++ ++ *done = 0; ++ if(mp->fr.error_protection) ++ getbits(16); ++ do_layer3(&mp->fr,(unsigned char *) out,done); ++ ++ mp->fsizeold = mp->framesize; ++ mp->framesize = 0; ++ return MP3_OK; ++ } ++ ++ int set_pointer(long backstep) ++ { ++ unsigned char *bsbufold; ++ if(gmp->fsizeold < 0 && backstep > 0) { ++ fprintf(stderr,"Can't step back %ld!\n",backstep); ++ return MP3_ERR; ++ } ++ bsbufold = gmp->bsspace[gmp->bsnum] + 512; ++ wordpointer -= backstep; ++ if (backstep) ++ memcpy(wordpointer,bsbufold+gmp->fsizeold-backstep,backstep); ++ bitindex = 0; ++ return MP3_OK; ++ } ++ ++ ++ ++ ++ ++ #endif +diff -r -c -N encoder/mpglib/layer3.c lame3.70/mpglib/layer3.c +*** encoder/mpglib/layer3.c Wed Dec 31 17:00:00 1969 +--- lame3.70/mpglib/layer3.c Thu Apr 6 12:50:53 2000 +*************** +*** 0 **** +--- 1,1738 ---- ++ #ifdef HAVEMPGLIB ++ /* ++ * Mpeg Layer-3 audio decoder ++ * -------------------------- ++ * copyright (c) 1995,1996,1997 by Michael Hipp. ++ * All rights reserved. See also 'README' ++ */ ++ ++ #include <stdlib.h> ++ #include "mpg123.h" ++ #include "mpglib.h" ++ #include "huffman.h" ++ #ifdef HAVEGTK ++ #include "../gtkanal.h" ++ #endif ++ ++ extern struct mpstr *gmp; ++ ++ #define MPEG1 ++ ++ ++ static real ispow[8207]; ++ static real aa_ca[8],aa_cs[8]; ++ static real COS1[12][6]; ++ static real win[4][36]; ++ static real win1[4][36]; ++ static real gainpow2[256+118+4]; ++ static real COS9[9]; ++ static real COS6_1,COS6_2; ++ static real tfcos36[9]; ++ static real tfcos12[3]; ++ ++ struct bandInfoStruct { ++ short longIdx[23]; ++ short longDiff[22]; ++ short shortIdx[14]; ++ short shortDiff[13]; ++ }; ++ ++ int longLimit[9][23]; ++ int shortLimit[9][14]; ++ ++ struct bandInfoStruct bandInfo[9] = { ++ ++ /* MPEG 1.0 */ ++ { {0,4,8,12,16,20,24,30,36,44,52,62,74, 90,110,134,162,196,238,288,342,418,576}, ++ {4,4,4,4,4,4,6,6,8, 8,10,12,16,20,24,28,34,42,50,54, 76,158}, ++ {0,4*3,8*3,12*3,16*3,22*3,30*3,40*3,52*3,66*3, 84*3,106*3,136*3,192*3}, ++ {4,4,4,4,6,8,10,12,14,18,22,30,56} } , ++ ++ { {0,4,8,12,16,20,24,30,36,42,50,60,72, 88,106,128,156,190,230,276,330,384,576}, ++ {4,4,4,4,4,4,6,6,6, 8,10,12,16,18,22,28,34,40,46,54, 54,192}, ++ {0,4*3,8*3,12*3,16*3,22*3,28*3,38*3,50*3,64*3, 80*3,100*3,126*3,192*3}, ++ {4,4,4,4,6,6,10,12,14,16,20,26,66} } , ++ ++ { {0,4,8,12,16,20,24,30,36,44,54,66,82,102,126,156,194,240,296,364,448,550,576} , ++ {4,4,4,4,4,4,6,6,8,10,12,16,20,24,30,38,46,56,68,84,102, 26} , ++ {0,4*3,8*3,12*3,16*3,22*3,30*3,42*3,58*3,78*3,104*3,138*3,180*3,192*3} , ++ {4,4,4,4,6,8,12,16,20,26,34,42,12} } , ++ ++ /* MPEG 2.0 */ ++ { {0,6,12,18,24,30,36,44,54,66,80,96,116,140,168,200,238,284,336,396,464,522,576}, ++ {6,6,6,6,6,6,8,10,12,14,16,20,24,28,32,38,46,52,60,68,58,54 } , ++ {0,4*3,8*3,12*3,18*3,24*3,32*3,42*3,56*3,74*3,100*3,132*3,174*3,192*3} , ++ {4,4,4,6,6,8,10,14,18,26,32,42,18 } } , ++ /* docs: 332. mpg123: 330 */ ++ { {0,6,12,18,24,30,36,44,54,66,80,96,114,136,162,194,232,278,332,394,464,540,576}, ++ {6,6,6,6,6,6,8,10,12,14,16,18,22,26,32,38,46,52,64,70,76,36 } , ++ {0,4*3,8*3,12*3,18*3,26*3,36*3,48*3,62*3,80*3,104*3,136*3,180*3,192*3} , ++ {4,4,4,6,8,10,12,14,18,24,32,44,12 } } , ++ ++ { {0,6,12,18,24,30,36,44,54,66,80,96,116,140,168,200,238,284,336,396,464,522,576}, ++ {6,6,6,6,6,6,8,10,12,14,16,20,24,28,32,38,46,52,60,68,58,54 }, ++ {0,4*3,8*3,12*3,18*3,26*3,36*3,48*3,62*3,80*3,104*3,134*3,174*3,192*3}, ++ {4,4,4,6,8,10,12,14,18,24,30,40,18 } } , ++ /* MPEG 2.5 */ ++ { {0,6,12,18,24,30,36,44,54,66,80,96,116,140,168,200,238,284,336,396,464,522,576} , ++ {6,6,6,6,6,6,8,10,12,14,16,20,24,28,32,38,46,52,60,68,58,54}, ++ {0,12,24,36,54,78,108,144,186,240,312,402,522,576}, ++ {4,4,4,6,8,10,12,14,18,24,30,40,18} }, ++ { {0,6,12,18,24,30,36,44,54,66,80,96,116,140,168,200,238,284,336,396,464,522,576} , ++ {6,6,6,6,6,6,8,10,12,14,16,20,24,28,32,38,46,52,60,68,58,54}, ++ {0,12,24,36,54,78,108,144,186,240,312,402,522,576}, ++ {4,4,4,6,8,10,12,14,18,24,30,40,18} }, ++ { {0,12,24,36,48,60,72,88,108,132,160,192,232,280,336,400,476,566,568,570,572,574,576}, ++ {12,12,12,12,12,12,16,20,24,28,32,40,48,56,64,76,90,2,2,2,2,2}, ++ {0, 24, 48, 72,108,156,216,288,372,480,486,492,498,576}, ++ {8,8,8,12,16,20,24,28,36,2,2,2,26} } , ++ }; ++ ++ static int mapbuf0[9][152]; ++ static int mapbuf1[9][156]; ++ static int mapbuf2[9][44]; ++ static int *map[9][3]; ++ static int *mapend[9][3]; ++ ++ static unsigned int n_slen2[512]; /* MPEG 2.0 slen for 'normal' mode */ ++ static unsigned int i_slen2[256]; /* MPEG 2.0 slen for intensity stereo */ ++ ++ static real tan1_1[16],tan2_1[16],tan1_2[16],tan2_2[16]; ++ static real pow1_1[2][16],pow2_1[2][16],pow1_2[2][16],pow2_2[2][16]; ++ ++ static unsigned int get1bit(void) ++ { ++ unsigned char rval; ++ rval = *wordpointer << bitindex; ++ ++ bitindex++; ++ wordpointer += (bitindex>>3); ++ bitindex &= 7; ++ ++ return rval>>7; ++ } ++ ++ ++ ++ ++ /* ++ * init tables for layer-3 ++ */ ++ void init_layer3(int down_sample_sblimit) ++ { ++ int i,j,k,l; ++ ++ for(i=-256;i<118+4;i++) ++ gainpow2[i+256] = pow((double)2.0,-0.25 * (double) (i+210) ); ++ ++ for(i=0;i<8207;i++) ++ ispow[i] = pow((double)i,(double)4.0/3.0); ++ ++ for (i=0;i<8;i++) ++ { ++ static double Ci[8]={-0.6,-0.535,-0.33,-0.185,-0.095,-0.041,-0.0142,-0.0037}; ++ double sq=sqrt(1.0+Ci[i]*Ci[i]); ++ aa_cs[i] = 1.0/sq; ++ aa_ca[i] = Ci[i]/sq; ++ } ++ ++ for(i=0;i<18;i++) ++ { ++ win[0][i] = win[1][i] = 0.5 * sin( M_PI / 72.0 * (double) (2*(i+0) +1) ) / cos ( M_PI * (double) (2*(i+0) +19) / 72.0 ); ++ win[0][i+18] = win[3][i+18] = 0.5 * sin( M_PI / 72.0 * (double) (2*(i+18)+1) ) / cos ( M_PI * (double) (2*(i+18)+19) / 72.0 ); ++ } ++ for(i=0;i<6;i++) ++ { ++ win[1][i+18] = 0.5 / cos ( M_PI * (double) (2*(i+18)+19) / 72.0 ); ++ win[3][i+12] = 0.5 / cos ( M_PI * (double) (2*(i+12)+19) / 72.0 ); ++ win[1][i+24] = 0.5 * sin( M_PI / 24.0 * (double) (2*i+13) ) / cos ( M_PI * (double) (2*(i+24)+19) / 72.0 ); ++ win[1][i+30] = win[3][i] = 0.0; ++ win[3][i+6 ] = 0.5 * sin( M_PI / 24.0 * (double) (2*i+1) ) / cos ( M_PI * (double) (2*(i+6 )+19) / 72.0 ); ++ } ++ ++ for(i=0;i<9;i++) ++ COS9[i] = cos( M_PI / 18.0 * (double) i); ++ ++ for(i=0;i<9;i++) ++ tfcos36[i] = 0.5 / cos ( M_PI * (double) (i*2+1) / 36.0 ); ++ for(i=0;i<3;i++) ++ tfcos12[i] = 0.5 / cos ( M_PI * (double) (i*2+1) / 12.0 ); ++ ++ COS6_1 = cos( M_PI / 6.0 * (double) 1); ++ COS6_2 = cos( M_PI / 6.0 * (double) 2); ++ ++ for(i=0;i<12;i++) ++ { ++ win[2][i] = 0.5 * sin( M_PI / 24.0 * (double) (2*i+1) ) / cos ( M_PI * (double) (2*i+7) / 24.0 ); ++ for(j=0;j<6;j++) ++ COS1[i][j] = cos( M_PI / 24.0 * (double) ((2*i+7)*(2*j+1)) ); ++ } ++ ++ for(j=0;j<4;j++) { ++ static int len[4] = { 36,36,12,36 }; ++ for(i=0;i<len[j];i+=2) ++ win1[j][i] = + win[j][i]; ++ for(i=1;i<len[j];i+=2) ++ win1[j][i] = - win[j][i]; ++ } ++ ++ for(i=0;i<16;i++) ++ { ++ double t = tan( (double) i * M_PI / 12.0 ); ++ tan1_1[i] = t / (1.0+t); ++ tan2_1[i] = 1.0 / (1.0 + t); ++ tan1_2[i] = M_SQRT2 * t / (1.0+t); ++ tan2_2[i] = M_SQRT2 / (1.0 + t); ++ ++ for(j=0;j<2;j++) { ++ double base = pow(2.0,-0.25*(j+1.0)); ++ double p1=1.0,p2=1.0; ++ if(i > 0) { ++ if( i & 1 ) ++ p1 = pow(base,(i+1.0)*0.5); ++ else ++ p2 = pow(base,i*0.5); ++ } ++ pow1_1[j][i] = p1; ++ pow2_1[j][i] = p2; ++ pow1_2[j][i] = M_SQRT2 * p1; ++ pow2_2[j][i] = M_SQRT2 * p2; ++ } ++ } ++ ++ for(j=0;j<9;j++) ++ { ++ struct bandInfoStruct *bi = &bandInfo[j]; ++ int *mp; ++ int cb,lwin; ++ short *bdf; ++ ++ mp = map[j][0] = mapbuf0[j]; ++ bdf = bi->longDiff; ++ for(i=0,cb = 0; cb < 8 ; cb++,i+=*bdf++) { ++ *mp++ = (*bdf) >> 1; ++ *mp++ = i; ++ *mp++ = 3; ++ *mp++ = cb; ++ } ++ bdf = bi->shortDiff+3; ++ for(cb=3;cb<13;cb++) { ++ int l = (*bdf++) >> 1; ++ for(lwin=0;lwin<3;lwin++) { ++ *mp++ = l; ++ *mp++ = i + lwin; ++ *mp++ = lwin; ++ *mp++ = cb; ++ } ++ i += 6*l; ++ } ++ mapend[j][0] = mp; ++ ++ mp = map[j][1] = mapbuf1[j]; ++ bdf = bi->shortDiff+0; ++ for(i=0,cb=0;cb<13;cb++) { ++ int l = (*bdf++) >> 1; ++ for(lwin=0;lwin<3;lwin++) { ++ *mp++ = l; ++ *mp++ = i + lwin; ++ *mp++ = lwin; ++ *mp++ = cb; ++ } ++ i += 6*l; ++ } ++ mapend[j][1] = mp; ++ ++ mp = map[j][2] = mapbuf2[j]; ++ bdf = bi->longDiff; ++ for(cb = 0; cb < 22 ; cb++) { ++ *mp++ = (*bdf++) >> 1; ++ *mp++ = cb; ++ } ++ mapend[j][2] = mp; ++ ++ } ++ ++ for(j=0;j<9;j++) { ++ for(i=0;i<23;i++) { ++ longLimit[j][i] = (bandInfo[j].longIdx[i] - 1 + 8) / 18 + 1; ++ if(longLimit[j][i] > (down_sample_sblimit) ) ++ longLimit[j][i] = down_sample_sblimit; ++ } ++ for(i=0;i<14;i++) { ++ shortLimit[j][i] = (bandInfo[j].shortIdx[i] - 1) / 18 + 1; ++ if(shortLimit[j][i] > (down_sample_sblimit) ) ++ shortLimit[j][i] = down_sample_sblimit; ++ } ++ } ++ ++ for(i=0;i<5;i++) { ++ for(j=0;j<6;j++) { ++ for(k=0;k<6;k++) { ++ int n = k + j * 6 + i * 36; ++ i_slen2[n] = i|(j<<3)|(k<<6)|(3<<12); ++ } ++ } ++ } ++ for(i=0;i<4;i++) { ++ for(j=0;j<4;j++) { ++ for(k=0;k<4;k++) { ++ int n = k + j * 4 + i * 16; ++ i_slen2[n+180] = i|(j<<3)|(k<<6)|(4<<12); ++ } ++ } ++ } ++ for(i=0;i<4;i++) { ++ for(j=0;j<3;j++) { ++ int n = j + i * 3; ++ i_slen2[n+244] = i|(j<<3) | (5<<12); ++ n_slen2[n+500] = i|(j<<3) | (2<<12) | (1<<15); ++ } ++ } ++ ++ for(i=0;i<5;i++) { ++ for(j=0;j<5;j++) { ++ for(k=0;k<4;k++) { ++ for(l=0;l<4;l++) { ++ int n = l + k * 4 + j * 16 + i * 80; ++ n_slen2[n] = i|(j<<3)|(k<<6)|(l<<9)|(0<<12); ++ } ++ } ++ } ++ } ++ for(i=0;i<5;i++) { ++ for(j=0;j<5;j++) { ++ for(k=0;k<4;k++) { ++ int n = k + j * 4 + i * 20; ++ n_slen2[n+400] = i|(j<<3)|(k<<6)|(1<<12); ++ } ++ } ++ } ++ } ++ ++ /* ++ * read additional side information ++ */ ++ #ifdef MPEG1 ++ static void III_get_side_info_1(struct III_sideinfo *si,int stereo, ++ int ms_stereo,long sfreq,int single) ++ { ++ int ch, gr; ++ int powdiff = (single == 3) ? 4 : 0; ++ ++ si->main_data_begin = getbits(9); ++ if (stereo == 1) ++ si->private_bits = getbits_fast(5); ++ else ++ si->private_bits = getbits_fast(3); ++ ++ for (ch=0; ch<stereo; ch++) { ++ si->ch[ch].gr[0].scfsi = -1; ++ si->ch[ch].gr[1].scfsi = getbits_fast(4); ++ } ++ ++ for (gr=0; gr<2; gr++) ++ { ++ for (ch=0; ch<stereo; ch++) ++ { ++ register struct gr_info_s *gr_info = &(si->ch[ch].gr[gr]); ++ ++ gr_info->part2_3_length = getbits(12); ++ gr_info->big_values = getbits_fast(9); ++ if(gr_info->big_values > 288) { ++ fprintf(stderr,"big_values too large!\n"); ++ gr_info->big_values = 288; ++ } ++ { ++ unsigned int qss = getbits_fast(8); ++ gr_info->pow2gain = gainpow2+256 - qss + powdiff; ++ #ifdef HAVEGTK ++ if (gtkflag) { ++ pinfo->qss[gr][ch]=qss; ++ pinfo->big_values[gr][ch]=gr_info->big_values; ++ } ++ #endif ++ } ++ if(ms_stereo) ++ gr_info->pow2gain += 2; ++ gr_info->scalefac_compress = getbits_fast(4); ++ /* window-switching flag == 1 for block_Type != 0 .. and block-type == 0 -> win-sw-flag = 0 */ ++ if(get1bit()) ++ { ++ int i; ++ gr_info->block_type = getbits_fast(2); ++ gr_info->mixed_block_flag = get1bit(); ++ gr_info->table_select[0] = getbits_fast(5); ++ gr_info->table_select[1] = getbits_fast(5); ++ ++ ++ /* ++ * table_select[2] not needed, because there is no region2, ++ * but to satisfy some verifications tools we set it either. ++ */ ++ gr_info->table_select[2] = 0; ++ for(i=0;i<3;i++) { ++ unsigned int sbg = (getbits_fast(3)<<3); ++ gr_info->full_gain[i] = gr_info->pow2gain + sbg; ++ #ifdef HAVEGTK ++ if (gtkflag) ++ pinfo->sub_gain[gr][ch][i]=sbg/8; ++ #endif ++ } ++ ++ ++ if(gr_info->block_type == 0) { ++ fprintf(stderr,"Blocktype == 0 and window-switching == 1 not allowed.\n"); ++ exit(1); ++ } ++ /* region_count/start parameters are implicit in this case. */ ++ gr_info->region1start = 36>>1; ++ gr_info->region2start = 576>>1; ++ } ++ else ++ { ++ int i,r0c,r1c; ++ for (i=0; i<3; i++) ++ gr_info->table_select[i] = getbits_fast(5); ++ r0c = getbits_fast(4); ++ r1c = getbits_fast(3); ++ gr_info->region1start = bandInfo[sfreq].longIdx[r0c+1] >> 1 ; ++ gr_info->region2start = bandInfo[sfreq].longIdx[r0c+1+r1c+1] >> 1; ++ gr_info->block_type = 0; ++ gr_info->mixed_block_flag = 0; ++ } ++ gr_info->preflag = get1bit(); ++ gr_info->scalefac_scale = get1bit(); ++ gr_info->count1table_select = get1bit(); ++ #ifdef HAVEGTK ++ if (gtkflag) ++ pinfo->scalefac_scale[gr][ch]=gr_info->scalefac_scale; ++ #endif ++ } ++ } ++ } ++ #endif ++ ++ /* ++ * Side Info for MPEG 2.0 / LSF ++ */ ++ static void III_get_side_info_2(struct III_sideinfo *si,int stereo, ++ int ms_stereo,long sfreq,int single) ++ { ++ int ch; ++ int powdiff = (single == 3) ? 4 : 0; ++ ++ si->main_data_begin = getbits(8); ++ if (stereo == 1) ++ si->private_bits = get1bit(); ++ else ++ si->private_bits = getbits_fast(2); ++ ++ for (ch=0; ch<stereo; ch++) ++ { ++ register struct gr_info_s *gr_info = &(si->ch[ch].gr[0]); ++ unsigned int qss; ++ ++ gr_info->part2_3_length = getbits(12); ++ gr_info->big_values = getbits_fast(9); ++ if(gr_info->big_values > 288) { ++ fprintf(stderr,"big_values too large!\n"); ++ gr_info->big_values = 288; ++ } ++ qss=getbits_fast(8); ++ gr_info->pow2gain = gainpow2+256 - qss + powdiff; ++ #ifdef HAVEGTK ++ if (gtkflag) { ++ pinfo->qss[0][ch]=qss; ++ pinfo->big_values[0][ch]=gr_info->big_values; ++ } ++ #endif ++ ++ if(ms_stereo) ++ gr_info->pow2gain += 2; ++ gr_info->scalefac_compress = getbits(9); ++ /* window-switching flag == 1 for block_Type != 0 .. and block-type == 0 -> win-sw-flag = 0 */ ++ if(get1bit()) ++ { ++ int i; ++ gr_info->block_type = getbits_fast(2); ++ gr_info->mixed_block_flag = get1bit(); ++ gr_info->table_select[0] = getbits_fast(5); ++ gr_info->table_select[1] = getbits_fast(5); ++ /* ++ * table_select[2] not needed, because there is no region2, ++ * but to satisfy some verifications tools we set it either. ++ */ ++ gr_info->table_select[2] = 0; ++ for(i=0;i<3;i++) { ++ unsigned int sbg = (getbits_fast(3)<<3); ++ gr_info->full_gain[i] = gr_info->pow2gain + sbg; ++ #ifdef HAVEGTK ++ if (gtkflag) ++ pinfo->sub_gain[0][ch][i]=sbg/8; ++ #endif ++ } ++ ++ if(gr_info->block_type == 0) { ++ fprintf(stderr,"Blocktype == 0 and window-switching == 1 not allowed.\n"); ++ exit(1); ++ } ++ /* region_count/start parameters are implicit in this case. */ ++ /* check this again! */ ++ if(gr_info->block_type == 2) ++ gr_info->region1start = 36>>1; ++ else if(sfreq == 8) ++ /* check this for 2.5 and sfreq=8 */ ++ gr_info->region1start = 108>>1; ++ else ++ gr_info->region1start = 54>>1; ++ gr_info->region2start = 576>>1; ++ } ++ else ++ { ++ int i,r0c,r1c; ++ for (i=0; i<3; i++) ++ gr_info->table_select[i] = getbits_fast(5); ++ r0c = getbits_fast(4); ++ r1c = getbits_fast(3); ++ gr_info->region1start = bandInfo[sfreq].longIdx[r0c+1] >> 1 ; ++ gr_info->region2start = bandInfo[sfreq].longIdx[r0c+1+r1c+1] >> 1; ++ gr_info->block_type = 0; ++ gr_info->mixed_block_flag = 0; ++ } ++ gr_info->scalefac_scale = get1bit(); ++ gr_info->count1table_select = get1bit(); ++ #ifdef HAVEGTK ++ if (gtkflag) ++ pinfo->scalefac_scale[0][ch]=gr_info->scalefac_scale; ++ #endif ++ } ++ } ++ ++ /* ++ * read scalefactors ++ */ ++ #ifdef MPEG1 ++ static int III_get_scale_factors_1(int *scf,struct gr_info_s *gr_info) ++ { ++ static unsigned char slen[2][16] = { ++ {0, 0, 0, 0, 3, 1, 1, 1, 2, 2, 2, 3, 3, 3, 4, 4}, ++ {0, 1, 2, 3, 0, 1, 2, 3, 1, 2, 3, 1, 2, 3, 2, 3} ++ }; ++ int numbits; ++ int num0 = slen[0][gr_info->scalefac_compress]; ++ int num1 = slen[1][gr_info->scalefac_compress]; ++ ++ if (gr_info->block_type == 2) ++ { ++ int i=18; ++ numbits = (num0 + num1) * 18; ++ ++ if (gr_info->mixed_block_flag) { ++ for (i=8;i;i--) ++ *scf++ = getbits_fast(num0); ++ i = 9; ++ numbits -= num0; /* num0 * 17 + num1 * 18 */ ++ } ++ ++ for (;i;i--) ++ *scf++ = getbits_fast(num0); ++ for (i = 18; i; i--) ++ *scf++ = getbits_fast(num1); ++ *scf++ = 0; *scf++ = 0; *scf++ = 0; /* short[13][0..2] = 0 */ ++ } ++ else ++ { ++ int i; ++ int scfsi = gr_info->scfsi; ++ ++ if(scfsi < 0) { /* scfsi < 0 => granule == 0 */ ++ for(i=11;i;i--) ++ *scf++ = getbits_fast(num0); ++ for(i=10;i;i--) ++ *scf++ = getbits_fast(num1); ++ numbits = (num0 + num1) * 10 + num0; ++ } ++ else { ++ numbits = 0; ++ if(!(scfsi & 0x8)) { ++ for (i=6;i;i--) ++ *scf++ = getbits_fast(num0); ++ numbits += num0 * 6; ++ } ++ else { ++ scf += 6; ++ } ++ ++ if(!(scfsi & 0x4)) { ++ for (i=5;i;i--) ++ *scf++ = getbits_fast(num0); ++ numbits += num0 * 5; ++ } ++ else { ++ scf += 5; ++ } ++ ++ if(!(scfsi & 0x2)) { ++ for(i=5;i;i--) ++ *scf++ = getbits_fast(num1); ++ numbits += num1 * 5; ++ } ++ else { ++ scf += 5; ++ } ++ ++ if(!(scfsi & 0x1)) { ++ for (i=5;i;i--) ++ *scf++ = getbits_fast(num1); ++ numbits += num1 * 5; ++ } ++ else { ++ scf += 5; ++ } ++ } ++ ++ *scf++ = 0; /* no l[21] in original sources */ ++ } ++ return numbits; ++ } ++ #endif ++ ++ static int III_get_scale_factors_2(int *scf,struct gr_info_s *gr_info,int i_stereo) ++ { ++ unsigned char *pnt; ++ int i,j; ++ unsigned int slen; ++ int n = 0; ++ int numbits = 0; ++ ++ static unsigned char stab[3][6][4] = { ++ { { 6, 5, 5,5 } , { 6, 5, 7,3 } , { 11,10,0,0} , ++ { 7, 7, 7,0 } , { 6, 6, 6,3 } , { 8, 8,5,0} } , ++ { { 9, 9, 9,9 } , { 9, 9,12,6 } , { 18,18,0,0} , ++ {12,12,12,0 } , {12, 9, 9,6 } , { 15,12,9,0} } , ++ { { 6, 9, 9,9 } , { 6, 9,12,6 } , { 15,18,0,0} , ++ { 6,15,12,0 } , { 6,12, 9,6 } , { 6,18,9,0} } }; ++ ++ if(i_stereo) /* i_stereo AND second channel -> do_layer3() checks this */ ++ slen = i_slen2[gr_info->scalefac_compress>>1]; ++ else ++ slen = n_slen2[gr_info->scalefac_compress]; ++ ++ gr_info->preflag = (slen>>15) & 0x1; ++ ++ n = 0; ++ if( gr_info->block_type == 2 ) { ++ n++; ++ if(gr_info->mixed_block_flag) ++ n++; ++ } ++ ++ pnt = stab[n][(slen>>12)&0x7]; ++ ++ for(i=0;i<4;i++) { ++ int num = slen & 0x7; ++ slen >>= 3; ++ if(num) { ++ for(j=0;j<(int)(pnt[i]);j++) ++ *scf++ = getbits_fast(num); ++ numbits += pnt[i] * num; ++ } ++ else { ++ for(j=0;j<(int)(pnt[i]);j++) ++ *scf++ = 0; ++ } ++ } ++ ++ n = (n << 1) + 1; ++ for(i=0;i<n;i++) ++ *scf++ = 0; ++ ++ return numbits; ++ } ++ ++ static int pretab1[22] = {0,0,0,0,0,0,0,0,0,0,0,1,1,1,1,2,2,3,3,3,2,0}; ++ static int pretab2[22] = {0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0}; ++ ++ /* ++ * don't forget to apply the same changes to III_dequantize_sample_ms() !!! ++ */ ++ static int III_dequantize_sample(real xr[SBLIMIT][SSLIMIT],int *scf, ++ struct gr_info_s *gr_info,int sfreq,int part2bits) ++ { ++ int shift = 1 + gr_info->scalefac_scale; ++ real *xrpnt = (real *) xr; ++ int l[3],l3; ++ int part2remain = gr_info->part2_3_length - part2bits; ++ int *me; ++ ++ { ++ int bv = gr_info->big_values; ++ int region1 = gr_info->region1start; ++ int region2 = gr_info->region2start; ++ ++ l3 = ((576>>1)-bv)>>1; ++ /* ++ * we may lose the 'odd' bit here !! ++ * check this later again ++ */ ++ if(bv <= region1) { ++ l[0] = bv; l[1] = 0; l[2] = 0; ++ } ++ else { ++ l[0] = region1; ++ if(bv <= region2) { ++ l[1] = bv - l[0]; l[2] = 0; ++ } ++ else { ++ l[1] = region2 - l[0]; l[2] = bv - region2; ++ } ++ } ++ } ++ ++ if(gr_info->block_type == 2) { ++ /* ++ * decoding with short or mixed mode BandIndex table ++ */ ++ int i,max[4]; ++ int step=0,lwin=0,cb=0; ++ register real v = 0.0; ++ register int *m,mc; ++ ++ if(gr_info->mixed_block_flag) { ++ max[3] = -1; ++ max[0] = max[1] = max[2] = 2; ++ m = map[sfreq][0]; ++ me = mapend[sfreq][0]; ++ } ++ else { ++ max[0] = max[1] = max[2] = max[3] = -1; ++ /* max[3] not really needed in this case */ ++ m = map[sfreq][1]; ++ me = mapend[sfreq][1]; ++ } ++ ++ mc = 0; ++ for(i=0;i<2;i++) { ++ int lp = l[i]; ++ struct newhuff *h = ht+gr_info->table_select[i]; ++ for(;lp;lp--,mc--) { ++ register int x,y; ++ if( (!mc) ) { ++ mc = *m++; ++ xrpnt = ((real *) xr) + (*m++); ++ lwin = *m++; ++ cb = *m++; ++ if(lwin == 3) { ++ v = gr_info->pow2gain[(*scf++) << shift]; ++ step = 1; ++ } ++ else { ++ v = gr_info->full_gain[lwin][(*scf++) << shift]; ++ step = 3; ++ } ++ } ++ { ++ register short *val = h->table; ++ while((y=*val++)<0) { ++ if (get1bit()) ++ val -= y; ++ part2remain--; ++ } ++ x = y >> 4; ++ y &= 0xf; ++ } ++ if(x == 15) { ++ max[lwin] = cb; ++ part2remain -= h->linbits+1; ++ x += getbits(h->linbits); ++ if(get1bit()) ++ *xrpnt = -ispow[x] * v; ++ else ++ *xrpnt = ispow[x] * v; ++ } ++ else if(x) { ++ max[lwin] = cb; ++ if(get1bit()) ++ *xrpnt = -ispow[x] * v; ++ else ++ *xrpnt = ispow[x] * v; ++ part2remain--; ++ } ++ else ++ *xrpnt = 0.0; ++ xrpnt += step; ++ if(y == 15) { ++ max[lwin] = cb; ++ part2remain -= h->linbits+1; ++ y += getbits(h->linbits); ++ if(get1bit()) ++ *xrpnt = -ispow[y] * v; ++ else ++ *xrpnt = ispow[y] * v; ++ } ++ else if(y) { ++ max[lwin] = cb; ++ if(get1bit()) ++ *xrpnt = -ispow[y] * v; ++ else ++ *xrpnt = ispow[y] * v; ++ part2remain--; ++ } ++ else ++ *xrpnt = 0.0; ++ xrpnt += step; ++ } ++ } ++ for(;l3 && (part2remain > 0);l3--) { ++ struct newhuff *h = htc+gr_info->count1table_select; ++ register short *val = h->table,a; ++ ++ while((a=*val++)<0) { ++ part2remain--; ++ if(part2remain < 0) { ++ part2remain++; ++ a = 0; ++ break; ++ } ++ if (get1bit()) ++ val -= a; ++ } ++ for(i=0;i<4;i++) { ++ if(!(i & 1)) { ++ if(!mc) { ++ mc = *m++; ++ xrpnt = ((real *) xr) + (*m++); ++ lwin = *m++; ++ cb = *m++; ++ if(lwin == 3) { ++ v = gr_info->pow2gain[(*scf++) << shift]; ++ step = 1; ++ } ++ else { ++ v = gr_info->full_gain[lwin][(*scf++) << shift]; ++ step = 3; ++ } ++ } ++ mc--; ++ } ++ if( (a & (0x8>>i)) ) { ++ max[lwin] = cb; ++ part2remain--; ++ if(part2remain < 0) { ++ part2remain++; ++ break; ++ } ++ if(get1bit()) ++ *xrpnt = -v; ++ else ++ *xrpnt = v; ++ } ++ else ++ *xrpnt = 0.0; ++ xrpnt += step; ++ } ++ } ++ ++ while( m < me ) { ++ if(!mc) { ++ mc = *m++; ++ xrpnt = ((real *) xr) + *m++; ++ if( (*m++) == 3) ++ step = 1; ++ else ++ step = 3; ++ m++; /* cb */ ++ } ++ mc--; ++ *xrpnt = 0.0; ++ xrpnt += step; ++ *xrpnt = 0.0; ++ xrpnt += step; ++ /* we could add a little opt. here: ++ * if we finished a band for window 3 or a long band ++ * further bands could copied in a simple loop without a ++ * special 'map' decoding ++ */ ++ } ++ ++ gr_info->maxband[0] = max[0]+1; ++ gr_info->maxband[1] = max[1]+1; ++ gr_info->maxband[2] = max[2]+1; ++ gr_info->maxbandl = max[3]+1; ++ ++ { ++ int rmax = max[0] > max[1] ? max[0] : max[1]; ++ rmax = (rmax > max[2] ? rmax : max[2]) + 1; ++ gr_info->maxb = rmax ? shortLimit[sfreq][rmax] : longLimit[sfreq][max[3]+1]; ++ } ++ ++ } ++ else { ++ /* ++ * decoding with 'long' BandIndex table (block_type != 2) ++ */ ++ int *pretab = gr_info->preflag ? pretab1 : pretab2; ++ int i,max = -1; ++ int cb = 0; ++ register int *m = map[sfreq][2]; ++ register real v = 0.0; ++ register int mc = 0; ++ #if 0 ++ me = mapend[sfreq][2]; ++ #endif ++ ++ /* ++ * long hash table values ++ */ ++ for(i=0;i<3;i++) { ++ int lp = l[i]; ++ struct newhuff *h = ht+gr_info->table_select[i]; ++ ++ for(;lp;lp--,mc--) { ++ int x,y; ++ ++ if(!mc) { ++ mc = *m++; ++ v = gr_info->pow2gain[((*scf++) + (*pretab++)) << shift]; ++ cb = *m++; ++ } ++ { ++ register short *val = h->table; ++ while((y=*val++)<0) { ++ if (get1bit()) ++ val -= y; ++ part2remain--; ++ } ++ x = y >> 4; ++ y &= 0xf; ++ } ++ if (x == 15) { ++ max = cb; ++ part2remain -= h->linbits+1; ++ x += getbits(h->linbits); ++ if(get1bit()) ++ *xrpnt++ = -ispow[x] * v; ++ else ++ *xrpnt++ = ispow[x] * v; ++ } ++ else if(x) { ++ max = cb; ++ if(get1bit()) ++ *xrpnt++ = -ispow[x] * v; ++ else ++ *xrpnt++ = ispow[x] * v; ++ part2remain--; ++ } ++ else ++ *xrpnt++ = 0.0; ++ ++ if (y == 15) { ++ max = cb; ++ part2remain -= h->linbits+1; ++ y += getbits(h->linbits); ++ if(get1bit()) ++ *xrpnt++ = -ispow[y] * v; ++ else ++ *xrpnt++ = ispow[y] * v; ++ } ++ else if(y) { ++ max = cb; ++ if(get1bit()) ++ *xrpnt++ = -ispow[y] * v; ++ else ++ *xrpnt++ = ispow[y] * v; ++ part2remain--; ++ } ++ else ++ *xrpnt++ = 0.0; ++ } ++ } ++ ++ /* ++ * short (count1table) values ++ */ ++ for(;l3 && (part2remain > 0);l3--) { ++ struct newhuff *h = htc+gr_info->count1table_select; ++ register short *val = h->table,a; ++ ++ while((a=*val++)<0) { ++ part2remain--; ++ if(part2remain < 0) { ++ part2remain++; ++ a = 0; ++ break; ++ } ++ if (get1bit()) ++ val -= a; ++ } ++ for(i=0;i<4;i++) { ++ if(!(i & 1)) { ++ if(!mc) { ++ mc = *m++; ++ cb = *m++; ++ v = gr_info->pow2gain[((*scf++) + (*pretab++)) << shift]; ++ } ++ mc--; ++ } ++ if ( (a & (0x8>>i)) ) { ++ max = cb; ++ part2remain--; ++ if(part2remain < 0) { ++ part2remain++; ++ break; ++ } ++ if(get1bit()) ++ *xrpnt++ = -v; ++ else ++ *xrpnt++ = v; ++ } ++ else ++ *xrpnt++ = 0.0; ++ } ++ } ++ ++ /* ++ * zero part ++ */ ++ for(i=(&xr[SBLIMIT][0]-xrpnt)>>1;i;i--) { ++ *xrpnt++ = 0.0; ++ *xrpnt++ = 0.0; ++ } ++ ++ gr_info->maxbandl = max+1; ++ gr_info->maxb = longLimit[sfreq][gr_info->maxbandl]; ++ } ++ ++ while( part2remain > 16 ) { ++ getbits(16); /* Dismiss stuffing Bits */ ++ part2remain -= 16; ++ } ++ if(part2remain > 0) ++ getbits(part2remain); ++ else if(part2remain < 0) { ++ fprintf(stderr,"mpg123: Can't rewind stream by %d bits!\n",-part2remain); ++ return 1; /* -> error */ ++ } ++ return 0; ++ } ++ ++ ++ /* ++ * III_stereo: calculate real channel values for Joint-I-Stereo-mode ++ */ ++ static void III_i_stereo(real xr_buf[2][SBLIMIT][SSLIMIT],int *scalefac, ++ struct gr_info_s *gr_info,int sfreq,int ms_stereo,int lsf) ++ { ++ real (*xr)[SBLIMIT*SSLIMIT] = (real (*)[SBLIMIT*SSLIMIT] ) xr_buf; ++ struct bandInfoStruct *bi = &bandInfo[sfreq]; ++ real *tab1,*tab2; ++ ++ if(lsf) { ++ int p = gr_info->scalefac_compress & 0x1; ++ if(ms_stereo) { ++ tab1 = pow1_2[p]; tab2 = pow2_2[p]; ++ } ++ else { ++ tab1 = pow1_1[p]; tab2 = pow2_1[p]; ++ } ++ } ++ else { ++ if(ms_stereo) { ++ tab1 = tan1_2; tab2 = tan2_2; ++ } ++ else { ++ tab1 = tan1_1; tab2 = tan2_1; ++ } ++ } ++ ++ if (gr_info->block_type == 2) ++ { ++ int lwin,do_l = 0; ++ if( gr_info->mixed_block_flag ) ++ do_l = 1; ++ ++ for (lwin=0;lwin<3;lwin++) /* process each window */ ++ { ++ /* get first band with zero values */ ++ int is_p,sb,idx,sfb = gr_info->maxband[lwin]; /* sfb is minimal 3 for mixed mode */ ++ if(sfb > 3) ++ do_l = 0; ++ ++ for(;sfb<12;sfb++) ++ { ++ is_p = scalefac[sfb*3+lwin-gr_info->mixed_block_flag]; /* scale: 0-15 */ ++ if(is_p != 7) { ++ real t1,t2; ++ sb = bi->shortDiff[sfb]; ++ idx = bi->shortIdx[sfb] + lwin; ++ t1 = tab1[is_p]; t2 = tab2[is_p]; ++ for (; sb > 0; sb--,idx+=3) ++ { ++ real v = xr[0][idx]; ++ xr[0][idx] = v * t1; ++ xr[1][idx] = v * t2; ++ } ++ } ++ } ++ ++ #if 1 ++ /* in the original: copy 10 to 11 , here: copy 11 to 12 ++ maybe still wrong??? (copy 12 to 13?) */ ++ is_p = scalefac[11*3+lwin-gr_info->mixed_block_flag]; /* scale: 0-15 */ ++ sb = bi->shortDiff[12]; ++ idx = bi->shortIdx[12] + lwin; ++ #else ++ is_p = scalefac[10*3+lwin-gr_info->mixed_block_flag]; /* scale: 0-15 */ ++ sb = bi->shortDiff[11]; ++ idx = bi->shortIdx[11] + lwin; ++ #endif ++ if(is_p != 7) ++ { ++ real t1,t2; ++ t1 = tab1[is_p]; t2 = tab2[is_p]; ++ for ( ; sb > 0; sb--,idx+=3 ) ++ { ++ real v = xr[0][idx]; ++ xr[0][idx] = v * t1; ++ xr[1][idx] = v * t2; ++ } ++ } ++ } /* end for(lwin; .. ; . ) */ ++ ++ if (do_l) ++ { ++ /* also check l-part, if ALL bands in the three windows are 'empty' ++ * and mode = mixed_mode ++ */ ++ int sfb = gr_info->maxbandl; ++ int idx = bi->longIdx[sfb]; ++ ++ for ( ; sfb<8; sfb++ ) ++ { ++ int sb = bi->longDiff[sfb]; ++ int is_p = scalefac[sfb]; /* scale: 0-15 */ ++ if(is_p != 7) { ++ real t1,t2; ++ t1 = tab1[is_p]; t2 = tab2[is_p]; ++ for ( ; sb > 0; sb--,idx++) ++ { ++ real v = xr[0][idx]; ++ xr[0][idx] = v * t1; ++ xr[1][idx] = v * t2; ++ } ++ } ++ else ++ idx += sb; ++ } ++ } ++ } ++ else /* ((gr_info->block_type != 2)) */ ++ { ++ int sfb = gr_info->maxbandl; ++ int is_p,idx = bi->longIdx[sfb]; ++ for ( ; sfb<21; sfb++) ++ { ++ int sb = bi->longDiff[sfb]; ++ is_p = scalefac[sfb]; /* scale: 0-15 */ ++ if(is_p != 7) { ++ real t1,t2; ++ t1 = tab1[is_p]; t2 = tab2[is_p]; ++ for ( ; sb > 0; sb--,idx++) ++ { ++ real v = xr[0][idx]; ++ xr[0][idx] = v * t1; ++ xr[1][idx] = v * t2; ++ } ++ } ++ else ++ idx += sb; ++ } ++ ++ is_p = scalefac[20]; /* copy l-band 20 to l-band 21 */ ++ if(is_p != 7) ++ { ++ int sb; ++ real t1 = tab1[is_p],t2 = tab2[is_p]; ++ ++ for ( sb = bi->longDiff[21]; sb > 0; sb--,idx++ ) ++ { ++ real v = xr[0][idx]; ++ xr[0][idx] = v * t1; ++ xr[1][idx] = v * t2; ++ } ++ } ++ } /* ... */ ++ } ++ ++ static void III_antialias(real xr[SBLIMIT][SSLIMIT],struct gr_info_s *gr_info) ++ { ++ int sblim; ++ ++ if(gr_info->block_type == 2) ++ { ++ if(!gr_info->mixed_block_flag) ++ return; ++ sblim = 1; ++ } ++ else { ++ sblim = gr_info->maxb-1; ++ } ++ ++ /* 31 alias-reduction operations between each pair of sub-bands */ ++ /* with 8 butterflies between each pair */ ++ ++ { ++ int sb; ++ real *xr1=(real *) xr[1]; ++ ++ for(sb=sblim;sb;sb--,xr1+=10) ++ { ++ int ss; ++ real *cs=aa_cs,*ca=aa_ca; ++ real *xr2 = xr1; ++ ++ for(ss=7;ss>=0;ss--) ++ { /* upper and lower butterfly inputs */ ++ register real bu = *--xr2,bd = *xr1; ++ *xr2 = (bu * (*cs) ) - (bd * (*ca) ); ++ *xr1++ = (bd * (*cs++) ) + (bu * (*ca++) ); ++ } ++ } ++ } ++ } ++ ++ /* ++ DCT insipired by Jeff Tsay's DCT from the maplay package ++ this is an optimized version with manual unroll. ++ ++ References: ++ [1] S. Winograd: "On Computing the Discrete Fourier Transform", ++ Mathematics of Computation, Volume 32, Number 141, January 1978, ++ Pages 175-199 ++ */ ++ ++ static void dct36(real *inbuf,real *o1,real *o2,real *wintab,real *tsbuf) ++ { ++ { ++ register real *in = inbuf; ++ ++ in[17]+=in[16]; in[16]+=in[15]; in[15]+=in[14]; ++ in[14]+=in[13]; in[13]+=in[12]; in[12]+=in[11]; ++ in[11]+=in[10]; in[10]+=in[9]; in[9] +=in[8]; ++ in[8] +=in[7]; in[7] +=in[6]; in[6] +=in[5]; ++ in[5] +=in[4]; in[4] +=in[3]; in[3] +=in[2]; ++ in[2] +=in[1]; in[1] +=in[0]; ++ ++ in[17]+=in[15]; in[15]+=in[13]; in[13]+=in[11]; in[11]+=in[9]; ++ in[9] +=in[7]; in[7] +=in[5]; in[5] +=in[3]; in[3] +=in[1]; ++ ++ ++ { ++ ++ #define MACRO0(v) { \ ++ real tmp; \ ++ out2[9+(v)] = (tmp = sum0 + sum1) * w[27+(v)]; \ ++ out2[8-(v)] = tmp * w[26-(v)]; } \ ++ sum0 -= sum1; \ ++ ts[SBLIMIT*(8-(v))] = out1[8-(v)] + sum0 * w[8-(v)]; \ ++ ts[SBLIMIT*(9+(v))] = out1[9+(v)] + sum0 * w[9+(v)]; ++ #define MACRO1(v) { \ ++ real sum0,sum1; \ ++ sum0 = tmp1a + tmp2a; \ ++ sum1 = (tmp1b + tmp2b) * tfcos36[(v)]; \ ++ MACRO0(v); } ++ #define MACRO2(v) { \ ++ real sum0,sum1; \ ++ sum0 = tmp2a - tmp1a; \ ++ sum1 = (tmp2b - tmp1b) * tfcos36[(v)]; \ ++ MACRO0(v); } ++ ++ register const real *c = COS9; ++ register real *out2 = o2; ++ register real *w = wintab; ++ register real *out1 = o1; ++ register real *ts = tsbuf; ++ ++ real ta33,ta66,tb33,tb66; ++ ++ ta33 = in[2*3+0] * c[3]; ++ ta66 = in[2*6+0] * c[6]; ++ tb33 = in[2*3+1] * c[3]; ++ tb66 = in[2*6+1] * c[6]; ++ ++ { ++ real tmp1a,tmp2a,tmp1b,tmp2b; ++ tmp1a = in[2*1+0] * c[1] + ta33 + in[2*5+0] * c[5] + in[2*7+0] * c[7]; ++ tmp1b = in[2*1+1] * c[1] + tb33 + in[2*5+1] * c[5] + in[2*7+1] * c[7]; ++ tmp2a = in[2*0+0] + in[2*2+0] * c[2] + in[2*4+0] * c[4] + ta66 + in[2*8+0] * c[8]; ++ tmp2b = in[2*0+1] + in[2*2+1] * c[2] + in[2*4+1] * c[4] + tb66 + in[2*8+1] * c[8]; ++ ++ MACRO1(0); ++ MACRO2(8); ++ } ++ ++ { ++ real tmp1a,tmp2a,tmp1b,tmp2b; ++ tmp1a = ( in[2*1+0] - in[2*5+0] - in[2*7+0] ) * c[3]; ++ tmp1b = ( in[2*1+1] - in[2*5+1] - in[2*7+1] ) * c[3]; ++ tmp2a = ( in[2*2+0] - in[2*4+0] - in[2*8+0] ) * c[6] - in[2*6+0] + in[2*0+0]; ++ tmp2b = ( in[2*2+1] - in[2*4+1] - in[2*8+1] ) * c[6] - in[2*6+1] + in[2*0+1]; ++ ++ MACRO1(1); ++ MACRO2(7); ++ } ++ ++ { ++ real tmp1a,tmp2a,tmp1b,tmp2b; ++ tmp1a = in[2*1+0] * c[5] - ta33 - in[2*5+0] * c[7] + in[2*7+0] * c[1]; ++ tmp1b = in[2*1+1] * c[5] - tb33 - in[2*5+1] * c[7] + in[2*7+1] * c[1]; ++ tmp2a = in[2*0+0] - in[2*2+0] * c[8] - in[2*4+0] * c[2] + ta66 + in[2*8+0] * c[4]; ++ tmp2b = in[2*0+1] - in[2*2+1] * c[8] - in[2*4+1] * c[2] + tb66 + in[2*8+1] * c[4]; ++ ++ MACRO1(2); ++ MACRO2(6); ++ } ++ ++ { ++ real tmp1a,tmp2a,tmp1b,tmp2b; ++ tmp1a = in[2*1+0] * c[7] - ta33 + in[2*5+0] * c[1] - in[2*7+0] * c[5]; ++ tmp1b = in[2*1+1] * c[7] - tb33 + in[2*5+1] * c[1] - in[2*7+1] * c[5]; ++ tmp2a = in[2*0+0] - in[2*2+0] * c[4] + in[2*4+0] * c[8] + ta66 - in[2*8+0] * c[2]; ++ tmp2b = in[2*0+1] - in[2*2+1] * c[4] + in[2*4+1] * c[8] + tb66 - in[2*8+1] * c[2]; ++ ++ MACRO1(3); ++ MACRO2(5); ++ } ++ ++ { ++ real sum0,sum1; ++ sum0 = in[2*0+0] - in[2*2+0] + in[2*4+0] - in[2*6+0] + in[2*8+0]; ++ sum1 = (in[2*0+1] - in[2*2+1] + in[2*4+1] - in[2*6+1] + in[2*8+1] ) * tfcos36[4]; ++ MACRO0(4); ++ } ++ } ++ ++ } ++ } ++ ++ /* ++ * new DCT12 ++ */ ++ static void dct12(real *in,real *rawout1,real *rawout2,register real *wi,register real *ts) ++ { ++ #define DCT12_PART1 \ ++ in5 = in[5*3]; \ ++ in5 += (in4 = in[4*3]); \ ++ in4 += (in3 = in[3*3]); \ ++ in3 += (in2 = in[2*3]); \ ++ in2 += (in1 = in[1*3]); \ ++ in1 += (in0 = in[0*3]); \ ++ \ ++ in5 += in3; in3 += in1; \ ++ \ ++ in2 *= COS6_1; \ ++ in3 *= COS6_1; \ ++ ++ #define DCT12_PART2 \ ++ in0 += in4 * COS6_2; \ ++ \ ++ in4 = in0 + in2; \ ++ in0 -= in2; \ ++ \ ++ in1 += in5 * COS6_2; \ ++ \ ++ in5 = (in1 + in3) * tfcos12[0]; \ ++ in1 = (in1 - in3) * tfcos12[2]; \ ++ \ ++ in3 = in4 + in5; \ ++ in4 -= in5; \ ++ \ ++ in2 = in0 + in1; \ ++ in0 -= in1; ++ ++ ++ { ++ real in0,in1,in2,in3,in4,in5; ++ register real *out1 = rawout1; ++ ts[SBLIMIT*0] = out1[0]; ts[SBLIMIT*1] = out1[1]; ts[SBLIMIT*2] = out1[2]; ++ ts[SBLIMIT*3] = out1[3]; ts[SBLIMIT*4] = out1[4]; ts[SBLIMIT*5] = out1[5]; ++ ++ DCT12_PART1 ++ ++ { ++ real tmp0,tmp1 = (in0 - in4); ++ { ++ real tmp2 = (in1 - in5) * tfcos12[1]; ++ tmp0 = tmp1 + tmp2; ++ tmp1 -= tmp2; ++ } ++ ts[(17-1)*SBLIMIT] = out1[17-1] + tmp0 * wi[11-1]; ++ ts[(12+1)*SBLIMIT] = out1[12+1] + tmp0 * wi[6+1]; ++ ts[(6 +1)*SBLIMIT] = out1[6 +1] + tmp1 * wi[1]; ++ ts[(11-1)*SBLIMIT] = out1[11-1] + tmp1 * wi[5-1]; ++ } ++ ++ DCT12_PART2 ++ ++ ts[(17-0)*SBLIMIT] = out1[17-0] + in2 * wi[11-0]; ++ ts[(12+0)*SBLIMIT] = out1[12+0] + in2 * wi[6+0]; ++ ts[(12+2)*SBLIMIT] = out1[12+2] + in3 * wi[6+2]; ++ ts[(17-2)*SBLIMIT] = out1[17-2] + in3 * wi[11-2]; ++ ++ ts[(6+0)*SBLIMIT] = out1[6+0] + in0 * wi[0]; ++ ts[(11-0)*SBLIMIT] = out1[11-0] + in0 * wi[5-0]; ++ ts[(6+2)*SBLIMIT] = out1[6+2] + in4 * wi[2]; ++ ts[(11-2)*SBLIMIT] = out1[11-2] + in4 * wi[5-2]; ++ } ++ ++ in++; ++ ++ { ++ real in0,in1,in2,in3,in4,in5; ++ register real *out2 = rawout2; ++ ++ DCT12_PART1 ++ ++ { ++ real tmp0,tmp1 = (in0 - in4); ++ { ++ real tmp2 = (in1 - in5) * tfcos12[1]; ++ tmp0 = tmp1 + tmp2; ++ tmp1 -= tmp2; ++ } ++ out2[5-1] = tmp0 * wi[11-1]; ++ out2[0+1] = tmp0 * wi[6+1]; ++ ts[(12+1)*SBLIMIT] += tmp1 * wi[1]; ++ ts[(17-1)*SBLIMIT] += tmp1 * wi[5-1]; ++ } ++ ++ DCT12_PART2 ++ ++ out2[5-0] = in2 * wi[11-0]; ++ out2[0+0] = in2 * wi[6+0]; ++ out2[0+2] = in3 * wi[6+2]; ++ out2[5-2] = in3 * wi[11-2]; ++ ++ ts[(12+0)*SBLIMIT] += in0 * wi[0]; ++ ts[(17-0)*SBLIMIT] += in0 * wi[5-0]; ++ ts[(12+2)*SBLIMIT] += in4 * wi[2]; ++ ts[(17-2)*SBLIMIT] += in4 * wi[5-2]; ++ } ++ ++ in++; ++ ++ { ++ real in0,in1,in2,in3,in4,in5; ++ register real *out2 = rawout2; ++ out2[12]=out2[13]=out2[14]=out2[15]=out2[16]=out2[17]=0.0; ++ ++ DCT12_PART1 ++ ++ { ++ real tmp0,tmp1 = (in0 - in4); ++ { ++ real tmp2 = (in1 - in5) * tfcos12[1]; ++ tmp0 = tmp1 + tmp2; ++ tmp1 -= tmp2; ++ } ++ out2[11-1] = tmp0 * wi[11-1]; ++ out2[6 +1] = tmp0 * wi[6+1]; ++ out2[0+1] += tmp1 * wi[1]; ++ out2[5-1] += tmp1 * wi[5-1]; ++ } ++ ++ DCT12_PART2 ++ ++ out2[11-0] = in2 * wi[11-0]; ++ out2[6 +0] = in2 * wi[6+0]; ++ out2[6 +2] = in3 * wi[6+2]; ++ out2[11-2] = in3 * wi[11-2]; ++ ++ out2[0+0] += in0 * wi[0]; ++ out2[5-0] += in0 * wi[5-0]; ++ out2[0+2] += in4 * wi[2]; ++ out2[5-2] += in4 * wi[5-2]; ++ } ++ } ++ ++ /* ++ * III_hybrid ++ */ ++ static void III_hybrid(real fsIn[SBLIMIT][SSLIMIT],real tsOut[SSLIMIT][SBLIMIT], ++ int ch,struct gr_info_s *gr_info) ++ { ++ real *tspnt = (real *) tsOut; ++ real (*block)[2][SBLIMIT*SSLIMIT] = gmp->hybrid_block; ++ int *blc = gmp->hybrid_blc; ++ real *rawout1,*rawout2; ++ int bt; ++ int sb = 0; ++ ++ { ++ int b = blc[ch]; ++ rawout1=block[b][ch]; ++ b=-b+1; ++ rawout2=block[b][ch]; ++ blc[ch] = b; ++ } ++ ++ ++ if(gr_info->mixed_block_flag) { ++ sb = 2; ++ dct36(fsIn[0],rawout1,rawout2,win[0],tspnt); ++ dct36(fsIn[1],rawout1+18,rawout2+18,win1[0],tspnt+1); ++ rawout1 += 36; rawout2 += 36; tspnt += 2; ++ } ++ ++ bt = gr_info->block_type; ++ if(bt == 2) { ++ for (; sb<gr_info->maxb; sb+=2,tspnt+=2,rawout1+=36,rawout2+=36) { ++ dct12(fsIn[sb],rawout1,rawout2,win[2],tspnt); ++ dct12(fsIn[sb+1],rawout1+18,rawout2+18,win1[2],tspnt+1); ++ } ++ } ++ else { ++ for (; sb<gr_info->maxb; sb+=2,tspnt+=2,rawout1+=36,rawout2+=36) { ++ dct36(fsIn[sb],rawout1,rawout2,win[bt],tspnt); ++ dct36(fsIn[sb+1],rawout1+18,rawout2+18,win1[bt],tspnt+1); ++ } ++ } ++ ++ for(;sb<SBLIMIT;sb++,tspnt++) { ++ int i; ++ for(i=0;i<SSLIMIT;i++) { ++ tspnt[i*SBLIMIT] = *rawout1++; ++ *rawout2++ = 0.0; ++ } ++ } ++ } ++ ++ /* ++ * main layer3 handler ++ */ ++ int do_layer3(struct frame *fr,unsigned char *pcm_sample,int *pcm_point) ++ { ++ int gr, ch, ss,clip=0; ++ int scalefacs[2][39]; /* max 39 for short[13][3] mode, mixed: 38, long: 22 */ ++ struct III_sideinfo sideinfo; ++ int stereo = fr->stereo; ++ int single = fr->single; ++ int ms_stereo,i_stereo; ++ int sfreq = fr->sampling_frequency; ++ int stereo1,granules; ++ ++ ++ ++ if(stereo == 1) { /* stream is mono */ ++ stereo1 = 1; ++ single = 0; ++ } ++ else if(single >= 0) /* stream is stereo, but force to mono */ ++ stereo1 = 1; ++ else ++ stereo1 = 2; ++ ++ if(fr->mode == MPG_MD_JOINT_STEREO) { ++ ms_stereo = fr->mode_ext & 0x2; ++ i_stereo = fr->mode_ext & 0x1; ++ } ++ else ++ ms_stereo = i_stereo = 0; ++ ++ ++ if(fr->lsf) { ++ granules = 1; ++ III_get_side_info_2(&sideinfo,stereo,ms_stereo,sfreq,single); ++ } ++ else { ++ granules = 2; ++ #ifdef MPEG1 ++ III_get_side_info_1(&sideinfo,stereo,ms_stereo,sfreq,single); ++ #else ++ fprintf(stderr,"Not supported\n"); ++ #endif ++ } ++ ++ if(set_pointer(sideinfo.main_data_begin) == MP3_ERR) ++ return 0; ++ ++ ++ for (gr=0;gr<granules;gr++) ++ { ++ static real hybridIn[2][SBLIMIT][SSLIMIT]; ++ static real hybridOut[2][SSLIMIT][SBLIMIT]; ++ ++ { ++ struct gr_info_s *gr_info = &(sideinfo.ch[0].gr[gr]); ++ long part2bits; ++ if(fr->lsf) ++ part2bits = III_get_scale_factors_2(scalefacs[0],gr_info,0); ++ else { ++ #ifdef MPEG1 ++ part2bits = III_get_scale_factors_1(scalefacs[0],gr_info); ++ #else ++ fprintf(stderr,"Not supported\n"); ++ #endif ++ } ++ #ifdef HAVEGTK ++ if (gtkflag) { ++ int i; ++ for (i=0; i<39; i++) ++ pinfo->sfb_s[gr][0][i]=scalefacs[0][i]; ++ } ++ #endif ++ if(III_dequantize_sample(hybridIn[0], scalefacs[0],gr_info,sfreq,part2bits)) ++ return clip; ++ } ++ if(stereo == 2) { ++ struct gr_info_s *gr_info = &(sideinfo.ch[1].gr[gr]); ++ long part2bits; ++ if(fr->lsf) ++ part2bits = III_get_scale_factors_2(scalefacs[1],gr_info,i_stereo); ++ else { ++ #ifdef MPEG1 ++ part2bits = III_get_scale_factors_1(scalefacs[1],gr_info); ++ #else ++ fprintf(stderr,"Not supported\n"); ++ #endif ++ } ++ #ifdef HAVEGTK ++ if (gtkflag) { ++ int i; ++ for (i=0; i<39; i++) ++ pinfo->sfb_s[gr][1][i]=scalefacs[1][i]; ++ } ++ #endif ++ ++ if(III_dequantize_sample(hybridIn[1],scalefacs[1],gr_info,sfreq,part2bits)) ++ return clip; ++ ++ if(ms_stereo) { ++ int i; ++ for(i=0;i<SBLIMIT*SSLIMIT;i++) { ++ real tmp0,tmp1; ++ tmp0 = ((real *) hybridIn[0])[i]; ++ tmp1 = ((real *) hybridIn[1])[i]; ++ ((real *) hybridIn[1])[i] = tmp0 - tmp1; ++ ((real *) hybridIn[0])[i] = tmp0 + tmp1; ++ } ++ } ++ ++ if(i_stereo) ++ III_i_stereo(hybridIn,scalefacs[1],gr_info,sfreq,ms_stereo,fr->lsf); ++ ++ if(ms_stereo || i_stereo || (single == 3) ) { ++ if(gr_info->maxb > sideinfo.ch[0].gr[gr].maxb) ++ sideinfo.ch[0].gr[gr].maxb = gr_info->maxb; ++ else ++ gr_info->maxb = sideinfo.ch[0].gr[gr].maxb; ++ } ++ ++ switch(single) { ++ case 3: ++ { ++ register int i; ++ register real *in0 = (real *) hybridIn[0],*in1 = (real *) hybridIn[1]; ++ for(i=0;i<SSLIMIT*gr_info->maxb;i++,in0++) ++ *in0 = (*in0 + *in1++); /* *0.5 done by pow-scale */ ++ } ++ break; ++ case 1: ++ { ++ register int i; ++ register real *in0 = (real *) hybridIn[0],*in1 = (real *) hybridIn[1]; ++ for(i=0;i<SSLIMIT*gr_info->maxb;i++) ++ *in0++ = *in1++; ++ } ++ break; ++ } ++ } ++ ++ #ifdef HAVEGTK ++ if (gtkflag) { ++ extern int tabsel_123[2][3][16]; ++ extern int pretab[21]; ++ int i,j,sb; ++ float ifqstep; ++ ++ for (ch=0;ch<stereo1;ch++) { ++ struct gr_info_s *gr_info = &(sideinfo.ch[ch].gr[gr]); ++ ifqstep = ( pinfo->scalefac_scale[gr][ch] == 0 ) ? .5 : 1.0; ++ if (2==gr_info->block_type) { ++ for (i=0; i<3; i++) { ++ for (sb=0; sb<12; sb++) { ++ j = 3*sb+i; ++ /* ++ is_p = scalefac[sfb*3+lwin-gr_info->mixed_block_flag]; ++ */ ++ /* scalefac was copied into pinfo->sfb_s[] above */ ++ pinfo->sfb_s[gr][ch][j] = -ifqstep*pinfo->sfb_s[gr][ch][j-gr_info->mixed_block_flag]; ++ pinfo->sfb_s[gr][ch][j] -= 2*(pinfo->sub_gain[gr][ch][i]); ++ } ++ pinfo->sfb_s[gr][ch][3*sb+i] = - 2*(pinfo->sub_gain[gr][ch][i]); ++ } ++ }else{ ++ for (sb=0; sb<21; sb++) { ++ /* scalefac was copied into pinfo->sfb[] above */ ++ pinfo->sfb[gr][ch][sb] = pinfo->sfb_s[gr][ch][sb]; ++ if (gr_info->preflag) pinfo->sfb[gr][ch][sb] += pretab[sb]; ++ pinfo->sfb[gr][ch][sb] *= -ifqstep; ++ } ++ } ++ } ++ ++ ++ ++ pinfo->bitrate = ++ tabsel_123[fr->lsf][fr->lay-1][fr->bitrate_index]; ++ pinfo->sampfreq = freqs[sfreq]; ++ pinfo->emph = fr->emphasis; ++ pinfo->crc = fr->error_protection; ++ pinfo->padding = fr->padding; ++ pinfo->stereo = fr->stereo; ++ pinfo->js = (fr->mode == MPG_MD_JOINT_STEREO); ++ pinfo->ms_stereo = ms_stereo; ++ pinfo->i_stereo = i_stereo; ++ pinfo->maindata = sideinfo.main_data_begin; ++ ++ for(ch=0;ch<stereo1;ch++) { ++ struct gr_info_s *gr_info = &(sideinfo.ch[ch].gr[gr]); ++ pinfo->mixed[gr][ch] = gr_info->mixed_block_flag; ++ pinfo->mpg123blocktype[gr][ch]=gr_info->block_type; ++ pinfo->mainbits[gr][ch] = gr_info->part2_3_length; ++ if (gr==1) pinfo->scfsi[ch] = gr_info->scfsi; ++ } ++ for(ch=0;ch<stereo1;ch++) { ++ int j=0; ++ for (sb=0;sb<SBLIMIT;sb++) ++ for(ss=0;ss<SSLIMIT;ss++,j++) ++ pinfo->mpg123xr[gr][ch][j]=hybridIn[ch][sb][ss]; ++ } ++ } ++ ++ #endif ++ ++ for(ch=0;ch<stereo1;ch++) { ++ struct gr_info_s *gr_info = &(sideinfo.ch[ch].gr[gr]); ++ III_antialias(hybridIn[ch],gr_info); ++ III_hybrid(hybridIn[ch], hybridOut[ch], ch,gr_info); ++ } ++ ++ for(ss=0;ss<SSLIMIT;ss++) { ++ if(single >= 0) { ++ clip += synth_1to1_mono(hybridOut[0][ss],pcm_sample,pcm_point); ++ } ++ else { ++ int p1 = *pcm_point; ++ clip += synth_1to1(hybridOut[0][ss],0,pcm_sample,&p1); ++ clip += synth_1to1(hybridOut[1][ss],1,pcm_sample,pcm_point); ++ } ++ } ++ } ++ ++ return clip; ++ } ++ ++ ++ ++ #endif +diff -r -c -N encoder/mpglib/main.c lame3.70/mpglib/main.c +*** encoder/mpglib/main.c Wed Dec 31 17:00:00 1969 +--- lame3.70/mpglib/main.c Thu Mar 16 17:05:38 2000 +*************** +*** 0 **** +--- 1,203 ---- ++ #ifdef HAVEMPGLIB ++ ++ #include "mpg123.h" ++ #include "mpglib.h" ++ ++ #ifdef OS_AMIGAOS ++ #include "/lame.h" ++ #include "/util.h" ++ #include "/VbrTag.h" ++ #else ++ #include "../lame.h" ++ #include "../util.h" ++ #include "../VbrTag.h" ++ #endif /* OS_AMIGAOS */ ++ ++ #include <stdlib.h> ++ ++ static char buf[16384]; ++ #define FSIZE 8192 ++ static char out[FSIZE]; ++ struct mpstr mp; ++ ++ ++ int is_syncword(char *header) ++ { ++ ++ /* ++ unsigned int s0,s1; ++ s0 = (unsigned char) header[0]; ++ s1 = (unsigned char) header[1] ; ++ printf(" syncword: %2X %2X \n ",s0, s1); ++ */ ++ ++ /* ++ printf(" integer %i \n",(int) ( header[0] == (char) 0xFF)); ++ printf(" integer %i \n",(int) ( (header[1] & (char) 0xF0) == (char) 0xF0)); ++ */ ++ ++ return ++ ((int) ( header[0] == (char) 0xFF)) && ++ ((int) ( (header[1] & (char) 0xF0) == (char) 0xF0)); ++ ++ ++ } ++ ++ ++ int lame_decode_initfile(FILE *fd, int *stereo, int *samp, int *bitrate, ++ unsigned long *num_samples) ++ { ++ extern int tabsel_123[2][3][16]; ++ VBRTAGDATA pTagData; ++ int ret,size,framesize; ++ unsigned long num_frames=0; ++ size_t len; ++ int xing_header; ++ ++ ++ InitMP3(&mp); ++ memset(buf, 0, sizeof(buf)); ++ ++ /* skip RIFF type proprietary headers */ ++ /* look for sync word FFF */ ++ while (!is_syncword(buf)) { ++ buf[0]=buf[1]; ++ if (fread(&buf[1],1,1,fd) == 0) return -1; /* failed */ ++ } ++ /* ret = decodeMP3(&mp,buf,2,out,FSIZE,&size); */ ++ ++ /* read the header */ ++ len = fread(&buf[2],1,46,fd); ++ if (len ==0 ) return -1; ++ len +=2; ++ ++ /* check for Xing header */ ++ xing_header = GetVbrTag(&pTagData,(unsigned char*)buf); ++ if (xing_header) { ++ num_frames=pTagData.frames; ++ } ++ ++ size=0; ++ ret = decodeMP3(&mp,buf,len,out,FSIZE,&size); ++ if (size>0 && !xing_header) { ++ fprintf(stderr,"Opps: first frame of mpglib output will be lost \n"); ++ } ++ ++ *stereo = mp.fr.stereo; ++ *samp = freqs[mp.fr.sampling_frequency]; ++ *bitrate = tabsel_123[mp.fr.lsf][mp.fr.lay-1][mp.fr.bitrate_index]; ++ framesize = (mp.fr.lsf == 0) ? 1152 : 576; ++ *num_samples=MAX_U_32_NUM; ++ if (xing_header && num_frames) { ++ *num_samples=framesize * num_frames; ++ } ++ ++ /* ++ printf("ret = %i NEED_MORE=%i \n",ret,MP3_NEED_MORE); ++ printf("stereo = %i \n",mp.fr.stereo); ++ printf("samp = %i \n",(int)freqs[mp.fr.sampling_frequency]); ++ printf("framesize = %i \n",framesize); ++ printf("num frames = %i \n",(int)num_frames); ++ printf("num samp = %i \n",(int)*num_samples); ++ */ ++ return 0; ++ } ++ ++ ++ int lame_decode_init(void) ++ { ++ InitMP3(&mp); ++ memset(buf, 0, sizeof(buf)); ++ return 0; ++ } ++ ++ ++ /* ++ For lame_decode_fromfile: return code ++ -1 error ++ 0 ok, but need more data before outputing any samples ++ n number of samples output. either 576 or 1152 depending on MP3 file. ++ */ ++ int lame_decode_fromfile(FILE *fd, short pcm_l[], short pcm_r[]) ++ { ++ int size,stereo; ++ int outsize=0,j,i,ret; ++ size_t len; ++ ++ size=0; ++ len = fread(buf,1,64,fd); ++ if (len ==0 ) return 0; ++ ret = decodeMP3(&mp,buf,len,out,FSIZE,&size); ++ ++ /* read more until we get a valid output frame */ ++ while((ret == MP3_NEED_MORE) || !size) { ++ len = fread(buf,1,100,fd); ++ if (len ==0 ) return -1; ++ ret = decodeMP3(&mp,buf,len,out,FSIZE,&size); ++ /* if (ret ==MP3_ERR) return -1; lets ignore errors and keep reading... */ ++ /* ++ printf("ret = %i size= %i %i %i %i \n",ret,size, ++ MP3_NEED_MORE,MP3_ERR,MP3_OK); ++ */ ++ } ++ ++ stereo=mp.fr.stereo; ++ ++ if (ret == MP3_OK) ++ { ++ /* write(1,out,size); */ ++ outsize = size/(2*(stereo)); ++ if ((outsize!=576) && (outsize!=1152)) { ++ fprintf(stderr,"Opps: mpg123 returned more than one frame! Cant handle this... \n"); ++ exit(-50); ++ } ++ ++ for (j=0; j<stereo; j++) ++ for (i=0; i<outsize; i++) ++ if (j==0) pcm_l[i] = ((short *) out)[mp.fr.stereo*i+j]; ++ else pcm_r[i] = ((short *) out)[mp.fr.stereo*i+j]; ++ ++ } ++ if (ret==MP3_ERR) return -1; ++ else return outsize; ++ } ++ ++ ++ ++ ++ /* ++ For lame_decode: return code ++ -1 error ++ 0 ok, but need more data before outputing any samples ++ n number of samples output. either 576 or 1152 depending on MP3 file. ++ */ ++ int lame_decode(char *buf,int len,short pcm_l[],short pcm_r[]) ++ { ++ int size; ++ int outsize=0,j,i,ret; ++ ++ ret = decodeMP3(&mp,buf,len,out,FSIZE,&size); ++ if (ret==MP3_OK) { ++ /* printf("mpg123 output one frame out=%i \n",size/4); */ ++ outsize = size/(2*mp.fr.stereo); ++ if (outsize > 1152) { ++ fprintf(stderr,"Opps: mpg123 returned more than one frame! shouldn't happen... \n"); ++ exit(-50); ++ } ++ ++ for (j=0; j<mp.fr.stereo; j++) ++ for (i=0; i<outsize; i++) ++ if (j==0) pcm_l[i] = ((short *) out)[mp.fr.stereo*i+j]; ++ else pcm_r[i] = ((short *) out)[mp.fr.stereo*i+j]; ++ ++ } ++ /* ++ printf("ok, more, err: %i %i %i \n",MP3_OK, MP3_NEED_MORE, MP3_ERR); ++ printf("ret = %i out=%i \n",ret,outsize); ++ */ ++ if (ret==MP3_ERR) return -1; ++ else return outsize; ++ } ++ ++ #endif /* HAVEMPGLIB */ ++ +diff -r -c -N encoder/mpglib/mpg123.h lame3.70/mpglib/mpg123.h +*** encoder/mpglib/mpg123.h Wed Dec 31 17:00:00 1969 +--- lame3.70/mpglib/mpg123.h Sat Feb 19 06:33:12 2000 +*************** +*** 0 **** +--- 1,174 ---- ++ #include <stdio.h> ++ #include <string.h> ++ #include <signal.h> ++ ++ ++ #include <math.h> ++ ++ #ifndef M_PI ++ #define M_PI 3.14159265358979323846 ++ #endif ++ #ifndef M_SQRT2 ++ #define M_SQRT2 1.41421356237309504880 ++ #endif ++ ++ #ifdef _MSC_VER ++ #pragma warning( disable : 4244 ) ++ #pragma warning( disable : 4305 ) ++ # define REAL_IS_FLOAT ++ # define NEW_DCT9 ++ ++ # define random rand ++ # define srandom srand ++ ++ #endif ++ ++ #ifdef REAL_IS_FLOAT ++ # define real float ++ #elif defined(REAL_IS_LONG_DOUBLE) ++ # define real long double ++ #else ++ # define real double ++ #endif ++ ++ /* AUDIOBUFSIZE = n*64 with n=1,2,3 ... */ ++ #define AUDIOBUFSIZE 16384 ++ ++ #define FALSE 0 ++ #define TRUE 1 ++ ++ #define SBLIMIT 32 ++ #define SSLIMIT 18 ++ ++ #define MPG_MD_STEREO 0 ++ #define MPG_MD_JOINT_STEREO 1 ++ #define MPG_MD_DUAL_CHANNEL 2 ++ #define MPG_MD_MONO 3 ++ ++ #define MAXFRAMESIZE 1792 ++ ++ ++ /* Pre Shift fo 16 to 8 bit converter table */ ++ #define AUSHIFT (3) ++ ++ struct frame { ++ int stereo; ++ int jsbound; ++ int single; ++ int lsf; ++ int mpeg25; ++ int header_change; ++ int lay; ++ int error_protection; ++ int bitrate_index; ++ int sampling_frequency; ++ int padding; ++ int extension; ++ int mode; ++ int mode_ext; ++ int copyright; ++ int original; ++ int emphasis; ++ int framesize; /* computed framesize */ ++ }; ++ ++ struct parameter { ++ int quiet; /* shut up! */ ++ int tryresync; /* resync stream after error */ ++ int verbose; /* verbose level */ ++ int checkrange; ++ }; ++ ++ /* extern unsigned int get1bit(void); */ ++ extern unsigned int getbits(int); ++ extern unsigned int getbits_fast(int); ++ extern int set_pointer(long); ++ ++ extern unsigned char *wordpointer; ++ extern int bitindex; ++ ++ extern void make_decode_tables(long scaleval); ++ extern int do_layer3(struct frame *fr,unsigned char *,int *); ++ extern int decode_header(struct frame *fr,unsigned long newhead); ++ ++ ++ ++ struct gr_info_s { ++ int scfsi; ++ unsigned part2_3_length; ++ unsigned big_values; ++ unsigned scalefac_compress; ++ unsigned block_type; ++ unsigned mixed_block_flag; ++ unsigned table_select[3]; ++ unsigned subblock_gain[3]; ++ unsigned maxband[3]; ++ unsigned maxbandl; ++ unsigned maxb; ++ unsigned region1start; ++ unsigned region2start; ++ unsigned preflag; ++ unsigned scalefac_scale; ++ unsigned count1table_select; ++ real *full_gain[3]; ++ real *pow2gain; ++ }; ++ ++ struct III_sideinfo ++ { ++ unsigned main_data_begin; ++ unsigned private_bits; ++ struct { ++ struct gr_info_s gr[2]; ++ } ch[2]; ++ }; ++ ++ extern int synth_1to1 (real *,int,unsigned char *,int *); ++ extern int synth_1to1_8bit (real *,int,unsigned char *,int *); ++ extern int synth_1to1_mono (real *,unsigned char *,int *); ++ extern int synth_1to1_mono2stereo (real *,unsigned char *,int *); ++ extern int synth_1to1_8bit_mono (real *,unsigned char *,int *); ++ extern int synth_1to1_8bit_mono2stereo (real *,unsigned char *,int *); ++ ++ extern int synth_2to1 (real *,int,unsigned char *,int *); ++ extern int synth_2to1_8bit (real *,int,unsigned char *,int *); ++ extern int synth_2to1_mono (real *,unsigned char *,int *); ++ extern int synth_2to1_mono2stereo (real *,unsigned char *,int *); ++ extern int synth_2to1_8bit_mono (real *,unsigned char *,int *); ++ extern int synth_2to1_8bit_mono2stereo (real *,unsigned char *,int *); ++ ++ extern int synth_4to1 (real *,int,unsigned char *,int *); ++ extern int synth_4to1_8bit (real *,int,unsigned char *,int *); ++ extern int synth_4to1_mono (real *,unsigned char *,int *); ++ extern int synth_4to1_mono2stereo (real *,unsigned char *,int *); ++ extern int synth_4to1_8bit_mono (real *,unsigned char *,int *); ++ extern int synth_4to1_8bit_mono2stereo (real *,unsigned char *,int *); ++ ++ extern int synth_ntom (real *,int,unsigned char *,int *); ++ extern int synth_ntom_8bit (real *,int,unsigned char *,int *); ++ extern int synth_ntom_mono (real *,unsigned char *,int *); ++ extern int synth_ntom_mono2stereo (real *,unsigned char *,int *); ++ extern int synth_ntom_8bit_mono (real *,unsigned char *,int *); ++ extern int synth_ntom_8bit_mono2stereo (real *,unsigned char *,int *); ++ ++ extern void rewindNbits(int bits); ++ extern int hsstell(void); ++ extern int get_songlen(struct frame *fr,int no); ++ ++ extern void init_layer3(int); ++ extern void init_layer2(void); ++ extern void make_decode_tables(long scale); ++ extern void make_conv16to8_table(int); ++ extern void dct64(real *,real *,real *); ++ ++ extern void synth_ntom_set_step(long,long); ++ ++ extern unsigned char *conv16to8; ++ extern long freqs[9]; ++ extern real muls[27][64]; ++ extern real decwin[512+32]; ++ extern real *pnts[5]; ++ ++ extern struct parameter param; ++ ++ +diff -r -c -N encoder/mpglib/mpglib.dsp lame3.70/mpglib/mpglib.dsp +*** encoder/mpglib/mpglib.dsp Wed Dec 31 17:00:00 1969 +--- lame3.70/mpglib/mpglib.dsp Thu Mar 16 17:05:38 2000 +*************** +*** 0 **** +--- 1,164 ---- ++ # Microsoft Developer Studio Project File - Name="mpglib" - Package Owner=<4> ++ # Microsoft Developer Studio Generated Build File, Format Version 5.00 ++ # ** DO NOT EDIT ** ++ ++ # TARGTYPE "Win32 (x86) Static Library" 0x0104 ++ ++ CFG=mpglib - Win32 Debug GTK ++ !MESSAGE This is not a valid makefile. To build this project using NMAKE, ++ !MESSAGE use the Export Makefile command and run ++ !MESSAGE ++ !MESSAGE NMAKE /f "mpglib.mak". ++ !MESSAGE ++ !MESSAGE You can specify a configuration when running NMAKE ++ !MESSAGE by defining the macro CFG on the command line. For example: ++ !MESSAGE ++ !MESSAGE NMAKE /f "mpglib.mak" CFG="mpglib - Win32 Debug GTK" ++ !MESSAGE ++ !MESSAGE Possible choices for configuration are: ++ !MESSAGE ++ !MESSAGE "mpglib - Win32 Release" (based on "Win32 (x86) Static Library") ++ !MESSAGE "mpglib - Win32 Debug" (based on "Win32 (x86) Static Library") ++ !MESSAGE "mpglib - Win32 Debug GTK" (based on "Win32 (x86) Static Library") ++ !MESSAGE "mpglib - Win32 Release GTK" (based on "Win32 (x86) Static Library") ++ !MESSAGE ++ ++ # Begin Project ++ # PROP Scc_ProjName "" ++ # PROP Scc_LocalPath "" ++ CPP=xicl.exe ++ ++ !IF "$(CFG)" == "mpglib - Win32 Release" ++ ++ # PROP BASE Use_MFC 0 ++ # PROP BASE Use_Debug_Libraries 0 ++ # PROP BASE Output_Dir "Release" ++ # PROP BASE Intermediate_Dir "Release" ++ # PROP BASE Target_Dir "" ++ # PROP Use_MFC 0 ++ # PROP Use_Debug_Libraries 0 ++ # PROP Output_Dir "Release" ++ # PROP Intermediate_Dir "Release" ++ # PROP Target_Dir "" ++ # ADD BASE CPP /nologo /W3 /GX /O2 /D "WIN32" /D "NDEBUG" /D "_WINDOWS" /YX /FD /c ++ # ADD CPP /nologo /W3 /GX /Ox /Ot /Og /Oy /D "WIN32" /D "NDEBUG" /D "_WINDOWS" /D "HAVEMPGLIB" /YX /FD /c ++ BSC32=bscmake.exe ++ # ADD BASE BSC32 /nologo ++ # ADD BSC32 /nologo ++ LIB32=link.exe -lib ++ # ADD BASE LIB32 /nologo ++ # ADD LIB32 /nologo ++ ++ !ELSEIF "$(CFG)" == "mpglib - Win32 Debug" ++ ++ # PROP BASE Use_MFC 0 ++ # PROP BASE Use_Debug_Libraries 1 ++ # PROP BASE Output_Dir "Debug" ++ # PROP BASE Intermediate_Dir "Debug" ++ # PROP BASE Target_Dir "" ++ # PROP Use_MFC 0 ++ # PROP Use_Debug_Libraries 1 ++ # PROP Output_Dir "Debug" ++ # PROP Intermediate_Dir "Debug" ++ # PROP Target_Dir "" ++ # ADD BASE CPP /nologo /W3 /GX /Z7 /Od /D "WIN32" /D "_DEBUG" /D "_WINDOWS" /YX /FD /c ++ # ADD CPP /nologo /W3 /GX /Z7 /Od /D "WIN32" /D "_DEBUG" /D "_WINDOWS" /YX /FD /c ++ BSC32=bscmake.exe ++ # ADD BASE BSC32 /nologo ++ # ADD BSC32 /nologo ++ LIB32=link.exe -lib ++ # ADD BASE LIB32 /nologo ++ # ADD LIB32 /nologo ++ ++ !ELSEIF "$(CFG)" == "mpglib - Win32 Debug GTK" ++ ++ # PROP BASE Use_MFC 0 ++ # PROP BASE Use_Debug_Libraries 1 ++ # PROP BASE Output_Dir "mpglib__" ++ # PROP BASE Intermediate_Dir "mpglib__" ++ # PROP BASE Target_Dir "" ++ # PROP Use_MFC 0 ++ # PROP Use_Debug_Libraries 1 ++ # PROP Output_Dir "DebugGTK" ++ # PROP Intermediate_Dir "DebugGTK" ++ # PROP Target_Dir "" ++ # ADD BASE CPP /nologo /W3 /GX /Z7 /Od /D "WIN32" /D "_DEBUG" /D "_WINDOWS" /YX /FD /c ++ # ADD CPP /nologo /W3 /GX /Z7 /Od /I "../WinGTK/glib-1.2" /I "../WinGTK/gtk-plus" /D "WIN32" /D "_DEBUG" /D "_WINDOWS" /D "HAVEGTK" /YX /FD /c ++ BSC32=bscmake.exe ++ # ADD BASE BSC32 /nologo ++ # ADD BSC32 /nologo ++ LIB32=link.exe -lib ++ # ADD BASE LIB32 /nologo ++ # ADD LIB32 /nologo ++ ++ !ELSEIF "$(CFG)" == "mpglib - Win32 Release GTK" ++ ++ # PROP BASE Use_MFC 0 ++ # PROP BASE Use_Debug_Libraries 0 ++ # PROP BASE Output_Dir "mpglib_0" ++ # PROP BASE Intermediate_Dir "mpglib_0" ++ # PROP BASE Target_Dir "" ++ # PROP Use_MFC 0 ++ # PROP Use_Debug_Libraries 0 ++ # PROP Output_Dir "ReleaseGTK" ++ # PROP Intermediate_Dir "ReleaseGTK" ++ # PROP Target_Dir "" ++ # ADD BASE CPP /nologo /W3 /GX /O2 /D "WIN32" /D "NDEBUG" /D "_WINDOWS" /YX /FD /c ++ # ADD CPP /nologo /W3 /GX /O2 /I "../WinGTK/glib-1.2" /I "../WinGTK/gtk-plus" /D "WIN32" /D "NDEBUG" /D "_WINDOWS" /D "HAVEGTK" /D "HAVEMPGLIB" /YX /FD /c ++ BSC32=bscmake.exe ++ # ADD BASE BSC32 /nologo ++ # ADD BSC32 /nologo ++ LIB32=link.exe -lib ++ # ADD BASE LIB32 /nologo ++ # ADD LIB32 /nologo ++ ++ !ENDIF ++ ++ # Begin Target ++ ++ # Name "mpglib - Win32 Release" ++ # Name "mpglib - Win32 Debug" ++ # Name "mpglib - Win32 Debug GTK" ++ # Name "mpglib - Win32 Release GTK" ++ # Begin Source File ++ ++ SOURCE=.\common.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\dct64_i386.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\decode_i386.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\huffman.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\interface.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\layer3.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\main.c ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\mpg123.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\mpglib.h ++ # End Source File ++ # Begin Source File ++ ++ SOURCE=.\tabinit.c ++ # End Source File ++ # End Target ++ # End Project +diff -r -c -N encoder/mpglib/mpglib.h lame3.70/mpglib/mpglib.h +*** encoder/mpglib/mpglib.h Wed Dec 31 17:00:00 1969 +--- lame3.70/mpglib/mpglib.h Wed Jan 5 10:40:35 2000 +*************** +*** 0 **** +--- 1,49 ---- ++ ++ struct buf { ++ unsigned char *pnt; ++ long size; ++ long pos; ++ struct buf *next; ++ struct buf *prev; ++ }; ++ ++ struct framebuf { ++ struct buf *buf; ++ long pos; ++ struct frame *next; ++ struct frame *prev; ++ }; ++ ++ struct mpstr { ++ struct buf *head,*tail; ++ int bsize; ++ int framesize; ++ int fsizeold; ++ struct frame fr; ++ unsigned char bsspace[2][MAXFRAMESIZE+512]; /* MAXFRAMESIZE */ ++ real hybrid_block[2][2][SBLIMIT*SSLIMIT]; ++ int hybrid_blc[2]; ++ unsigned long header; ++ int bsnum; ++ real synth_buffs[2][2][0x110]; ++ int synth_bo; ++ ++ }; ++ ++ ++ #if ( defined(_MSC_VER) || defined(__BORLANDC__) ) ++ typedef int BOOL; /* windef.h contains the same definition */ ++ #else ++ #define BOOL int ++ #endif ++ ++ #define MP3_ERR -1 ++ #define MP3_OK 0 ++ #define MP3_NEED_MORE 1 ++ ++ ++ BOOL InitMP3(struct mpstr *mp); ++ int decodeMP3(struct mpstr *mp,char *inmemory,int inmemsize, ++ char *outmemory,int outmemsize,int *done); ++ void ExitMP3(struct mpstr *mp); ++ +diff -r -c -N encoder/mpglib/tabinit.c lame3.70/mpglib/tabinit.c +*** encoder/mpglib/tabinit.c Wed Dec 31 17:00:00 1969 +--- lame3.70/mpglib/tabinit.c Mon Mar 6 12:53:03 2000 +*************** +*** 0 **** +--- 1,82 ---- ++ #ifdef HAVEMPGLIB ++ #include <stdlib.h> ++ ++ #include "mpg123.h" ++ ++ real decwin[512+32]; ++ static real cos64[16],cos32[8],cos16[4],cos8[2],cos4[1]; ++ real *pnts[] = { cos64,cos32,cos16,cos8,cos4 }; ++ ++ #if 0 ++ static unsigned char *conv16to8_buf = NULL; ++ unsigned char *conv16to8; ++ #endif ++ ++ static long intwinbase[] = { ++ 0, -1, -1, -1, -1, -1, -1, -2, -2, -2, ++ -2, -3, -3, -4, -4, -5, -5, -6, -7, -7, ++ -8, -9, -10, -11, -13, -14, -16, -17, -19, -21, ++ -24, -26, -29, -31, -35, -38, -41, -45, -49, -53, ++ -58, -63, -68, -73, -79, -85, -91, -97, -104, -111, ++ -117, -125, -132, -139, -147, -154, -161, -169, -176, -183, ++ -190, -196, -202, -208, -213, -218, -222, -225, -227, -228, ++ -228, -227, -224, -221, -215, -208, -200, -189, -177, -163, ++ -146, -127, -106, -83, -57, -29, 2, 36, 72, 111, ++ 153, 197, 244, 294, 347, 401, 459, 519, 581, 645, ++ 711, 779, 848, 919, 991, 1064, 1137, 1210, 1283, 1356, ++ 1428, 1498, 1567, 1634, 1698, 1759, 1817, 1870, 1919, 1962, ++ 2001, 2032, 2057, 2075, 2085, 2087, 2080, 2063, 2037, 2000, ++ 1952, 1893, 1822, 1739, 1644, 1535, 1414, 1280, 1131, 970, ++ 794, 605, 402, 185, -45, -288, -545, -814, -1095, -1388, ++ -1692, -2006, -2330, -2663, -3004, -3351, -3705, -4063, -4425, -4788, ++ -5153, -5517, -5879, -6237, -6589, -6935, -7271, -7597, -7910, -8209, ++ -8491, -8755, -8998, -9219, -9416, -9585, -9727, -9838, -9916, -9959, ++ -9966, -9935, -9863, -9750, -9592, -9389, -9139, -8840, -8492, -8092, ++ -7640, -7134, -6574, -5959, -5288, -4561, -3776, -2935, -2037, -1082, ++ -70, 998, 2122, 3300, 4533, 5818, 7154, 8540, 9975, 11455, ++ 12980, 14548, 16155, 17799, 19478, 21189, 22929, 24694, 26482, 28289, ++ 30112, 31947, 33791, 35640, 37489, 39336, 41176, 43006, 44821, 46617, ++ 48390, 50137, 51853, 53534, 55178, 56778, 58333, 59838, 61289, 62684, ++ 64019, 65290, 66494, 67629, 68692, 69679, 70590, 71420, 72169, 72835, ++ 73415, 73908, 74313, 74630, 74856, 74992, 75038 }; ++ ++ void make_decode_tables(long scaleval) ++ { ++ int i,j,k,kr,divv; ++ real *table,*costab; ++ ++ ++ for(i=0;i<5;i++) ++ { ++ kr=0x10>>i; divv=0x40>>i; ++ costab = pnts[i]; ++ for(k=0;k<kr;k++) ++ costab[k] = 1.0 / (2.0 * cos(M_PI * ((double) k * 2.0 + 1.0) / (double) divv)); ++ } ++ ++ table = decwin; ++ scaleval = -scaleval; ++ for(i=0,j=0;i<256;i++,j++,table+=32) ++ { ++ if(table < decwin+512+16) ++ table[16] = table[0] = (double) intwinbase[j] / 65536.0 * (double) scaleval; ++ if(i % 32 == 31) ++ table -= 1023; ++ if(i % 64 == 63) ++ scaleval = - scaleval; ++ } ++ ++ for( /* i=256 */ ;i<512;i++,j--,table+=32) ++ { ++ if(table < decwin+512+16) ++ table[16] = table[0] = (double) intwinbase[j] / 65536.0 * (double) scaleval; ++ if(i % 32 == 31) ++ table -= 1023; ++ if(i % 64 == 63) ++ scaleval = - scaleval; ++ } ++ } ++ ++ ++ ++ #endif +diff -r -c -N encoder/musicin.c lame3.70/musicin.c +*** encoder/musicin.c Wed Jan 22 02:43:17 1997 +--- lame3.70/musicin.c Wed Dec 31 17:00:00 1969 +*************** +*** 1,1151 **** +- /********************************************************************** +- * ISO MPEG Audio Subgroup Software Simulation Group (1996) +- * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- * +- * $Id: musicin.c,v 1.2 1997/01/19 22:28:29 rowlands Exp $ +- * +- * $Log: musicin.c,v $ +- * Revision 1.2 1997/01/19 22:28:29 rowlands +- * Layer 3 bug fixes from Seymour Shlien +- * +- * Revision 1.1 1996/02/14 04:04:23 rowlands +- * Initial revision +- * +- * Received from Mike Coleman +- **********************************************************************/ +- /********************************************************************** +- * date programmers comment * +- * 3/01/91 Douglas Wong, start of version 1.1 records * +- * Davis Pan * +- * 3/06/91 Douglas Wong, rename: setup.h to endef.h * +- * removed extraneous variables * +- * 3/21/91 J.Georges Fritsch introduction of the bit-stream * +- * package. This package allows you * +- * to generate the bit-stream in a * +- * binary or ascii format * +- * 3/31/91 Bill Aspromonte replaced the read of the SB matrix * +- * by an "code generated" one * +- * 5/10/91 W. Joseph Carter Ported to Macintosh and Unix. * +- * Incorporated Jean-Georges Fritsch's * +- * "bitstream.c" package. * +- * Modified to strictly adhere to * +- * encoded bitstream specs, including * +- * "Berlin changes". * +- * Modified user interface dialog & code * +- * to accept any input & output * +- * filenames desired. Also added * +- * de-emphasis prompt and final bail-out * +- * opportunity before encoding. * +- * Added AIFF PCM sound file reading * +- * capability. * +- * Modified PCM sound file handling to * +- * process all incoming samples and fill * +- * out last encoded frame with zeros * +- * (silence) if needed. * +- * Located and fixed numerous software * +- * bugs and table data errors. * +- * 27jun91 dpwe (Aware Inc) Used new frame_params struct. * +- * Clear all automatic arrays. * +- * Changed some variable names, * +- * simplified some code. * +- * Track number of bits actually sent. * +- * Fixed padding slot, stereo bitrate * +- * Added joint-stereo : scales L+R. * +- * 6/12/91 Earle Jennings added fix for MS_DOS in obtain_param * +- * 6/13/91 Earle Jennings added stack length adjustment before * +- * main for MS_DOS * +- * 7/10/91 Earle Jennings conversion of all float to FLOAT * +- * port to MsDos from MacIntosh completed* +- * 8/ 8/91 Jens Spille Change for MS-C6.00 * +- * 8/22/91 Jens Spille new obtain_parameters() * +- *10/ 1/91 S.I. Sudharsanan, Ported to IBM AIX platform. * +- * Don H. Lee, * +- * Peter W. Farrett * +- *10/ 3/91 Don H. Lee implemented CRC-16 error protection * +- * newly introduced functions are * +- * I_CRC_calc, II_CRC_calc and encode_CRC* +- * Additions and revisions are marked * +- * with "dhl" for clarity * +- *11/11/91 Katherine Wang Documentation of code. * +- * (variables in documentation are * +- * surround by the # symbol, and an '*'* +- * denotes layer I or II versions) * +- * 2/11/92 W. Joseph Carter Ported new code to Macintosh. Most * +- * important fixes involved changing * +- * 16-bit ints to long or unsigned in * +- * bit alloc routines for quant of 65535 * +- * and passing proper function args. * +- * Removed "Other Joint Stereo" option * +- * and made bitrate be total channel * +- * bitrate, irrespective of the mode. * +- * Fixed many small bugs & reorganized. * +- * 2/25/92 Masahiro Iwadare made code cleaner and more consistent * +- * 8/07/92 Mike Coleman make exit() codes return error status * +- * made slight changes for portability * +- *19 aug 92 Soren H. Nielsen Changed MS-DOS file name extensions. * +- * 8/25/92 Shaun Astarabadi Replaced rint() function with explicit* +- * rounding for portability with MSDOS. * +- * 9/22/92 jddevine@aware.com Fixed _scale_factor_calc() calls. * +- *10/19/92 Masahiro Iwadare added info->mode and info->mode_ext * +- * updates for AIFF format files * +- * 3/10/93 Kevin Peterson In parse_args, only set non default * +- * bit rate if specified in arg list. * +- * Use return value from aiff_read_hdrs * +- * to fseek to start of sound data * +- * 7/26/93 Davis Pan fixed bug in printing info->mode_ext * +- * value for joint stereo condition * +- * 8/27/93 Seymour Shlien, Fixes in Unix and MSDOS ports, * +- * Daniel Lauzon, and * +- * Bill Truerniet * +- * 11/7/95 Soeren H. Nielsen LSF added. Bug fix in MSDOS ext. * +- * 8/02/95 mc@fivebats.com Changed default bitrate selection so * +- * it works with the new LSF stuff * +- *10/01/95 mc@fivebats.com Added layer3 * +- **********************************************************************/ +- +- #ifdef MS_DOS +- #include <dos.h> +- #endif +- #include <stdlib.h> +- #include "common.h" +- #include "encoder.h" +- #include "l3psy.h" +- #include "mdct.h" +- #include "loop.h" +- #include "l3bitstream.h" +- #include <assert.h> +- +- /* Global variable definitions for "musicin.c" */ +- +- FILE *musicin; +- Bit_stream_struc bs; +- char *programName; +- +- /* Implementations */ +- +- /************************************************************************ +- * +- * obtain_parameters +- * +- * PURPOSE: Prompts for and reads user input for encoding parameters +- * +- * SEMANTICS: The parameters read are: +- * - input and output filenames +- * - sampling frequency (if AIFF file, will read from the AIFF file header) +- * - layer number +- * - mode (stereo, joint stereo, dual channel or mono) +- * - psychoacoustic model (I or II) +- * - total bitrate, irrespective of the mode +- * - de-emphasis, error protection, copyright and original or copy flags +- * +- ************************************************************************/ +- +- void +- obtain_parameters(fr_ps,psy,num_samples,original_file_name,encoded_file_name) +- frame_params *fr_ps; +- int *psy; +- unsigned long *num_samples; +- char original_file_name[MAX_NAME_SIZE]; +- char encoded_file_name[MAX_NAME_SIZE]; +- { +- int j; +- long int freq; +- int model, brt; +- char t[50]; +- IFF_AIFF pcm_aiff_data; +- layer *info = fr_ps->header; +- long soundPosition; +- #ifdef MS_DOS +- char temp_str[MAX_NAME_SIZE]; +- #endif +- +- do { +- printf("Enter PCM input file name <required>: "); +- gets(original_file_name); +- if (original_file_name[0] == NULL_CHAR) +- printf("PCM input file name is required.\n"); +- } while (original_file_name[0] == NULL_CHAR); +- printf(">>> PCM input file name is: %s\n", original_file_name); +- +- if ((musicin = fopen(original_file_name, "rb")) == NULL) { +- printf("Could not find \"%s\".\n", original_file_name); +- exit(1); +- } +- +- #ifdef MS_DOS +- /* replace old extension with new one, 1992-08-19, 1995-06-12 shn */ +- new_ext(original_file_name, DFLT_EXT, temp_str); +- printf("Enter MPEG encoded output file name <%s>: ", +- temp_str); +- #else +- printf("Enter MPEG encoded output file name <%s%s>: ", +- original_file_name, DFLT_EXT); +- #endif +- +- gets(encoded_file_name); +- +- if (encoded_file_name[0] == NULL_CHAR) { +- #ifdef MS_DOS +- strcpy(encoded_file_name, temp_str); +- #else +- strcat(strcpy(encoded_file_name, original_file_name), DFLT_EXT); +- #endif +- } +- +- +- printf(">>> MPEG encoded output file name is: %s\n", encoded_file_name); +- +- open_bit_stream_w(&bs, encoded_file_name, BUFFER_SIZE); +- +- if ((soundPosition = aiff_read_headers(musicin, &pcm_aiff_data)) != -1) { +- +- printf(">>> Using Audio IFF sound file headers\n"); +- +- aiff_check(original_file_name, &pcm_aiff_data, &info->version); +- +- if (fseek(musicin, soundPosition, SEEK_SET) != 0) { +- printf("Could not seek to PCM sound data in \"%s\".\n", +- original_file_name); +- exit(1); +- } +- +- info->sampling_frequency = +- SmpFrqIndex((long)pcm_aiff_data.sampleRate, &info->version); +- printf(">>> %f Hz sampling frequency selected\n", +- pcm_aiff_data.sampleRate); +- +- /* Determine number of samples in sound file */ +- #ifndef MS_DOS +- *num_samples = pcm_aiff_data.numChannels * +- pcm_aiff_data.numSampleFrames; +- #else +- *num_samples = (long)(pcm_aiff_data.numChannels) * +- (long)(pcm_aiff_data.numSampleFrames); +- #endif +- +- } +- else { /* Not using Audio IFF sound file headers. */ +- +- printf("What is the sampling frequency? <44100>[Hz]: "); +- gets(t); +- freq = atol(t); +- switch (freq) { +- case 48000 : info->sampling_frequency = 1; +- printf(">>> %ld Hz sampling freq selected\n", freq); +- break; +- case 44100 : info->sampling_frequency = 0; +- printf(">>> %ld Hz sampling freq selected\n", freq); +- break; +- case 32000 : info->sampling_frequency = 2; +- printf(">>> %ld Hz sampling freq selected\n", freq); +- break; +- case 24000 : info->sampling_frequency = 1; +- info->version = MPEG_PHASE2_LSF; +- printf(">>> %ld Hz sampling freq selected\n", freq); +- break; +- case 22050 : info->sampling_frequency = 0; +- info->version = MPEG_PHASE2_LSF; +- printf(">>> %ld Hz sampling freq selected\n", freq); +- break; +- case 16000 : info->sampling_frequency = 2; +- info->version = MPEG_PHASE2_LSF; +- printf(">>> %ld Hz sampling freq selected\n", freq); +- break; +- default: info->sampling_frequency = 0; +- printf(">>> Default 44.1 kHz samp freq selected\n"); +- } +- printf(">>> Encoding algorithm is %s\n", version_names[info->version]); +- if (fseek(musicin, 0, SEEK_SET) != 0) { +- printf("Could not seek to PCM sound data in \"%s\".\n", +- original_file_name); +- exit(1); +- } +- +- /* Declare sound file to have "infinite" number of samples. */ +- *num_samples = MAX_U_32_NUM; +- +- } +- +- printf("Which layer do you want to use?\n"); +- printf("Available: Layer (1), Layer (<2>), Layer (3): "); +- gets(t); +- switch(*t){ +- case '1': info->lay = 1; printf(">>> Using Layer %s\n",t); break; +- case '2': info->lay = 2; printf(">>> Using Layer %s\n",t); break; +- case '3': info->lay = 3; printf(">>> Using Layer %s\n",t); break; +- default: info->lay = 2; printf(">>> Using default Layer 2\n"); break; +- } +- +- printf("Which mode do you want?\n"); +- printf("Available: (<s>)tereo, (j)oint stereo, "); +- printf("(d)ual channel, s(i)ngle Channel: "); +- gets(t); +- switch(*t){ +- case 's': +- case 'S': +- info->mode = MPG_MD_STEREO; info->mode_ext = 0; +- printf(">>> Using mode %s\n",t); +- break; +- case 'j': +- case 'J': +- info->mode = MPG_MD_JOINT_STEREO; +- printf(">>> Using mode %s\n",t); +- break; +- case 'd': +- case 'D': +- info->mode = MPG_MD_DUAL_CHANNEL; info->mode_ext = 0; +- printf(">>> Using mode %s\n",t); +- break; +- case 'i': +- case 'I': +- info->mode = MPG_MD_MONO; info->mode_ext = 0; +- printf(">>> Using mode %s\n",t); +- break; +- default: +- info->mode = MPG_MD_STEREO; info->mode_ext = 0; +- printf(">>> Using default stereo mode\n"); +- break; +- } +- +- printf("Which psychoacoustic model do you want to use? <1>: "); +- gets(t); +- model = atoi(t); +- if (model > 2 || model < 1) { +- printf(">>> Default model 1 selected\n"); +- *psy = 1; +- } +- else { +- *psy = model; +- printf(">>> Using psychoacoustic model %d\n", model); +- } +- +- /* set default bitrate to highest allowed, which is index 14 */ +- brt = bitrate[info->version][info->lay-1][14]; +- printf( "What is the total bitrate? <%u>[kbps]: ", brt ); +- gets( t ); +- brt = atoi( t ); +- if ( brt == 0 ) +- j = 15; +- else +- j = 0; +- while ( j < 15 ) +- { +- if ( bitrate[info->version][info->lay-1][j] == brt ) +- break; +- j++; +- } +- if ( j == 15 ) +- { +- brt = bitrate[info->version][info->lay-1][14]; +- printf( ">>> Using default %u kbps\n", brt ); +- info->bitrate_index = 14; +- } +- else +- { +- info->bitrate_index = j; +- printf( ">>> Bitrate = %d kbps\n", bitrate[info->version][info->lay-1][j] ); +- } +- +- printf("What type of de-emphasis should the decoder use?\n"); +- printf("Available: (<n>)one, (5)0/15 microseconds, (c)citt j.17: "); +- gets(t); +- if (*t != 'n' && *t != '5' && *t != 'c') { +- printf(">>> Using default no de-emphasis\n"); +- info->emphasis = 0; +- } +- else { +- if (*t == 'n') info->emphasis = 0; +- else if (*t == '5') info->emphasis = 1; +- else if (*t == 'c') info->emphasis = 3; +- printf(">>> Using de-emphasis %s\n",t); +- } +- +- /* Start 2. Part changes for CD Ver 3.2; jsp; 22-Aug-1991 */ +- +- printf("Do you want to set the private bit? (y/<n>): "); +- gets(t); +- if (*t == 'y' || *t == 'Y') info->extension = 1; +- else info->extension = 0; +- if(info->extension) printf(">>> Private bit set\n"); +- else printf(">>> Private bit not set\n"); +- +- /* End changes for CD Ver 3.2; jsp; 22-Aug-1991 */ +- +- printf("Do you want error protection? (y/<n>): "); +- gets(t); +- if (*t == 'y' || *t == 'Y') info->error_protection = TRUE; +- else info->error_protection = FALSE; +- if(info->error_protection) printf(">>> Error protection used\n"); +- else printf(">>> Error protection not used\n"); +- +- printf("Is the material copyrighted? (y/<n>): "); +- gets(t); +- if (*t == 'y' || *t == 'Y') info->copyright = 1; +- else info->copyright = 0; +- if(info->copyright) printf(">>> Copyrighted material\n"); +- else printf(">>> Material not copyrighted\n"); +- +- printf("Is this the original? (y/<n>): "); +- gets(t); +- if (*t == 'y' || *t == 'Y') info->original = 1; +- else info->original = 0; +- if(info->original) printf(">>> Original material\n"); +- else printf(">>> Material not original\n"); +- +- printf("Do you wish to exit (last chance before encoding)? (y/<n>): "); +- gets(t); +- if (*t == 'y' || *t == 'Y') exit(0); +- } +- +- /************************************************************************ +- * +- * parse_args +- * +- * PURPOSE: Sets encoding parameters to the specifications of the +- * command line. Default settings are used for parameters +- * not specified in the command line. +- * +- * SEMANTICS: The command line is parsed according to the following +- * syntax: +- * +- * -l is followed by the layer number +- * -m is followed by the mode +- * -p is followed by the psychoacoustic model number +- * -s is followed by the sampling rate +- * -b is followed by the total bitrate, irrespective of the mode +- * -d is followed by the emphasis flag +- * -c is followed by the copyright/no_copyright flag +- * -o is followed by the original/not_original flag +- * -e is followed by the error_protection on/off flag +- * +- * If the input file is in AIFF format, the sampling frequency is read +- * from the AIFF header. +- * +- * The input and output filenames are read into #inpath# and #outpath#. +- * +- ************************************************************************/ +- +- void +- parse_args(argc, argv, fr_ps, psy, num_samples, inPath, outPath) +- int argc; +- char **argv; +- frame_params *fr_ps; +- int *psy; +- unsigned long *num_samples; +- char inPath[MAX_NAME_SIZE]; +- char outPath[MAX_NAME_SIZE]; +- { +- FLOAT srate; +- int brate; +- layer *info = fr_ps->header; +- int err = 0, i = 0; +- IFF_AIFF pcm_aiff_data; +- long samplerate; +- long soundPosition; +- +- /* preset defaults */ +- inPath[0] = '\0'; outPath[0] = '\0'; +- info->lay = DFLT_LAY; +- switch(DFLT_MOD) { +- case 's': info->mode = MPG_MD_STEREO; info->mode_ext = 0; break; +- case 'd': info->mode = MPG_MD_DUAL_CHANNEL; info->mode_ext=0; break; +- case 'j': info->mode = MPG_MD_JOINT_STEREO; break; +- case 'm': info->mode = MPG_MD_MONO; info->mode_ext = 0; break; +- default: +- fprintf(stderr, "%s: Bad mode dflt %c\n", programName, DFLT_MOD); +- abort(); +- } +- *psy = DFLT_PSY; +- if((info->sampling_frequency = SmpFrqIndex((long)(1000*DFLT_SFQ), &info->version)) < 0) { +- fprintf(stderr, "%s: bad sfrq default %.2f\n", programName, DFLT_SFQ); +- abort(); +- } +- info->bitrate_index = 14; +- brate = 0; +- switch(DFLT_EMP) { +- case 'n': info->emphasis = 0; break; +- case '5': info->emphasis = 1; break; +- case 'c': info->emphasis = 3; break; +- default: +- fprintf(stderr, "%s: Bad emph dflt %c\n", programName, DFLT_EMP); +- abort(); +- } +- info->copyright = 0; info->original = 0; info->error_protection = FALSE; +- +- /* process args */ +- while(++i<argc && err == 0) { +- char c, *token, *arg, *nextArg; +- int argUsed; +- +- token = argv[i]; +- if(*token++ == '-') { +- if(i+1 < argc) nextArg = argv[i+1]; +- else nextArg = ""; +- argUsed = 0; +- while( (c = *token++) ) { +- if(*token /* NumericQ(token) */) arg = token; +- else arg = nextArg; +- switch(c) { +- case 'l': info->lay = atoi(arg); argUsed = 1; +- if(info->lay<1 || info->lay>3) { +- fprintf(stderr,"%s: -l layer must be 1, 2, or 3, not %s\n", +- programName, arg); +- err = 1; +- } +- break; +- case 'm': argUsed = 1; +- if (*arg == 's') +- { info->mode = MPG_MD_STEREO; info->mode_ext = 0; } +- else if (*arg == 'd') +- { info->mode = MPG_MD_DUAL_CHANNEL; info->mode_ext=0; } +- else if (*arg == 'j') +- { info->mode = MPG_MD_JOINT_STEREO; } +- else if (*arg == 'm') +- { info->mode = MPG_MD_MONO; info->mode_ext = 0; } +- else { +- fprintf(stderr,"%s: -m mode must be s/d/j/m not %s\n", +- programName, arg); +- err = 1; +- } +- break; +- case 'p': *psy = atoi(arg); argUsed = 1; +- if(*psy<1 || *psy>2) { +- fprintf(stderr,"%s: -p model must be 1 or 2, not %s\n", +- programName, arg); +- err = 1; +- } +- break; +- +- case 's': +- argUsed = 1; +- srate = atof( arg ); +- /* samplerate = rint( 1000.0 * srate ); $A */ +- samplerate = (long) (( 1000.0 * srate ) + 0.5); +- if( (info->sampling_frequency = +- SmpFrqIndex((long) samplerate, &info->version)) < 0 ) +- err = 1; +- break; +- +- case 'b': +- argUsed = 1; +- brate = atoi(arg); +- break; +- case 'd': argUsed = 1; +- if (*arg == 'n') info->emphasis = 0; +- else if (*arg == '5') info->emphasis = 1; +- else if (*arg == 'c') info->emphasis = 3; +- else { +- fprintf(stderr,"%s: -d emp must be n/5/c not %s\n", +- programName, arg); +- err = 1; +- } +- break; +- case 'c': info->copyright = 1; break; +- case 'o': info->original = 1; break; +- case 'e': info->error_protection = TRUE; break; +- default: fprintf(stderr,"%s: unrec option %c\n", +- programName, c); +- err = 1; break; +- } +- if(argUsed) { +- if(arg == token) token = ""; /* no more from token */ +- else ++i; /* skip arg we used */ +- arg = ""; argUsed = 0; +- } +- } +- } +- else { +- if(inPath[0] == '\0') strcpy(inPath, argv[i]); +- else if(outPath[0] == '\0') strcpy(outPath, argv[i]); +- else { +- fprintf(stderr,"%s: excess arg %s\n", programName, argv[i]); +- err = 1; +- } +- } +- } +- +- if(err || inPath[0] == '\0') usage(); /* never returns */ +- +- if(outPath[0] == '\0') { +- #ifdef MS_DOS +- /* replace old extension with new one, 1992-08-19, 1995-06-12 shn */ +- new_ext(inPath, DFLT_EXT, outPath); +- #else +- strcpy(outPath, inPath); +- strcat(outPath, DFLT_EXT); +- #endif +- } +- +- if ((musicin = fopen(inPath, "rb")) == NULL) { +- printf("Could not find \"%s\".\n", inPath); +- exit(1); +- } +- +- open_bit_stream_w(&bs, outPath, BUFFER_SIZE); +- +- if ((soundPosition = aiff_read_headers(musicin, &pcm_aiff_data)) != -1) { +- +- printf(">>> Using Audio IFF sound file headers\n"); +- +- aiff_check(inPath, &pcm_aiff_data, &info->version); +- +- if (fseek(musicin, soundPosition, SEEK_SET) != 0) { +- printf("Could not seek to PCM sound data in \"%s\".\n", inPath); +- exit(1); +- } +- +- info->sampling_frequency = SmpFrqIndex((long)pcm_aiff_data.sampleRate, &info->version); +- printf(">>> %f Hz sampling frequency selected\n", +- pcm_aiff_data.sampleRate); +- +- /* Determine number of samples in sound file */ +- #ifndef MS_DOS +- *num_samples = pcm_aiff_data.numChannels * +- pcm_aiff_data.numSampleFrames; +- #else +- *num_samples = (long)(pcm_aiff_data.numChannels) * +- (long)(pcm_aiff_data.numSampleFrames); +- #endif +- if ( pcm_aiff_data.numChannels == 1 ) { +- info->mode = MPG_MD_MONO; +- info->mode_ext = 0; +- } +- } +- else { /* Not using Audio IFF sound file headers. */ +- +- if (fseek(musicin, 0, SEEK_SET) != 0) { +- printf("Could not seek to PCM sound data in \"%s\".\n", inPath); +- exit(1); +- } +- +- /* Declare sound file to have "infinite" number of samples. */ +- *num_samples = MAX_U_32_NUM; +- +- } +- if ( brate == 0 ) +- brate = bitrate[info->version][info->lay-1][14]; +- if( (info->bitrate_index = BitrateIndex(info->lay, brate, info->version)) < 0) err=1; +- if(err || inPath[0] == '\0') usage(); /* never returns */ +- +- } +- +- /************************************************************************ +- * +- * print_config +- * +- * PURPOSE: Prints the encoding parameters used +- * +- ************************************************************************/ +- +- void print_config( frame_params *fr_ps, int *psy, char *inPath, char *outPath) +- { +- layer *info = fr_ps->header; +- +- printf("Encoding configuration:\n"); +- printf("Algorithm=%s\n", version_names[info->version]); +- if(info->mode != MPG_MD_JOINT_STEREO) +- printf("Layer=%s mode=%s extn=%d psy model=%d\n", +- layer_names[info->lay-1], mode_names[info->mode], +- info->mode_ext, *psy); +- else printf("Layer=%s mode=%s extn=data dependant psy model=%d\n", +- layer_names[info->lay-1], mode_names[info->mode], *psy); +- printf("samp frq=%.1f kHz total bitrate=%d kbps\n", +- s_freq[info->version][info->sampling_frequency], +- bitrate[info->version][info->lay-1][info->bitrate_index]); +- printf("de-emph=%d c/right=%d orig=%d errprot=%s\n", +- info->emphasis, info->copyright, info->original, +- ((info->error_protection) ? "on" : "off")); +- printf("input file: '%s' output file: '%s'\n", inPath, outPath); +- } +- +- +- +- +- +- /************************************************************************ +- * +- * main +- * +- * PURPOSE: MPEG I Encoder supporting layers 1 and 2, and 3, with +- * psychoacoustic models 1 (MUSICAM) and 2 (AT&T) +- * +- * SEMANTICS: One overlapping frame of audio of up to 2 channels are +- * processed at a time in the following order: +- * (associated routines are in parentheses) +- * +- * 1. Filter sliding window of data to get 32 subband +- * samples per channel. +- * (window_subband,filter_subband) +- * +- * 2. If joint stereo mode, combine left and right channels +- * for subbands above #jsbound#. +- * (*_combine_LR) +- * +- * 3. Calculate scalefactors for the frame, and if layer 2, +- * also calculate scalefactor select information. +- * (*_scale_factor_calc) +- * +- * 4. Calculate psychoacoustic masking levels using selected +- * psychoacoustic model. +- * (*_Psycho_One, psycho_anal) +- * +- * 5. Perform iterative bit allocation for subbands with low +- * mask_to_noise ratios using masking levels from step 4. +- * (*_main_bit_allocation) +- * +- * 6. If error protection flag is active, add redundancy for +- * error protection. +- * (*_CRC_calc) +- * +- * 7. Pack bit allocation, scalefactors, and scalefactor select +- * information (layer 2) onto bitstream. +- * (*_encode_bit_alloc,*_encode_scale,II_transmission_pattern) +- * +- * 8. Quantize subbands and pack them into bitstream +- * (*_subband_quantization, *_sample_encoding) +- * +- ************************************************************************/ +- +- int frameNum=0; +- +- void main(argc, argv) +- int argc; +- char **argv; +- { +- typedef double SBS[2][3][SCALE_BLOCK][SBLIMIT]; +- SBS FAR *sb_sample; +- L3SBS FAR *l3_sb_sample; +- typedef double JSBS[3][SCALE_BLOCK][SBLIMIT]; +- JSBS FAR *j_sample; +- typedef double IN[2][HAN_SIZE]; +- IN FAR *win_que; +- typedef unsigned int SUB[2][3][SCALE_BLOCK][SBLIMIT]; +- SUB FAR *subband; +- +- frame_params fr_ps; +- layer info; +- char original_file_name[MAX_NAME_SIZE]; +- char encoded_file_name[MAX_NAME_SIZE]; +- short FAR **win_buf; +- static short FAR buffer[2][1152]; +- static unsigned int bit_alloc[2][SBLIMIT], scfsi[2][SBLIMIT]; +- static unsigned int scalar[2][3][SBLIMIT], j_scale[3][SBLIMIT]; +- static double FAR ltmin[2][SBLIMIT], lgmin[2][SBLIMIT], max_sc[2][SBLIMIT]; +- FLOAT snr32[32]; +- short sam[2][1344]; /* was [1056]; */ +- int whole_SpF, extra_slot = 0; +- double avg_slots_per_frame, frac_SpF, slot_lag; +- int model, stereo, error_protection; +- static unsigned int crc; +- int i, j, k, adb; +- unsigned long bitsPerSlot, samplesPerFrame; +- unsigned long frameBits, sentBits = 0; +- unsigned long num_samples; +- +- #ifdef MACINTOSH +- argc = ccommand( &argv ); +- #endif +- +- +- /* Most large variables are declared dynamically to ensure +- compatibility with smaller machines */ +- +- sb_sample = (SBS FAR *) mem_alloc(sizeof(SBS), "sb_sample"); +- l3_sb_sample = (L3SBS FAR *) mem_alloc(sizeof(SBS), "l3_sb_sample"); +- j_sample = (JSBS FAR *) mem_alloc(sizeof(JSBS), "j_sample"); +- win_que = (IN FAR *) mem_alloc(sizeof(IN), "Win_que"); +- subband = (SUB FAR *) mem_alloc(sizeof(SUB),"subband"); +- win_buf = (short FAR **) mem_alloc(sizeof(short *)*2, "win_buf"); +- +- /* clear buffers */ +- memset((char *) buffer, 0, sizeof(buffer)); +- memset((char *) bit_alloc, 0, sizeof(bit_alloc)); +- memset((char *) scalar, 0, sizeof(scalar)); +- memset((char *) j_scale, 0, sizeof(j_scale)); +- memset((char *) scfsi, 0, sizeof(scfsi)); +- memset((char *) ltmin, 0, sizeof(ltmin)); +- memset((char *) lgmin, 0, sizeof(lgmin)); +- memset((char *) max_sc, 0, sizeof(max_sc)); +- memset((char *) snr32, 0, sizeof(snr32)); +- memset((char *) sam, 0, sizeof(sam)); +- +- fr_ps.header = &info; +- fr_ps.tab_num = -1; /* no table loaded */ +- fr_ps.alloc = NULL; +- info.version = MPEG_AUDIO_ID; /* Default: MPEG-1 */ +- +- programName = argv[0]; +- if(argc==1) /* no command-line args */ +- obtain_parameters(&fr_ps, &model, &num_samples, +- original_file_name, encoded_file_name); +- else +- parse_args(argc, argv, &fr_ps, &model, &num_samples, +- original_file_name, encoded_file_name); +- print_config(&fr_ps, &model, +- original_file_name, encoded_file_name); +- +- hdr_to_frps(&fr_ps); +- stereo = fr_ps.stereo; +- error_protection = info.error_protection; +- +- if (info.lay == 1) +- { bitsPerSlot = 32; samplesPerFrame = 384; } +- else +- if ( info.lay == 2 ) +- { bitsPerSlot = 8; samplesPerFrame = 1152; } +- else +- { /* layer 3 */ +- bitsPerSlot = 8; +- samplesPerFrame = info.version == 1 ? 1152 : 576; +- +- /* Apologize for missing features */ +- if ( info.mode == MPG_MD_JOINT_STEREO ) +- { +- fprintf( stderr, "Sorry, joint stereo not yet available for layer3\n" ); +- exit( 1 ); +- } +- #if 0 +- if ( info.version != MPEG_AUDIO_ID ) +- { +- fprintf( stderr, "Sorry, MPEG2-LSF not yet available for layer3\n" ); +- exit( 1 ); +- } +- #endif +- if ( model != 2 ) +- { +- fprintf( stderr, "Sorry, psycho model 1 not available for layer3\n" ); +- exit( 1 ); +- } +- } +- /* Figure average number of 'slots' per frame. */ +- /* Bitrate means TOTAL for both channels, not per side. */ +- avg_slots_per_frame = ((double)samplesPerFrame / +- s_freq[info.version][info.sampling_frequency]) * +- ((double)bitrate[info.version][info.lay-1][info.bitrate_index] / +- (double)bitsPerSlot); +- whole_SpF = (int) avg_slots_per_frame; +- printf("slots/frame = %d\n",whole_SpF); +- frac_SpF = avg_slots_per_frame - (double)whole_SpF; +- slot_lag = -frac_SpF; +- printf("frac SpF=%.3f, tot bitrate=%d kbps, s freq=%.1f kHz\n", +- frac_SpF, bitrate[info.version][info.lay-1][info.bitrate_index], +- s_freq[info.version][info.sampling_frequency]); +- +- if (frac_SpF != 0) +- printf("Fractional number of slots, padding required\n"); +- else info.padding = 0; +- +- while ( get_audio(musicin, buffer, num_samples, stereo, &info) > 0 ) +- { +- fprintf(stderr, "{%4lu}", frameNum++); fflush(stderr); +- win_buf[0] = &buffer[0][0]; +- win_buf[1] = &buffer[1][0]; +- if (frac_SpF != 0) { +- if (slot_lag > (frac_SpF-1.0) ) { +- slot_lag -= frac_SpF; +- extra_slot = 0; +- info.padding = 0; +- /* printf("No padding for this frame\n"); */ +- } +- else { +- extra_slot = 1; +- info.padding = 1; +- slot_lag += (1-frac_SpF); +- /* printf("Padding for this frame\n"); */ +- } +- } +- adb = (whole_SpF+extra_slot) * bitsPerSlot; +- +- switch (info.lay) +- { +- +- /***************************** Layer I **********************************/ +- +- case 1 : +- for (j=0;j<SCALE_BLOCK;j++) +- for (k=0;k<stereo;k++) { +- window_subband(&win_buf[k], &(*win_que)[k][0], k); +- filter_subband(&(*win_que)[k][0], &(*sb_sample)[k][0][j][0]); +- } +- +- I_scale_factor_calc(*sb_sample, scalar, stereo); +- if(fr_ps.actual_mode == MPG_MD_JOINT_STEREO) { +- I_combine_LR(*sb_sample, *j_sample); +- I_scale_factor_calc(j_sample, &j_scale, 1); +- } +- +- put_scale(scalar, &fr_ps, max_sc); +- +- if (model == 1) I_Psycho_One(buffer, max_sc, ltmin, &fr_ps); +- else { +- for (k=0;k<stereo;k++) { +- psycho_anal(&buffer[k][0],&sam[k][0], k, info.lay, snr32, +- (FLOAT)s_freq[info.version][info.sampling_frequency]*1000); +- for (i=0;i<SBLIMIT;i++) ltmin[k][i] = (double) snr32[i]; +- } +- } +- +- I_main_bit_allocation(ltmin, bit_alloc, &adb, &fr_ps); +- +- if (error_protection) I_CRC_calc(&fr_ps, bit_alloc, &crc); +- +- encode_info(&fr_ps, &bs); +- +- if (error_protection) encode_CRC(crc, &bs); +- +- I_encode_bit_alloc(bit_alloc, &fr_ps, &bs); +- I_encode_scale(scalar, bit_alloc, &fr_ps, &bs); +- I_subband_quantization(scalar, *sb_sample, j_scale, *j_sample, +- bit_alloc, *subband, &fr_ps); +- I_sample_encoding(*subband, bit_alloc, &fr_ps, &bs); +- for (i=0;i<adb;i++) put1bit(&bs, 0); +- break; +- +- /***************************** Layer 2 **********************************/ +- +- case 2 : +- for (i=0;i<3;i++) for (j=0;j<SCALE_BLOCK;j++) +- for (k=0;k<stereo;k++) { +- window_subband(&win_buf[k], &(*win_que)[k][0], k); +- filter_subband(&(*win_que)[k][0], &(*sb_sample)[k][i][j][0]); +- } +- +- II_scale_factor_calc(*sb_sample, scalar, stereo, fr_ps.sblimit); +- pick_scale(scalar, &fr_ps, max_sc); +- if(fr_ps.actual_mode == MPG_MD_JOINT_STEREO) { +- II_combine_LR(*sb_sample, *j_sample, fr_ps.sblimit); +- II_scale_factor_calc(j_sample, &j_scale, 1, fr_ps.sblimit); +- } /* this way we calculate more mono than we need */ +- /* but it is cheap */ +- +- if (model == 1) II_Psycho_One(buffer, max_sc, ltmin, &fr_ps); +- else { +- for (k=0;k<stereo;k++) { +- psycho_anal(&buffer[k][0],&sam[k][0], k, +- info.lay, snr32, +- (FLOAT)s_freq[info.version][info.sampling_frequency]*1000); +- for (i=0;i<SBLIMIT;i++) ltmin[k][i] = (double) snr32[i]; +- } +- } +- +- II_transmission_pattern(scalar, scfsi, &fr_ps); +- II_main_bit_allocation(ltmin, scfsi, bit_alloc, &adb, &fr_ps); +- +- if (error_protection) +- II_CRC_calc(&fr_ps, bit_alloc, scfsi, &crc); +- +- encode_info(&fr_ps, &bs); +- +- if (error_protection) encode_CRC(crc, &bs); +- +- II_encode_bit_alloc(bit_alloc, &fr_ps, &bs); +- II_encode_scale(bit_alloc, scfsi, scalar, &fr_ps, &bs); +- II_subband_quantization(scalar, *sb_sample, j_scale, +- *j_sample, bit_alloc, *subband, &fr_ps); +- II_sample_encoding(*subband, bit_alloc, &fr_ps, &bs); +- for (i=0;i<adb;i++) put1bit(&bs, 0); +- break; +- +- /***************************** Layer 3 **********************************/ +- +- case 3: +- { +- /* +- large "auto" vars are static due to the Macintosh linker +- */ +- static double xr[2][2][576]; +- static double xr_dec[2][2][576]; +- static double pe[2][2]; +- static int l3_enc[2][2][576]; +- static III_psy_ratio ratio; +- static III_side_info_t l3_side; +- static III_scalefac_t scalefac; +- int gr, mode_gr, ch; +- int mean_bits, sideinfo_len; +- +- int bitsPerFrame = 8 * whole_SpF + (info.padding * 8); +- mode_gr = (info.version == 1) ? 2 : 1; +- +- /* +- determine the mean bitrate for main data +- */ +- sideinfo_len = 32; +- if ( info.version == 1 ) +- { /* MPEG 1 */ +- if ( stereo == 1 ) +- sideinfo_len += 136; +- else +- sideinfo_len += 256; +- } +- else +- { /* MPEG 2 */ +- if ( stereo == 1 ) +- sideinfo_len += 72; +- else +- sideinfo_len += 136; +- } +- if ( info.error_protection ) +- sideinfo_len += 16; +- mean_bits = (bitsPerFrame - sideinfo_len) / mode_gr; +- +- /* +- psychoacoustic model +- */ +- for ( gr = 0; gr < mode_gr; gr++ ) +- for ( ch = 0; ch < stereo; ch++ ) +- { +- L3psycho_anal( &buffer[ch][gr*576], &sam[ch][0], ch, info.lay, +- snr32, s_freq[info.version][info.sampling_frequency] * 1000.0, +- &ratio.l[gr][ch][0], &ratio.s[gr][ch][0], +- &pe[gr][ch], &l3_side.gr[gr].ch[ch].tt ); +- } +- +- /* +- polyphase filtering +- */ +- for( gr = 0; gr < mode_gr; gr++ ) +- for ( ch = 0; ch < stereo; ch++ ) +- for ( j = 0; j < 18; j++ ) +- { +- window_subband( &win_buf[ch], &(*win_que)[ch][0], ch ); +- filter_subband( &(*win_que)[ch][0], &(*l3_sb_sample)[ch][gr+1][j][0] ); +- } +- +- /* +- apply mdct to the polyphase outputs +- */ +- mdct_sub( l3_sb_sample, xr, stereo, &l3_side, mode_gr ); +- #if 0 +- delay( xr, stereo ); +- #endif +- /* +- bit and noise allocation +- */ +- iteration_loop( pe, xr, &ratio, &l3_side, l3_enc, mean_bits, +- stereo, xr_dec, &scalefac, &fr_ps, 0, bitsPerFrame ); +- +- /* +- write the frame to the bitstream +- */ +- III_format_bitstream( bitsPerFrame, &fr_ps, l3_enc, &l3_side, &scalefac, &bs, +- xr, NULL, 0 ); +- } +- break; /* end of layer 3 */ +- +- +- } /* end switch */ +- +- frameBits = sstell( &bs ) - sentBits; +- if ( frameBits % bitsPerSlot ) /* a program failure */ +- fprintf( stderr, "Sent %ld bits = %ld slots plus %ld\n", +- frameBits, frameBits/bitsPerSlot, +- frameBits%bitsPerSlot ); +- sentBits += frameBits; +- +- } +- +- if ( info.lay == 3 ) +- III_FlushBitstream(); +- +- close_bit_stream_w( &bs ); +- +- printf("Avg slots/frame = %.3f; b/smp = %.2f; br = %.3f kbps\n", +- (FLOAT) sentBits / (frameNum * bitsPerSlot), +- (FLOAT) sentBits / (frameNum * samplesPerFrame), +- (FLOAT) sentBits / (frameNum * samplesPerFrame) * +- s_freq[info.version][info.sampling_frequency]); +- +- if (fclose(musicin) != 0){ +- printf("Could not close \"%s\".\n", original_file_name); +- exit(2); +- } +- +- #ifdef MACINTOSH +- set_mac_file_attr( encoded_file_name, VOL_REF_NUM, CREATOR_ENCODE, +- FILETYPE_ENCODE ); +- #endif +- +- printf("Encoding of \"%s\" with psychoacoustic model %d is finished\n", +- original_file_name, model); +- printf("The MPEG encoded output file name is \"%s\"\n", +- encoded_file_name); +- exit(0); +- } +- +- /************************************************************************ +- * +- * usage +- * +- * PURPOSE: Writes command line syntax to the file specified by #stderr# +- * +- ************************************************************************/ +- +- void usage() /* print syntax & exit */ +- { +- fprintf(stderr, +- "usage: %s queries for all arguments, or\n", +- programName); +- fprintf(stderr, +- " %s [-l lay][-m mode][-p psy][-s sfrq][-b br][-d emp]\n", +- programName); +- fprintf(stderr, +- " [-c][-o][-e] inputPCM [outBS]\n"); +- fprintf(stderr,"where\n"); +- fprintf(stderr," -l lay use layer <lay> coding (dflt %4u)\n",DFLT_LAY); +- fprintf(stderr," -m mode channel mode : s/d/j/m (dflt %4c)\n",DFLT_MOD); +- fprintf(stderr," -p psy psychoacoustic model 1/2 (dflt %4u)\n",DFLT_PSY); +- fprintf(stderr," -s sfrq input smpl rate in kHz (dflt %4.1f)\n",DFLT_SFQ); +- fprintf(stderr," -b br total bitrate in kbps (dflt highest)\n"); +- fprintf(stderr," -d emp de-emphasis n/5/c (dflt %4c)\n",DFLT_EMP); +- fprintf(stderr," -c mark as copyright\n"); +- fprintf(stderr," -o mark as original\n"); +- fprintf(stderr," -e add error protection\n"); +- fprintf(stderr," inputPCM input PCM sound file (standard or AIFF)\n"); +- fprintf(stderr," outBS output bit stream of encoded audio (dflt inName+%s)\n", +- DFLT_EXT); +- exit(1); +- } +- +- /************************************************************************ +- * +- * aiff_check +- * +- * PURPOSE: Checks AIFF header information to make sure it is valid. +- * Exits if not. +- * +- ************************************************************************/ +- +- void aiff_check( char *file_name, IFF_AIFF *pcm_aiff_data, int *version) +- { +- if (pcm_aiff_data->sampleType != IFF_ID_SSND) { +- printf("Sound data is not PCM in \"%s\".\n", file_name); +- exit(1); +- } +- +- if(SmpFrqIndex((long)pcm_aiff_data->sampleRate, version) < 0) { +- printf("in \"%s\".\n", file_name); +- exit(1); +- } +- +- if (pcm_aiff_data->sampleSize != sizeof(short) * BITS_IN_A_BYTE) { +- printf("Sound data is not %d bits in \"%s\".\n", +- sizeof(short) * BITS_IN_A_BYTE, file_name); +- exit(1); +- } +- +- if (pcm_aiff_data->numChannels != MONO && +- pcm_aiff_data->numChannels != STEREO) { +- printf("Sound data is not mono or stereo in \"%s\".\n", file_name); +- exit(1); +- } +- +- if (pcm_aiff_data->blkAlgn.blockSize != 0) { +- printf("Block size is not %d bytes in \"%s\".\n", 0, file_name); +- exit(1); +- } +- +- if (pcm_aiff_data->blkAlgn.offset != 0) { +- printf("Block offset is not %d bytes in \"%s\".\n", 0, file_name); +- exit(1); +- } +- } +--- 0 ---- +diff -r -c -N encoder/newmdct.c lame3.70/newmdct.c +*** encoder/newmdct.c Wed Dec 31 17:00:00 1969 +--- lame3.70/newmdct.c Thu Apr 6 12:50:51 2000 +*************** +*** 0 **** +--- 1,631 ---- ++ /* ++ * MP3 window subband -> subband filtering -> mdct routine ++ * ++ * Copyright (c) 1999 Takehiro TOMINAGA ++ * ++ * ++ * This library is free software; you can redistribute it and/or ++ * modify it under the terms of the GNU Library General Public ++ * License as published by the Free Software Foundation; either ++ * version 2 of the License, or (at your option) any later version. ++ * ++ * This library is distributed in the hope that it will be useful, ++ * but WITHOUT ANY WARRANTY; without even the implied warranty of ++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU ++ * Library General Public License for more details. ++ * ++ * You should have received a copy of the GNU Library General Public ++ * License along with this library; if not, write to the ++ * Free Software Foundation, Inc., 59 Temple Place - Suite 330, ++ * Boston, MA 02111-1307, USA. ++ */ ++ ++ /* ++ * Special Thanks to Patrick De Smet for your advices. ++ */ ++ ++ ++ #include "util.h" ++ #include "l3side.h" ++ #include "newmdct.h" ++ ++ #define SCALE 32768 ++ ++ static FLOAT8 enwindow[] = ++ { ++ 3.5780907e-02,1.7876148e-02,3.134727e-03,2.457142e-03, ++ 9.71317e-04, 2.18868e-04, 1.01566e-04, 1.3828e-05, ++ ++ 3.5758972e-02, 3.401756e-03, 9.83715e-04, 9.9182e-05, ++ -4.77e-07, 1.03951e-04, 9.53674e-04, 2.841473e-03, ++ 1.2398e-05, 1.91212e-04, 2.283096e-03,1.6994476e-02, ++ 1.8756866e-02, 2.630711e-03, 2.47478e-04, 1.4782e-05, ++ ++ 3.5694122e-02, 3.643036e-03, 9.91821e-04, 9.6321e-05, ++ -4.77e-07, 1.05858e-04, 9.30786e-04, 2.521515e-03, ++ 1.1444e-05, 1.65462e-04, 2.110004e-03,1.6112804e-02, ++ 1.9634247e-02, 2.803326e-03, 2.77042e-04, 1.6689e-05, ++ ++ 3.5586357e-02, 3.858566e-03, 9.95159e-04, 9.3460e-05, ++ -4.77e-07, 1.07288e-04, 9.02653e-04, 2.174854e-03, ++ 1.0014e-05, 1.40190e-04, 1.937389e-03,1.5233517e-02, ++ 2.0506859e-02, 2.974033e-03, 3.07560e-04, 1.8120e-05, ++ ++ 3.5435200e-02, 4.049301e-03, 9.94205e-04, 9.0599e-05, ++ -4.77e-07, 1.08242e-04, 8.68797e-04, 1.800537e-03, ++ 9.060e-06, 1.16348e-04, 1.766682e-03,1.4358521e-02, ++ 2.1372318e-02, 3.14188e-03, 3.39031e-04, 1.9550e-05, ++ ++ 3.5242081e-02, 4.215240e-03, 9.89437e-04, 8.7261e-05, ++ -4.77e-07, 1.08719e-04, 8.29220e-04, 1.399517e-03, ++ 8.106e-06, 9.3937e-05, 1.597881e-03,1.3489246e-02, ++ 2.2228718e-02, 3.306866e-03, 3.71456e-04, 2.1458e-05, ++ ++ 3.5007000e-02, 4.357815e-03, 9.80854e-04, 8.3923e-05, ++ -4.77e-07, 1.08719e-04, 7.8392e-04, 9.71317e-04, ++ 7.629e-06, 7.2956e-05, 1.432419e-03,1.2627602e-02, ++ 2.3074150e-02, 3.467083e-03, 4.04358e-04, 2.3365e-05, ++ ++ 3.4730434e-02, 4.477024e-03, 9.68933e-04, 8.0585e-05, ++ -9.54e-07, 1.08242e-04, 7.31945e-04, 5.15938e-04, ++ 6.676e-06, 5.2929e-05, 1.269817e-03,1.1775017e-02, ++ 2.3907185e-02, 3.622532e-03, 4.38213e-04, 2.5272e-05, ++ ++ 3.4412861e-02, 4.573822e-03, 9.54151e-04, 7.6771e-05, ++ -9.54e-07, 1.06812e-04, 6.74248e-04, 3.3379e-05, ++ 6.199e-06, 3.4332e-05, 1.111031e-03,1.0933399e-02, ++ 2.4725437e-02, 3.771782e-03, 4.72546e-04, 2.7657e-05, ++ ++ 3.4055710e-02, 4.649162e-03, 9.35555e-04, 7.3433e-05, ++ -9.54e-07, 1.05381e-04, 6.10352e-04, -4.75883e-04, ++ 5.245e-06, 1.7166e-05, 9.56535e-04,1.0103703e-02, ++ 2.5527000e-02, 3.914356e-03, 5.07355e-04, 3.0041e-05, ++ ++ 3.3659935e-02, 4.703045e-03, 9.15051e-04, 7.0095e-05, ++ -9.54e-07, 1.02520e-04, 5.39303e-04,-1.011848e-03, ++ 4.768e-06, 9.54e-07, 8.06808e-04, 9.287834e-03, ++ 2.6310921e-02, 4.048824e-03, 5.42164e-04, 3.2425e-05, ++ ++ 3.3225536e-02, 4.737377e-03, 8.91685e-04, 6.6280e-05, ++ -1.431e-06, 9.9182e-05, 4.62532e-04,-1.573563e-03, ++ 4.292e-06, -1.3828e-05, 6.61850e-04, 8.487225e-03, ++ 2.7073860e-02, 4.174709e-03, 5.76973e-04, 3.4809e-05, ++ ++ 3.2754898e-02, 4.752159e-03, 8.66413e-04, 6.2943e-05, ++ -1.431e-06, 9.5367e-05, 3.78609e-04,-2.161503e-03, ++ 3.815e-06, -2.718e-05, 5.22137e-04, 7.703304e-03, ++ 2.7815342e-02, 4.290581e-03, 6.11782e-04, 3.7670e-05, ++ ++ 3.2248020e-02, 4.748821e-03, 8.38757e-04, 5.9605e-05, ++ -1.907e-06, 9.0122e-05, 2.88486e-04,-2.774239e-03, ++ 3.338e-06, -3.9577e-05, 3.88145e-04, 6.937027e-03, ++ 2.8532982e-02, 4.395962e-03, 6.46591e-04, 4.0531e-05, ++ ++ 3.1706810e-02, 4.728317e-03, 8.09669e-04, 5.579e-05, ++ -1.907e-06, 8.4400e-05, 1.91689e-04,-3.411293e-03, ++ 3.338e-06, -5.0545e-05, 2.59876e-04, 6.189346e-03, ++ 2.9224873e-02, 4.489899e-03, 6.80923e-04, 4.3392e-05, ++ ++ 3.1132698e-02, 4.691124e-03, 7.79152e-04, 5.2929e-05, ++ -2.384e-06, 7.7724e-05, 8.8215e-05,-4.072189e-03, ++ 2.861e-06, -6.0558e-05, 1.37329e-04, 5.462170e-03, ++ 2.9890060e-02, 4.570484e-03, 7.14302e-04, 4.6253e-05, ++ ++ 3.0526638e-02, 4.638195e-03, 7.47204e-04, 4.9591e-05, ++ 4.756451e-03, 2.1458e-05, -6.9618e-05, 2.384e-06 ++ }; ++ ++ static FLOAT8 sb_sample[2][2][18][SBLIMIT]; ++ static FLOAT8 mm[16][SBLIMIT - 1]; ++ ++ #define NS 12 ++ #define NL 36 ++ ++ static const int all[] = {0,2,3,5,6,8,9,11,12,14,15,17}; ++ static FLOAT8 ca[8], cs[8]; ++ static FLOAT8 cos_s[NS / 2][NS / 2]; ++ static FLOAT8 cos_l[(NL / 2) * 12 + (NL / 6) * 4 + (NL / 18) * 2]; ++ static FLOAT8 win[4][36]; ++ ++ #define work (&win[2][4]) ++ ++ /************************************************************************ ++ * ++ * window_subband() ++ * ++ * PURPOSE: Overlapping window on PCM samples ++ * ++ * SEMANTICS: ++ * 32 16-bit pcm samples are scaled to fractional 2's complement and ++ * concatenated to the end of the window buffer #x#. The updated window ++ * buffer #x# is then windowed by the analysis window #c# to produce the ++ * windowed sample #z# ++ * ++ ************************************************************************/ ++ ++ static void window_subband(short *xk, FLOAT8 d[SBLIMIT], FLOAT8 *in) ++ { ++ int i; ++ FLOAT8 s, t, *wp; ++ wp = enwindow; ++ { ++ t = xk[255]; ++ t += (xk[223] - xk[287]) * *wp++; ++ t += (xk[191] + xk[319]) * *wp++; ++ t += (xk[159] - xk[351]) * *wp++; ++ t += (xk[127] + xk[383]) * *wp++; ++ t += (xk[ 95] - xk[415]) * *wp++; ++ t += (xk[ 63] + xk[447]) * *wp++; ++ t += (xk[ 31] - xk[479]) * *wp++; ++ in[15] = t; ++ } ++ ++ for (i = 14; i >= 0; --i) { ++ short *x1 = &xk[i]; ++ short *x2 = &xk[-i]; ++ FLOAT8 w; ++ ++ s = x2[270]; t = x1[240]; ++ w = *wp++; s += x2[334] * w; t += x1[176] * w; ++ w = *wp++; s += x2[398] * w; t += x1[112] * w; ++ w = *wp++; s += x2[462] * w; t += x1[ 48] * w; ++ w = *wp++; s += x2[ 14] * w; t += x1[496] * w; ++ w = *wp++; s += x2[ 78] * w; t += x1[432] * w; ++ w = *wp++; s += x2[142] * w; t += x1[368] * w; ++ w = *wp++; s += x2[206] * w; t += x1[304] * w; ++ ++ w = *wp++; s += x1[ 16] * w; t -= x2[494] * w; ++ w = *wp++; s += x1[ 80] * w; t -= x2[430] * w; ++ w = *wp++; s += x1[144] * w; t -= x2[366] * w; ++ w = *wp++; s += x1[208] * w; t -= x2[302] * w; ++ w = *wp++; s -= x1[272] * w; t += x2[238] * w; ++ w = *wp++; s -= x1[336] * w; t += x2[174] * w; ++ w = *wp++; s -= x1[400] * w; t += x2[110] * w; ++ w = *wp++; s -= x1[464] * w; t += x2[ 46] * w; ++ ++ in[30 - i] = s; ++ in[i] = t; ++ } ++ ++ { ++ s = xk[239]; ++ s += xk[175] * *wp++; ++ s += xk[111] * *wp++; ++ s += xk[ 47] * *wp++; ++ s -= xk[303] * *wp++; ++ s -= xk[367] * *wp++; ++ s -= xk[431] * *wp++; ++ s -= xk[495] * *wp++; ++ /* in[-1] = s; */ ++ } ++ ++ in++; ++ wp = &mm[0][0]; ++ for (i = 15; i >= 0; --i) { ++ int j; ++ FLOAT8 s0 = s; /* mm[i][0] is always 1 */ ++ FLOAT8 s1 = t * *wp++; ++ for (j = 14; j >= 0; j--) { ++ s0 += *wp++ * *in++; ++ s1 += *wp++ * *in++; ++ } ++ in -= 30; ++ d[i ] = s0 + s1; ++ d[31 - i] = s0 - s1; ++ } ++ } ++ ++ ++ /*-------------------------------------------------------------------*/ ++ /* */ ++ /* Function: Calculation of the MDCT */ ++ /* In the case of long blocks (type 0,1,3) there are */ ++ /* 36 coefficents in the time domain and 18 in the frequency */ ++ /* domain. */ ++ /* In the case of short blocks (type 2) there are 3 */ ++ /* transformations with short length. This leads to 12 coefficents */ ++ /* in the time and 6 in the frequency domain. In this case the */ ++ /* results are stored side by side in the vector out[]. */ ++ /* */ ++ /* New layer3 */ ++ /* */ ++ /*-------------------------------------------------------------------*/ ++ ++ static void mdct_short(FLOAT8 *out, FLOAT8 *in) ++ { ++ int m; ++ for (m = NS / 2 - 1; m >= 0; --m) { ++ int l; ++ FLOAT8 a0, a1, a2, a3, a4, a5; ++ a0 = cos_s[m][0]; ++ a1 = cos_s[m][1]; ++ a2 = cos_s[m][2]; ++ a3 = cos_s[m][3]; ++ a4 = cos_s[m][4]; ++ a5 = cos_s[m][5]; ++ for (l = 2; l >= 0; l--) { ++ out[3 * m + l] = ++ a0 * in[6 * l ] + ++ a1 * in[6 * l + 1] + ++ a2 * in[6 * l + 2] + ++ a3 * in[6 * l + 3] + ++ a4 * in[6 * l + 4] + ++ a5 * in[6 * l + 5]; ++ } ++ } ++ } ++ ++ static void mdct_long(FLOAT8 *out, FLOAT8 *in) ++ { ++ FLOAT8 s0, s1, s2, s3, s4, s5; ++ int j = sizeof(all) / sizeof(int) - 1; ++ FLOAT8 *cos_l0 = cos_l; ++ do { ++ out[all[j]] = ++ in[ 0] * cos_l0[ 0] + ++ in[ 1] * cos_l0[ 1] + ++ in[ 2] * cos_l0[ 2] + ++ in[ 3] * cos_l0[ 3] + ++ in[ 4] * cos_l0[ 4] + ++ in[ 5] * cos_l0[ 5] + ++ in[ 6] * cos_l0[ 6] + ++ in[ 7] * cos_l0[ 7] + ++ in[ 8] * cos_l0[ 8] + ++ in[ 9] * cos_l0[ 9] + ++ in[10] * cos_l0[10] + ++ in[11] * cos_l0[11] + ++ in[12] * cos_l0[12] + ++ in[13] * cos_l0[13] + ++ in[14] * cos_l0[14] + ++ in[15] * cos_l0[15] + ++ in[16] * cos_l0[16] + ++ in[17] * cos_l0[17]; ++ cos_l0 += 18; ++ } while (--j >= 0); ++ ++ s0 = in[0] + in[ 5] + in[15]; ++ s1 = in[1] + in[ 4] + in[16]; ++ s2 = in[2] + in[ 3] + in[17]; ++ s3 = in[6] - in[ 9] + in[14]; ++ s4 = in[7] - in[10] + in[13]; ++ s5 = in[8] - in[11] + in[12]; ++ ++ /* 16 */ ++ out[16] = ++ s0 * cos_l0[0] + s1 * cos_l0[1] + s2 * cos_l0[2] + ++ s3 * cos_l0[3] + s4 * cos_l0[4] + s5 * cos_l0[5]; ++ cos_l0 += 6; ++ ++ /* 10 */ ++ out[10] = ++ s0 * cos_l0[0] + s1 * cos_l0[1] + s2 * cos_l0[2] + ++ s3 * cos_l0[3] + s4 * cos_l0[4] + s5 * cos_l0[5]; ++ cos_l0 += 6; ++ ++ /* 7 */ ++ out[7] = ++ s0 * cos_l0[0] + s1 * cos_l0[1] + s2 * cos_l0[2] + ++ s3 * cos_l0[3] + s4 * cos_l0[4] + s5 * cos_l0[5]; ++ cos_l0 += 6; ++ ++ /* 1 */ ++ out[1] = ++ s0 * cos_l0[0] + s1 * cos_l0[1] + s2 * cos_l0[2] + ++ s3 * cos_l0[3] + s4 * cos_l0[4] + s5 * cos_l0[5]; ++ cos_l0 += 6; ++ ++ s0 = s0 - s1 + s5; ++ s2 = s2 - s3 - s4; ++ /* 13 */ ++ out[13] = s0 * cos_l0[0] + s2 * cos_l0[1]; ++ ++ /* 4 */ ++ out[4] = s0 * cos_l0[2] + s2 * cos_l0[3]; ++ } ++ ++ ++ void mdct_sub48(lame_global_flags *gfp, ++ short *w0, short *w1, ++ FLOAT8 mdct_freq[2][2][576], ++ III_side_info_t *l3_side) ++ { ++ int gr, k, ch; ++ short *wk; ++ static int init = 0; ++ ++ if ( init == 0 ) { ++ void mdct_init48(void); ++ mdct_init48(); ++ init++; ++ } ++ ++ wk = w0; ++ /* thinking cache performance, ch->gr loop is better than gr->ch loop */ ++ for (ch = 0; ch < gfp->stereo; ch++) { ++ for (gr = 0; gr < gfp->mode_gr; gr++) { ++ int band; ++ FLOAT8 *mdct_enc = mdct_freq[gr][ch]; ++ gr_info *gi = &(l3_side->gr[gr].ch[ch].tt); ++ FLOAT8 *samp = sb_sample[ch][1 - gr][0]; ++ ++ for (k = 0; k < 18 / 2; k++) { ++ window_subband(wk, samp, work); ++ window_subband(wk + 32, samp + 32, work); ++ /* ++ * Compensate for inversion in the analysis filter ++ */ ++ for (band = 1; band < 32; band += 2) ++ samp[band + 32] *= -1.0; ++ samp += 64; ++ wk += 64; ++ } ++ ++ ++ /* apply filters on the polyphase filterbank outputs */ ++ /* bands <= gfp->highpass_band will be zeroed out below */ ++ /* bands >= gfp->lowpass_band will be zeroed out below */ ++ if (gfp->filter_type==0) { ++ FLOAT8 amp,freq; ++ for (band=gfp->highpass_band+1; band < gfp->lowpass_band ; band++) { ++ freq = band/31.0; ++ if (gfp->lowpass1 < freq && freq < gfp->lowpass2) { ++ amp = cos((PI/2)*(gfp->lowpass1-freq)/(gfp->lowpass2-gfp->lowpass1)); ++ for (k=0; k<18; k++) ++ sb_sample[ch][1-gr][k][band]*=amp; ++ } ++ if (gfp->highpass1 < freq && freq < gfp->highpass2) { ++ amp = cos((PI/2)*(gfp->highpass2-freq)/(gfp->highpass2-gfp->highpass1)); ++ for (k=0; k<18; k++) ++ sb_sample[ch][1-gr][k][band]*=amp; ++ } ++ } ++ } ++ ++ ++ ++ /* ++ * Perform imdct of 18 previous subband samples ++ * + 18 current subband samples ++ */ ++ for (band = 0; band < 32; band++, mdct_enc += 18) ++ { ++ int type = gi->block_type; ++ #ifdef ALLOW_MIXED ++ if (gi->mixed_block_flag && band < 2) ++ type = 0; ++ #endif ++ if (band >= gfp->lowpass_band || band <= gfp->highpass_band) { ++ memset((char *)mdct_enc,0,18*sizeof(FLOAT8)); ++ }else { ++ if (type == SHORT_TYPE) { ++ for (k = 2; k >= 0; --k) { ++ FLOAT8 w1 = win[SHORT_TYPE][k]; ++ work[k] = ++ sb_sample[ch][gr][k+6][band] * w1 - ++ sb_sample[ch][gr][11-k][band]; ++ work[k+3] = ++ sb_sample[ch][gr][k+12][band] + ++ sb_sample[ch][gr][17-k][band] * w1; ++ ++ work[k+6] = ++ sb_sample[ch][gr][k+12][band] * w1 - ++ sb_sample[ch][gr][17-k][band]; ++ work[k+9] = ++ sb_sample[ch][1-gr][k][band] + ++ sb_sample[ch][1-gr][5-k][band] * w1; ++ ++ work[k+12] = ++ sb_sample[ch][1-gr][k][band] * w1 - ++ sb_sample[ch][1-gr][5-k][band]; ++ work[k+15] = ++ sb_sample[ch][1-gr][k+6][band] + ++ sb_sample[ch][1-gr][11-k][band] * w1; ++ } ++ mdct_short(mdct_enc, work); ++ } else { ++ for (k = 8; k >= 0; --k) { ++ work[k] = ++ win[type][k ] * sb_sample[ch][gr][k ][band] ++ - win[type][k+9] * sb_sample[ch][gr][17-k][band]; ++ ++ work[9+k] = ++ win[type][k+18] * sb_sample[ch][1-gr][k ][band] ++ + win[type][k+27] * sb_sample[ch][1-gr][17-k][band]; ++ } ++ mdct_long(mdct_enc, work); ++ } ++ } ++ ++ ++ /* ++ Perform aliasing reduction butterfly ++ */ ++ if (type != SHORT_TYPE) { ++ if (band == 0) ++ continue; ++ for (k = 7; k >= 0; --k) { ++ FLOAT8 bu,bd; ++ bu = mdct_enc[k] * ca[k] + mdct_enc[-1-k] * cs[k]; ++ bd = mdct_enc[k] * cs[k] - mdct_enc[-1-k] * ca[k]; ++ ++ mdct_enc[-1-k] = bu; ++ mdct_enc[k] = bd; ++ } ++ } ++ } ++ } ++ wk = w1; ++ if (gfp->mode_gr == 1) { ++ memcpy(sb_sample[ch][0], sb_sample[ch][1], 576 * sizeof(FLOAT8)); ++ } ++ } ++ } ++ ++ ++ ++ void mdct_init48(void) ++ { ++ int i, k, m; ++ FLOAT8 sq; ++ FLOAT8 max; ++ ++ /* prepare the aliasing reduction butterflies */ ++ for (k = 0; k < 8; k++) { ++ /* ++ This is table B.9: coefficients for aliasing reduction ++ */ ++ static const FLOAT8 c[8] = { ++ -0.6,-0.535,-0.33,-0.185,-0.095,-0.041,-0.0142, -0.0037 ++ }; ++ sq = 1.0 + c[k] * c[k]; ++ sq = sqrt(sq); ++ ca[k] = c[k] / sq; ++ cs[k] = 1.0 / sq; ++ } ++ ++ /* type 0*/ ++ for (i = 0; i < 36; i++) ++ win[0][i] = sin(PI/36 * (i + 0.5)); ++ /* type 1*/ ++ for (i = 0; i < 18; i++) ++ win[1][i] = win[0][i]; ++ for (; i < 24; i++) ++ win[1][i] = 1.0; ++ for (; i < 30; i++) ++ win[1][i] = cos(PI/12 * (i + 0.5)); ++ for (; i < 36; i++) ++ win[1][i] = 0.0; ++ /* type 3*/ ++ for (i = 0; i < 36; i++) ++ win[3][i] = win[1][35 - i]; ++ ++ sq = 4.0 / NL; ++ { ++ FLOAT8 *cos_l0 = cos_l; ++ static const int d3[] = {1,7,10,16}; ++ static const int d9[] = {4,13}; ++ ++ int j = sizeof(all) / sizeof(int) - 1; ++ do { ++ m = all[j]; ++ for (k = 0; k < NL / 4; k++) { ++ *cos_l0++ = sq * ++ cos((PI / (4 * NL)) * (2 * m + 1) * (4 * k + 2 + NL)); ++ } ++ for (k = 0; k < NL / 4; k++) { ++ *cos_l0++ = sq * ++ cos((PI / (4 * NL)) * (2 * m + 1) * (4 * k + 2 + NL * 3)); ++ } ++ } while (--j >= 0); ++ ++ j = sizeof(d3) / sizeof(int) - 1; ++ do { ++ m = d3[j]; ++ for (k = 0; k < 3; k++) { ++ *cos_l0++ = sq * ++ cos((PI / (4 * NL)) * (2 * m + 1) * (4 * k + 2 + NL)); ++ } ++ for (k = 6; k < 9; k++) { ++ *cos_l0++ = sq * ++ cos((PI / (4 * NL)) * (2 * m + 1) * (4 * k + 2 + NL)); ++ } ++ } while (--j >= 0); ++ ++ j = sizeof(d9) / sizeof(int) - 1; ++ do { ++ m = d9[j]; ++ *cos_l0++ = sq * ++ cos((PI / (4 * NL)) * (2 * m + 1) * (2 + NL)); ++ *cos_l0++ = sq * ++ cos((PI / (4 * NL)) * (2 * m + 1) * (4 * 2 + 2 + NL)); ++ } while (--j >= 0); ++ } ++ ++ max = enwindow[256 - 8]; ++ { ++ FLOAT8 *wp = enwindow; ++ FLOAT8 *wr = enwindow; ++ FLOAT8 mmax[32 - 1]; ++ ++ { ++ FLOAT8 w = *wp++; ++ mmax[15] = w / max; ++ ++ for (k = 0; k < 7; k++) { ++ *wr++ = *wp++ / w; ++ } ++ } ++ ++ for (i = 14; i >= 0; --i) { ++ FLOAT8 w = *wp++; ++ mmax[i] = mmax[30 - i] = w / max; ++ ++ for (k = 0; k < 15; k++) { ++ *wr++ = *wp++ / w; ++ } ++ } ++ ++ { ++ wp++; ++ for (k = 0; k < 7; k++) { ++ *wr++ = *wp++ / max; ++ } ++ } ++ ++ wp = &mm[0][0]; ++ for (i = 15; i >= 0; --i) { ++ for (k = 1; k < 32; k++) { ++ *wp++ = cos((2 * i + 1) * k * PI/64) * mmax[k - 1]; ++ } ++ } ++ } ++ ++ /* swap window data*/ ++ for (k = 0; k < 4; k++) { ++ FLOAT8 a; ++ ++ a = win[0][17-k]; ++ win[0][17-k] = win[0][9+k]; ++ win[0][9+k] = a; ++ ++ a = win[0][35-k]; ++ win[0][35-k] = win[0][27+k]; ++ win[0][27+k] = a; ++ ++ a = win[1][17-k]; ++ win[1][17-k] = win[1][9+k]; ++ win[1][9+k] = a; ++ ++ a = win[1][35-k]; ++ win[1][35-k] = win[1][27+k]; ++ win[1][27+k] = a; ++ ++ a = win[3][17-k]; ++ win[3][17-k] = win[3][9+k]; ++ win[3][9+k] = a; ++ ++ a = win[3][35-k]; ++ win[3][35-k] = win[3][27+k]; ++ win[3][27+k] = a; ++ } ++ ++ for (i = 0; i < 36; i++) { ++ win[0][i] *= max / SCALE; ++ win[1][i] *= max / SCALE; ++ win[3][i] *= max / SCALE; ++ } ++ ++ /* type 2(short)*/ ++ sq = 4.0 / NS; ++ for (i = 0; i < NS / 4; i++) { ++ FLOAT8 w2 = cos(PI/12 * (i + 0.5)) * max / SCALE * sq; ++ win[SHORT_TYPE][i] = tan(PI/12 * (i + 0.5)); ++ ++ for (m = 0; m < NS / 2; m++) { ++ cos_s[m][i] = w2 * ++ cos((PI / (4 * NS)) * (2 * m + 1) * (4 * i + 2 + NS)); ++ cos_s[m][i + NS / 4] = w2 * ++ cos((PI / (4 * NS)) * (2 * m + 1) * (4 * i + 2 + NS * 3)); ++ } ++ } ++ } +diff -r -c -N encoder/newmdct.h lame3.70/newmdct.h +*** encoder/newmdct.h Wed Dec 31 17:00:00 1969 +--- lame3.70/newmdct.h Tue Mar 21 16:02:17 2000 +*************** +*** 0 **** +--- 1,6 ---- ++ #ifndef MDCT_DOT_H ++ #define MDCT_DOT_H ++ void mdct_sub48(lame_global_flags *gfp,short *w0, short *w1, ++ FLOAT8 mdct_freq[2][2][576], ++ III_side_info_t *l3_side); ++ #endif +diff -r -c -N encoder/parse.c lame3.70/parse.c +*** encoder/parse.c Wed Dec 31 17:00:00 1969 +--- lame3.70/parse.c Thu Apr 6 12:50:51 2000 +*************** +*** 0 **** +--- 1,724 ---- ++ #ifdef LAMEPARSE ++ ++ #include "util.h" ++ #include "id3tag.h" ++ #include "get_audio.h" ++ #include "brhist.h" ++ #include "version.h" ++ ++ ++ ++ #define MAX_NAME_SIZE 300 ++ char inPath[MAX_NAME_SIZE]; ++ char outPath[MAX_NAME_SIZE]; ++ ++ ++ /************************************************************************ ++ * ++ * usage ++ * ++ * PURPOSE: Writes command line syntax to the file specified by #stderr# ++ * ++ ************************************************************************/ ++ ++ void lame_usage(lame_global_flags *gfp,char *name) /* print syntax & exit */ ++ { ++ lame_print_version(stderr); ++ fprintf(stderr,"\n"); ++ fprintf(stderr,"USAGE : %s [options] <infile> [outfile]\n",name); ++ fprintf(stderr,"\n<infile> and/or <outfile> can be \"-\", which means stdin/stdout.\n"); ++ fprintf(stderr,"\n"); ++ fprintf(stderr,"Try \"%s --help\" for more information\n",name); ++ exit(1); ++ } ++ ++ ++ ++ /************************************************************************ ++ * ++ * usage ++ * ++ * PURPOSE: Writes command line syntax to the file specified by #stdout# ++ * ++ ************************************************************************/ ++ ++ void lame_help(lame_global_flags *gfp,char *name) /* print syntax & exit */ ++ { ++ lame_print_version(stdout); ++ fprintf(stdout,"\n"); ++ fprintf(stdout,"USAGE : %s [options] <infile> [outfile]\n",name); ++ fprintf(stdout,"\n<infile> and/or <outfile> can be \"-\", which means stdin/stdout.\n"); ++ fprintf(stdout,"\n"); ++ fprintf(stdout,"OPTIONS :\n"); ++ fprintf(stdout," Input options:\n"); ++ fprintf(stdout," -r input is raw pcm\n"); ++ fprintf(stdout," -x force byte-swapping of input\n"); ++ fprintf(stdout," -s sfreq sampling frequency of input file(kHz) - default 44.1kHz\n"); ++ fprintf(stdout," --mp3input input file is a MP3 file\n"); ++ fprintf(stdout,"\n"); ++ fprintf(stdout," Filter options:\n"); ++ fprintf(stdout," -k keep ALL frequencies (disables all filters)\n"); ++ fprintf(stdout," --lowpass freq frequency(kHz), lowpass filter cutoff above freq\n"); ++ fprintf(stdout," --lowpass-width freq frequency(kHz) - default 15%% of lowpass freq\n"); ++ fprintf(stdout," --highpass freq frequency(kHz), highpass filter cutoff below freq\n"); ++ fprintf(stdout," --highpass-width freq frequency(kHz) - default 15%% of highpass freq\n"); ++ fprintf(stdout," --resample sfreq sampling frequency of output file(kHz)- default=input sfreq\n"); ++ fprintf(stdout," --cwlimit freq compute tonality up to freq (in kHz) default 8.8717\n"); ++ fprintf(stdout,"\n"); ++ fprintf(stdout," Operational options:\n"); ++ fprintf(stdout," -m mode (s)tereo, (j)oint, (f)orce or (m)ono (default j)\n"); ++ fprintf(stdout," force = force ms_stereo on all frames. Faster\n"); ++ fprintf(stdout," -a downmix from stereo to mono file for mono encoding\n"); ++ fprintf(stdout," -d allow channels to have different blocktypes\n"); ++ fprintf(stdout," -S don't print progress report, VBR histograms\n"); ++ fprintf(stdout," --athonly only use the ATH for masking\n"); ++ fprintf(stdout," --noath disable the ATH for masking\n"); ++ fprintf(stdout," --noshort do not use short blocks\n"); ++ fprintf(stdout," --voice experimental voice mode\n"); ++ fprintf(stdout," --preset type type must be phone, voice, fm, tape, hifi, cd or studio\n"); ++ fprintf(stdout," help gives some more infos on these\n"); ++ fprintf(stdout,"\n"); ++ fprintf(stdout," CBR (constant bitrate, the default) options:\n"); ++ fprintf(stdout," -h higher quality, but a little slower. Recommended.\n"); ++ fprintf(stdout," -f fast mode (very low quality)\n"); ++ fprintf(stdout," -b bitrate set the bitrate, default 128kbps\n"); ++ fprintf(stdout,"\n"); ++ fprintf(stdout," VBR options:\n"); ++ fprintf(stdout," -v use variable bitrate (VBR)\n"); ++ fprintf(stdout," -V n quality setting for VBR. default n=%i\n",gfp->VBR_q); ++ fprintf(stdout," 0=high quality,bigger files. 9=smaller files\n"); ++ fprintf(stdout," -b bitrate specify minimum allowed bitrate, default 32kbs\n"); ++ fprintf(stdout," -B bitrate specify maximum allowed bitrate, default 256kbs\n"); ++ fprintf(stdout," -t disable Xing VBR informational tag\n"); ++ fprintf(stdout," --nohist disable VBR histogram display\n"); ++ fprintf(stdout,"\n"); ++ fprintf(stdout," MP3 header/stream options:\n"); ++ fprintf(stdout," -e emp de-emphasis n/5/c (obsolete)\n"); ++ fprintf(stdout," -c mark as copyright\n"); ++ fprintf(stdout," -o mark as non-original\n"); ++ fprintf(stdout," -p error protection. adds 16bit checksum to every frame\n"); ++ fprintf(stdout," (the checksum is computed correctly)\n"); ++ fprintf(stdout," --nores disable the bit reservoir\n"); ++ fprintf(stdout,"\n"); ++ fprintf(stdout," Specifying any of the following options will add an ID3 tag:\n"); ++ fprintf(stdout," --tt \"title\" title of song (max 30 chars)\n"); ++ fprintf(stdout," --ta \"artist\" artist who did the song (max 30 chars)\n"); ++ fprintf(stdout," --tl \"album\" album where it came from (max 30 chars)\n"); ++ fprintf(stdout," --ty \"year\" year in which the song/album was made (max 4 chars)\n"); ++ fprintf(stdout," --tc \"comment\" additional info (max 30 chars)\n"); ++ fprintf(stdout," (or max 28 chars if using the \"track\" option)\n"); ++ fprintf(stdout," --tn \"track\" track number of the song on the CD (1 to 99)\n"); ++ fprintf(stdout," (using this option will add an ID3v1.1 tag)\n"); ++ fprintf(stdout," --tg \"genre\" genre of song (name or number)\n"); ++ fprintf(stdout,"\n"); ++ #ifdef HAVEGTK ++ fprintf(stdout," -g run graphical analysis on <infile>\n"); ++ #endif ++ display_bitrates(stdout); ++ exit(0); ++ } ++ ++ ++ ++ /************************************************************************ ++ * ++ * usage ++ * ++ * PURPOSE: Writes presetting info to #stdout# ++ * ++ ************************************************************************/ ++ ++ void lame_presets_info(lame_global_flags *gfp,char *name) /* print syntax & exit */ ++ { ++ lame_print_version(stdout); ++ fprintf(stdout,"\n"); ++ fprintf(stdout,"Presets are some shortcuts for common settings.\n"); ++ fprintf(stdout,"They can be combined with -v if you want VBR MP3s.\n"); ++ fprintf(stdout,"\n"); ++ fprintf(stdout," --preset phone => --resample 16\n"); ++ fprintf(stdout," --highpass 0.260\n"); ++ fprintf(stdout," --highpasswidth 0.040\n"); ++ fprintf(stdout," --lowpass 3.700\n"); ++ fprintf(stdout," --lowpasswidth 0.300\n"); ++ fprintf(stdout," --noshort\n"); ++ fprintf(stdout," -m m\n"); ++ fprintf(stdout," -b 16\n"); ++ fprintf(stdout," plus -b 8 \\\n"); ++ fprintf(stdout," -B 56 > in combination with -v\n"); ++ fprintf(stdout," -V 5 /\n"); ++ fprintf(stdout,"\n"); ++ fprintf(stdout," --preset voice: => --resample 24\n"); ++ fprintf(stdout," --highpass 0.100\n"); ++ fprintf(stdout," --highpasswidth 0.020\n"); ++ fprintf(stdout," --lowpass 11\n"); ++ fprintf(stdout," --lowpasswidth 2\n"); ++ fprintf(stdout," --noshort\n"); ++ fprintf(stdout," -m m\n"); ++ fprintf(stdout," -b 32\n"); ++ fprintf(stdout," plus -b 8 \\\n"); ++ fprintf(stdout," -B 96 > in combination with -v\n"); ++ fprintf(stdout," -V 4 /\n"); ++ fprintf(stdout,"\n"); ++ fprintf(stdout," --preset fm: => --resample 32\n"); ++ fprintf(stdout," --highpass 0.030\n"); ++ fprintf(stdout," --highpasswidth 0\n"); ++ fprintf(stdout," --lowpass 11.4\n"); ++ fprintf(stdout," --lowpasswidth 0\n"); ++ fprintf(stdout," -m j\n"); ++ fprintf(stdout," -b 96\n"); ++ fprintf(stdout," plus -b 32 \\\n"); ++ fprintf(stdout," -B 192 > in combination with -v\n"); ++ fprintf(stdout," -V 4 /\n"); ++ fprintf(stdout,"\n"); ++ fprintf(stdout," --preset tape: => --lowpass 17\n"); ++ fprintf(stdout," --lowpasswidth 2\n"); ++ fprintf(stdout," --highpass 0.015\n"); ++ fprintf(stdout," --highpasswidth 0.015\n"); ++ fprintf(stdout," -m j\n"); ++ fprintf(stdout," -b 128\n"); ++ fprintf(stdout," plus -b 32 \\\n"); ++ fprintf(stdout," -B 192 > in combination with -v\n"); ++ fprintf(stdout," -V 4 /\n"); ++ fprintf(stdout,"\n"); ++ fprintf(stdout," --preset hifi: => --lowpass 20\n"); ++ fprintf(stdout," --lowpasswidth 3\n"); ++ fprintf(stdout," --highpass 0.015\n"); ++ fprintf(stdout," --highpasswidth 0.015\n"); ++ fprintf(stdout," -h\n"); ++ fprintf(stdout," -m j\n"); ++ fprintf(stdout," -b 160\n"); ++ fprintf(stdout," plus -b 32 \\\n"); ++ fprintf(stdout," -B 224 > in combination with -v\n"); ++ fprintf(stdout," -V 3 /\n"); ++ fprintf(stdout,"\n"); ++ fprintf(stdout," --preset cd: => -k\n"); ++ fprintf(stdout," -h\n"); ++ fprintf(stdout," -m s\n"); ++ fprintf(stdout," -b 192\n"); ++ fprintf(stdout," plus -b 80 \\\n"); ++ fprintf(stdout," -B 256 > in combination with -v\n"); ++ fprintf(stdout," -V 2 /\n"); ++ fprintf(stdout,"\n"); ++ fprintf(stdout," --preset studio: => -k\n"); ++ fprintf(stdout," -h\n"); ++ fprintf(stdout," -m s\n"); ++ fprintf(stdout," -b 256\n"); ++ fprintf(stdout," plus -b 112 \\\n"); ++ fprintf(stdout," -B 320 > in combination with -v\n"); ++ fprintf(stdout," -V 0 /\n"); ++ fprintf(stdout,"\n"); ++ ++ exit(0); ++ } ++ ++ ++ ++ /************************************************************************ ++ * ++ * parse_args ++ * ++ * PURPOSE: Sets encoding parameters to the specifications of the ++ * command line. Default settings are used for parameters ++ * not specified in the command line. ++ * ++ * If the input file is in WAVE or AIFF format, the sampling frequency is read ++ * from the AIFF header. ++ * ++ * The input and output filenames are read into #inpath# and #outpath#. ++ * ++ ************************************************************************/ ++ void lame_parse_args(lame_global_flags *gfp,int argc, char **argv) ++ { ++ FLOAT srate; ++ int err = 0, i = 0; ++ int autoconvert=0; ++ int user_quality=0; ++ ++ char *programName = argv[0]; ++ int track = 0; ++ ++ inPath[0] = '\0'; ++ outPath[0] = '\0'; ++ gfp->inPath=inPath; ++ gfp->outPath=outPath; ++ ++ id3_inittag(&id3tag); ++ id3tag.used = 0; ++ ++ /* process args */ ++ while(++i<argc && err == 0) { ++ char c, *token, *arg, *nextArg; ++ int argUsed; ++ ++ token = argv[i]; ++ if(*token++ == '-') { ++ if(i+1 < argc) nextArg = argv[i+1]; ++ else nextArg = ""; ++ argUsed = 0; ++ if (! *token) { ++ /* The user wants to use stdin and/or stdout. */ ++ if(inPath[0] == '\0') strncpy(inPath, argv[i],MAX_NAME_SIZE); ++ else if(outPath[0] == '\0') strncpy(outPath, argv[i],MAX_NAME_SIZE); ++ } ++ if (*token == '-') { ++ /* GNU style */ ++ token++; ++ ++ if (strcmp(token, "resample")==0) { ++ argUsed=1; ++ srate = atof( nextArg ); ++ /* samplerate = rint( 1000.0 * srate ); $A */ ++ gfp->out_samplerate = (( 1000.0 * srate ) + 0.5); ++ if (srate < 1) { ++ fprintf(stderr,"Must specify a samplerate with --resample\n"); ++ exit(1); ++ } ++ } ++ else if (strcmp(token, "mp3input")==0) { ++ gfp->input_format=sf_mp3; ++ } ++ else if (strcmp(token, "voice")==0) { ++ gfp->lowpassfreq=12000; ++ gfp->VBR_max_bitrate_kbps=160; ++ gfp->no_short_blocks=1; ++ } ++ else if (strcmp(token, "noshort")==0) { ++ gfp->no_short_blocks=1; ++ } ++ else if (strcmp(token, "noath")==0) { ++ gfp->noATH=1; ++ } ++ else if (strcmp(token, "nores")==0) { ++ gfp->disable_reservoir=1; ++ gfp->padding=0; ++ } ++ else if (strcmp(token, "athonly")==0) { ++ gfp->ATHonly=1; ++ } ++ else if (strcmp(token, "nohist")==0) { ++ #ifdef BRHIST ++ disp_brhist = 0; ++ #endif ++ } ++ /* options for ID3 tag */ ++ else if (strcmp(token, "tt")==0) { ++ id3tag.used=1; argUsed = 1; ++ strncpy(id3tag.title, nextArg, 30); ++ } ++ else if (strcmp(token, "ta")==0) { ++ id3tag.used=1; argUsed = 1; ++ strncpy(id3tag.artist, nextArg, 30); ++ } ++ else if (strcmp(token, "tl")==0) { ++ id3tag.used=1; argUsed = 1; ++ strncpy(id3tag.album, nextArg, 30); ++ } ++ else if (strcmp(token, "ty")==0) { ++ id3tag.used=1; argUsed = 1; ++ strncpy(id3tag.year, nextArg, 4); ++ } ++ else if (strcmp(token, "tc")==0) { ++ id3tag.used=1; argUsed = 1; ++ strncpy(id3tag.comment, nextArg, 30); ++ } ++ else if (strcmp(token, "tn")==0) { ++ id3tag.used=1; argUsed = 1; ++ track = atoi(nextArg); ++ if (track < 1) { track = 1; } ++ if (track > 99) { track = 99; } ++ id3tag.track = track; ++ } ++ else if (strcmp(token, "tg")==0) { ++ argUsed = strtol (nextArg, &token, 10); ++ if (nextArg==token) { ++ /* Genere was given as a string, so it's number*/ ++ for (argUsed=0; argUsed<=genre_last; argUsed++) { ++ if (!strcmp (genre_list[argUsed], nextArg)) { break; } ++ } ++ } ++ if (argUsed>genre_last) { ++ argUsed=255; ++ fprintf(stderr,"Unknown genre: %s. Specifiy genre number \n", nextArg); ++ } ++ argUsed &= 255; c=(char)(argUsed); ++ ++ id3tag.used=1; argUsed = 1; ++ strncpy(id3tag.genre, &c, 1); ++ } ++ else if (strcmp(token, "lowpass")==0) { ++ argUsed=1; ++ gfp->lowpassfreq = (( 1000.0 * atof( nextArg ) ) + 0.5); ++ if (gfp->lowpassfreq < 1) { ++ fprintf(stderr,"Must specify lowpass with --lowpass freq, freq >= 0.001 kHz\n"); ++ exit(1); ++ } ++ } ++ else if (strcmp(token, "lowpass-width")==0) { ++ argUsed=1; ++ gfp->lowpasswidth = (( 1000.0 * atof( nextArg ) ) + 0.5); ++ if (gfp->lowpasswidth < 0) { ++ fprintf(stderr,"Must specify lowpass width with --lowpass-width freq, freq >= 0 kHz\n"); ++ exit(1); ++ } ++ } ++ else if (strcmp(token, "highpass")==0) { ++ argUsed=1; ++ gfp->highpassfreq = (( 1000.0 * atof( nextArg ) ) + 0.5); ++ if (gfp->highpassfreq < 1) { ++ fprintf(stderr,"Must specify highpass with --highpass freq, freq >= 0.001 kHz\n"); ++ exit(1); ++ } ++ } ++ else if (strcmp(token, "highpass-width")==0) { ++ argUsed=1; ++ gfp->highpasswidth = (( 1000.0 * atof( nextArg ) ) + 0.5); ++ if (gfp->highpasswidth < 0) { ++ fprintf(stderr,"Must specify highpass width with --highpass-width freq, freq >= 0 kHz\n"); ++ exit(1); ++ } ++ } ++ else if (strcmp(token, "cwlimit")==0) { ++ argUsed=1; ++ gfp->cwlimit = atof( nextArg ); ++ if (gfp->cwlimit <= 0 ) { ++ fprintf(stderr,"Must specify cwlimit in kHz\n"); ++ exit(1); ++ } ++ } /* some more GNU-ish options could be added ++ * version => complete name, version and license info (normal exit) ++ * quiet/silent => no messages on screen ++ * brief => few messages on screen (name, status report) ++ * verbose => all infos to screen (brhist, internal flags/filters) ++ * o/output file => specifies output filename ++ * O => stdout ++ * i/input file => specifies input filename ++ * I => stdin ++ */ ++ else if (strcmp(token, "help") ==0 ++ ||strcmp(token, "usage")==0){ ++ lame_help(gfp,programName); /* doesn't return */ ++ } ++ else if (strcmp(token, "preset")==0) { ++ argUsed=1; ++ if (strcmp(nextArg,"phone")==0) ++ { /* when making changes, please update help text too */ ++ gfp->brate = 16; ++ gfp->highpassfreq=260; ++ gfp->highpasswidth=40; ++ gfp->lowpassfreq=3700; ++ gfp->lowpasswidth=300; ++ gfp->VBR_q=5; ++ gfp->VBR_min_bitrate_kbps=8; ++ gfp->VBR_max_bitrate_kbps=56; ++ gfp->no_short_blocks=1; ++ gfp->out_samplerate = 16000; ++ gfp->mode = MPG_MD_MONO; ++ gfp->mode_fixed = 1; ++ gfp->quality = 5; ++ } ++ else if (strcmp(nextArg,"voice")==0) ++ { /* when making changes, please update help text too */ ++ gfp->brate = 56; ++ gfp->highpassfreq=100; ++ gfp->highpasswidth=20; ++ gfp->lowpasswidth=2000; ++ gfp->lowpassfreq=11000; ++ gfp->VBR_q=4; ++ gfp->VBR_min_bitrate_kbps=8; ++ gfp->VBR_max_bitrate_kbps=96; ++ gfp->no_short_blocks=1; ++ gfp->mode = MPG_MD_MONO; ++ gfp->mode_fixed = 1; ++ gfp->out_samplerate = 24000; ++ gfp->quality = 5; ++ } ++ else if (strcmp(nextArg,"fm")==0) ++ { /* when making changes, please update help text too */ ++ gfp->brate = 96; ++ gfp->highpassfreq=30; ++ gfp->highpasswidth=0; ++ gfp->lowpassfreq=15000; ++ gfp->lowpasswidth=0; ++ gfp->VBR_q=4; ++ gfp->VBR_min_bitrate_kbps=32; ++ gfp->VBR_max_bitrate_kbps=192; ++ gfp->mode = MPG_MD_JOINT_STEREO; ++ gfp->mode_fixed = 1; ++ /*gfp->out_samplerate = 32000; */ /* determined automatically based on bitrate & sample freq. */ ++ gfp->quality = 5; ++ } ++ else if (strcmp(nextArg,"tape")==0) ++ { /* when making changes, please update help text too */ ++ gfp->brate = 128; ++ gfp->highpassfreq=15; ++ gfp->highpasswidth=15; ++ gfp->lowpassfreq=17000; ++ gfp->lowpasswidth=2000; ++ gfp->VBR_q=4; ++ gfp->VBR_min_bitrate_kbps=32; ++ gfp->VBR_max_bitrate_kbps=192; ++ gfp->mode = MPG_MD_JOINT_STEREO; ++ gfp->mode_fixed = 1; ++ gfp->quality = 5; ++ } ++ else if (strcmp(nextArg,"hifi")==0) ++ { /* when making changes, please update help text too */ ++ gfp->brate = 160; ++ gfp->highpassfreq=15; ++ gfp->highpasswidth=15; ++ gfp->lowpassfreq=20000; ++ gfp->lowpasswidth=3000; ++ gfp->VBR_q=3; ++ gfp->VBR_min_bitrate_kbps=32; ++ gfp->VBR_max_bitrate_kbps=224; ++ gfp->mode = MPG_MD_JOINT_STEREO; ++ gfp->mode_fixed = 1; ++ gfp->quality = 2; ++ } ++ else if (strcmp(nextArg,"cd")==0) ++ { /* when making changes, please update help text too */ ++ gfp->brate = 192; ++ gfp->lowpassfreq=-1; ++ gfp->highpassfreq=-1; ++ gfp->VBR_q=2; ++ gfp->VBR_min_bitrate_kbps=80; ++ gfp->VBR_max_bitrate_kbps=256; ++ gfp->mode = MPG_MD_STEREO; ++ gfp->mode_fixed = 1; ++ gfp->quality = 2; ++ } ++ else if (strcmp(nextArg,"studio")==0) ++ { /* when making changes, please update help text too */ ++ gfp->brate = 256; ++ gfp->lowpassfreq=-1; ++ gfp->highpassfreq=-1; ++ gfp->VBR_q=0; ++ gfp->VBR_min_bitrate_kbps=112; ++ gfp->VBR_max_bitrate_kbps=320; ++ gfp->mode = MPG_MD_STEREO; ++ gfp->mode_fixed = 1; ++ gfp->quality = 2; /* should be 0, but does not work now */ ++ } ++ else if (strcmp(nextArg,"help")==0) ++ { ++ lame_presets_info(gfp,programName); /* doesn't return */ ++ } ++ else ++ { ++ fprintf(stderr,"%s: --preset type, type must be phone, voice, fm, tape, hifi, cd or studio, not %s\n", ++ programName, nextArg); ++ exit(1); ++ } ++ } /* --preset */ ++ else ++ { ++ fprintf(stderr,"%s: unrec option --%s\n", ++ programName, token); ++ } ++ i += argUsed; ++ ++ } else while( (c = *token++) ) { ++ if(*token ) arg = token; ++ else arg = nextArg; ++ switch(c) { ++ case 'm': argUsed = 1; gfp->mode_fixed = 1; ++ if (*arg == 's') ++ { gfp->mode = MPG_MD_STEREO; } ++ else if (*arg == 'd') ++ { gfp->mode = MPG_MD_DUAL_CHANNEL; } ++ else if (*arg == 'j') ++ { gfp->mode = MPG_MD_JOINT_STEREO; } ++ else if (*arg == 'f') ++ { gfp->mode = MPG_MD_JOINT_STEREO; gfp->force_ms=1; } ++ else if (*arg == 'm') ++ { gfp->mode = MPG_MD_MONO; } ++ else { ++ fprintf(stderr,"%s: -m mode must be s/d/j/f/m not %s\n", ++ programName, arg); ++ err = 1; ++ } ++ break; ++ case 'V': argUsed = 1; gfp->VBR = 1; ++ gfp->VBR_q = atoi(arg); ++ if (gfp->VBR_q <0) gfp->VBR_q=0; ++ if (gfp->VBR_q >9) gfp->VBR_q=9; ++ break; ++ case 'q': argUsed = 1; ++ user_quality = atoi(arg); ++ if (user_quality<0) user_quality=0; ++ if (user_quality>9) user_quality=9; ++ break; ++ case 's': ++ argUsed = 1; ++ srate = atof( arg ); ++ /* samplerate = rint( 1000.0 * srate ); $A */ ++ gfp->in_samplerate = (( 1000.0 * srate ) + 0.5); ++ break; ++ case 'b': ++ argUsed = 1; ++ gfp->brate = atoi(arg); ++ gfp->VBR_min_bitrate_kbps=gfp->brate; ++ break; ++ case 'B': ++ argUsed = 1; ++ gfp->VBR_max_bitrate_kbps=atoi(arg); ++ break; ++ case 't': /* dont write VBR tag */ ++ gfp->bWriteVbrTag=0; ++ break; ++ case 'r': /* force raw pcm input file */ ++ #ifdef LIBSNDFILE ++ fprintf(stderr,"WARNING: libsndfile may ignore -r and perform fseek's on the input.\n"); ++ fprintf(stderr,"Compile without libsndfile if this is a problem.\n"); ++ #endif ++ gfp->input_format=sf_raw; ++ break; ++ case 'x': /* force byte swapping */ ++ gfp->swapbytes=TRUE; ++ break; ++ case 'p': /* (jo) error_protection: add crc16 information to stream */ ++ gfp->error_protection = 1; ++ break; ++ case 'a': /* autoconvert input file from stereo to mono - for mono mp3 encoding */ ++ autoconvert=1; ++ gfp->mode=MPG_MD_MONO; ++ gfp->mode_fixed=1; ++ break; ++ case 'h': ++ gfp->quality = 2; ++ break; ++ case 'k': ++ gfp->lowpassfreq=-1; ++ gfp->highpassfreq=-1; ++ break; ++ case 'd': ++ gfp->allow_diff_short = 1; ++ break; ++ case 'v': ++ gfp->VBR = 1; ++ break; ++ case 'S': ++ gfp->silent = TRUE; ++ break; ++ case 'X': argUsed = 1; gfp->experimentalX = 0; ++ if (*arg == '0') ++ { gfp->experimentalX=0; } ++ else if (*arg == '1') ++ { gfp->experimentalX=1; } ++ else if (*arg == '2') ++ { gfp->experimentalX=2; } ++ else if (*arg == '3') ++ { gfp->experimentalX=3; } ++ else if (*arg == '4') ++ { gfp->experimentalX=4; } ++ else if (*arg == '5') ++ { gfp->experimentalX=5; } ++ else if (*arg == '6') ++ { gfp->experimentalX=6; } ++ else { ++ fprintf(stderr,"%s: -X n must be 0-6, not %s\n", ++ programName, arg); ++ err = 1; ++ } ++ break; ++ ++ ++ case 'Y': ++ gfp->experimentalY = TRUE; ++ break; ++ case 'Z': ++ gfp->experimentalZ = TRUE; ++ break; ++ case 'f': ++ gfp->quality= 9; ++ break; ++ case 'g': /* turn on gtk analysis */ ++ #ifdef HAVEGTK ++ gfp->gtkflag = TRUE; ++ #else ++ fprintf(stderr,"LAME not compiled with GTK support, -g not supported.\n", ++ programName, arg); ++ #endif ++ break; ++ ++ case 'e': argUsed = 1; ++ if (*arg == 'n') gfp->emphasis = 0; ++ else if (*arg == '5') gfp->emphasis = 1; ++ else if (*arg == 'c') gfp->emphasis = 3; ++ else { ++ fprintf(stderr,"%s: -e emp must be n/5/c not %s\n", ++ programName, arg); ++ err = 1; ++ } ++ break; ++ case 'c': gfp->copyright = 1; break; ++ case 'o': gfp->original = 0; break; ++ ++ case '?': lame_help(gfp,programName); /* doesn't return */ ++ default: fprintf(stderr,"%s: unrec option %c\n", ++ programName, c); ++ err = 1; break; ++ } ++ if(argUsed) { ++ if(arg == token) token = ""; /* no more from token */ ++ else ++i; /* skip arg we used */ ++ arg = ""; argUsed = 0; ++ } ++ } ++ } else { ++ if(inPath[0] == '\0') strncpy(inPath, argv[i], MAX_NAME_SIZE); ++ else if(outPath[0] == '\0') strncpy(outPath, argv[i], MAX_NAME_SIZE); ++ else { ++ fprintf(stderr,"%s: excess arg %s\n", programName, argv[i]); ++ err = 1; ++ } ++ } ++ } /* loop over args */ ++ ++ ++ ++ if(err || inPath[0] == '\0') lame_usage(gfp,programName); /* never returns */ ++ if (inPath[0]=='-') gfp->silent=1; /* turn off status - it's broken for stdin */ ++ if(outPath[0] == '\0') { ++ if (inPath[0]=='-') { ++ /* if input is stdin, default output is stdout */ ++ strcpy(outPath,"-"); ++ }else { ++ strncpy(outPath, inPath, MAX_NAME_SIZE - 4); ++ strncat(outPath, ".mp3", 4 ); ++ } ++ } ++ /* some file options not allowed with stdout */ ++ if (outPath[0]=='-') { ++ gfp->bWriteVbrTag=0; /* turn off VBR tag */ ++ if (id3tag.used) { ++ id3tag.used=0; /* turn of id3 tagging */ ++ fprintf(stderr,"id3tag ignored: id3 tagging not supported for stdout.\n"); ++ } ++ } ++ ++ ++ /* if user did not explicitly specify input is mp3, check file name */ ++ if (gfp->input_format != sf_mp3) ++ if (!(strcmp((char *) &inPath[strlen(inPath)-4],".mp3"))) ++ gfp->input_format = sf_mp3; ++ ++ #if !(defined HAVEMPGLIB || defined AMIGA_MPEGA) ++ if (gfp->input_format == sf_mp3) { ++ fprintf(stderr,"Error: libmp3lame not compiled with mp3 *decoding* support \n"); ++ exit(1); ++ } ++ #endif ++ /* default guess for number of channels */ ++ if (autoconvert) gfp->num_channels=2; ++ else if (gfp->mode == MPG_MD_MONO) gfp->num_channels=1; ++ else gfp->num_channels=2; ++ ++ /* user specified a quality value. override any defaults set above */ ++ if (user_quality) gfp->quality=user_quality; ++ ++ } ++ ++ ++ ++ #endif +diff -r -c -N encoder/portableio.c lame3.70/portableio.c +*** encoder/portableio.c Wed Jan 22 02:43:12 1997 +--- lame3.70/portableio.c Wed Nov 24 01:43:35 1999 +*************** +*** 43,51 **** + * conversions, and accommodated conversions involving +/- infinity, + * NaN's, and denormalized numbers. + * +! * $Id: portableio.c,v 2.6 1991/04/30 17:06:02 malcolm Exp $ + * + * $Log: portableio.c,v $ + * Revision 2.6 91/04/30 17:06:02 malcolm + */ + +--- 43,55 ---- + * conversions, and accommodated conversions involving +/- infinity, + * NaN's, and denormalized numbers. + * +! * $Id: portableio.c,v 1.1.1.1 1999/11/24 08:43:35 markt Exp $ + * + * $Log: portableio.c,v $ ++ * Revision 1.1.1.1 1999/11/24 08:43:35 markt ++ * initial checkin of LAME ++ * Starting with LAME 3.57beta with some modifications ++ * + * Revision 2.6 91/04/30 17:06:02 malcolm + */ + +*************** +*** 59,66 **** + + + int +! ReadByte(fp) +! FILE *fp; + { + int result; + +--- 63,69 ---- + + + int +! ReadByte(FILE *fp) + { + int result; + +*************** +*** 72,79 **** + + + int +! Read16BitsLowHigh(fp) +! FILE *fp; + { + int first, second, result; + +--- 75,81 ---- + + + int +! Read16BitsLowHigh(FILE *fp) + { + int first, second, result; + +*************** +*** 90,97 **** + + + int +! Read16BitsHighLow(fp) +! FILE *fp; + { + int first, second, result; + +--- 92,98 ---- + + + int +! Read16BitsHighLow(FILE *fp) + { + int first, second, result; + +*************** +*** 108,125 **** + + + void +! Write8Bits(fp, i) +! FILE *fp; +! int i; + { + putc(i&0xff,fp); + } + + + void +! Write16BitsLowHigh(fp, i) +! FILE *fp; +! int i; + { + putc(i&0xff,fp); + putc((i>>8)&0xff,fp); +--- 109,122 ---- + + + void +! Write8Bits(FILE *fp, int i) + { + putc(i&0xff,fp); + } + + + void +! Write16BitsLowHigh(FILE *fp, int i) + { + putc(i&0xff,fp); + putc((i>>8)&0xff,fp); +*************** +*** 127,135 **** + + + void +! Write16BitsHighLow(fp, i) +! FILE *fp; +! int i; + { + putc((i>>8)&0xff,fp); + putc(i&0xff,fp); +--- 124,130 ---- + + + void +! Write16BitsHighLow(FILE *fp, int i) + { + putc((i>>8)&0xff,fp); + putc(i&0xff,fp); +*************** +*** 137,144 **** + + + int +! Read24BitsHighLow(fp) +! FILE *fp; + { + int first, second, third; + int result; +--- 132,138 ---- + + + int +! Read24BitsHighLow(FILE *fp) + { + int first, second, third; + int result; +*************** +*** 157,164 **** + + + int +! Read32Bits(fp) +! FILE *fp; + { + int first, second, result; + +--- 151,157 ---- + + + int +! Read32Bits(FILE *fp) + { + int first, second, result; + +*************** +*** 175,182 **** + + + int +! Read32BitsHighLow(fp) +! FILE *fp; + { + int first, second, result; + +--- 168,174 ---- + + + int +! Read32BitsHighLow(FILE *fp) + { + int first, second, result; + +*************** +*** 193,201 **** + + + void +! Write32Bits(fp, i) +! FILE *fp; +! int i; + { + Write16BitsLowHigh(fp,(int)(i&0xffffL)); + Write16BitsLowHigh(fp,(int)((i>>16)&0xffffL)); +--- 185,191 ---- + + + void +! Write32Bits(FILE *fp, int i) + { + Write16BitsLowHigh(fp,(int)(i&0xffffL)); + Write16BitsLowHigh(fp,(int)((i>>16)&0xffffL)); +*************** +*** 203,211 **** + + + void +! Write32BitsLowHigh(fp, i) +! FILE *fp; +! int i; + { + Write16BitsLowHigh(fp,(int)(i&0xffffL)); + Write16BitsLowHigh(fp,(int)((i>>16)&0xffffL)); +--- 193,199 ---- + + + void +! Write32BitsLowHigh(FILE *fp, int i) + { + Write16BitsLowHigh(fp,(int)(i&0xffffL)); + Write16BitsLowHigh(fp,(int)((i>>16)&0xffffL)); +*************** +*** 213,243 **** + + + void +! Write32BitsHighLow(fp, i) +! FILE *fp; +! int i; + { + Write16BitsHighLow(fp,(int)((i>>16)&0xffffL)); + Write16BitsHighLow(fp,(int)(i&0xffffL)); + } + +! void ReadBytes(fp, p, n) +! FILE *fp; +! char *p; +! int n; + { +! while (!feof(fp) & n-- > 0) + *p++ = getc(fp); + } + +! void ReadBytesSwapped(fp, p, n) +! FILE *fp; +! char *p; +! int n; + { + register char *q = p; + +! while (!feof(fp) & n-- > 0) + *q++ = getc(fp); + + for (q--; p < q; p++, q--){ +--- 201,223 ---- + + + void +! Write32BitsHighLow(FILE *fp, int i) + { + Write16BitsHighLow(fp,(int)((i>>16)&0xffffL)); + Write16BitsHighLow(fp,(int)(i&0xffffL)); + } + +! void ReadBytes(FILE *fp, char *p, int n) + { +! while (!feof(fp) & (n-- > 0)) + *p++ = getc(fp); + } + +! void ReadBytesSwapped(FILE *fp, char *p, int n) + { + register char *q = p; + +! while (!feof(fp) & (n-- > 0)) + *q++ = getc(fp); + + for (q--; p < q; p++, q--){ +*************** +*** 247,265 **** + } + } + +! void WriteBytes(fp, p, n) +! FILE *fp; +! char *p; +! int n; + { + while (n-- > 0) + putc(*p++, fp); + } + +! void WriteBytesSwapped(fp, p, n) +! FILE *fp; +! char *p; +! int n; + { + p += n-1; + while (n-- > 0) +--- 227,239 ---- + } + } + +! void WriteBytes(FILE *fp, char *p, int n) + { + while (n-- > 0) + putc(*p++, fp); + } + +! void WriteBytesSwapped(FILE *fp, char *p, int n) + { + p += n-1; + while (n-- > 0) +*************** +*** 267,274 **** + } + + defdouble +! ReadIeeeFloatHighLow(fp) +! FILE *fp; + { + char bits[kFloatLength]; + +--- 241,247 ---- + } + + defdouble +! ReadIeeeFloatHighLow(FILE *fp) + { + char bits[kFloatLength]; + +*************** +*** 277,284 **** + } + + defdouble +! ReadIeeeFloatLowHigh(fp) +! FILE *fp; + { + char bits[kFloatLength]; + +--- 250,256 ---- + } + + defdouble +! ReadIeeeFloatLowHigh(FILE *fp) + { + char bits[kFloatLength]; + +*************** +*** 287,294 **** + } + + defdouble +! ReadIeeeDoubleHighLow(fp) +! FILE *fp; + { + char bits[kDoubleLength]; + +--- 259,265 ---- + } + + defdouble +! ReadIeeeDoubleHighLow(FILE *fp) + { + char bits[kDoubleLength]; + +*************** +*** 297,304 **** + } + + defdouble +! ReadIeeeDoubleLowHigh(fp) +! FILE *fp; + { + char bits[kDoubleLength]; + +--- 268,274 ---- + } + + defdouble +! ReadIeeeDoubleLowHigh(FILE *fp) + { + char bits[kDoubleLength]; + +*************** +*** 307,314 **** + } + + defdouble +! ReadIeeeExtendedHighLow(fp) +! FILE *fp; + { + char bits[kExtendedLength]; + +--- 277,283 ---- + } + + defdouble +! ReadIeeeExtendedHighLow(FILE *fp) + { + char bits[kExtendedLength]; + +*************** +*** 317,324 **** + } + + defdouble +! ReadIeeeExtendedLowHigh(fp) +! FILE *fp; + { + char bits[kExtendedLength]; + +--- 286,292 ---- + } + + defdouble +! ReadIeeeExtendedLowHigh(FILE *fp) + { + char bits[kExtendedLength]; + +*************** +*** 327,335 **** + } + + void +! WriteIeeeFloatLowHigh(fp, num) +! FILE *fp; +! defdouble num; + { + char bits[kFloatLength]; + +--- 295,301 ---- + } + + void +! WriteIeeeFloatLowHigh(FILE *fp, defdouble num) + { + char bits[kFloatLength]; + +*************** +*** 338,346 **** + } + + void +! WriteIeeeFloatHighLow(fp, num) +! FILE *fp; +! defdouble num; + { + char bits[kFloatLength]; + +--- 304,310 ---- + } + + void +! WriteIeeeFloatHighLow(FILE *fp, defdouble num) + { + char bits[kFloatLength]; + +*************** +*** 349,357 **** + } + + void +! WriteIeeeDoubleLowHigh(fp, num) +! FILE *fp; +! defdouble num; + { + char bits[kDoubleLength]; + +--- 313,319 ---- + } + + void +! WriteIeeeDoubleLowHigh(FILE *fp, defdouble num) + { + char bits[kDoubleLength]; + +*************** +*** 360,368 **** + } + + void +! WriteIeeeDoubleHighLow(fp, num) +! FILE *fp; +! defdouble num; + { + char bits[kDoubleLength]; + +--- 322,328 ---- + } + + void +! WriteIeeeDoubleHighLow(FILE *fp, defdouble num) + { + char bits[kDoubleLength]; + +*************** +*** 371,379 **** + } + + void +! WriteIeeeExtendedLowHigh(fp, num) +! FILE *fp; +! defdouble num; + { + char bits[kExtendedLength]; + +--- 331,337 ---- + } + + void +! WriteIeeeExtendedLowHigh(FILE *fp, defdouble num) + { + char bits[kExtendedLength]; + +*************** +*** 383,391 **** + + + void +! WriteIeeeExtendedHighLow(fp, num) +! FILE *fp; +! defdouble num; + { + char bits[kExtendedLength]; + +--- 341,347 ---- + + + void +! WriteIeeeExtendedHighLow(FILE *fp, defdouble num) + { + char bits[kExtendedLength]; + +diff -r -c -N encoder/portableio.h lame3.70/portableio.h +*** encoder/portableio.h Wed Jan 22 02:43:15 1997 +--- lame3.70/portableio.h Wed Nov 24 01:43:37 1999 +*************** +*** 46,54 **** + * conversions, and accommodated conversions involving +/- infinity, + * NaN's, and denormalized numbers. + * +! * $Id: portableio.h,v 2.6 1991/04/30 17:06:02 malcolm Exp $ + * + * $Log: portableio.h,v $ + * Revision 2.6 91/04/30 17:06:02 malcolm + */ + +--- 46,58 ---- + * conversions, and accommodated conversions involving +/- infinity, + * NaN's, and denormalized numbers. + * +! * $Id: portableio.h,v 1.1.1.1 1999/11/24 08:43:37 markt Exp $ + * + * $Log: portableio.h,v $ ++ * Revision 1.1.1.1 1999/11/24 08:43:37 markt ++ * initial checkin of LAME ++ * Starting with LAME 3.57beta with some modifications ++ * + * Revision 2.6 91/04/30 17:06:02 malcolm + */ + +diff -r -c -N encoder/psy.c lame3.70/psy.c +*** encoder/psy.c Wed Jan 22 02:43:18 1997 +--- lame3.70/psy.c Wed Dec 31 17:00:00 1969 +*************** +*** 1,454 **** +- /********************************************************************** +- * ISO MPEG Audio Subgroup Software Simulation Group (1996) +- * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- * +- * $Id: psy.c,v 1.1 1996/02/14 04:04:23 rowlands Exp $ +- * +- * $Log: psy.c,v $ +- * Revision 1.1 1996/02/14 04:04:23 rowlands +- * Initial revision +- * +- * Received from Mike Coleman +- **********************************************************************/ +- /********************************************************************** +- * date programmers comment * +- * 2/25/91 Davis Pan start of version 1.0 records * +- * 5/10/91 W. Joseph Carter Ported to Macintosh and Unix. * +- * 7/10/91 Earle Jennings Ported to MsDos. * +- * replace of floats with FLOAT * +- * 2/11/92 W. Joseph Carter Fixed mem_alloc() arg for "absthr". * +- * 7/24/92 M. Iwadare HANN window coefficients modified. * +- * 7/27/92 Masahiro Iwadare Bug fix, FFT modification for Layer 3 * +- * 7/27/92 Masahiro Iwadare Bug fix, "new", "old", and "oldest" * +- * updates * +- * 8/07/92 Mike Coleman Bug fix, read_absthr() * +- * 95/3/21 Jon Rowlands Removed extra debug statements * +- **********************************************************************/ +- +- #include "common.h" +- #include "encoder.h" +- FILE *fpo; /* file pointer */ +- void psycho_anal(buffer,savebuf,chn,lay,snr32,sfreq) +- short int *buffer; +- short int savebuf[1056]; +- int chn, lay; +- FLOAT snr32[32]; +- double sfreq; /* to match prototype : float args are always double */ +- { +- unsigned int i, j, k; +- FLOAT r_prime, phi_prime; +- FLOAT freq_mult, bval_lo, minthres, sum_energy; +- double tb, temp1, temp2, temp3; +- +- /* The static variables "r", "phi_sav", "new", "old" and "oldest" have */ +- /* to be remembered for the unpredictability measure. For "r" and */ +- /* "phi_sav", the first index from the left is the channel select and */ +- /* the second index is the "age" of the data. */ +- +- static int new = 0, old = 1, oldest = 0; +- static int init = 0, flush, sync_flush, syncsize, sfreq_idx; +- +- /* The following static variables are constants. */ +- +- static double nmt = 5.5; +- +- static FLOAT crit_band[27] = {0, 100, 200, 300, 400, 510, 630, 770, +- 920, 1080, 1270,1480,1720,2000,2320, 2700, +- 3150, 3700, 4400,5300,6400,7700,9500,12000, +- 15500,25000,30000}; +- +- static FLOAT bmax[27] = {20.0, 20.0, 20.0, 20.0, 20.0, 17.0, 15.0, +- 10.0, 7.0, 4.4, 4.5, 4.5, 4.5, 4.5, +- 4.5, 4.5, 4.5, 4.5, 4.5, 4.5, 4.5, +- 4.5, 4.5, 4.5, 3.5, 3.5, 3.5}; +- +- /* The following pointer variables point to large areas of memory */ +- /* dynamically allocated by the mem_alloc() function. Dynamic memory */ +- /* allocation is used in order to avoid stack frame or data area */ +- /* overflow errors that otherwise would have occurred at compile time */ +- /* on the Macintosh computer. */ +- +- FLOAT *grouped_c, *grouped_e, *nb, *cb, *ecb, *bc; +- FLOAT *wsamp_r, *wsamp_i, *phi, *energy; +- FLOAT *c, *fthr; +- F32 *snrtmp; +- +- static int *numlines; +- static int *partition; +- static FLOAT *cbval, *rnorm; +- static FLOAT *window; +- static FLOAT *absthr; +- static double *tmn; +- static FCB *s; +- static FHBLK *lthr; +- static F2HBLK *r, *phi_sav; +- +- /* These dynamic memory allocations simulate "automatic" variables */ +- /* placed on the stack. For each mem_alloc() call here, there must be */ +- /* a corresponding mem_free() call at the end of this function. */ +- +- grouped_c = (FLOAT *) mem_alloc(sizeof(FCB), "grouped_c"); +- grouped_e = (FLOAT *) mem_alloc(sizeof(FCB), "grouped_e"); +- nb = (FLOAT *) mem_alloc(sizeof(FCB), "nb"); +- cb = (FLOAT *) mem_alloc(sizeof(FCB), "cb"); +- ecb = (FLOAT *) mem_alloc(sizeof(FCB), "ecb"); +- bc = (FLOAT *) mem_alloc(sizeof(FCB), "bc"); +- wsamp_r = (FLOAT *) mem_alloc(sizeof(FBLK), "wsamp_r"); +- wsamp_i = (FLOAT *) mem_alloc(sizeof(FBLK), "wsamp_i"); +- phi = (FLOAT *) mem_alloc(sizeof(FBLK), "phi"); +- energy = (FLOAT *) mem_alloc(sizeof(FBLK), "energy"); +- c = (FLOAT *) mem_alloc(sizeof(FHBLK), "c"); +- fthr = (FLOAT *) mem_alloc(sizeof(FHBLK), "fthr"); +- snrtmp = (F32 *) mem_alloc(sizeof(F2_32), "snrtmp"); +- +- if(init==0){ +- +- /* These dynamic memory allocations simulate "static" variables placed */ +- /* in the data space. Each mem_alloc() call here occurs only once at */ +- /* initialization time. The mem_free() function must not be called. */ +- +- numlines = (int *) mem_alloc(sizeof(ICB), "numlines"); +- partition = (int *) mem_alloc(sizeof(IHBLK), "partition"); +- fpo = fopen("out.dat", "wb"); +- if(fpo==NULL) { +- puts("\t The attempt to open the output file failed.\n"); +- exit(-1);} +- cbval = (FLOAT *) mem_alloc(sizeof(FCB), "cbval"); +- rnorm = (FLOAT *) mem_alloc(sizeof(FCB), "rnorm"); +- window = (FLOAT *) mem_alloc(sizeof(FBLK), "window"); +- absthr = (FLOAT *) mem_alloc(sizeof(FHBLK), "absthr"); +- tmn = (double *) mem_alloc(sizeof(DCB), "tmn"); +- s = (FCB *) mem_alloc(sizeof(FCBCB), "s"); +- lthr = (FHBLK *) mem_alloc(sizeof(F2HBLK), "lthr"); +- r = (F2HBLK *) mem_alloc(sizeof(F22HBLK), "r"); +- phi_sav = (F2HBLK *) mem_alloc(sizeof(F22HBLK), "phi_sav"); +- +- i = sfreq + 0.5; +- switch(i){ +- case 32000: sfreq_idx = 0; break; +- case 44100: sfreq_idx = 1; break; +- case 48000: sfreq_idx = 2; break; +- default: printf("error, invalid sampling frequency: %d Hz\n",i); +- exit(-1); +- } +- printf("absthr[][] sampling frequency index: %d\n",sfreq_idx); +- read_absthr(absthr, sfreq_idx); +- if(lay==1){ +- flush = 384; +- syncsize = 1024; +- sync_flush = 576; +- } +- else { +- flush = 384*3.0/2.0; +- syncsize = 1056; +- sync_flush = syncsize - flush; +- } +- /* calculate HANN window coefficients */ +- /* for(i=0;i<BLKSIZE;i++)window[i]=0.5*(1-cos(2.0*PI*i/(BLKSIZE-1.0))); */ +- for(i=0;i<BLKSIZE;i++)window[i]=0.5*(1-cos(2.0*PI*(i-0.5)/BLKSIZE)); +- /* reset states used in unpredictability measure */ +- for(i=0;i<HBLKSIZE;i++){ +- r[0][0][i]=r[1][0][i]=r[0][1][i]=r[1][1][i]=0; +- phi_sav[0][0][i]=phi_sav[1][0][i]=0; +- phi_sav[0][1][i]=phi_sav[1][1][i]=0; +- lthr[0][i] = 60802371420160.0; +- lthr[1][i] = 60802371420160.0; +- } +- /***************************************************************************** +- * Initialization: Compute the following constants for use later * +- * partition[HBLKSIZE] = the partition number associated with each * +- * frequency line * +- * cbval[CBANDS] = the center (average) bark value of each * +- * partition * +- * numlines[CBANDS] = the number of frequency lines in each partition * +- * tmn[CBANDS] = tone masking noise * +- *****************************************************************************/ +- /* compute fft frequency multiplicand */ +- freq_mult = sfreq/BLKSIZE; +- +- /* calculate fft frequency, then bval of each line (use fthr[] as tmp storage)*/ +- for(i=0;i<HBLKSIZE;i++){ +- temp1 = i*freq_mult; +- j = 1; +- while(temp1>crit_band[j])j++; +- fthr[i]=j-1+(temp1-crit_band[j-1])/(crit_band[j]-crit_band[j-1]); +- } +- partition[0] = 0; +- /* temp2 is the counter of the number of frequency lines in each partition */ +- temp2 = 1; +- cbval[0]=fthr[0]; +- bval_lo=fthr[0]; +- for(i=1;i<HBLKSIZE;i++){ +- if((fthr[i]-bval_lo)>0.33){ +- partition[i]=partition[i-1]+1; +- cbval[partition[i-1]] = cbval[partition[i-1]]/temp2; +- cbval[partition[i]] = fthr[i]; +- bval_lo = fthr[i]; +- numlines[partition[i-1]] = temp2; +- temp2 = 1; +- } +- else { +- partition[i]=partition[i-1]; +- cbval[partition[i]] += fthr[i]; +- temp2++; +- } +- } +- numlines[partition[i-1]] = temp2; +- cbval[partition[i-1]] = cbval[partition[i-1]]/temp2; +- +- /************************************************************************ +- * Now compute the spreading function, s[j][i], the value of the spread-* +- * ing function, centered at band j, for band i, store for later use * +- ************************************************************************/ +- for(j=0;j<CBANDS;j++){ +- for(i=0;i<CBANDS;i++){ +- temp1 = (cbval[i] - cbval[j])*1.05; +- if(temp1>=0.5 && temp1<=2.5){ +- temp2 = temp1 - 0.5; +- temp2 = 8.0 * (temp2*temp2 - 2.0 * temp2); +- } +- else temp2 = 0; +- temp1 += 0.474; +- temp3 = 15.811389+7.5*temp1-17.5*sqrt((double) (1.0+temp1*temp1)); +- if(temp3 <= -100) s[i][j] = 0; +- else { +- temp3 = (temp2 + temp3)*LN_TO_LOG10; +- s[i][j] = exp(temp3); +- } +- } +- } +- +- /* Calculate Tone Masking Noise values */ +- for(j=0;j<CBANDS;j++){ +- temp1 = 15.5 + cbval[j]; +- tmn[j] = (temp1>24.5) ? temp1 : 24.5; +- /* Calculate normalization factors for the net spreading functions */ +- rnorm[j] = 0; +- for(i=0;i<CBANDS;i++){ +- rnorm[j] += s[j][i]; +- } +- } +- init++; +- } +- +- /************************* End of Initialization *****************************/ +- switch(lay) { +- case 1: +- case 2: +- for(i=0; i<lay; i++){ +- /***************************************************************************** +- * Net offset is 480 samples (1056-576) for layer 2; this is because one must* +- * stagger input data by 256 samples to synchronize psychoacoustic model with* +- * filter bank outputs, then stagger so that center of 1024 FFT window lines * +- * up with center of 576 "new" audio samples. * +- * * +- * For layer 1, the input data still needs to be staggered by 256 samples, * +- * then it must be staggered again so that the 384 "new" samples are centered* +- * in the 1024 FFT window. The net offset is then 576 and you need 448 "new"* +- * samples for each iteration to keep the 384 samples of interest centered * +- *****************************************************************************/ +- for(j=0; j<syncsize; j++){ +- if(j<(sync_flush))savebuf[j] = savebuf[j+flush]; +- else savebuf[j] = *buffer++; +- if(j<BLKSIZE){ +- /**window data with HANN window***********************************************/ +- wsamp_r[j] = window[j]*((FLOAT) savebuf[j]); +- wsamp_i[j] = 0; +- } +- } +- /**Compute FFT****************************************************************/ +- fft(wsamp_r,wsamp_i,energy,phi,1024); +- /***************************************************************************** +- * calculate the unpredictability measure, given energy[f] and phi[f] * +- *****************************************************************************/ +- /*only update data "age" pointers after you are done with both channels */ +- /*for layer 1 computations, for the layer 2 double computations, the pointers*/ +- /*are reset automatically on the second pass */ +- if(lay==2 || (lay==1 && chn==0) ){ +- if(new==0){new = 1; oldest = 1;} +- else {new = 0; oldest = 0;} +- if(old==0)old = 1; else old = 0; +- } +- for(j=0; j<HBLKSIZE; j++){ +- r_prime = 2.0 * r[chn][old][j] - r[chn][oldest][j]; +- phi_prime = 2.0 * phi_sav[chn][old][j] - phi_sav[chn][oldest][j]; +- r[chn][new][j] = sqrt((double) energy[j]); +- phi_sav[chn][new][j] = phi[j]; +- temp1=r[chn][new][j] * cos((double) phi[j]) - r_prime * cos((double) phi_prime); +- temp2=r[chn][new][j] * sin((double) phi[j]) - r_prime * sin((double) phi_prime); +- temp3=r[chn][new][j] + fabs((double)r_prime); +- if(temp3 != 0)c[j]=sqrt(temp1*temp1+temp2*temp2)/temp3; +- else c[j] = 0; +- } +- /***************************************************************************** +- * Calculate the grouped, energy-weighted, unpredictability measure, * +- * grouped_c[], and the grouped energy. grouped_e[] * +- *****************************************************************************/ +- for(j=1;j<CBANDS;j++){ +- grouped_e[j] = 0; +- grouped_c[j] = 0; +- } +- grouped_e[0] = energy[0]; +- grouped_c[0] = energy[0]*c[0]; +- for(j=1;j<HBLKSIZE;j++){ +- grouped_e[partition[j]] += energy[j]; +- grouped_c[partition[j]] += energy[j]*c[j]; +- } +- +- /***************************************************************************** +- * convolve the grouped energy-weighted unpredictability measure * +- * and the grouped energy with the spreading function, s[j][k] * +- *****************************************************************************/ +- for(j=0;j<CBANDS;j++){ +- ecb[j] = 0; +- cb[j] = 0; +- for(k=0;k<CBANDS;k++){ +- if(s[j][k] != 0.0){ +- ecb[j] += s[j][k]*grouped_e[k]; +- cb[j] += s[j][k]*grouped_c[k]; +- } +- } +- if(ecb[j] !=0)cb[j] = cb[j]/ecb[j]; +- else cb[j] = 0; +- } +- +- /***************************************************************************** +- * Calculate the required SNR for each of the frequency partitions * +- * this whole section can be accomplished by a table lookup * +- *****************************************************************************/ +- for(j=0;j<CBANDS;j++){ +- if(cb[j]<.05)cb[j]=0.05; +- else if(cb[j]>.5)cb[j]=0.5; +- tb = -0.434294482*log((double) cb[j])-0.301029996; +- cb[j]=tb; +- bc[j] = tmn[j]*tb + nmt*(1.0-tb); +- k = cbval[j] + 0.5; +- bc[j] = (bc[j] > bmax[k]) ? bc[j] : bmax[k]; +- bc[j] = exp((double) -bc[j]*LN_TO_LOG10); +- } +- +- /***************************************************************************** +- * Calculate the permissible noise energy level in each of the frequency * +- * partitions. Include absolute threshold and pre-echo controls * +- * this whole section can be accomplished by a table lookup * +- *****************************************************************************/ +- for(j=0;j<CBANDS;j++) +- if(rnorm[j] && numlines[j]) +- nb[j] = ecb[j]*bc[j]/(rnorm[j]*numlines[j]); +- else nb[j] = 0; +- for(j=0;j<HBLKSIZE;j++){ +- /*temp1 is the preliminary threshold */ +- temp1=nb[partition[j]]; +- temp1=(temp1>absthr[j])?temp1:absthr[j]; +- /*do not use pre-echo control for layer 2 because it may do bad things to the*/ +- /* MUSICAM bit allocation algorithm */ +- if(lay==1){ +- fthr[j] = (temp1 < lthr[chn][j]) ? temp1 : lthr[chn][j]; +- temp2 = temp1 * 0.00316; +- fthr[j] = (temp2 > fthr[j]) ? temp2 : fthr[j]; +- } +- else fthr[j] = temp1; +- lthr[chn][j] = LXMIN*temp1; +- } +- +- /***************************************************************************** +- * Translate the 512 threshold values to the 32 filter bands of the coder * +- *****************************************************************************/ +- for(j=0;j<193;j += 16){ +- minthres = 60802371420160.0; +- sum_energy = 0.0; +- for(k=0;k<17;k++){ +- if(minthres>fthr[j+k])minthres = fthr[j+k]; +- sum_energy += energy[j+k]; +- } +- snrtmp[i][j/16] = sum_energy/(minthres * 17.0); +- snrtmp[i][j/16] = 4.342944819 * log((double)snrtmp[i][j/16]); +- } +- for(j=208;j<(HBLKSIZE-1);j += 16){ +- minthres = 0.0; +- sum_energy = 0.0; +- for(k=0;k<17;k++){ +- minthres += fthr[j+k]; +- sum_energy += energy[j+k]; +- } +- snrtmp[i][j/16] = sum_energy/minthres; +- snrtmp[i][j/16] = 4.342944819 * log((double)snrtmp[i][j/16]); +- } +- /***************************************************************************** +- * End of Psychoacuostic calculation loop * +- *****************************************************************************/ +- } +- for(i=0; i<32; i++){ +- if(lay==2) +- snr32[i]=(snrtmp[0][i]>snrtmp[1][i])?snrtmp[0][i]:snrtmp[1][i]; +- else snr32[i]=snrtmp[0][i]; +- } +- break; +- case 3: +- printf("layer 3 is not currently supported\n"); +- break; +- default: +- printf("error, invalid MPEG/audio coding layer: %d\n",lay); +- } +- +- /* These mem_free() calls must correspond with the mem_alloc() calls */ +- /* used at the beginning of this function to simulate "automatic" */ +- /* variables placed on the stack. */ +- +- mem_free((void **) &grouped_c); +- mem_free((void **) &grouped_e); +- mem_free((void **) &nb); +- mem_free((void **) &cb); +- mem_free((void **) &ecb); +- mem_free((void **) &bc); +- mem_free((void **) &wsamp_r); +- mem_free((void **) &wsamp_i); +- mem_free((void **) &phi); +- mem_free((void **) &energy); +- mem_free((void **) &c); +- mem_free((void **) &fthr); +- mem_free((void **) &snrtmp); +- } +- +- /****************************************************************************** +- routine to read in absthr table from a file. +- ******************************************************************************/ +- +- void read_absthr(absthr, table) +- FLOAT *absthr; +- int table; +- { +- FILE *fp; +- long j,index; +- float a; +- char t[80]; +- char ta[16]; +- +- strcpy( ta, "absthr_0" ); +- +- switch(table){ +- case 0 : ta[7] = '0'; +- break; +- case 1 : ta[7] = '1'; +- break; +- case 2 : ta[7] = '2'; +- break; +- default : printf("absthr table: Not valid table number\n"); +- } +- if(!(fp = OpenTableFile(ta) ) ){ +- printf("Please check %s table\n", ta); +- exit(1); +- } +- fgets(t, 150, fp); +- sscanf(t, "table %ld", &index); +- if(index != table){ +- printf("error in absthr table %s",ta); +- exit(1); +- } +- for(j=0; j<HBLKSIZE; j++){ +- fgets(t,80,fp); +- sscanf(t,"%f", &a); +- absthr[j] = a; +- } +- fclose(fp); +- } +--- 0 ---- +diff -r -c -N encoder/psymodel.c lame3.70/psymodel.c +*** encoder/psymodel.c Wed Dec 31 17:00:00 1969 +--- lame3.70/psymodel.c Thu Apr 6 12:50:51 2000 +*************** +*** 0 **** +--- 1,1253 ---- ++ /********************************************************************** ++ * date programmers comment * ++ * 2/25/91 Davis Pan start of version 1.0 records * ++ * 5/10/91 W. Joseph Carter Ported to Macintosh and Unix. * ++ * 7/10/91 Earle Jennings Ported to MsDos. * ++ * replace of floats with FLOAT * ++ * 2/11/92 W. Joseph Carter Fixed mem_alloc() arg for "absthr". * ++ * 3/16/92 Masahiro Iwadare Modification for Layer III * ++ * 17/4/93 Masahiro Iwadare Updated for IS Modification * ++ **********************************************************************/ ++ ++ #include "util.h" ++ #include "encoder.h" ++ #include "psymodel.h" ++ #include "l3side.h" ++ #include <assert.h> ++ #ifdef HAVEGTK ++ #include "gtkanal.h" ++ #endif ++ #include "tables.h" ++ #include "fft.h" ++ ++ #ifdef M_LN10 ++ #define LN_TO_LOG10 (M_LN10/10) ++ #else ++ #define LN_TO_LOG10 0.2302585093 ++ #endif ++ ++ ++ void L3para_read( FLOAT8 sfreq, int numlines[CBANDS],int numlines_s[CBANDS], int partition_l[HBLKSIZE], ++ FLOAT8 minval[CBANDS], FLOAT8 qthr_l[CBANDS], ++ FLOAT8 s3_l[CBANDS + 1][CBANDS + 1], ++ FLOAT8 s3_s[CBANDS + 1][CBANDS + 1], ++ FLOAT8 qthr_s[CBANDS], ++ FLOAT8 SNR_s[CBANDS], ++ int bu_l[SBPSY_l], int bo_l[SBPSY_l], ++ FLOAT8 w1_l[SBPSY_l], FLOAT8 w2_l[SBPSY_l], ++ int bu_s[SBPSY_s], int bo_s[SBPSY_s], ++ FLOAT8 w1_s[SBPSY_s], FLOAT8 w2_s[SBPSY_s] ); ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ void L3psycho_anal( lame_global_flags *gfp, ++ short int *buffer[2],int gr_out , ++ FLOAT8 *ms_ratio, ++ FLOAT8 *ms_ratio_next, ++ FLOAT8 *ms_ener_ratio, ++ III_psy_ratio masking_ratio[2][2], ++ III_psy_ratio masking_MS_ratio[2][2], ++ FLOAT8 percep_entropy[2],FLOAT8 percep_MS_entropy[2], ++ int blocktype_d[2]) ++ { ++ ++ /* to get a good cache performance, one has to think about ++ * the sequence, in which the variables are used ++ */ ++ ++ /* The static variables "r", "phi_sav", "new", "old" and "oldest" have */ ++ /* to be remembered for the unpredictability measure. For "r" and */ ++ /* "phi_sav", the first index from the left is the channel select and */ ++ /* the second index is the "age" of the data. */ ++ static FLOAT8 minval[CBANDS],qthr_l[CBANDS]; ++ static FLOAT8 qthr_s[CBANDS]; ++ static FLOAT8 nb_1[4][CBANDS], nb_2[4][CBANDS]; ++ static FLOAT8 s3_s[CBANDS + 1][CBANDS + 1]; ++ static FLOAT8 s3_l[CBANDS + 1][CBANDS + 1]; ++ ++ static III_psy_xmin thm[4]; ++ static III_psy_xmin en[4]; ++ ++ /* unpredictability calculation ++ */ ++ static int cw_upper_index; ++ static int cw_lower_index; ++ static FLOAT ax_sav[4][2][HBLKSIZE]; ++ static FLOAT bx_sav[4][2][HBLKSIZE]; ++ static FLOAT rx_sav[4][2][HBLKSIZE]; ++ static FLOAT cw[HBLKSIZE]; ++ ++ /* fft and energy calculation ++ */ ++ FLOAT (*wsamp_l)[BLKSIZE]; ++ FLOAT (*wsamp_s)[3][BLKSIZE_s]; ++ FLOAT tot_ener[4]; ++ static FLOAT wsamp_L[2][BLKSIZE]; ++ static FLOAT energy[HBLKSIZE]; ++ static FLOAT wsamp_S[2][3][BLKSIZE_s]; ++ static FLOAT energy_s[3][HBLKSIZE_s]; ++ ++ /* convolution ++ */ ++ static FLOAT8 eb[CBANDS]; ++ static FLOAT8 cb[CBANDS]; ++ static FLOAT8 thr[CBANDS]; ++ ++ /* Scale Factor Bands ++ */ ++ static FLOAT8 w1_l[SBPSY_l], w2_l[SBPSY_l]; ++ static FLOAT8 w1_s[SBPSY_s], w2_s[SBPSY_s]; ++ static FLOAT8 mld_l[SBPSY_l],mld_s[SBPSY_s]; ++ static int bu_l[SBPSY_l],bo_l[SBPSY_l] ; ++ static int bu_s[SBPSY_s],bo_s[SBPSY_s] ; ++ static int npart_l,npart_s; ++ static int npart_l_orig,npart_s_orig; ++ ++ static int s3ind[CBANDS][2]; ++ static int s3ind_s[CBANDS][2]; ++ ++ static int numlines_s[CBANDS] ; ++ static int numlines_l[CBANDS]; ++ static int partition_l[HBLKSIZE]; ++ ++ /* frame analyzer ++ */ ++ #ifdef HAVEGTK ++ static FLOAT energy_save[4][HBLKSIZE]; ++ static FLOAT8 pe_save[4]; ++ static FLOAT8 ers_save[4]; ++ #endif ++ ++ /* ratios ++ */ ++ static FLOAT8 pe[4]={0,0,0,0}; ++ static FLOAT8 ms_ratio_s_old=0,ms_ratio_l_old=0; ++ static FLOAT8 ms_ener_ratio_old=.25; ++ FLOAT8 ms_ratio_l=0,ms_ratio_s=0; ++ ++ /* block type ++ */ ++ static int blocktype_old[2]; ++ int blocktype[2],uselongblock[2]; ++ ++ /* usual variables like loop indices, etc.. ++ */ ++ int numchn, chn; ++ int b, i, j, k; ++ int sb,sblock; ++ FLOAT cwlimit; ++ ++ ++ /* initialization of static variables ++ */ ++ if((gfp->frameNum==0) && (gr_out==0)){ ++ FLOAT8 SNR_s[CBANDS]; ++ ++ blocktype_old[0]=STOP_TYPE; ++ blocktype_old[1]=STOP_TYPE; ++ i = gfp->out_samplerate; ++ switch(i){ ++ case 32000: break; ++ case 44100: break; ++ case 48000: break; ++ case 16000: break; ++ case 22050: break; ++ case 24000: break; ++ default: fprintf(stderr,"error, invalid sampling frequency: %d Hz\n",i); ++ exit(-1); ++ } ++ ++ /* reset states used in unpredictability measure */ ++ memset (rx_sav,0, sizeof(rx_sav)); ++ memset (ax_sav,0, sizeof(ax_sav)); ++ memset (bx_sav,0, sizeof(bx_sav)); ++ memset (en,0, sizeof(en)); ++ memset (thm,0, sizeof(thm)); ++ ++ ++ /* gfp->cwlimit = sfreq*j/1024.0; */ ++ cw_lower_index=6; ++ if (gfp->cwlimit>0) ++ cwlimit=gfp->cwlimit; ++ else ++ cwlimit=8.8717; ++ cw_upper_index = cwlimit*1000.0*1024.0/((FLOAT8) gfp->out_samplerate); ++ cw_upper_index=Min(HBLKSIZE-4,cw_upper_index); /* j+3 < HBLKSIZE-1 */ ++ cw_upper_index=Max(6,cw_upper_index); ++ ++ for ( j = 0; j < HBLKSIZE; j++ ) ++ cw[j] = 0.4; ++ ++ /* setup stereo demasking thresholds */ ++ /* formula reverse enginerred from plot in paper */ ++ for ( sb = 0; sb < SBPSY_s; sb++ ) { ++ FLOAT8 mld = 1.25*(1-cos(PI*sb/SBPSY_s))-2.5; ++ mld_s[sb] = pow(10.0,mld); ++ } ++ for ( sb = 0; sb < SBPSY_l; sb++ ) { ++ FLOAT8 mld = 1.25*(1-cos(PI*sb/SBPSY_l))-2.5; ++ mld_l[sb] = pow(10.0,mld); ++ } ++ ++ for (i=0;i<HBLKSIZE;i++) partition_l[i]=-1; ++ ++ L3para_read( (FLOAT8) gfp->out_samplerate,numlines_l,numlines_s,partition_l,minval,qthr_l,s3_l,s3_s, ++ qthr_s,SNR_s, ++ bu_l,bo_l,w1_l,w2_l, bu_s,bo_s,w1_s,w2_s ); ++ ++ ++ /* npart_l_orig = number of partition bands before convolution */ ++ /* npart_l = number of partition bands after convolution */ ++ npart_l_orig=0; npart_s_orig=0; ++ for (i=0;i<HBLKSIZE;i++) ++ if (partition_l[i]>npart_l_orig) npart_l_orig=partition_l[i]; ++ npart_l_orig++; ++ ++ for (i=0;numlines_s[i]>=0;i++) ++ ; ++ npart_s_orig = i; ++ ++ npart_l=bo_l[SBPSY_l-1]+1; ++ npart_s=bo_s[SBPSY_s-1]+1; ++ ++ /* MPEG2 tables are screwed up ++ * the mapping from paritition bands to scalefactor bands will use ++ * more paritition bands than we have. ++ * So we will not compute these fictitious partition bands by reducing ++ * npart_l below. */ ++ if (npart_l > npart_l_orig) { ++ npart_l=npart_l_orig; ++ bo_l[SBPSY_l-1]=npart_l-1; ++ w2_l[SBPSY_l-1]=1.0; ++ } ++ if (npart_s > npart_s_orig) { ++ npart_s=npart_s_orig; ++ bo_s[SBPSY_s-1]=npart_s-1; ++ w2_s[SBPSY_s-1]=1.0; ++ } ++ ++ ++ ++ for (i=0; i<npart_l; i++) { ++ for (j = 0; j < npart_l_orig; j++) { ++ if (s3_l[i][j] != 0.0) ++ break; ++ } ++ s3ind[i][0] = j; ++ ++ for (j = npart_l_orig - 1; j > 0; j--) { ++ if (s3_l[i][j] != 0.0) ++ break; ++ } ++ s3ind[i][1] = j; ++ } ++ ++ ++ for (i=0; i<npart_s; i++) { ++ for (j = 0; j < npart_s_orig; j++) { ++ if (s3_s[i][j] != 0.0) ++ break; ++ } ++ s3ind_s[i][0] = j; ++ ++ for (j = npart_s_orig - 1; j > 0; j--) { ++ if (s3_s[i][j] != 0.0) ++ break; ++ } ++ s3ind_s[i][1] = j; ++ } ++ ++ ++ /* ++ #include "debugscalefac.c" ++ */ ++ ++ ++ #define AACS3 ++ #define NEWS3XX ++ ++ #ifdef AACS3 ++ /* AAC values, results in more masking over MP3 values */ ++ # define TMN 18 ++ # define NMT 6 ++ #else ++ /* MP3 values */ ++ # define TMN 29 ++ # define NMT 6 ++ #endif ++ ++ #define rpelev 2 ++ #define rpelev2 16 ++ ++ /* compute norm_l, norm_s instead of relying on table data */ ++ for ( b = 0;b < npart_l; b++ ) { ++ FLOAT8 norm=0; ++ for ( k = s3ind[b][0]; k <= s3ind[b][1]; k++ ) { ++ norm += s3_l[b][k]; ++ } ++ for ( k = s3ind[b][0]; k <= s3ind[b][1]; k++ ) { ++ s3_l[b][k] *= exp(-LN_TO_LOG10 * NMT) / norm; ++ } ++ /*printf("%i norm=%f norm_l=%f \n",b,1/norm,norm_l[b]);*/ ++ } ++ ++ /* MPEG1 SNR_s data is given in db, convert to energy */ ++ if (gfp->version == 1) { ++ for ( b = 0;b < npart_s; b++ ) { ++ SNR_s[b]=exp( (FLOAT8) SNR_s[b] * LN_TO_LOG10 ); ++ } ++ } ++ ++ for ( b = 0;b < npart_s; b++ ) { ++ FLOAT8 norm=0; ++ for ( k = s3ind_s[b][0]; k <= s3ind_s[b][1]; k++ ) { ++ norm += s3_s[b][k]; ++ } ++ for ( k = s3ind_s[b][0]; k <= s3ind_s[b][1]; k++ ) { ++ s3_s[b][k] *= SNR_s[b] / norm; ++ } ++ /*printf("%i norm=%f norm_s=%f \n",b,1/norm,norm_l[b]);*/ ++ } ++ ++ init_fft(); ++ } ++ /************************* End of Initialization *****************************/ ++ ++ ++ ++ ++ ++ numchn = gfp->stereo; ++ /* chn=2 and 3 = Mid and Side channels */ ++ if (gfp->mode == MPG_MD_JOINT_STEREO) numchn=4; ++ for (chn=0; chn<numchn; chn++) { ++ ++ wsamp_s = wsamp_S+(chn & 1); ++ wsamp_l = wsamp_L+(chn & 1); ++ ++ ++ if (chn<2) { ++ /********************************************************************** ++ * compute FFTs ++ **********************************************************************/ ++ fft_long ( *wsamp_l, chn, buffer); ++ fft_short( *wsamp_s, chn, buffer); ++ ++ /* LR maskings */ ++ percep_entropy[chn] = pe[chn]; ++ masking_ratio[gr_out][chn].thm = thm[chn]; ++ masking_ratio[gr_out][chn].en = en[chn]; ++ }else{ ++ /* MS maskings */ ++ percep_MS_entropy[chn-2] = pe[chn]; ++ masking_MS_ratio[gr_out][chn-2].en = en[chn]; ++ masking_MS_ratio[gr_out][chn-2].thm = thm[chn]; ++ ++ if (chn == 2) ++ { ++ for (j = BLKSIZE-1; j >=0 ; --j) ++ { ++ FLOAT l = wsamp_L[0][j]; ++ FLOAT r = wsamp_L[1][j]; ++ wsamp_L[0][j] = (l+r)*(FLOAT)(SQRT2*0.5); ++ wsamp_L[1][j] = (l-r)*(FLOAT)(SQRT2*0.5); ++ } ++ for (b = 2; b >= 0; --b) ++ { ++ for (j = BLKSIZE_s-1; j >= 0 ; --j) ++ { ++ FLOAT l = wsamp_S[0][b][j]; ++ FLOAT r = wsamp_S[1][b][j]; ++ wsamp_S[0][b][j] = (l+r)*(FLOAT)(SQRT2*0.5); ++ wsamp_S[1][b][j] = (l-r)*(FLOAT)(SQRT2*0.5); ++ } ++ } ++ } ++ } ++ ++ /********************************************************************** ++ * compute energies ++ **********************************************************************/ ++ ++ ++ ++ energy[0] = (*wsamp_l)[0]; ++ energy[0] *= energy[0]; ++ ++ tot_ener[chn] = energy[0]; /* sum total energy at nearly no extra cost */ ++ ++ for (j=BLKSIZE/2-1; j >= 0; --j) ++ { ++ FLOAT re = (*wsamp_l)[BLKSIZE/2-j]; ++ FLOAT im = (*wsamp_l)[BLKSIZE/2+j]; ++ energy[BLKSIZE/2-j] = (re * re + im * im) * (FLOAT)0.5; ++ ++ tot_ener[chn] += energy[BLKSIZE/2-j]; ++ } ++ for (b = 2; b >= 0; --b) ++ { ++ energy_s[b][0] = (*wsamp_s)[b][0]; ++ energy_s[b][0] *= energy_s [b][0]; ++ for (j=BLKSIZE_s/2-1; j >= 0; --j) ++ { ++ FLOAT re = (*wsamp_s)[b][BLKSIZE_s/2-j]; ++ FLOAT im = (*wsamp_s)[b][BLKSIZE_s/2+j]; ++ energy_s[b][BLKSIZE_s/2-j] = (re * re + im * im) * (FLOAT)0.5; ++ } ++ } ++ ++ ++ #ifdef HAVEGTK ++ if(gfp->gtkflag) { ++ for (j=0; j<HBLKSIZE ; j++) { ++ pinfo->energy[gr_out][chn][j]=energy_save[chn][j]; ++ energy_save[chn][j]=energy[j]; ++ } ++ } ++ #endif ++ ++ /********************************************************************** ++ * compute unpredicatability of first six spectral lines * ++ **********************************************************************/ ++ for ( j = 0; j < cw_lower_index; j++ ) ++ { /* calculate unpredictability measure cw */ ++ FLOAT an, a1, a2; ++ FLOAT bn, b1, b2; ++ FLOAT rn, r1, r2; ++ FLOAT numre, numim, den; ++ ++ a2 = ax_sav[chn][1][j]; ++ b2 = bx_sav[chn][1][j]; ++ r2 = rx_sav[chn][1][j]; ++ a1 = ax_sav[chn][1][j] = ax_sav[chn][0][j]; ++ b1 = bx_sav[chn][1][j] = bx_sav[chn][0][j]; ++ r1 = rx_sav[chn][1][j] = rx_sav[chn][0][j]; ++ an = ax_sav[chn][0][j] = (*wsamp_l)[j]; ++ bn = bx_sav[chn][0][j] = j==0 ? (*wsamp_l)[0] : (*wsamp_l)[BLKSIZE-j]; ++ rn = rx_sav[chn][0][j] = sqrt(energy[j]); ++ ++ { /* square (x1,y1) */ ++ if( r1 != 0 ) { ++ numre = (a1*b1); ++ numim = (a1*a1-b1*b1)*(FLOAT)0.5; ++ den = r1*r1; ++ } else { ++ numre = 1; ++ numim = 0; ++ den = 1; ++ } ++ } ++ ++ { /* multiply by (x2,-y2) */ ++ if( r2 != 0 ) { ++ FLOAT tmp2 = (numim+numre)*(a2+b2)*(FLOAT)0.5; ++ FLOAT tmp1 = -a2*numre+tmp2; ++ numre = -b2*numim+tmp2; ++ numim = tmp1; ++ den *= r2; ++ } else { ++ /* do nothing */ ++ } ++ } ++ ++ { /* r-prime factor */ ++ FLOAT tmp = (2*r1-r2)/den; ++ numre *= tmp; ++ numim *= tmp; ++ } ++ den=rn+fabs(2*r1-r2); ++ if( den != 0 ) { ++ numre = (an+bn)*(FLOAT)0.5-numre; ++ numim = (an-bn)*(FLOAT)0.5-numim; ++ den = sqrt(numre*numre+numim*numim)/den; ++ } ++ cw[j] = den; ++ } ++ ++ ++ ++ /********************************************************************** ++ * compute unpredicatibility of next 200 spectral lines * ++ **********************************************************************/ ++ for ( j = cw_lower_index; j < cw_upper_index; j += 4 ) ++ {/* calculate unpredictability measure cw */ ++ FLOAT rn, r1, r2; ++ FLOAT numre, numim, den; ++ ++ k = (j+2) / 4; ++ ++ { /* square (x1,y1) */ ++ r1 = energy_s[0][k]; ++ if( r1 != 0 ) { ++ FLOAT a1 = (*wsamp_s)[0][k]; ++ FLOAT b1 = (*wsamp_s)[0][BLKSIZE_s-k]; /* k is never 0 */ ++ numre = (a1*b1); ++ numim = (a1*a1-b1*b1)*(FLOAT)0.5; ++ den = r1; ++ r1 = sqrt(r1); ++ } else { ++ numre = 1; ++ numim = 0; ++ den = 1; ++ } ++ } ++ ++ ++ { /* multiply by (x2,-y2) */ ++ r2 = energy_s[2][k]; ++ if( r2 != 0 ) { ++ FLOAT a2 = (*wsamp_s)[2][k]; ++ FLOAT b2 = (*wsamp_s)[2][BLKSIZE_s-k]; ++ ++ ++ FLOAT tmp2 = (numim+numre)*(a2+b2)*(FLOAT)0.5; ++ FLOAT tmp1 = -a2*numre+tmp2; ++ numre = -b2*numim+tmp2; ++ numim = tmp1; ++ ++ r2 = sqrt(r2); ++ den *= r2; ++ } else { ++ /* do nothing */ ++ } ++ } ++ ++ { /* r-prime factor */ ++ FLOAT tmp = (2*r1-r2)/den; ++ numre *= tmp; ++ numim *= tmp; ++ } ++ ++ rn = sqrt(energy_s[1][k]); ++ den=rn+fabs(2*r1-r2); ++ if( den != 0 ) { ++ FLOAT an = (*wsamp_s)[1][k]; ++ FLOAT bn = (*wsamp_s)[1][BLKSIZE_s-k]; ++ numre = (an+bn)*(FLOAT)0.5-numre; ++ numim = (an-bn)*(FLOAT)0.5-numim; ++ den = sqrt(numre*numre+numim*numim)/den; ++ } ++ ++ cw[j+1] = cw[j+2] = cw[j+3] = cw[j] = den; ++ } ++ ++ #if 0 ++ for ( j = 14; j < HBLKSIZE-4; j += 4 ) ++ {/* calculate energy from short ffts */ ++ FLOAT8 tot,ave; ++ k = (j+2) / 4; ++ for (tot=0, sblock=0; sblock < 3; sblock++) ++ tot+=energy_s[sblock][k]; ++ ave = energy[j+1]+ energy[j+2]+ energy[j+3]+ energy[j]; ++ ave /= 4.; ++ /* ++ printf("energy / tot %i %5.2f %e %e\n",j,ave/(tot*16./3.), ++ ave,tot*16./3.); ++ */ ++ energy[j+1] = energy[j+2] = energy[j+3] = energy[j]=tot; ++ } ++ #endif ++ ++ ++ ++ ++ ++ ++ ++ ++ /********************************************************************** ++ * Calculate the energy and the unpredictability in the threshold * ++ * calculation partitions * ++ **********************************************************************/ ++ #if 0 ++ for ( b = 0; b < CBANDS; b++ ) ++ { ++ eb[b] = 0; ++ cb[b] = 0; ++ } ++ for ( j = 0; j < HBLKSIZE; j++ ) ++ { ++ int tp = partition_l[j]; ++ ++ if ( tp >= 0 ) ++ { ++ eb[tp] += energy[j]; ++ cb[tp] += cw[j] * energy[j]; ++ } ++ assert(tp<npart_l_orig); ++ } ++ #else ++ b = 0; ++ for (j = 0; j < cw_upper_index;) ++ { ++ FLOAT8 ebb, cbb; ++ int i; ++ ++ ebb = energy[j]; ++ cbb = energy[j] * cw[j]; ++ j++; ++ ++ for (i = numlines_l[b] - 1; i > 0; i--) ++ { ++ ebb += energy[j]; ++ cbb += energy[j] * cw[j]; ++ j++; ++ } ++ eb[b] = ebb; ++ cb[b] = cbb; ++ b++; ++ } ++ ++ for (; b < npart_l_orig; b++ ) ++ { ++ int i; ++ FLOAT8 ebb = energy[j++]; ++ ++ for (i = numlines_l[b] - 1; i > 0; i--) ++ { ++ ebb += energy[j++]; ++ } ++ eb[b] = ebb; ++ cb[b] = ebb * 0.4; ++ } ++ #endif ++ ++ /********************************************************************** ++ * convolve the partitioned energy and unpredictability * ++ * with the spreading function, s3_l[b][k] * ++ ******************************************************************** */ ++ pe[chn] = 0; /* calculate percetual entropy */ ++ for ( b = 0;b < npart_l; b++ ) ++ { ++ FLOAT8 tbb,ecb,ctb; ++ FLOAT8 temp_1; /* BUG of IS */ ++ ++ ecb = 0; ++ ctb = 0; ++ for ( k = s3ind[b][0]; k <= s3ind[b][1]; k++ ) ++ { ++ ecb += s3_l[b][k] * eb[k]; /* sprdngf for Layer III */ ++ ctb += s3_l[b][k] * cb[k]; ++ } ++ ++ /* calculate the tonality of each threshold calculation partition */ ++ /* calculate the SNR in each threshhold calculation partition */ ++ ++ tbb = ecb; ++ if (tbb != 0) ++ { ++ tbb = ctb / tbb; ++ if (tbb <= 0.04875584301) ++ { ++ tbb = exp(-LN_TO_LOG10 * (TMN - NMT)); ++ } ++ else if (tbb > 0.4989003827) ++ { ++ tbb = 1; ++ } ++ else ++ { ++ tbb = log(tbb); ++ tbb = exp(((TMN - NMT)*(LN_TO_LOG10*0.299)) ++ + ((TMN - NMT)*(LN_TO_LOG10*0.43 ))*tbb); /* conv1=-0.299, conv2=-0.43 */ ++ } ++ } ++ ++ tbb = Min(minval[b], tbb); ++ ecb *= tbb; ++ ++ /* pre-echo control */ ++ /* rpelev=2.0, rpelev2=16.0 */ ++ temp_1 = Min(ecb, Min(rpelev*nb_1[chn][b],rpelev2*nb_2[chn][b]) ); ++ thr[b] = Max( qthr_l[b], temp_1 ); ++ nb_2[chn][b] = nb_1[chn][b]; ++ nb_1[chn][b] = ecb; ++ ++ /* note: all surges in PE are because of the above pre-echo formula ++ * for temp_1. it this is not used, PE is always around 600 ++ */ ++ ++ if (thr[b] < eb[b]) ++ { ++ /* there's no non sound portition, because thr[b] is ++ maximum of qthr_l and temp_1 */ ++ pe[chn] -= numlines_l[b] * log(thr[b] / eb[b]); ++ } ++ } ++ ++ ++ #ifdef HAVEGTK ++ if (gfp->gtkflag) { ++ FLOAT mn,mx,ma=0,mb=0,mc=0; ++ ++ for ( j = HBLKSIZE_s/2; j < HBLKSIZE_s; j ++) ++ { ++ ma += energy_s[0][j]; ++ mb += energy_s[1][j]; ++ mc += energy_s[2][j]; ++ } ++ mn = Min(ma,mb); ++ mn = Min(mn,mc); ++ mx = Max(ma,mb); ++ mx = Max(mx,mc); ++ ++ pinfo->ers[gr_out][chn]=ers_save[chn]; ++ ers_save[chn]=mx/(1e-12+mn); ++ pinfo->pe[gr_out][chn]=pe_save[chn]; ++ pe_save[chn]=pe[chn]; ++ } ++ #endif ++ ++ /*************************************************************** ++ * determine the block type (window type) based on L & R channels ++ * ++ ***************************************************************/ ++ if (chn<2) { ++ if (gfp->no_short_blocks){ ++ uselongblock[chn]=1; ++ } else { ++ /* tuned for t1.wav. doesnt effect most other samples */ ++ if (pe[chn] > 3000) { ++ uselongblock[chn]=0; ++ } else { ++ FLOAT mn,mx,ma=0,mb=0,mc=0; ++ ++ for ( j = HBLKSIZE_s/2; j < HBLKSIZE_s; j ++) ++ { ++ ma += energy_s[0][j]; ++ mb += energy_s[1][j]; ++ mc += energy_s[2][j]; ++ } ++ mn = Min(ma,mb); ++ mn = Min(mn,mc); ++ mx = Max(ma,mb); ++ mx = Max(mx,mc); ++ ++ uselongblock[chn] = 1; ++ ++ if ( mx > 30*mn ) ++ {/* big surge of energy - always use short blocks */ ++ uselongblock[chn] = 0; ++ } ++ else if ((mx > 10*mn) && (pe[chn] > 1000)) ++ {/* medium surge, medium pe - use short blocks */ ++ uselongblock[chn] = 0; ++ } ++ } ++ } ++ } ++ ++ ++ ++ /*************************************************************** ++ * compute masking thresholds for both short and long blocks ++ ***************************************************************/ ++ /* longblock threshold calculation (part 2) */ ++ for ( sb = 0; sb < SBPSY_l; sb++ ) ++ { ++ FLOAT8 enn = w1_l[sb] * eb[bu_l[sb]] + w2_l[sb] * eb[bo_l[sb]]; ++ FLOAT8 thmm = w1_l[sb] *thr[bu_l[sb]] + w2_l[sb] * thr[bo_l[sb]]; ++ for ( b = bu_l[sb]+1; b < bo_l[sb]; b++ ) ++ { ++ enn += eb[b]; ++ thmm += thr[b]; ++ } ++ en[chn].l[sb] = enn; ++ thm[chn].l[sb] = thmm; ++ } ++ ++ ++ /* threshold calculation for short blocks */ ++ for ( sblock = 0; sblock < 3; sblock++ ) ++ { ++ j = 0; ++ for ( b = 0; b < npart_s_orig; b++ ) ++ { ++ int i; ++ FLOAT ecb = energy_s[sblock][j++]; ++ for (i = numlines_s[b]; i > 0; i--) ++ { ++ ecb += energy_s[sblock][j++]; ++ } ++ eb[b] = ecb; ++ } ++ ++ for ( b = 0; b < npart_s; b++ ) ++ { ++ FLOAT8 ecb = 0; ++ for ( k = s3ind_s[b][0]; k <= s3ind_s[b][1]; k++ ) ++ { ++ ecb += s3_s[b][k] * eb[k]; ++ } ++ thr[b] = Max (qthr_s[b], ecb); ++ } ++ ++ for ( sb = 0; sb < SBPSY_s; sb++ ) ++ { ++ FLOAT8 enn = w1_s[sb] * eb[bu_s[sb]] + w2_s[sb] * eb[bo_s[sb]]; ++ FLOAT8 thmm = w1_s[sb] *thr[bu_s[sb]] + w2_s[sb] * thr[bo_s[sb]]; ++ for ( b = bu_s[sb]+1; b < bo_s[sb]; b++ ) ++ { ++ enn += eb[b]; ++ thmm += thr[b]; ++ } ++ en[chn].s[sb][sblock] = enn; ++ thm[chn].s[sb][sblock] = thmm; ++ } ++ } ++ } /* end loop over chn */ ++ ++ ++ /* compute M/S thresholds from Johnston & Ferreira 1992 ICASSP paper */ ++ if ( numchn==4 /* mid/side and r/l */) { ++ FLOAT8 rside,rmid,mld; ++ int chmid=2,chside=3; ++ ++ for ( sb = 0; sb < SBPSY_l; sb++ ) { ++ /* use this fix if L & R masking differs by 2db or less */ ++ /* if db = 10*log10(x2/x1) < 2 */ ++ /* if (x2 < 1.58*x1) { */ ++ if (thm[0].l[sb] <= 1.58*thm[1].l[sb] ++ && thm[1].l[sb] <= 1.58*thm[0].l[sb]) { ++ ++ mld = mld_l[sb]*en[chside].l[sb]; ++ rmid = Max(thm[chmid].l[sb], Min(thm[chside].l[sb],mld)); ++ ++ mld = mld_l[sb]*en[chmid].l[sb]; ++ rside = Max(thm[chside].l[sb],Min(thm[chmid].l[sb],mld)); ++ ++ thm[chmid].l[sb]=rmid; ++ thm[chside].l[sb]=rside; ++ } ++ } ++ for ( sb = 0; sb < SBPSY_s; sb++ ) { ++ for ( sblock = 0; sblock < 3; sblock++ ) { ++ if (thm[0].s[sb][sblock] <= 1.58*thm[1].s[sb][sblock] ++ && thm[1].s[sb][sblock] <= 1.58*thm[0].s[sb][sblock]) { ++ ++ mld = mld_s[sb]*en[chside].s[sb][sblock]; ++ rmid = Max(thm[chmid].s[sb][sblock],Min(thm[chside].s[sb][sblock],mld)); ++ ++ mld = mld_s[sb]*en[chmid].s[sb][sblock]; ++ rside = Max(thm[chside].s[sb][sblock],Min(thm[chmid].s[sb][sblock],mld)); ++ ++ thm[chmid].s[sb][sblock]=rmid; ++ thm[chside].s[sb][sblock]=rside; ++ } ++ } ++ } ++ } ++ ++ ++ ++ ++ ++ ++ if (gfp->mode == MPG_MD_JOINT_STEREO) { ++ /* determin ms_ratio from masking thresholds*/ ++ /* use ms_stereo (ms_ratio < .35) if average thresh. diff < 5 db */ ++ FLOAT8 db,x1,x2,sidetot=0,tot=0; ++ for (sb= SBPSY_l/4 ; sb< SBPSY_l; sb ++ ) { ++ x1 = Min(thm[0].l[sb],thm[1].l[sb]); ++ x2 = Max(thm[0].l[sb],thm[1].l[sb]); ++ /* thresholds difference in db */ ++ if (x2 >= 1000*x1) db=3; ++ else db = log10(x2/x1); ++ /* printf("db = %f %e %e \n",db,thm[0].l[sb],thm[1].l[sb]);*/ ++ sidetot += db; ++ tot++; ++ } ++ ms_ratio_l= (sidetot/tot)*0.7; /* was .35*(sidetot/tot)/5.0*10 */ ++ ms_ratio_l = Min(ms_ratio_l,0.5); ++ ++ sidetot=0; tot=0; ++ for ( sblock = 0; sblock < 3; sblock++ ) ++ for ( sb = SBPSY_s/4; sb < SBPSY_s; sb++ ) { ++ x1 = Min(thm[0].s[sb][sblock],thm[1].s[sb][sblock]); ++ x2 = Max(thm[0].s[sb][sblock],thm[1].s[sb][sblock]); ++ /* thresholds difference in db */ ++ if (x2 >= 1000*x1) db=3; ++ else db = log10(x2/x1); ++ sidetot += db; ++ tot++; ++ } ++ ms_ratio_s = (sidetot/tot)*0.7; /* was .35*(sidetot/tot)/5.0*10 */ ++ ms_ratio_s = Min(ms_ratio_s,.5); ++ } ++ ++ /*************************************************************** ++ * determin final block type ++ ***************************************************************/ ++ ++ for (chn=0; chn<gfp->stereo; chn++) { ++ blocktype[chn] = NORM_TYPE; ++ } ++ ++ ++ if (gfp->stereo==2) { ++ if (!gfp->allow_diff_short || gfp->mode==MPG_MD_JOINT_STEREO) { ++ /* force both channels to use the same block type */ ++ /* this is necessary if the frame is to be encoded in ms_stereo. */ ++ /* But even without ms_stereo, FhG does this */ ++ int bothlong= (uselongblock[0] && uselongblock[1]); ++ if (!bothlong) { ++ uselongblock[0]=0; ++ uselongblock[1]=0; ++ } ++ } ++ } ++ ++ ++ ++ /* update the blocktype of the previous granule, since it depends on what ++ * happend in this granule */ ++ for (chn=0; chn<gfp->stereo; chn++) { ++ if ( uselongblock[chn]) ++ { /* no attack : use long blocks */ ++ switch( blocktype_old[chn] ) ++ { ++ case NORM_TYPE: ++ case STOP_TYPE: ++ blocktype[chn] = NORM_TYPE; ++ break; ++ case SHORT_TYPE: ++ blocktype[chn] = STOP_TYPE; ++ break; ++ case START_TYPE: ++ fprintf( stderr, "Error in block selecting\n" ); ++ abort(); ++ break; /* problem */ ++ } ++ } else { ++ /* attack : use short blocks */ ++ blocktype[chn] = SHORT_TYPE; ++ if ( blocktype_old[chn] == NORM_TYPE ) { ++ blocktype_old[chn] = START_TYPE; ++ } ++ if ( blocktype_old[chn] == STOP_TYPE ) { ++ blocktype_old[chn] = SHORT_TYPE ; ++ } ++ } ++ ++ blocktype_d[chn] = blocktype_old[chn]; /* value returned to calling program */ ++ blocktype_old[chn] = blocktype[chn]; /* save for next call to l3psy_anal */ ++ } ++ ++ if (blocktype_d[0]==2) ++ *ms_ratio = ms_ratio_s_old; ++ else ++ *ms_ratio = ms_ratio_l_old; ++ ++ ms_ratio_s_old = ms_ratio_s; ++ ms_ratio_l_old = ms_ratio_l; ++ ++ /* we dont know the block type of this frame yet - assume long */ ++ *ms_ratio_next = ms_ratio_l; ++ ++ ++ ++ /*********************************************************************/ ++ /* compute side_energy / (side+mid)_energy */ ++ /* 0 = no energy in side channel */ ++ /* .5 = half of total energy in side channel */ ++ /*********************************************************************/ ++ if (numchn==4) { ++ FLOAT tmp = tot_ener[3]+tot_ener[2]; ++ *ms_ener_ratio = ms_ener_ratio_old; ++ ms_ener_ratio_old=0; ++ if (tmp>0) ms_ener_ratio_old=tot_ener[3]/tmp; ++ } else ++ /* we didn't compute ms_ener_ratios */ ++ *ms_ener_ratio = 0; ++ ++ } ++ ++ ++ ++ ++ ++ ++ void L3para_read(FLOAT8 sfreq, int *numlines_l,int *numlines_s, int *partition_l, FLOAT8 *minval, ++ FLOAT8 *qthr_l, FLOAT8 s3_l[64][64], FLOAT8 s3_s[CBANDS + 1][CBANDS + 1], ++ FLOAT8 *qthr_s, FLOAT8 *SNR, ++ int *bu_l, int *bo_l, FLOAT8 *w1_l, FLOAT8 *w2_l, ++ int *bu_s, int *bo_s, FLOAT8 *w1_s, FLOAT8 *w2_s) ++ { ++ FLOAT8 freq_tp; ++ FLOAT8 bval_l[CBANDS], bval_s[CBANDS]; ++ int cbmax=0, cbmax_tp; ++ FLOAT8 *p = psy_data; ++ ++ int sbmax ; ++ int i,j,k,k2,loop, part_max ; ++ int freq_scale=1; ++ ++ ++ /* use MPEG1 tables. The MPEG2 tables in tables.c appear to be ++ * junk. MPEG2 doc claims data for these tables is the same as the ++ * MPEG1 data for 2x sampling frequency */ ++ /* if (sfreq<32000) freq_scale=2; */ ++ ++ ++ ++ /* Read long block data */ ++ ++ for(loop=0;loop<6;loop++) ++ { ++ freq_tp = *p++; ++ cbmax_tp = (int) *p++; ++ cbmax_tp++; ++ ++ if (sfreq == freq_tp/freq_scale ) ++ { ++ cbmax = cbmax_tp; ++ for(i=0,k2=0;i<cbmax_tp;i++) ++ { ++ j = (int) *p++; ++ numlines_l[i] = (int) *p++; ++ minval[i] = exp(-((*p++) - NMT) * LN_TO_LOG10); ++ qthr_l[i] = *p++; ++ /* norm_l[i] = *p++*/ p++; ++ bval_l[i] = *p++; ++ if (j!=i) ++ { ++ fprintf(stderr,"1. please check \"psy_data\""); ++ exit(-1); ++ } ++ for(k=0;k<numlines_l[i];k++) ++ partition_l[k2++] = i ; ++ } ++ } ++ else ++ p += cbmax_tp * 6; ++ } ++ ++ #define NEWBARKXXX ++ #ifdef NEWBARK ++ /* compute bark values of each critical band */ ++ j = 0; ++ for(i=0;i<cbmax;i++) ++ { ++ FLOAT8 ji, freq, bark; ++ ++ ji = j + (numlines_l[i]-1)/2.0; ++ freq = sfreq*ji/1024000.0; ++ bark = 13*atan(.76*freq) + 3.5*atan(freq*freq/(7.5*7.5)); ++ ++ printf("%i %i bval_l table=%f f=%f formaula=%f \n",i,j,bval_l[i],freq,bark); ++ bval_l[i]=bark; ++ j += numlines_l[i]; ++ } ++ #endif ++ ++ /************************************************************************ ++ * Now compute the spreading function, s[j][i], the value of the spread-* ++ * ing function, centered at band j, for band i, store for later use * ++ ************************************************************************/ ++ /* i.e.: sum over j to spread into signal barkval=i ++ NOTE: i and j are used opposite as in the ISO docs */ ++ part_max = cbmax ; ++ for(i=0;i<part_max;i++) ++ { ++ FLOAT8 tempx,x,tempy,temp; ++ for(j=0;j<part_max;j++) ++ { ++ /*tempx = (bval_l[i] - bval_l[j])*1.05;*/ ++ if (j>=i) tempx = (bval_l[i] - bval_l[j])*3.0; ++ else tempx = (bval_l[i] - bval_l[j])*1.5; ++ ++ #ifdef AACS3 ++ if (i>=j) tempx = (bval_l[i] - bval_l[j])*3.0; ++ else tempx = (bval_l[i] - bval_l[j])*1.5; ++ #endif ++ ++ if(tempx>=0.5 && tempx<=2.5) ++ { ++ temp = tempx - 0.5; ++ x = 8.0 * (temp*temp - 2.0 * temp); ++ } ++ else x = 0.0; ++ tempx += 0.474; ++ tempy = 15.811389 + 7.5*tempx - 17.5*sqrt(1.0+tempx*tempx); ++ ++ #ifdef NEWS3 ++ if (j>=i) tempy = (bval_l[j] - bval_l[i])*(-15); ++ else tempy = (bval_l[j] - bval_l[i])*25; ++ x=0; ++ #endif ++ /* ++ if ((i==part_max/2) && (fabs(bval_l[j] - bval_l[i])) < 3) { ++ printf("bark=%f x+tempy = %f \n",bval_l[j] - bval_l[i],x+tempy); ++ } ++ */ ++ ++ if (tempy <= -60.0) s3_l[i][j] = 0.0; ++ else s3_l[i][j] = exp( (x + tempy)*LN_TO_LOG10 ); ++ } ++ } ++ ++ /* Read short block data */ ++ for(loop=0;loop<6;loop++) ++ { ++ freq_tp = *p++; ++ cbmax_tp = (int) *p++; ++ cbmax_tp++; ++ ++ if (sfreq == freq_tp/freq_scale ) ++ { ++ cbmax = cbmax_tp; ++ for(i=0,k2=0;i<cbmax_tp;i++) ++ { ++ j = (int) *p++; ++ numlines_s[i] = (int) *p++; ++ qthr_s[i] = *p++; ++ /* norm_s[i] =*p++ */ p++; ++ SNR[i] = *p++; ++ bval_s[i] = *p++; ++ if (j!=i) ++ { ++ fprintf(stderr,"3. please check \"psy_data\""); ++ exit(-1); ++ } ++ numlines_s[i]--; ++ } ++ numlines_s[i] = -1; ++ } ++ else ++ p += cbmax_tp * 6; ++ } ++ ++ ++ #ifdef NEWBARK ++ /* compute bark values of each critical band */ ++ j = 0; ++ for(i=0;i<cbmax;i++) ++ { ++ FLOAT8 ji, freq, bark; ++ ji = (j * 2 + numlines_s[i]) / 2.0; ++ freq = sfreq*ji/256000.0; ++ bark = 13*atan(.76*freq) + 3.5*atan(freq*freq/(7.5*7.5)); ++ printf("%i %i bval_s = %f %f %f \n",i,j,bval_s[i],freq,bark); ++ bval_s[i]=bark; ++ j += numlines_s[i] + 1; ++ } ++ #endif ++ ++ ++ ++ /************************************************************************ ++ * Now compute the spreading function, s[j][i], the value of the spread-* ++ * ing function, centered at band j, for band i, store for later use * ++ ************************************************************************/ ++ part_max = cbmax ; ++ for(i=0;i<part_max;i++) ++ { ++ FLOAT8 tempx,x,tempy,temp; ++ for(j=0;j<part_max;j++) ++ { ++ /* tempx = (bval_s[i] - bval_s[j])*1.05;*/ ++ if (j>=i) tempx = (bval_s[i] - bval_s[j])*3.0; ++ else tempx = (bval_s[i] - bval_s[j])*1.5; ++ #ifdef AACS3 ++ if (i>=j) tempx = (bval_s[i] - bval_s[j])*3.0; ++ else tempx = (bval_s[i] - bval_s[j])*1.5; ++ #endif ++ if(tempx>=0.5 && tempx<=2.5) ++ { ++ temp = tempx - 0.5; ++ x = 8.0 * (temp*temp - 2.0 * temp); ++ } ++ else x = 0.0; ++ tempx += 0.474; ++ tempy = 15.811389 + 7.5*tempx - 17.5*sqrt(1.0+tempx*tempx); ++ #ifdef NEWS3 ++ if (j>=i) tempy = (bval_s[j] - bval_s[i])*(-15); ++ else tempy = (bval_s[j] - bval_s[i])*25; ++ x=0; ++ #endif ++ if (tempy <= -60.0) s3_s[i][j] = 0.0; ++ else s3_s[i][j] = exp( (x + tempy)*LN_TO_LOG10 ); ++ } ++ } ++ /* Read long block data for converting threshold calculation ++ partitions to scale factor bands */ ++ ++ for(loop=0;loop<6;loop++) ++ { ++ freq_tp = *p++; ++ sbmax = (int) *p++; ++ sbmax++; ++ ++ if (sfreq == freq_tp/freq_scale) ++ { ++ for(i=0;i<sbmax;i++) ++ { ++ j = (int) *p++; ++ p++; ++ bu_l[i] = (int) *p++; ++ bo_l[i] = (int) *p++; ++ w1_l[i] = (FLOAT8) *p++; ++ w2_l[i] = (FLOAT8) *p++; ++ if (j!=i) ++ { fprintf(stderr,"30:please check \"psy_data\"\n"); ++ exit(-1); ++ } ++ ++ if (i!=0) ++ if ( (fabs(1.0-w1_l[i]-w2_l[i-1]) > 0.01 ) ) ++ { ++ fprintf(stderr,"31l: please check \"psy_data.\"\n"); ++ fprintf(stderr,"w1,w2: %f %f \n",w1_l[i],w2_l[i-1]); ++ exit(-1); ++ } ++ } ++ } ++ else ++ p += sbmax * 6; ++ } ++ ++ /* Read short block data for converting threshold calculation ++ partitions to scale factor bands */ ++ ++ for(loop=0;loop<6;loop++) ++ { ++ freq_tp = *p++; ++ sbmax = (int) *p++; ++ sbmax++; ++ ++ if (sfreq == freq_tp/freq_scale) ++ { ++ for(i=0;i<sbmax;i++) ++ { ++ j = (int) *p++; ++ p++; ++ bu_s[i] = (int) *p++; ++ bo_s[i] = (int) *p++; ++ w1_s[i] = *p++; ++ w2_s[i] = *p++; ++ if (j!=i) ++ { fprintf(stderr,"30:please check \"psy_data\"\n"); ++ exit(-1); ++ } ++ ++ if (i!=0) ++ if ( (fabs(1.0-w1_s[i]-w2_s[i-1]) > 0.01 ) ) ++ { ++ fprintf(stderr,"31s: please check \"psy_data.\"\n"); ++ fprintf(stderr,"w1,w2: %f %f \n",w1_s[i],w2_s[i-1]); ++ exit(-1); ++ } ++ } ++ } ++ else ++ p += sbmax * 6; ++ } ++ ++ } +diff -r -c -N encoder/psymodel.h lame3.70/psymodel.h +*** encoder/psymodel.h Wed Dec 31 17:00:00 1969 +--- lame3.70/psymodel.h Fri Mar 24 11:07:30 2000 +*************** +*** 0 **** +--- 1,15 ---- ++ #ifndef L3PSY_DOT_H_ ++ #define L3PSY_DOT_H_ ++ ++ /* l3psy.c */ ++ #include "l3side.h" ++ void L3psycho_anal( lame_global_flags *gfp, ++ short int *buffer[2], int gr , ++ FLOAT8 *ms_ratio, ++ FLOAT8 *ms_ratio_next, ++ FLOAT8 *ms_ener_ratio, ++ III_psy_ratio ratio[2][2], ++ III_psy_ratio MS_ratio[2][2], ++ FLOAT8 pe[2], FLOAT8 pe_MS[2], ++ int blocktype_d[2]); ++ #endif +diff -r -c -N encoder/quantize-pvt.c lame3.70/quantize-pvt.c +*** encoder/quantize-pvt.c Wed Dec 31 17:00:00 1969 +--- lame3.70/quantize-pvt.c Thu Apr 6 12:50:51 2000 +*************** +*** 0 **** +--- 1,1318 ---- ++ #include <assert.h> ++ #include "util.h" ++ #include "tables.h" ++ #include "reservoir.h" ++ #include "quantize-pvt.h" ++ ++ FLOAT masking_lower=1; ++ int convert_mdct, reduce_sidechannel; ++ /* ++ mt 5/99. These global flags denote 4 possibilities: ++ mode l3_xmin ++ 1 MDCT input L/R, quantize L/R, psy-model thresholds: L/R -m s either ++ 2 MDCT input L/R, quantize M/S, psy-model thresholds: L/R -m j orig ++ 3 MDCT input M/S, quantize M/S, psy-model thresholds: M/S -m f either ++ 4 MDCT input L/R, quantize M/S, psy-model thresholds: M/S -m j -h m/s ++ ++ 1: convert_mdct = 0, convert_psy=0, reduce_sidechannel=0 ++ 2: convert_mdct = 1, convert_psy=1, reduce_sidechannel=1 ++ 3: convert_mdct = 0, convert_psy=0, reduce_sidechannel=1 (this mode no longer used) ++ 4: convert_mdct = 1, convert_psy=0, reduce_sidechannel=1 ++ ++ if (convert_mdct), then iteration_loop will quantize M/S data from ++ the L/R input MDCT coefficients. ++ ++ if (convert_psy), then calc_noise will compute the noise for the L/R ++ channels from M/S MDCT data and L/R psy-model threshold information. ++ Distortion in ether L or R channel will be marked as distortion in ++ both Mid and Side channels. ++ NOTE: 3/00: this mode has been removed. ++ ++ if (reduce_sidechannel) then outer_loop will allocate less bits ++ to the side channel and more bits to the mid channel based on relative ++ energies. ++ */ ++ ++ ++ ++ /* ++ The following table is used to implement the scalefactor ++ partitioning for MPEG2 as described in section ++ 2.4.3.2 of the IS. The indexing corresponds to the ++ way the tables are presented in the IS: ++ ++ [table_number][row_in_table][column of nr_of_sfb] ++ */ ++ unsigned nr_of_sfb_block[6][3][4] = ++ { ++ { ++ {6, 5, 5, 5}, ++ {9, 9, 9, 9}, ++ {6, 9, 9, 9} ++ }, ++ { ++ {6, 5, 7, 3}, ++ {9, 9, 12, 6}, ++ {6, 9, 12, 6} ++ }, ++ { ++ {11, 10, 0, 0}, ++ {18, 18, 0, 0}, ++ {15,18,0,0} ++ }, ++ { ++ {7, 7, 7, 0}, ++ {12, 12, 12, 0}, ++ {6, 15, 12, 0} ++ }, ++ { ++ {6, 6, 6, 3}, ++ {12, 9, 9, 6}, ++ {6, 12, 9, 6} ++ }, ++ { ++ {8, 8, 5, 0}, ++ {15,12,9,0}, ++ {6,18,9,0} ++ } ++ }; ++ ++ ++ /* Table B.6: layer3 preemphasis */ ++ int pretab[21] = ++ { ++ 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 1, 1, 1, 1, 2, 2, 3, 3, 3, 2 ++ }; ++ ++ /* ++ Here are MPEG1 Table B.8 and MPEG2 Table B.1 ++ -- Layer III scalefactor bands. ++ Index into this using a method such as: ++ idx = fr_ps->header->sampling_frequency ++ + (fr_ps->header->version * 3) ++ */ ++ ++ struct scalefac_struct sfBandIndex[6] = ++ { ++ { /* Table B.2.b: 22.05 kHz */ ++ {0,6,12,18,24,30,36,44,54,66,80,96,116,140,168,200,238,284,336,396,464,522,576}, ++ {0,4,8,12,18,24,32,42,56,74,100,132,174,192} ++ }, ++ { /* Table B.2.c: 24 kHz */ /* docs: 332. mpg123: 330 */ ++ {0,6,12,18,24,30,36,44,54,66,80,96,114,136,162,194,232,278, 332, 394,464,540,576}, ++ {0,4,8,12,18,26,36,48,62,80,104,136,180,192} ++ }, ++ { /* Table B.2.a: 16 kHz */ ++ {0,6,12,18,24,30,36,44,54,66,80,96,116,140,168,200,238,284,336,396,464,522,576}, ++ {0,4,8,12,18,26,36,48,62,80,104,134,174,192} ++ }, ++ { /* Table B.8.b: 44.1 kHz */ ++ {0,4,8,12,16,20,24,30,36,44,52,62,74,90,110,134,162,196,238,288,342,418,576}, ++ {0,4,8,12,16,22,30,40,52,66,84,106,136,192} ++ }, ++ { /* Table B.8.c: 48 kHz */ ++ {0,4,8,12,16,20,24,30,36,42,50,60,72,88,106,128,156,190,230,276,330,384,576}, ++ {0,4,8,12,16,22,28,38,50,64,80,100,126,192} ++ }, ++ { /* Table B.8.a: 32 kHz */ ++ {0,4,8,12,16,20,24,30,36,44,54,66,82,102,126,156,194,240,296,364,448,550,576}, ++ {0,4,8,12,16,22,30,42,58,78,104,138,180,192} ++ } ++ }; ++ ++ struct scalefac_struct scalefac_band; ++ ++ ++ FLOAT8 pow20[Q_MAX]; ++ FLOAT8 ipow20[Q_MAX]; ++ FLOAT8 pow43[PRECALC_SIZE]; ++ static FLOAT8 adj43[PRECALC_SIZE]; ++ static FLOAT8 adj43asm[PRECALC_SIZE]; ++ static FLOAT8 ATH_l[SBPSY_l]; ++ static FLOAT8 ATH_s[SBPSY_l]; ++ ++ FLOAT8 ATH_mdct_long[576]; ++ FLOAT8 ATH_mdct_short[192]; ++ ++ ++ /************************************************************************/ ++ /* initialization for iteration_loop */ ++ /************************************************************************/ ++ void ++ iteration_init( lame_global_flags *gfp,III_side_info_t *l3_side, int l3_enc[2][2][576]) ++ { ++ gr_info *cod_info; ++ int ch, gr, i; ++ ++ l3_side->resvDrain = 0; ++ ++ if ( gfp->frameNum==0 ) { ++ for (i = 0; i < SBMAX_l + 1; i++) { ++ scalefac_band.l[i] = ++ sfBandIndex[gfp->samplerate_index + (gfp->version * 3)].l[i]; ++ } ++ for (i = 0; i < SBMAX_s + 1; i++) { ++ scalefac_band.s[i] = ++ sfBandIndex[gfp->samplerate_index + (gfp->version * 3)].s[i]; ++ } ++ ++ l3_side->main_data_begin = 0; ++ compute_ath(gfp,ATH_l,ATH_s); ++ ++ for(i=0;i<PRECALC_SIZE;i++) ++ pow43[i] = pow((FLOAT8)i, 4.0/3.0); ++ ++ for (i = 0; i < PRECALC_SIZE-1; i++) ++ adj43[i] = (i + 1) - pow(0.5 * (pow43[i] + pow43[i + 1]), 0.75); ++ adj43[i] = 0.5; ++ ++ ++ adj43asm[0] = 0.0; ++ for (i = 1; i < PRECALC_SIZE; i++) ++ adj43asm[i] = i - 0.5 - pow(0.5 * (pow43[i - 1] + pow43[i]),0.75); ++ ++ for (i = 0; i < Q_MAX; i++) { ++ ipow20[i] = pow(2.0, (double)(i - 210) * -0.1875); ++ pow20[i] = pow(2.0, (double)(i - 210) * 0.25); ++ } ++ } ++ ++ ++ convert_mdct=0; ++ reduce_sidechannel=0; ++ if (gfp->mode_ext==MPG_MD_MS_LR) { ++ convert_mdct = 1; ++ reduce_sidechannel=1; ++ } ++ ++ /* some intializations. */ ++ for ( gr = 0; gr < gfp->mode_gr; gr++ ){ ++ for ( ch = 0; ch < gfp->stereo; ch++ ){ ++ cod_info = (gr_info *) &(l3_side->gr[gr].ch[ch]); ++ ++ if (cod_info->block_type == SHORT_TYPE) ++ { ++ cod_info->sfb_lmax = 0; /* No sb*/ ++ cod_info->sfb_smax = 0; ++ } ++ else ++ { ++ /* MPEG 1 doesnt use last scalefactor band */ ++ cod_info->sfb_lmax = SBPSY_l; ++ cod_info->sfb_smax = SBPSY_s; /* No sb */ ++ } ++ ++ } ++ } ++ ++ ++ /* dont bother with scfsi. */ ++ for ( ch = 0; ch < gfp->stereo; ch++ ) ++ for ( i = 0; i < 4; i++ ) ++ l3_side->scfsi[ch][i] = 0; ++ } ++ ++ ++ ++ ++ ++ /* ++ compute the ATH for each scalefactor band ++ cd range: 0..96db ++ ++ Input: 3.3kHz signal 32767 amplitude (3.3kHz is where ATH is smallest = -5db) ++ longblocks: sfb=12 en0/bw=-11db max_en0 = 1.3db ++ shortblocks: sfb=5 -9db 0db ++ ++ Input: 1 1 1 1 1 1 1 -1 -1 -1 -1 -1 -1 -1 (repeated) ++ longblocks: amp=1 sfb=12 en0/bw=-103 db max_en0 = -92db ++ amp=32767 sfb=12 -12 db -1.4db ++ ++ Input: 1 1 1 1 1 1 1 -1 -1 -1 -1 -1 -1 -1 (repeated) ++ shortblocks: amp=1 sfb=5 en0/bw= -99 -86 ++ amp=32767 sfb=5 -9 db 4db ++ ++ ++ MAX energy of largest wave at 3.3kHz = 1db ++ AVE energy of largest wave at 3.3kHz = -11db ++ Let's take AVE: -11db = maximum signal in sfb=12. ++ Dynamic range of CD: 96db. Therefor energy of smallest audible wave ++ in sfb=12 = -11 - 96 = -107db = ATH at 3.3kHz. ++ ++ ATH formula for this wave: -5db. To adjust to LAME scaling, we need ++ ATH = ATH_formula - 103 (db) ++ ATH = ATH * 2.5e-10 (ener) ++ ++ */ ++ FLOAT8 ATHformula(lame_global_flags *gfp,FLOAT8 f) ++ { ++ FLOAT8 ath; ++ f = Max(0.02, f); ++ /* from Painter & Spanias, 1997 */ ++ /* minimum: (i=77) 3.3kHz = -5db */ ++ ath=(3.640 * pow(f,-0.8) ++ - 6.500 * exp(-0.6*pow(f-3.3,2.0)) ++ + 0.001 * pow(f,4.0)); ++ /* convert to energy */ ++ if (gfp->noATH) ++ ath -= 200; /* disables ATH */ ++ else { ++ ath -= 114; /* MDCT scaling. From tests by macik and MUS420 code */ ++ /* ath -= 109; */ ++ } ++ #ifdef RH_QUALITY_CONTROL ++ /* purpose of RH_QUALITY_CONTROL: ++ * at higher quality lower ATH masking abilities => needs more bits ++ * at lower quality increase ATH masking abilities => needs less bits ++ * works together with adjusted masking lowering of GPSYCHO thresholds ++ * (Robert.Hegemann@gmx.de 2000-01-30) ++ */ ++ ath -= (4-gfp->VBR_q)*4.0; ++ #endif ++ ath = pow( 10.0, ath/10.0 ); ++ return ath; ++ } ++ ++ ++ void compute_ath(lame_global_flags *gfp,FLOAT8 ATH_l[SBPSY_l],FLOAT8 ATH_s[SBPSY_l]) ++ { ++ int sfb,i,start,end; ++ FLOAT8 ATH_f; ++ FLOAT8 samp_freq = gfp->out_samplerate/1000.0; ++ #ifdef RH_ATH ++ /* going from average to peak level ATH masking ++ */ ++ FLOAT8 adjust_mdct_scaling = 10.0; ++ #endif ++ ++ ++ /* last sfb is not used */ ++ for ( sfb = 0; sfb < SBPSY_l; sfb++ ) { ++ start = scalefac_band.l[ sfb ]; ++ end = scalefac_band.l[ sfb+1 ]; ++ ATH_l[sfb]=1e99; ++ for (i=start ; i < end; i++) { ++ ATH_f = ATHformula(gfp,samp_freq*i/(2*576)); /* freq in kHz */ ++ ATH_l[sfb]=Min(ATH_l[sfb],ATH_f); ++ #ifdef RH_ATH ++ ATH_mdct_long[i] = ATH_f*adjust_mdct_scaling; ++ #endif ++ } ++ /* ++ printf("sfb=%i %f ATH=%f %f %f \n",sfb,samp_freq*start/(2*576), ++ 10*log10(ATH_l[sfb]), ++ 10*log10( ATHformula(samp_freq*start/(2*576))) , ++ 10*log10(ATHformula(samp_freq*end/(2*576)))); ++ */ ++ } ++ ++ for ( sfb = 0; sfb < SBPSY_s; sfb++ ){ ++ start = scalefac_band.s[ sfb ]; ++ end = scalefac_band.s[ sfb+1 ]; ++ ATH_s[sfb]=1e99; ++ for (i=start ; i < end; i++) { ++ ATH_f = ATHformula(gfp,samp_freq*i/(2*192)); /* freq in kHz */ ++ ATH_s[sfb]=Min(ATH_s[sfb],ATH_f); ++ #ifdef RH_ATH ++ ATH_mdct_short[i] = ATH_f*adjust_mdct_scaling; ++ #endif ++ } ++ } ++ } ++ ++ ++ ++ ++ ++ /* convert from L/R <-> Mid/Side */ ++ void ms_convert(FLOAT8 xr[2][576],FLOAT8 xr_org[2][576]) ++ { ++ int i; ++ for ( i = 0; i < 576; i++ ) { ++ FLOAT8 l = xr_org[0][i]; ++ FLOAT8 r = xr_org[1][i]; ++ xr[0][i] = (l+r)*(SQRT2*0.5); ++ xr[1][i] = (l-r)*(SQRT2*0.5); ++ } ++ } ++ ++ ++ ++ /************************************************************************ ++ * allocate bits among 2 channels based on PE ++ * mt 6/99 ++ ************************************************************************/ ++ void on_pe(lame_global_flags *gfp,FLOAT8 pe[2][2],III_side_info_t *l3_side, ++ int targ_bits[2],int mean_bits, int gr) ++ { ++ gr_info *cod_info; ++ int extra_bits,tbits,bits; ++ int add_bits[2]; ++ int ch; ++ ++ /* allocate targ_bits for granule */ ++ ResvMaxBits( mean_bits, &tbits, &extra_bits, gr); ++ ++ ++ for (ch=0 ; ch < gfp->stereo ; ch ++) { ++ /****************************************************************** ++ * allocate bits for each channel ++ ******************************************************************/ ++ cod_info = &l3_side->gr[gr].ch[ch].tt; ++ ++ targ_bits[ch]=tbits/gfp->stereo; ++ ++ /* allocate extra bits from reservoir based on PE */ ++ bits=0; ++ ++ /* extra bits based on PE > 700 */ ++ add_bits[ch]=(pe[gr][ch]-750)/1.55; /* 1.4; */ ++ ++ /* short blocks need extra, no matter what the pe */ ++ if (cod_info->block_type==SHORT_TYPE) ++ if (add_bits[ch]<500) add_bits[ch]=500; ++ ++ if (add_bits[ch] < 0) add_bits[ch]=0; ++ bits += add_bits[ch]; ++ ++ if (bits > extra_bits) add_bits[ch] = (extra_bits*add_bits[ch])/bits; ++ if ((targ_bits[ch]+add_bits[ch]) > 4095) ++ add_bits[ch]=4095-targ_bits[ch]; ++ ++ targ_bits[ch] = targ_bits[ch] + add_bits[ch]; ++ extra_bits -= add_bits[ch]; ++ } ++ } ++ ++ void reduce_side(int targ_bits[2],FLOAT8 ms_ener_ratio,int mean_bits) ++ { ++ int ch; ++ int numchn=2; ++ /* ms_ener_ratio = 0: allocate 66/33 mid/side fac=.33 ++ * ms_ener_ratio =.5: allocate 50/50 mid/side fac= 0 */ ++ /* 75/25 split is fac=.5 */ ++ /* float fac = .50*(.5-ms_ener_ratio[gr])/.5;*/ ++ float fac = .33*(.5-ms_ener_ratio)/.5; ++ if (fac<0) fac=0; ++ ++ if (targ_bits[1] >= 125) { ++ /* dont reduce side channel below 125 bits */ ++ if (targ_bits[1]-targ_bits[1]*fac > 125) { ++ targ_bits[0] += targ_bits[1]*fac; ++ targ_bits[1] -= targ_bits[1]*fac; ++ } else { ++ targ_bits[0] += targ_bits[1] - 125; ++ targ_bits[1] = 125; ++ } ++ } ++ ++ /* dont allow to many bits per channel */ ++ for (ch=0; ch<numchn; ch++) { ++ int max_bits = Min(4095,mean_bits/2 + 1200); ++ if (targ_bits[ch] > max_bits) { ++ targ_bits[ch] = max_bits; ++ } ++ } ++ ++ } ++ ++ /*************************************************************************** ++ * inner_loop * ++ *************************************************************************** ++ * The code selects the best global gain for a particular set of scalefacs */ ++ ++ int ++ inner_loop( lame_global_flags *gfp,FLOAT8 xrpow[576], ++ int l3_enc[576], int max_bits, ++ gr_info *cod_info) ++ { ++ int bits; ++ assert( max_bits >= 0 ); ++ cod_info->global_gain--; ++ do ++ { ++ cod_info->global_gain++; ++ bits = count_bits(gfp,l3_enc, xrpow, cod_info); ++ } ++ while ( bits > max_bits ); ++ return bits; ++ } ++ ++ ++ ++ /*************************************************************************/ ++ /* scale_bitcount */ ++ /*************************************************************************/ ++ ++ /* Also calculates the number of bits necessary to code the scalefactors. */ ++ ++ int scale_bitcount( III_scalefac_t *scalefac, gr_info *cod_info) ++ { ++ int i, k, sfb, max_slen1 = 0, max_slen2 = 0, /*a, b, */ ep = 2; ++ ++ static int slen1[16] = { 1, 1, 1, 1, 8, 2, 2, 2, 4, 4, 4, 8, 8, 8,16,16 }; ++ static int slen2[16] = { 1, 2, 4, 8, 1, 2, 4, 8, 2, 4, 8, 2, 4, 8, 4, 8 }; ++ ++ static int slen1_tab[16] = {0, ++ 18, 36, 54, 54, 36, 54, 72, 54, 72, 90, 72, 90,108,108,126 ++ }; ++ static int slen2_tab[16] = {0, ++ 10, 20, 30, 33, 21, 31, 41, 32, 42, 52, 43, 53, 63, 64, 74 ++ }; ++ int *tab; ++ ++ ++ if ( cod_info->block_type == SHORT_TYPE ) ++ { ++ tab = slen1_tab; ++ /* a = 18; b = 18; */ ++ for ( i = 0; i < 3; i++ ) ++ { ++ for ( sfb = 0; sfb < 6; sfb++ ) ++ if (scalefac->s[sfb][i] > max_slen1 ) ++ max_slen1 = scalefac->s[sfb][i]; ++ for (sfb = 6; sfb < SBPSY_s; sfb++ ) ++ if ( scalefac->s[sfb][i] > max_slen2 ) ++ max_slen2 = scalefac->s[sfb][i]; ++ } ++ } ++ else ++ { /* block_type == 1,2,or 3 */ ++ tab = slen2_tab; ++ /* a = 11; b = 10; */ ++ for ( sfb = 0; sfb < 11; sfb++ ) ++ if ( scalefac->l[sfb] > max_slen1 ) ++ max_slen1 = scalefac->l[sfb]; ++ ++ if (!cod_info->preflag) { ++ for ( sfb = 11; sfb < SBPSY_l; sfb++ ) ++ if (scalefac->l[sfb] < pretab[sfb]) ++ break; ++ ++ if (sfb == SBPSY_l) { ++ cod_info->preflag = 1; ++ for ( sfb = 11; sfb < SBPSY_l; sfb++ ) ++ scalefac->l[sfb] -= pretab[sfb]; ++ } ++ } ++ ++ for ( sfb = 11; sfb < SBPSY_l; sfb++ ) ++ if ( scalefac->l[sfb] > max_slen2 ) ++ max_slen2 = scalefac->l[sfb]; ++ } ++ ++ ++ ++ /* from Takehiro TOMINAGA <tominaga@isoternet.org> 10/99 ++ * loop over *all* posible values of scalefac_compress to find the ++ * one which uses the smallest number of bits. ISO would stop ++ * at first valid index */ ++ cod_info->part2_length = LARGE_BITS; ++ for ( k = 0; k < 16; k++ ) ++ { ++ if ( (max_slen1 < slen1[k]) && (max_slen2 < slen2[k]) && ++ ((int)cod_info->part2_length > tab[k])) { ++ cod_info->part2_length=tab[k]; ++ cod_info->scalefac_compress=k; ++ ep=0; /* we found a suitable scalefac_compress */ ++ } ++ } ++ return ep; ++ } ++ ++ ++ ++ /* ++ table of largest scalefactors (number of bits) for MPEG2 ++ */ ++ /* ++ static unsigned max_sfac_tab[6][4] = ++ { ++ {4, 4, 3, 3}, ++ {4, 4, 3, 0}, ++ {3, 2, 0, 0}, ++ {4, 5, 5, 0}, ++ {3, 3, 3, 0}, ++ {2, 2, 0, 0} ++ }; ++ */ ++ /* ++ table of largest scalefactor values for MPEG2 ++ */ ++ static unsigned max_range_sfac_tab[6][4] = ++ { ++ { 15, 15, 7, 7}, ++ { 15, 15, 7, 0}, ++ { 7, 3, 0, 0}, ++ { 15, 31, 31, 0}, ++ { 7, 7, 7, 0}, ++ { 3, 3, 0, 0} ++ }; ++ ++ ++ ++ ++ ++ /*************************************************************************/ ++ /* scale_bitcount_lsf */ ++ /*************************************************************************/ ++ ++ /* Also counts the number of bits to encode the scalefacs but for MPEG 2 */ ++ /* Lower sampling frequencies (24, 22.05 and 16 kHz.) */ ++ ++ /* This is reverse-engineered from section 2.4.3.2 of the MPEG2 IS, */ ++ /* "Audio Decoding Layer III" */ ++ ++ int scale_bitcount_lsf(III_scalefac_t *scalefac, gr_info *cod_info) ++ { ++ int table_number, row_in_table, partition, nr_sfb, window, over; ++ int i, sfb, max_sfac[ 4 ]; ++ unsigned *partition_table; ++ ++ /* ++ Set partition table. Note that should try to use table one, ++ but do not yet... ++ */ ++ if ( cod_info->preflag ) ++ table_number = 2; ++ else ++ table_number = 0; ++ ++ for ( i = 0; i < 4; i++ ) ++ max_sfac[i] = 0; ++ ++ if ( cod_info->block_type == SHORT_TYPE ) ++ { ++ row_in_table = 1; ++ partition_table = &nr_of_sfb_block[table_number][row_in_table][0]; ++ for ( sfb = 0, partition = 0; partition < 4; partition++ ) ++ { ++ nr_sfb = partition_table[ partition ] / 3; ++ for ( i = 0; i < nr_sfb; i++, sfb++ ) ++ for ( window = 0; window < 3; window++ ) ++ if ( scalefac->s[sfb][window] > max_sfac[partition] ) ++ max_sfac[partition] = scalefac->s[sfb][window]; ++ } ++ } ++ else ++ { ++ row_in_table = 0; ++ partition_table = &nr_of_sfb_block[table_number][row_in_table][0]; ++ for ( sfb = 0, partition = 0; partition < 4; partition++ ) ++ { ++ nr_sfb = partition_table[ partition ]; ++ for ( i = 0; i < nr_sfb; i++, sfb++ ) ++ if ( scalefac->l[sfb] > max_sfac[partition] ) ++ max_sfac[partition] = scalefac->l[sfb]; ++ } ++ } ++ ++ for ( over = 0, partition = 0; partition < 4; partition++ ) ++ { ++ if ( max_sfac[partition] > (int)max_range_sfac_tab[table_number][partition] ) ++ over++; ++ } ++ if ( !over ) ++ { ++ /* ++ Since no bands have been over-amplified, we can set scalefac_compress ++ and slen[] for the formatter ++ */ ++ static int log2tab[] = { 0, 1, 2, 2, 3, 3, 3, 3, 4, 4, 4, 4, 4, 4, 4, 4 }; ++ ++ unsigned slen1, slen2, slen3, slen4; ++ ++ cod_info->sfb_partition_table = &nr_of_sfb_block[table_number][row_in_table][0]; ++ for ( partition = 0; partition < 4; partition++ ) ++ cod_info->slen[partition] = log2tab[max_sfac[partition]]; ++ ++ /* set scalefac_compress */ ++ slen1 = cod_info->slen[ 0 ]; ++ slen2 = cod_info->slen[ 1 ]; ++ slen3 = cod_info->slen[ 2 ]; ++ slen4 = cod_info->slen[ 3 ]; ++ ++ switch ( table_number ) ++ { ++ case 0: ++ cod_info->scalefac_compress = (((slen1 * 5) + slen2) << 4) ++ + (slen3 << 2) ++ + slen4; ++ break; ++ ++ case 1: ++ cod_info->scalefac_compress = 400 ++ + (((slen1 * 5) + slen2) << 2) ++ + slen3; ++ break; ++ ++ case 2: ++ cod_info->scalefac_compress = 500 + (slen1 * 3) + slen2; ++ break; ++ ++ default: ++ fprintf( stderr, "intensity stereo not implemented yet\n" ); ++ exit( EXIT_FAILURE ); ++ break; ++ } ++ } ++ #ifdef DEBUG ++ if ( over ) ++ printf( "---WARNING !! Amplification of some bands over limits\n" ); ++ #endif ++ if (!over) { ++ assert( cod_info->sfb_partition_table ); ++ cod_info->part2_length=0; ++ for ( partition = 0; partition < 4; partition++ ) ++ cod_info->part2_length += cod_info->slen[partition] * cod_info->sfb_partition_table[partition]; ++ } ++ return over; ++ } ++ ++ ++ ++ ++ ++ /*************************************************************************/ ++ /* calc_xmin */ ++ /*************************************************************************/ ++ ++ /* ++ Calculate the allowed distortion for each scalefactor band, ++ as determined by the psychoacoustic model. ++ xmin(sb) = ratio(sb) * en(sb) / bw(sb) ++ ++ returns number of sfb's with energy > ATH ++ */ ++ int calc_xmin( lame_global_flags *gfp,FLOAT8 xr[576], III_psy_ratio *ratio, ++ gr_info *cod_info, III_psy_xmin *l3_xmin) ++ { ++ int start, end, bw,l, b, ath_over=0; ++ u_int sfb; ++ FLOAT8 en0, xmin, ener; ++ ++ if (gfp->ATHonly) { ++ for ( sfb = cod_info->sfb_smax; sfb < SBPSY_s; sfb++ ) ++ for ( b = 0; b < 3; b++ ) ++ l3_xmin->s[sfb][b]=ATH_s[sfb]; ++ for ( sfb = 0; sfb < cod_info->sfb_lmax; sfb++ ) ++ l3_xmin->l[sfb]=ATH_l[sfb]; ++ ++ }else{ ++ ++ for ( sfb = cod_info->sfb_smax; sfb < SBPSY_s; sfb++ ) { ++ start = scalefac_band.s[ sfb ]; ++ end = scalefac_band.s[ sfb + 1 ]; ++ bw = end - start; ++ for ( b = 0; b < 3; b++ ) { ++ for (en0 = 0.0, l = start; l < end; l++) { ++ ener = xr[l * 3 + b]; ++ ener = ener * ener; ++ en0 += ener; ++ } ++ en0 /= bw; ++ ++ xmin = ratio->en.s[sfb][b]; ++ if (xmin > 0.0) ++ xmin = en0 * ratio->thm.s[sfb][b] * masking_lower / xmin; ++ ++ #ifdef RH_ATH ++ /* do not mix up ATH masking with GPSYCHO thresholds ++ */ ++ l3_xmin->s[sfb][b] = Max(1e-20, xmin); ++ #else ++ l3_xmin->s[sfb][b] = Max(ATH_s[sfb], xmin); ++ #endif ++ if (en0 > ATH_s[sfb]) ath_over++; ++ } ++ } ++ ++ for ( sfb = 0; sfb < cod_info->sfb_lmax; sfb++ ){ ++ start = scalefac_band.l[ sfb ]; ++ end = scalefac_band.l[ sfb+1 ]; ++ bw = end - start; ++ ++ for (en0 = 0.0, l = start; l < end; l++ ) { ++ ener = xr[l] * xr[l]; ++ en0 += ener; ++ } ++ en0 /= bw; ++ ++ xmin = ratio->en.l[sfb]; ++ if (xmin > 0.0) ++ xmin = en0 * ratio->thm.l[sfb] * masking_lower / xmin; ++ ++ ++ #ifdef RH_ATH ++ /* do not mix up ATH masking with GPSYCHO thresholds ++ */ ++ l3_xmin->l[sfb]=Max(1e-20, xmin); ++ #else ++ l3_xmin->l[sfb]=Max(ATH_l[sfb], xmin); ++ #endif ++ if (en0 > ATH_l[sfb]) ath_over++; ++ } ++ } ++ return ath_over; ++ } ++ ++ ++ ++ /*************************************************************************/ ++ /* loop_break */ ++ /*************************************************************************/ ++ ++ /* Function: Returns zero if there is a scalefac which has not been ++ amplified. Otherwise it returns one. ++ */ ++ ++ int loop_break( III_scalefac_t *scalefac, gr_info *cod_info) ++ { ++ int i; ++ u_int sfb; ++ ++ for ( sfb = 0; sfb < cod_info->sfb_lmax; sfb++ ) ++ if ( scalefac->l[sfb] == 0 ) ++ return 0; ++ ++ for ( sfb = cod_info->sfb_smax; sfb < SBPSY_s; sfb++ ) ++ for ( i = 0; i < 3; i++ ) ++ if ( scalefac->s[sfb][i] == 0 ) ++ return 0; ++ ++ return 1; ++ } ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ /* ++ ---------------------------------------------------------------------- ++ if someone wants to try to find a faster step search function, ++ here is some code which gives a lower bound for the step size: ++ ++ for (max_xrspow = 0, i = 0; i < 576; ++i) ++ { ++ max_xrspow = Max(max_xrspow, xrspow[i]); ++ } ++ lowerbound = 210+log10(max_xrspow/IXMAX_VAL)/(0.1875*LOG2); ++ ++ ++ Robert.Hegemann@gmx.de ++ ---------------------------------------------------------------------- ++ */ ++ ++ ++ typedef enum { ++ BINSEARCH_NONE, ++ BINSEARCH_UP, ++ BINSEARCH_DOWN ++ } binsearchDirection_t; ++ ++ /*-------------------------------------------------------------------------*/ ++ int ++ bin_search_StepSize2 (lame_global_flags *gfp,int desired_rate, int start, int *ix, ++ FLOAT8 xrspow[576], gr_info *cod_info) ++ /*-------------------------------------------------------------------------*/ ++ { ++ static int CurrentStep = 4; ++ int nBits; ++ int flag_GoneOver = 0; ++ int StepSize = start; ++ binsearchDirection_t Direction = BINSEARCH_NONE; ++ ++ do ++ { ++ cod_info->global_gain = StepSize; ++ nBits = count_bits(gfp,ix, xrspow, cod_info); ++ ++ if (CurrentStep == 1 ) ++ { ++ break; /* nothing to adjust anymore */ ++ } ++ if (flag_GoneOver) ++ { ++ CurrentStep /= 2; ++ } ++ if (nBits > desired_rate) /* increase Quantize_StepSize */ ++ { ++ if (Direction == BINSEARCH_DOWN && !flag_GoneOver) ++ { ++ flag_GoneOver = 1; ++ CurrentStep /= 2; /* late adjust */ ++ } ++ Direction = BINSEARCH_UP; ++ StepSize += CurrentStep; ++ if (StepSize > 255) break; ++ } ++ else if (nBits < desired_rate) ++ { ++ if (Direction == BINSEARCH_UP && !flag_GoneOver) ++ { ++ flag_GoneOver = 1; ++ CurrentStep /= 2; /* late adjust */ ++ } ++ Direction = BINSEARCH_DOWN; ++ StepSize -= CurrentStep; ++ if (StepSize < 0) break; ++ } ++ else break; /* nBits == desired_rate;; most unlikely to happen.*/ ++ } while (1); /* For-ever, break is adjusted. */ ++ ++ CurrentStep = abs(start - StepSize); ++ ++ if (CurrentStep >= 4) { ++ CurrentStep = 4; ++ } else { ++ CurrentStep = 2; ++ } ++ ++ return nBits; ++ } ++ ++ ++ ++ ++ ++ ++ ++ ++ #if (defined(__GNUC__) && defined(__i386__)) ++ #define USE_GNUC_ASM ++ #endif ++ #ifdef _MSC_VER ++ #define USE_MSC_ASM ++ #endif ++ ++ ++ ++ /********************************************************************* ++ * XRPOW_FTOI is a macro to convert floats to ints. ++ * if XRPOW_FTOI(x) = nearest_int(x), then QUANTFAC(x)=adj43asm[x] ++ * ROUNDFAC= -0.0946 ++ * ++ * if XRPOW_FTOI(x) = floor(x), then QUANTFAC(x)=asj43[x] ++ * ROUNDFAC=0.4054 ++ *********************************************************************/ ++ #ifdef USE_GNUC_ASM ++ # define QUANTFAC(rx) adj43asm[rx] ++ # define ROUNDFAC -0.0946 ++ # define XRPOW_FTOI(src, dest) \ ++ asm ("fistpl %0 " : "=m"(dest) : "t"(src) : "st") ++ #elif defined (USE_MSC_ASM) ++ # define QUANTFAC(rx) adj43asm[rx] ++ # define ROUNDFAC -0.0946 ++ # define XRPOW_FTOI(src, dest) do { \ ++ FLOAT8 src_ = (src); \ ++ int dest_; \ ++ { \ ++ __asm fld src_ \ ++ __asm fistp dest_ \ ++ } \ ++ (dest) = dest_; \ ++ } while (0) ++ #else ++ # define QUANTFAC(rx) adj43[rx] ++ # define ROUNDFAC 0.4054 ++ # define XRPOW_FTOI(src,dest) ((dest) = (int)(src)) ++ #endif ++ ++ #ifdef USE_MSC_ASM ++ /* define F8type and F8size according to type of FLOAT8 */ ++ # if defined FLOAT8_is_double ++ # define F8type qword ++ # define F8size 8 ++ # elif defined FLOAT8_is_float ++ # define F8type dword ++ # define F8size 4 ++ # else ++ /* only float and double supported */ ++ # error invalid FLOAT8 type for USE_MSC_ASM ++ # endif ++ #endif ++ ++ #ifdef USE_GNUC_ASM ++ /* define F8type and F8size according to type of FLOAT8 */ ++ # if defined FLOAT8_is_double ++ # define F8type "l" ++ # define F8size "8" ++ # elif defined FLOAT8_is_float ++ # define F8type "s" ++ # define F8size "4" ++ # else ++ /* only float and double supported */ ++ # error invalid FLOAT8 type for USE_GNUC_ASM ++ # endif ++ #endif ++ ++ /********************************************************************* ++ * nonlinear quantization of xr ++ * More accurate formula than the ISO formula. Takes into account ++ * the fact that we are quantizing xr -> ix, but we want ix^4/3 to be ++ * as close as possible to x^4/3. (taking the nearest int would mean ++ * ix is as close as possible to xr, which is different.) ++ * From Segher Boessenkool <segher@eastsite.nl> 11/1999 ++ * ASM optimization from ++ * Mathew Hendry <scampi@dial.pipex.com> 11/1999 ++ * Acy Stapp <AStapp@austin.rr.com> 11/1999 ++ * Takehiro Tominaga <tominaga@isoternet.org> 11/1999 ++ *********************************************************************/ ++ ++ void quantize_xrpow(FLOAT8 xr[576], int ix[576], gr_info *cod_info) { ++ /* quantize on xr^(3/4) instead of xr */ ++ const FLOAT8 istep = IPOW20(cod_info->global_gain); ++ ++ #if defined (USE_GNUC_ASM) ++ { ++ int rx[4]; ++ __asm__ __volatile__( ++ "\n\nloop1:\n\t" ++ ++ "fld" F8type " 0*" F8size "(%1)\n\t" ++ "fld" F8type " 1*" F8size "(%1)\n\t" ++ "fld" F8type " 2*" F8size "(%1)\n\t" ++ "fld" F8type " 3*" F8size "(%1)\n\t" ++ ++ "fxch %%st(3)\n\t" ++ "fmul %%st(4)\n\t" ++ "fxch %%st(2)\n\t" ++ "fmul %%st(4)\n\t" ++ "fxch %%st(1)\n\t" ++ "fmul %%st(4)\n\t" ++ "fxch %%st(3)\n\t" ++ "fmul %%st(4)\n\t" ++ ++ "addl $4*" F8size ", %1\n\t" ++ "addl $16, %3\n\t" ++ ++ "fxch %%st(2)\n\t" ++ "fistl %5\n\t" ++ "fxch %%st(1)\n\t" ++ "fistl 4+%5\n\t" ++ "fxch %%st(3)\n\t" ++ "fistl 8+%5\n\t" ++ "fxch %%st(2)\n\t" ++ "fistl 12+%5\n\t" ++ ++ "dec %4\n\t" ++ ++ "movl %5, %%eax\n\t" ++ "movl 4+%5, %%ebx\n\t" ++ "fxch %%st(1)\n\t" ++ "fadd" F8type " (%2,%%eax," F8size ")\n\t" ++ "fxch %%st(3)\n\t" ++ "fadd" F8type " (%2,%%ebx," F8size ")\n\t" ++ ++ "movl 8+%5, %%eax\n\t" ++ "movl 12+%5, %%ebx\n\t" ++ "fxch %%st(2)\n\t" ++ "fadd" F8type " (%2,%%eax," F8size ")\n\t" ++ "fxch %%st(1)\n\t" ++ "fadd" F8type " (%2,%%ebx," F8size ")\n\t" ++ ++ "fxch %%st(3)\n\t" ++ "fistpl -16(%3)\n\t" ++ "fxch %%st(1)\n\t" ++ "fistpl -12(%3)\n\t" ++ "fistpl -8(%3)\n\t" ++ "fistpl -4(%3)\n\t" ++ ++ "jnz loop1\n\n" ++ : /* no outputs */ ++ : "t" (istep), "r" (xr), "r" (adj43asm), "r" (ix), "r" (576 / 4), "m" (rx) ++ : "%eax", "%ebx", "memory", "cc" ++ ); ++ } ++ #elif defined (USE_MSC_ASM) ++ { ++ /* asm from Acy Stapp <AStapp@austin.rr.com> */ ++ int rx[4]; ++ _asm { ++ fld F8type ptr [istep] ++ mov esi, dword ptr [xr] ++ lea edi, dword ptr [adj43asm] ++ mov edx, dword ptr [ix] ++ mov ecx, 576/4 ++ } loop1: _asm { ++ fld F8type ptr [esi+(0*F8size)] // 0 ++ fld F8type ptr [esi+(1*F8size)] // 1 0 ++ fld F8type ptr [esi+(2*F8size)] // 2 1 0 ++ fld F8type ptr [esi+(3*F8size)] // 3 2 1 0 ++ fxch st(3) // 0 2 1 3 ++ fmul st(0), st(4) ++ fxch st(2) // 1 2 0 3 ++ fmul st(0), st(4) ++ fxch st(1) // 2 1 0 3 ++ fmul st(0), st(4) ++ fxch st(3) // 3 1 0 2 ++ fmul st(0), st(4) ++ ++ add esi, 4*F8size ++ add edx, 16 ++ ++ fxch st(2) // 0 1 3 2 ++ fist dword ptr [rx] ++ fxch st(1) // 1 0 3 2 ++ fist dword ptr [rx+4] ++ fxch st(3) // 2 0 3 1 ++ fist dword ptr [rx+8] ++ fxch st(2) // 3 0 2 1 ++ fist dword ptr [rx+12] ++ ++ dec ecx ++ ++ mov eax, dword ptr [rx] ++ mov ebx, dword ptr [rx+4] ++ fxch st(1) // 0 3 2 1 ++ fadd F8type ptr [edi+eax*F8size] ++ fxch st(3) // 1 3 2 0 ++ fadd F8type ptr [edi+ebx*F8size] ++ ++ mov eax, dword ptr [rx+8] ++ mov ebx, dword ptr [rx+12] ++ fxch st(2) // 2 3 1 0 ++ fadd F8type ptr [edi+eax*F8size] ++ fxch st(1) // 3 2 1 0 ++ fadd F8type ptr [edi+ebx*F8size] ++ fxch st(3) // 0 2 1 3 ++ fistp dword ptr [edx-16] // 2 1 3 ++ fxch st(1) // 1 2 3 ++ fistp dword ptr [edx-12] // 2 3 ++ fistp dword ptr [edx-8] // 3 ++ fistp dword ptr [edx-4] ++ ++ jnz loop1 ++ ++ mov dword ptr [xr], esi ++ mov dword ptr [ix], edx ++ fstp st(0) ++ } ++ } ++ #else ++ #if 0 ++ { /* generic code if you write ASM for XRPOW_FTOI() */ ++ FLOAT8 x; ++ int j, rx; ++ for (j = 576 / 4; j > 0; --j) { ++ x = *xr++ * istep; ++ XRPOW_FTOI(x, rx); ++ XRPOW_FTOI(x + QUANTFAC(rx), *ix++); ++ ++ x = *xr++ * istep; ++ XRPOW_FTOI(x, rx); ++ XRPOW_FTOI(x + QUANTFAC(rx), *ix++); ++ ++ x = *xr++ * istep; ++ XRPOW_FTOI(x, rx); ++ XRPOW_FTOI(x + QUANTFAC(rx), *ix++); ++ ++ x = *xr++ * istep; ++ XRPOW_FTOI(x, rx); ++ XRPOW_FTOI(x + QUANTFAC(rx), *ix++); ++ } ++ } ++ #endif ++ {/* from Wilfried.Behne@t-online.de. Reported to be 2x faster than ++ the above code (when not using ASM) on PowerPC */ ++ int j; ++ ++ for ( j = 576/8; j > 0; --j) ++ { ++ FLOAT8 x1, x2, x3, x4, x5, x6, x7, x8; ++ int rx1, rx2, rx3, rx4, rx5, rx6, rx7, rx8; ++ x1 = *xr++ * istep; ++ x2 = *xr++ * istep; ++ XRPOW_FTOI(x1, rx1); ++ x3 = *xr++ * istep; ++ XRPOW_FTOI(x2, rx2); ++ x4 = *xr++ * istep; ++ XRPOW_FTOI(x3, rx3); ++ x5 = *xr++ * istep; ++ XRPOW_FTOI(x4, rx4); ++ x6 = *xr++ * istep; ++ XRPOW_FTOI(x5, rx5); ++ x7 = *xr++ * istep; ++ XRPOW_FTOI(x6, rx6); ++ x8 = *xr++ * istep; ++ XRPOW_FTOI(x7, rx7); ++ x1 += QUANTFAC(rx1); ++ XRPOW_FTOI(x8, rx8); ++ x2 += QUANTFAC(rx2); ++ XRPOW_FTOI(x1,*ix++); ++ x3 += QUANTFAC(rx3); ++ XRPOW_FTOI(x2,*ix++); ++ x4 += QUANTFAC(rx4); ++ XRPOW_FTOI(x3,*ix++); ++ x5 += QUANTFAC(rx5); ++ XRPOW_FTOI(x4,*ix++); ++ x6 += QUANTFAC(rx6); ++ XRPOW_FTOI(x5,*ix++); ++ x7 += QUANTFAC(rx7); ++ XRPOW_FTOI(x6,*ix++); ++ x8 += QUANTFAC(rx8); ++ XRPOW_FTOI(x7,*ix++); ++ XRPOW_FTOI(x8,*ix++); ++ } ++ } ++ #endif ++ } ++ ++ ++ ++ ++ ++ ++ void quantize_xrpow_ISO( FLOAT8 xr[576], int ix[576], gr_info *cod_info ) ++ { ++ /* quantize on xr^(3/4) instead of xr */ ++ const FLOAT8 istep = IPOW20(cod_info->global_gain); ++ ++ #if defined(USE_GNUC_ASM) ++ { ++ __asm__ __volatile__ ( ++ "\n\nloop0:\n\t" ++ ++ "fld" F8type " 0*" F8size "(%3)\n\t" ++ "fld" F8type " 1*" F8size "(%3)\n\t" ++ "fld" F8type " 2*" F8size "(%3)\n\t" ++ "fld" F8type " 3*" F8size "(%3)\n\t" ++ ++ "addl $4*" F8size ", %3\n\t" ++ "addl $16, %4\n\t" ++ ++ "fxch %%st(3)\n\t" ++ "fmul %%st(4)\n\t" ++ "fxch %%st(2)\n\t" ++ "fmul %%st(4)\n\t" ++ "fxch %%st(1)\n\t" ++ "fmul %%st(4)\n\t" ++ "fxch %%st(3)\n\t" ++ "fmul %%st(4)\n\t" ++ ++ "dec %0\n\t" ++ ++ "fxch %%st(2)\n\t" ++ "fadd %%st(5)\n\t" ++ "fxch %%st(1)\n\t" ++ "fadd %%st(5)\n\t" ++ "fxch %%st(3)\n\t" ++ "fadd %%st(5)\n\t" ++ "fxch %%st(2)\n\t" ++ "fadd %%st(5)\n\t" ++ ++ "fxch %%st(1)\n\t" ++ "fistpl -16(%4)\n\t" ++ "fxch %%st(2)\n\t" ++ "fistpl -12(%4)\n\t" ++ "fistpl -8(%4)\n\t" ++ "fistpl -4(%4)\n\t" ++ ++ "jnz loop0\n\n" ++ ++ : /* no outputs */ ++ : "r" (576 / 4), "u" ((FLOAT8)(0.4054 - 0.5)), "t" (istep), "r" (xr), "r" (ix) ++ : "memory", "cc" ++ ); ++ } ++ #elif defined(USE_MSC_ASM) ++ { ++ /* asm from Acy Stapp <AStapp@austin.rr.com> */ ++ const FLOAT8 temp0 = 0.4054 - 0.5; ++ _asm { ++ mov ecx, 576/4; ++ fld F8type ptr [temp0]; ++ fld F8type ptr [istep]; ++ mov eax, dword ptr [xr]; ++ mov edx, dword ptr [ix]; ++ } loop0: _asm { ++ fld F8type ptr [eax+0*F8size]; // 0 ++ fld F8type ptr [eax+1*F8size]; // 1 0 ++ fld F8type ptr [eax+2*F8size]; // 2 1 0 ++ fld F8type ptr [eax+3*F8size]; // 3 2 1 0 ++ ++ add eax, 4*F8size; ++ add edx, 16; ++ ++ fxch st(3); // 0 2 1 3 ++ fmul st(0), st(4); ++ fxch st(2); // 1 2 0 3 ++ fmul st(0), st(4); ++ fxch st(1); // 2 1 0 3 ++ fmul st(0), st(4); ++ fxch st(3); // 3 1 0 2 ++ fmul st(0), st(4); ++ ++ dec ecx; ++ ++ fxch st(2); // 0 1 3 2 ++ fadd st(0), st(5); ++ fxch st(1); // 1 0 3 2 ++ fadd st(0), st(5); ++ fxch st(3); // 2 0 3 1 ++ fadd st(0), st(5); ++ fxch st(2); // 3 0 2 1 ++ fadd st(0), st(5); ++ ++ fxch st(1); // 0 3 2 1 ++ fistp dword ptr [edx-16]; // 3 2 1 ++ fxch st(2); // 1 2 3 ++ fistp dword ptr [edx-12]; ++ fistp dword ptr [edx-8]; ++ fistp dword ptr [edx-4]; ++ ++ jnz loop0; ++ ++ mov dword ptr [xr], eax; ++ mov dword ptr [ix], edx; ++ fstp st(0); ++ fstp st(0); ++ } ++ } ++ #else ++ #if 0 ++ /* generic ASM */ ++ register int j; ++ for (j=576/4;j>0;j--) { ++ XRPOW_FTOI(istep * (*xr++) + ROUNDFAC, *ix++); ++ XRPOW_FTOI(istep * (*xr++) + ROUNDFAC, *ix++); ++ XRPOW_FTOI(istep * (*xr++) + ROUNDFAC, *ix++); ++ XRPOW_FTOI(istep * (*xr++) + ROUNDFAC, *ix++); ++ } ++ #endif ++ { ++ register int j; ++ const FLOAT8 compareval0 = (1.0 - 0.4054)/istep; ++ /* depending on architecture, it may be worth calculating a few more compareval's. ++ eg. compareval1 = (2.0 - 0.4054/istep); ++ .. and then after the first compare do this ... ++ if compareval1>*xr then ix = 1; ++ On a pentium166, it's only worth doing the one compare (as done here), as the second ++ compare becomes more expensive than just calculating the value. Architectures with ++ slow FP operations may want to add some more comparevals. try it and send your diffs ++ statistically speaking ++ 73% of all xr*istep values give ix=0 ++ 16% will give 1 ++ 4% will give 2 ++ */ ++ for (j=576;j>0;j--) ++ { ++ if (compareval0 > *xr) { ++ *(ix++) = 0; ++ xr++; ++ } else ++ /* *(ix++) = (int)( istep*(*(xr++)) + 0.4054); */ ++ XRPOW_FTOI( istep*(*(xr++)) + ROUNDFAC , *(ix++) ); ++ } ++ } ++ #endif ++ } +diff -r -c -N encoder/quantize-pvt.h lame3.70/quantize-pvt.h +*** encoder/quantize-pvt.h Wed Dec 31 17:00:00 1969 +--- lame3.70/quantize-pvt.h Thu Apr 6 12:50:51 2000 +*************** +*** 0 **** +--- 1,124 ---- ++ #ifndef LOOP_PVT_H ++ #define LOOP_PVT_H ++ ++ #define IXMAX_VAL 8206 /* ix always <= 8191+15. see count_bits() */ ++ #define PRECALC_SIZE (IXMAX_VAL+2) ++ ++ extern FLOAT masking_lower; ++ extern int convert_mdct, convert_psy, reduce_sidechannel; ++ extern unsigned nr_of_sfb_block[6][3][4]; ++ extern int pretab[21]; ++ ++ struct scalefac_struct ++ { ++ int l[1+SBMAX_l]; ++ int s[1+SBMAX_s]; ++ }; ++ ++ extern struct scalefac_struct scalefac_band; ++ extern struct scalefac_struct sfBandIndex[6]; ++ ++ extern FLOAT8 pow43[PRECALC_SIZE]; ++ ++ #define Q_MAX 256 ++ ++ extern FLOAT8 pow20[Q_MAX]; ++ extern FLOAT8 ipow20[Q_MAX]; ++ ++ #ifdef RH_ATH ++ extern FLOAT8 ATH_mdct_long[576], ATH_mdct_short[192]; ++ #endif ++ ++ FLOAT8 ATHformula(lame_global_flags *gfp,FLOAT8 f); ++ void compute_ath(lame_global_flags *gfp,FLOAT8 ATH_l[SBPSY_l],FLOAT8 ATH_s[SBPSY_l]); ++ void ms_convert(FLOAT8 xr[2][576],FLOAT8 xr_org[2][576]); ++ void on_pe(lame_global_flags *gfp,FLOAT8 pe[2][2],III_side_info_t *l3_side, ++ int targ_bits[2],int mean_bits, int gr); ++ void reduce_side(int targ_bits[2],FLOAT8 ms_ener_ratio,int mean_bits); ++ ++ ++ void outer_loop( lame_global_flags *gfp, ++ FLOAT8 xr[576], /*vector of the magnitudees of the spectral values */ ++ int bits, ++ FLOAT8 noise[4], ++ III_psy_xmin *l3_xmin, /* the allowed distortion of the scalefactor */ ++ int l3_enc[576], /* vector of quantized values ix(0..575) */ ++ III_scalefac_t *scalefac, /* scalefactors */ ++ gr_info *, ++ FLOAT8 xfsf[4][SBPSY_l], ++ int ch); ++ ++ ++ ++ void iteration_init( lame_global_flags *gfp,III_side_info_t *l3_side, int l3_enc[2][2][576]); ++ ++ int inner_loop( lame_global_flags *gfp,FLOAT8 xrpow[576], ++ int l3_enc[576], ++ int max_bits, ++ gr_info *cod_info); ++ ++ int calc_xmin( lame_global_flags *gfp,FLOAT8 xr[576], ++ III_psy_ratio *ratio, ++ gr_info *cod_info, ++ III_psy_xmin *l3_xmin); ++ ++ ++ int scale_bitcount( III_scalefac_t *scalefac, gr_info *cod_info); ++ int scale_bitcount_lsf( III_scalefac_t *scalefac, gr_info *cod_info); ++ int calc_noise1( FLOAT8 xr[576], ++ int ix[576], ++ gr_info *cod_info, ++ FLOAT8 xfsf[4][SBPSY_l], ++ FLOAT8 distort[4][SBPSY_l], ++ III_psy_xmin *l3_xmin, ++ III_scalefac_t *, ++ FLOAT8 *noise, FLOAT8 *tot_noise, FLOAT8 *max_noise); ++ ++ int loop_break( III_scalefac_t *scalefac, gr_info *cod_info); ++ ++ void amp_scalefac_bands(FLOAT8 xrpow[576], ++ gr_info *cod_info, ++ III_scalefac_t *scalefac, ++ FLOAT8 distort[4][SBPSY_l]); ++ ++ void quantize_xrpow( FLOAT8 xr[576], ++ int ix[576], ++ gr_info *cod_info ); ++ void quantize_xrpow_ISO( FLOAT8 xr[576], ++ int ix[576], ++ gr_info *cod_info ); ++ ++ int ++ new_choose_table( int ix[576], ++ unsigned int begin, ++ unsigned int end, int * s ); ++ ++ int bin_search_StepSize2(lame_global_flags *gfp,int desired_rate, int start, int ix[576], ++ FLOAT8 xrspow[576], gr_info * cod_info); ++ int count_bits(lame_global_flags *gfp,int *ix, FLOAT8 xr[576], gr_info *cod_info); ++ ++ ++ int quant_compare(int type, ++ int best_over,FLOAT8 best_tot_noise,FLOAT8 best_over_noise,FLOAT8 best_max_over, ++ int over,FLOAT8 tot_noise, FLOAT8 over_noise,FLOAT8 max_noise); ++ ++ int VBR_compare( ++ int best_over,FLOAT8 best_tot_noise,FLOAT8 best_over_noise,FLOAT8 best_max_over, ++ int over,FLOAT8 tot_noise, FLOAT8 over_noise,FLOAT8 max_noise); ++ ++ void best_huffman_divide(int gr, int ch, gr_info *cod_info, int *ix); ++ ++ void best_scalefac_store(lame_global_flags *gfp,int gr, int ch, ++ int l3_enc[2][2][576], ++ III_side_info_t *l3_side, ++ III_scalefac_t scalefac[2][2]); ++ ++ int init_outer_loop( ++ lame_global_flags *gfp, ++ FLOAT8 xr[576], /* could be L/R OR MID/SIDE */ ++ gr_info *cod_info); ++ ++ #define LARGE_BITS 100000 ++ ++ #endif ++ +diff -r -c -N encoder/quantize.c lame3.70/quantize.c +*** encoder/quantize.c Wed Dec 31 17:00:00 1969 +--- lame3.70/quantize.c Thu Apr 6 12:50:51 2000 +*************** +*** 0 **** +--- 1,1201 ---- ++ #define MAXNOISEXX ++ /* ++ * MP3 quantization ++ * ++ * Copyright (c) 1999 Mark Taylor ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License as published by ++ * the Free Software Foundation; either version 2, or (at your option) ++ * any later version. ++ * ++ * This program is distributed in the hope that it will be useful, ++ * but WITHOUT ANY WARRANTY; without even the implied warranty of ++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the ++ * GNU General Public License for more details. ++ * ++ * You should have received a copy of the GNU General Public License ++ * along with this program; see the file COPYING. If not, write to ++ * the Free Software Foundation, 675 Mass Ave, Cambridge, MA 02139, USA. ++ */ ++ ++ ++ ++ #include <assert.h> ++ #include "util.h" ++ #include "l3side.h" ++ #include "quantize.h" ++ #include "l3bitstream.h" ++ #include "reservoir.h" ++ #include "quantize-pvt.h" ++ #ifdef HAVEGTK ++ #include "gtkanal.h" ++ #endif ++ ++ ++ ++ #ifdef HAVEGTK ++ ++ /************************************************************************/ ++ /* updates plotting data */ ++ /************************************************************************/ ++ void ++ set_pinfo ( ++ gr_info *cod_info, ++ III_psy_ratio *ratio, ++ III_scalefac_t *scalefac, ++ FLOAT8 xr[576], ++ FLOAT8 xfsf[4][SBPSY_l], ++ FLOAT8 noise[4], ++ int gr, ++ int ch ++ ) ++ { ++ int sfb; ++ FLOAT ifqstep; ++ int i,l,start,end,bw; ++ FLOAT8 en0; ++ D192_3 *xr_s = (D192_3 *)xr; ++ ifqstep = ( cod_info->scalefac_scale == 0 ) ? .5 : 1.0; ++ ++ if (cod_info->block_type == SHORT_TYPE) { ++ for ( i = 0; i < 3; i++ ) { ++ for ( sfb = 0; sfb < SBPSY_s; sfb++ ) { ++ start = scalefac_band.s[ sfb ]; ++ end = scalefac_band.s[ sfb + 1 ]; ++ bw = end - start; ++ for ( en0 = 0.0, l = start; l < end; l++ ) ++ en0 += (*xr_s)[l][i] * (*xr_s)[l][i]; ++ en0=Max(en0/bw,1e-20); ++ ++ /* conversion to FFT units */ ++ en0 = ratio->en.s[sfb][i]/en0; ++ ++ pinfo->xfsf_s[gr][ch][3*sfb+i] = xfsf[i+1][sfb]*en0; ++ pinfo->thr_s[gr][ch][3*sfb+i] = ratio->thm.s[sfb][i]; ++ pinfo->en_s[gr][ch][3*sfb+i] = ratio->en.s[sfb][i]; ++ ++ pinfo->LAMEsfb_s[gr][ch][3*sfb+i]= ++ -2*cod_info->subblock_gain[i]-ifqstep*scalefac->s[sfb][i]; ++ } ++ } ++ }else{ ++ for ( sfb = 0; sfb < SBPSY_l; sfb++ ) { ++ start = scalefac_band.l[ sfb ]; ++ end = scalefac_band.l[ sfb+1 ]; ++ bw = end - start; ++ for ( en0 = 0.0, l = start; l < end; l++ ) ++ en0 += xr[l] * xr[l]; ++ en0=Max(en0/bw,1e-20); ++ /* ++ printf("diff = %f \n",10*log10(Max(ratio[gr][ch].en.l[sfb],1e-20)) ++ -(10*log10(en0)+150)); ++ */ ++ ++ /* convert to FFT units */ ++ en0 = ratio->en.l[sfb]/en0; ++ ++ pinfo->xfsf[gr][ch][sfb] = xfsf[0][sfb]*en0; ++ pinfo->thr[gr][ch][sfb] = ratio->thm.l[sfb]; ++ pinfo->en[gr][ch][sfb] = ratio->en.l[sfb]; ++ ++ pinfo->LAMEsfb[gr][ch][sfb]=-ifqstep*scalefac->l[sfb]; ++ if (cod_info->preflag && sfb>=11) ++ pinfo->LAMEsfb[gr][ch][sfb]-=ifqstep*pretab[sfb]; ++ } ++ } ++ pinfo->LAMEqss[gr][ch] = cod_info->global_gain; ++ pinfo->LAMEmainbits[gr][ch] = cod_info->part2_3_length; ++ ++ pinfo->over [gr][ch] = noise[0]; ++ pinfo->max_noise [gr][ch] = noise[1]; ++ pinfo->over_noise[gr][ch] = noise[2]; ++ pinfo->tot_noise [gr][ch] = noise[3]; ++ } ++ ++ #endif ++ ++ ++ ++ /************************************************************************/ ++ /* iteration_loop() */ ++ /************************************************************************/ ++ void ++ iteration_loop( lame_global_flags *gfp, ++ FLOAT8 pe[2][2], FLOAT8 ms_ener_ratio[2], ++ FLOAT8 xr[2][2][576], III_psy_ratio ratio[2][2], ++ III_side_info_t *l3_side, int l3_enc[2][2][576], ++ III_scalefac_t scalefac[2][2]) ++ { ++ FLOAT8 xfsf[4][SBPSY_l]; ++ FLOAT8 noise[4]; /* over,max_noise,over_noise,tot_noise; */ ++ III_psy_xmin l3_xmin[2]; ++ gr_info *cod_info; ++ int bitsPerFrame; ++ int mean_bits; ++ int ch, gr, i, bit_rate; ++ ++ ++ iteration_init(gfp,l3_side,l3_enc); ++ bit_rate = bitrate_table[gfp->version][gfp->bitrate_index]; ++ ++ ++ getframebits(gfp,&bitsPerFrame, &mean_bits); ++ ResvFrameBegin(gfp, l3_side, mean_bits, bitsPerFrame ); ++ ++ /* quantize! */ ++ ++ ++ ++ for ( gr = 0; gr < gfp->mode_gr; gr++ ) { ++ int targ_bits[2]; ++ ++ if (convert_mdct) ++ ms_convert(xr[gr], xr[gr]); ++ ++ on_pe(gfp,pe,l3_side,targ_bits,mean_bits, gr); ++ #ifdef RH_SIDE_CBR ++ #else ++ if (reduce_sidechannel) ++ reduce_side(targ_bits,ms_ener_ratio[gr],mean_bits); ++ #endif ++ ++ for (ch=0 ; ch < gfp->stereo ; ch ++) { ++ cod_info = &l3_side->gr[gr].ch[ch].tt; ++ if (!init_outer_loop(gfp,xr[gr][ch], cod_info)) ++ { ++ /* xr contains no energy ++ * cod_info was set in init_outer_loop above ++ */ ++ memset(&scalefac[gr][ch],0,sizeof(III_scalefac_t)); ++ memset(l3_enc[gr][ch],0,576*sizeof(int)); ++ noise[0]=noise[1]=noise[2]=noise[3]=0; ++ } ++ else ++ { ++ calc_xmin(gfp,xr[gr][ch], &ratio[gr][ch], cod_info, &l3_xmin[ch]); ++ outer_loop( gfp,xr[gr][ch], targ_bits[ch], noise, ++ &l3_xmin[ch], l3_enc[gr][ch], ++ &scalefac[gr][ch], cod_info, xfsf, ch); ++ } ++ best_scalefac_store(gfp,gr, ch, l3_enc, l3_side, scalefac); ++ if (gfp->use_best_huffman==1 && cod_info->block_type == NORM_TYPE) { ++ best_huffman_divide(gr, ch, cod_info, l3_enc[gr][ch]); ++ } ++ #ifdef HAVEGTK ++ if (gfp->gtkflag) ++ set_pinfo (cod_info, &ratio[gr][ch], &scalefac[gr][ch], xr[gr][ch], xfsf, noise, gr, ch); ++ #endif ++ ++ /*#define NORES_TEST */ ++ #ifndef NORES_TEST ++ ResvAdjust(gfp,cod_info, l3_side, mean_bits ); ++ #endif ++ /* set the sign of l3_enc */ ++ for ( i = 0; i < 576; i++) { ++ if (xr[gr][ch][i] < 0) ++ l3_enc[gr][ch][i] *= -1; ++ } ++ } ++ } /* loop over gr */ ++ ++ #ifdef NORES_TEST ++ /* replace ResvAdjust above with this code if you do not want ++ the second granule to use bits saved by the first granule. ++ when combined with --nores, this is usefull for testing only */ ++ for ( gr = 0; gr < gfp->mode_gr; gr++ ) { ++ for ( ch = 0; ch < gfp->stereo; ch++ ) { ++ cod_info = &l3_side->gr[gr].ch[ch].tt; ++ ResvAdjust(gfp, cod_info, l3_side, mean_bits ); ++ } ++ } ++ #endif ++ ++ ++ ++ ResvFrameEnd(gfp,l3_side, mean_bits ); ++ } ++ ++ ++ void ++ set_masking_lower (int VBR_q,int nbits) ++ { ++ FLOAT masking_lower_db, adjust; ++ ++ /* quality setting */ ++ /* Adjust allowed masking based on quality setting */ ++ ++ #ifdef RH_QUALITY_CONTROL ++ /* - lower masking depending on Quality setting ++ * - quality control together with adjusted ATH MDCT scaling ++ * on lower quality setting allocate more noise from ++ * ATH masking, and on higher quality setting allocate ++ * less noise from ATH masking. ++ * - experiments show that going more than 2dB over GPSYCHO's ++ * limits ends up in very annoying artefacts ++ */ ++ static FLOAT dbQ[10]={-6.0,-4.5,-3.0,-1.5,0,0.3,0.6,1.0,1.5,2.0}; ++ ++ assert( VBR_q <= 9 ); ++ assert( VBR_q >= 0 ); ++ ++ masking_lower_db = dbQ[VBR_q]; ++ adjust = 0; ++ #else ++ /* masking_lower varies from -8 to +10 db */ ++ masking_lower_db = -6 + 2*VBR_q; ++ /* adjust by -6(min)..0(max) depending on bitrate */ ++ adjust = (nbits-125)/(2500.0-125.0); ++ adjust = 4*(adjust-1); ++ #endif ++ masking_lower_db += adjust; ++ masking_lower = pow(10.0,masking_lower_db/10); ++ } ++ ++ /************************************************************************ ++ * ++ * VBR_iteration_loop() ++ * ++ * tries to find out how many bits are needed for each granule and channel ++ * to get an acceptable quantization. An appropriate bitrate will then be ++ * choosed for quantization. rh 8/99 ++ * ++ ************************************************************************/ ++ void ++ VBR_iteration_loop (lame_global_flags *gfp, ++ FLOAT8 pe[2][2], FLOAT8 ms_ener_ratio[2], ++ FLOAT8 xr[2][2][576], III_psy_ratio ratio[2][2], ++ III_side_info_t * l3_side, int l3_enc[2][2][576], ++ III_scalefac_t scalefac[2][2]) ++ { ++ #ifdef HAVEGTK ++ plotting_data bst_pinfo; ++ #endif ++ gr_info bst_cod_info, clean_cod_info; ++ III_scalefac_t bst_scalefac; ++ int bst_l3_enc[576]; ++ ++ III_psy_xmin l3_xmin; ++ gr_info *cod_info = NULL; ++ int save_bits[2][2]; ++ FLOAT8 noise[4]; /* over,max_noise,over_noise,tot_noise; */ ++ FLOAT8 targ_noise[4]; /* over,max_noise,over_noise,tot_noise; */ ++ FLOAT8 xfsf[4][SBPSY_l]; ++ int this_bits, dbits; ++ int used_bits=0; ++ int min_bits,max_bits,min_mean_bits=0; ++ int frameBits[15]; ++ int bitsPerFrame; ++ int bits; ++ int mean_bits; ++ int i,ch, gr, analog_silence; ++ int reparted = 0; ++ ++ iteration_init(gfp,l3_side,l3_enc); ++ ++ #ifdef RH_QUALITY_CONTROL ++ /* with RH_QUALITY_CONTROL we have to set masking_lower only once */ ++ set_masking_lower(gfp->VBR_q, 0 ); ++ #endif ++ ++ /******************************************************************* ++ * how many bits are available for each bitrate? ++ *******************************************************************/ ++ for( gfp->bitrate_index = 1; ++ gfp->bitrate_index <= gfp->VBR_max_bitrate; ++ gfp->bitrate_index++ ) { ++ getframebits (gfp,&bitsPerFrame, &mean_bits); ++ if (gfp->bitrate_index == gfp->VBR_min_bitrate) { ++ /* always use at least this many bits per granule per channel */ ++ /* unless we detect analog silence, see below */ ++ min_mean_bits=mean_bits/gfp->stereo; ++ } ++ frameBits[gfp->bitrate_index]= ++ ResvFrameBegin (gfp,l3_side, mean_bits, bitsPerFrame); ++ } ++ ++ gfp->bitrate_index=gfp->VBR_max_bitrate; ++ ++ ++ /******************************************************************* ++ * how many bits would we use of it? ++ *******************************************************************/ ++ analog_silence=0; ++ for (gr = 0; gr < gfp->mode_gr; gr++) { ++ int num_chan=gfp->stereo; ++ #ifdef RH_SIDE_VBR ++ /* my experiences are, that side channel reduction ++ * does more harm than good when VBR encoding ++ * (Robert.Hegemann@gmx.de 2000-02-18) ++ */ ++ #else ++ /* determine quality based on mid channel only */ ++ if (reduce_sidechannel) num_chan=1; ++ #endif ++ ++ /* copy data to be quantized into xr */ ++ if (convert_mdct) ++ ms_convert(xr[gr],xr[gr]); ++ ++ for (ch = 0; ch < num_chan; ch++) { ++ int real_bits; ++ ++ /****************************************************************** ++ * find smallest number of bits for an allowable quantization ++ ******************************************************************/ ++ cod_info = &l3_side->gr[gr].ch[ch].tt; ++ min_bits = Max(125,min_mean_bits); ++ ++ if (!init_outer_loop(gfp,xr[gr][ch], cod_info)) ++ { ++ /* xr contains no energy ++ * cod_info was set in init_outer_loop above ++ */ ++ memset(&scalefac[gr][ch],0,sizeof(III_scalefac_t)); ++ memset(l3_enc[gr][ch],0,576*sizeof(int)); ++ save_bits[gr][ch] = 0; ++ #ifdef HAVEGTK ++ if (gfp->gtkflag) ++ set_pinfo(cod_info, &ratio[gr][ch], &scalefac[gr][ch], xr[gr][ch], xfsf, noise, gr, ch); ++ #endif ++ analog_silence=1; ++ continue; /* with next channel */ ++ } ++ ++ memcpy( &clean_cod_info, cod_info, sizeof(gr_info) ); ++ ++ #ifdef RH_QUALITY_CONTROL ++ /* ++ * masking lower already set in the beginning ++ */ ++ #else ++ /* ++ * has to be set before calculating l3_xmin ++ */ ++ set_masking_lower( gfp->VBR_q,2500 ); ++ #endif ++ /* check for analolg silence */ ++ /* if energy < ATH, set min_bits = 125 */ ++ if (0==calc_xmin(gfp,xr[gr][ch], &ratio[gr][ch], cod_info, &l3_xmin)) { ++ analog_silence=1; ++ min_bits=125; ++ } ++ ++ if (cod_info->block_type==SHORT_TYPE) { ++ min_bits += Max(1100,pe[gr][ch]); ++ min_bits=Min(min_bits,1800); ++ } ++ ++ max_bits = 1200 + frameBits[gfp->VBR_max_bitrate]/(gfp->stereo*gfp->mode_gr); ++ max_bits=Min(max_bits,2500); ++ max_bits=Max(max_bits,min_bits); ++ ++ dbits = (max_bits-min_bits)/4; ++ this_bits = (max_bits+min_bits)/2; ++ real_bits = max_bits+1; ++ ++ /* bin search to within +/- 10 bits of optimal */ ++ do { ++ int better; ++ assert(this_bits>=min_bits); ++ assert(this_bits<=max_bits); ++ ++ if( this_bits >= real_bits ){ ++ /* ++ * we already found a quantization with fewer bits ++ * so we can skip this try ++ */ ++ this_bits -= dbits; ++ dbits /= 2; ++ continue; /* skips the rest of this do-while loop */ ++ } ++ ++ /* VBR will look for a quantization which has better values ++ * then those specified below.*/ ++ targ_noise[0]=0; /* over */ ++ targ_noise[1]=0; /* max_noise */ ++ targ_noise[2]=0; /* over_noise */ ++ targ_noise[3]=0; /* tot_noise */ ++ ++ targ_noise[0]=Max(0,targ_noise[0]); ++ targ_noise[2]=Max(0,targ_noise[2]); ++ ++ /* ++ * OK, start with a fresh setting ++ * - scalefac will be set up by outer_loop ++ * - l3_enc will be set up by outer_loop ++ * + cod_info we will restore our initialized one, see below ++ */ ++ memcpy( cod_info, &clean_cod_info, sizeof(gr_info) ); ++ ++ #ifdef RH_QUALITY_CONTROL ++ /* ++ * there is no need for a recalculation of l3_xmin, ++ * because masking_lower did not change within this do-while ++ */ ++ #else ++ /* quality setting */ ++ set_masking_lower( gfp->VBR_q,this_bits ); ++ /* ++ * compute max allowed distortion, masking lower has changed ++ */ ++ calc_xmin(gfp,xr[gr][ch], &ratio[gr][ch], cod_info, &l3_xmin); ++ #endif ++ outer_loop( gfp,xr[gr][ch], this_bits, noise, ++ &l3_xmin, l3_enc[gr][ch], ++ &scalefac[gr][ch], cod_info, xfsf, ++ ch); ++ ++ /* is quantization as good as we are looking for ? */ ++ better=VBR_compare((int)targ_noise[0],targ_noise[3],targ_noise[2], ++ targ_noise[1],(int)noise[0],noise[3],noise[2], ++ noise[1]); ++ #ifdef HAVEGTK ++ if (gfp->gtkflag) ++ set_pinfo(cod_info, &ratio[gr][ch], &scalefac[gr][ch], xr[gr][ch], xfsf, noise, gr, ch); ++ #endif ++ if (better) { ++ /* ++ * we now know it can be done with "real_bits" ++ * and maybe we can skip some iterations ++ */ ++ real_bits = cod_info->part2_3_length; ++ /* ++ * save best quantization so far ++ */ ++ memcpy( &bst_scalefac, &scalefac[gr][ch], sizeof(III_scalefac_t) ); ++ memcpy( bst_l3_enc, l3_enc [gr][ch], sizeof(int)*576 ); ++ memcpy( &bst_cod_info, cod_info, sizeof(gr_info) ); ++ #ifdef HAVEGTK ++ if (gfp->gtkflag) ++ memcpy( &bst_pinfo, pinfo, sizeof(plotting_data) ); ++ #endif ++ /* ++ * try with fewer bits ++ */ ++ this_bits -= dbits; ++ } else { ++ /* ++ * try with more bits ++ */ ++ this_bits += dbits; ++ } ++ dbits /= 2; ++ } while (dbits>10) ; ++ ++ if (real_bits <= max_bits) ++ { ++ /* restore best quantization found */ ++ memcpy( cod_info, &bst_cod_info, sizeof(gr_info) ); ++ memcpy( &scalefac[gr][ch], &bst_scalefac, sizeof(III_scalefac_t) ); ++ memcpy( l3_enc [gr][ch], bst_l3_enc, sizeof(int)*576 ); ++ #ifdef HAVEGTK ++ if (gfp->gtkflag) ++ memcpy( pinfo, &bst_pinfo, sizeof(plotting_data) ); ++ #endif ++ } ++ assert((int)cod_info->part2_3_length <= max_bits); ++ save_bits[gr][ch] = cod_info->part2_3_length; ++ used_bits += save_bits[gr][ch]; ++ ++ } /* for ch */ ++ } /* for gr */ ++ ++ ++ #ifdef RH_SIDE_VBR ++ /* my experiences are, that side channel reduction ++ * does more harm than good when VBR encoding ++ * (Robert.Hegemann@gmx.de 2000-02-18) ++ */ ++ #else ++ if (reduce_sidechannel) { ++ /* number of bits needed was found for MID channel above. Use formula ++ * (fixed bitrate code) to set the side channel bits */ ++ for (gr = 0; gr < gfp->mode_gr; gr++) { ++ FLOAT8 fac = .33*(.5-ms_ener_ratio[gr])/.5; ++ save_bits[gr][1]=((1-fac)/(1+fac))*save_bits[gr][0]; ++ save_bits[gr][1]=Max(125,save_bits[gr][1]); ++ used_bits += save_bits[gr][1]; ++ } ++ } ++ #endif ++ ++ /****************************************************************** ++ * find lowest bitrate able to hold used bits ++ ******************************************************************/ ++ for( gfp->bitrate_index = (analog_silence ? 1 : gfp->VBR_min_bitrate ); ++ gfp->bitrate_index < gfp->VBR_max_bitrate; ++ gfp->bitrate_index++ ) ++ if( used_bits <= frameBits[gfp->bitrate_index] ) break; ++ ++ /******************************************************************* ++ * calculate quantization for this bitrate ++ *******************************************************************/ ++ getframebits (gfp,&bitsPerFrame, &mean_bits); ++ bits=ResvFrameBegin (gfp,l3_side, mean_bits, bitsPerFrame); ++ ++ /* repartion available bits in same proportion */ ++ if (used_bits > bits ) { ++ reparted = 1; ++ for( gr = 0; gr < gfp->mode_gr; gr++) { ++ for(ch = 0; ch < gfp->stereo; ch++) { ++ save_bits[gr][ch]=(save_bits[gr][ch]*frameBits[gfp->bitrate_index])/used_bits; ++ } ++ } ++ used_bits=0; ++ for( gr = 0; gr < gfp->mode_gr; gr++) { ++ for(ch = 0; ch < gfp->stereo; ch++) { ++ used_bits += save_bits[gr][ch]; ++ } ++ } ++ } ++ assert(used_bits <= bits); ++ ++ for(gr = 0; gr < gfp->mode_gr; gr++) { ++ for(ch = 0; ch < gfp->stereo; ch++) { ++ #ifdef RH_SIDE_VBR ++ if (reparted) ++ #else ++ if (reparted || (reduce_sidechannel && ch == 1)) ++ #endif ++ { ++ cod_info = &l3_side->gr[gr].ch[ch].tt; ++ ++ if (!init_outer_loop(gfp,xr[gr][ch], cod_info)) ++ { ++ /* xr contains no energy ++ * cod_info was set in init_outer_loop above ++ */ ++ memset(&scalefac[gr][ch],0,sizeof(III_scalefac_t)); ++ memset(l3_enc[gr][ch],0,576*sizeof(int)); ++ noise[0]=noise[1]=noise[2]=noise[3]=0; ++ } ++ else ++ { ++ #ifdef RH_QUALITY_CONTROL ++ /* ++ * masking lower already set in the beginning ++ */ ++ #else ++ /* quality setting */ ++ set_masking_lower( gfp->VBR_q,save_bits[gr][ch] ); ++ #endif ++ calc_xmin(gfp,xr[gr][ch], &ratio[gr][ch], cod_info, &l3_xmin); ++ ++ outer_loop( gfp,xr[gr][ch], save_bits[gr][ch], noise, ++ &l3_xmin, l3_enc[gr][ch], ++ &scalefac[gr][ch], cod_info, xfsf, ch); ++ } ++ #ifdef HAVEGTK ++ if (gfp->gtkflag) ++ set_pinfo(cod_info, &ratio[gr][ch], &scalefac[gr][ch], xr[gr][ch], xfsf, noise, gr, ch); ++ #endif ++ } ++ } ++ } ++ ++ /******************************************************************* ++ * update reservoir status after FINAL quantization/bitrate ++ *******************************************************************/ ++ for (gr = 0; gr < gfp->mode_gr; gr++) ++ for (ch = 0; ch < gfp->stereo; ch++) { ++ cod_info = &l3_side->gr[gr].ch[ch].tt; ++ best_scalefac_store(gfp,gr, ch, l3_enc, l3_side, scalefac); ++ if (cod_info->block_type == NORM_TYPE) { ++ best_huffman_divide(gr, ch, cod_info, l3_enc[gr][ch]); ++ } ++ #ifdef HAVEGTK ++ if (gfp->gtkflag) ++ pinfo->LAMEmainbits[gr][ch]=cod_info->part2_3_length; ++ #endif ++ ResvAdjust (gfp,cod_info, l3_side, mean_bits); ++ } ++ ++ /******************************************************************* ++ * set the sign of l3_enc ++ *******************************************************************/ ++ for (gr = 0; gr < gfp->mode_gr; gr++) ++ for (ch = 0; ch < gfp->stereo; ch++) { ++ /* ++ * is the following code correct? ++ * ++ int *pi = &l3_enc[gr][ch][0]; ++ ++ for (i = 0; i < 576; i++) { ++ FLOAT8 pr = xr[gr][ch][i]; ++ ++ if ((pr < 0) && (pi[i] > 0)) ++ pi[i] *= -1; ++ } ++ * ++ * or is the code used for CBR correct? ++ */ ++ for ( i = 0; i < 576; i++) { ++ if (xr[gr][ch][i] < 0) l3_enc[gr][ch][i] *= -1; ++ } ++ } ++ ++ ResvFrameEnd (gfp,l3_side, mean_bits); ++ } ++ ++ ++ ++ ++ /************************************************************************/ ++ /* init_outer_loop mt 6/99 */ ++ /* returns 0 if all energies in xr are zero, else 1 */ ++ /************************************************************************/ ++ int init_outer_loop(lame_global_flags *gfp, ++ FLOAT8 xr[576], /* could be L/R OR MID/SIDE */ ++ gr_info *cod_info) ++ { ++ int i; ++ ++ ++ for ( i = 0; i < 4; i++ ) ++ cod_info->slen[i] = 0; ++ cod_info->sfb_partition_table = &nr_of_sfb_block[0][0][0]; ++ ++ cod_info->part2_3_length = 0; ++ cod_info->big_values = 0; ++ cod_info->count1 = 0; ++ cod_info->scalefac_compress = 0; ++ cod_info->table_select[0] = 0; ++ cod_info->table_select[1] = 0; ++ cod_info->table_select[2] = 0; ++ cod_info->subblock_gain[0] = 0; ++ cod_info->subblock_gain[1] = 0; ++ cod_info->subblock_gain[2] = 0; ++ cod_info->region0_count = 0; ++ cod_info->region1_count = 0; ++ cod_info->part2_length = 0; ++ cod_info->preflag = 0; ++ cod_info->scalefac_scale = 0; ++ cod_info->global_gain = 210; ++ cod_info->count1table_select= 0; ++ cod_info->count1bits = 0; ++ ++ ++ if (gfp->experimentalZ) { ++ /* compute subblock gains */ ++ int j,b; FLOAT8 en[3],mx; ++ if ((cod_info->block_type==SHORT_TYPE) ) { ++ /* estimate energy within each subblock */ ++ for (b=0; b<3; b++) en[b]=0; ++ for ( i=0,j = 0; j < 192; j++ ) { ++ for (b=0; b<3; b++) { ++ en[b]+=xr[i] * xr[i]; ++ i++; ++ } ++ } ++ mx = 1e-12; ++ for (b=0; b<3; b++) mx=Max(mx,en[b]); ++ for (b=0; b<3; b++) en[b] = Max(en[b],1e-12)/mx; ++ /*printf("ener = %4.2f %4.2f %4.2f \n",en[0],en[1],en[2]);*/ ++ /* pick gain so that 2^(2gain)*en[0] = 1 */ ++ /* gain = .5* log( 1/en[0] )/LOG2 = -.5*log(en[])/LOG2 */ ++ for (b=0; b<3; b++) { ++ cod_info->subblock_gain[b] = (int)(-.5*log(en[b])/LOG2 + 0.5); ++ if (cod_info->subblock_gain[b] > 2) ++ cod_info->subblock_gain[b]=2; ++ if (cod_info->subblock_gain[b] < 0) ++ cod_info->subblock_gain[b]=0; ++ } ++ /* ++ * check if there is some energy we have to quantize ++ * if so, then return 1 else 0 ++ */ ++ if (1e-99 < en[0]+en[1]+en[2]) ++ return 1; ++ else ++ return 0; ++ } ++ } ++ /* ++ * check if there is some energy we have to quantize ++ * if so, then return 1 else 0 ++ */ ++ for (i=0; i<576; i++) ++ if ( 1e-99 < fabs (xr[i]) ) ++ return 1; ++ ++ return 0; ++ } ++ ++ ++ ++ ++ /************************************************************************/ ++ /* outer_loop */ ++ /************************************************************************/ ++ /* Function: The outer iteration loop controls the masking conditions */ ++ /* of all scalefactorbands. It computes the best scalefac and */ ++ /* global gain. This module calls the inner iteration loop ++ * ++ * mt 5/99 completely rewritten to allow for bit reservoir control, ++ * mid/side channels with L/R or mid/side masking thresholds, ++ * and chooses best quantization instead of last quantization when ++ * no distortion free quantization can be found. ++ * ++ * added VBR support mt 5/99 ++ ************************************************************************/ ++ void outer_loop( ++ lame_global_flags *gfp, ++ FLOAT8 xr[576], ++ int targ_bits, ++ FLOAT8 best_noise[4], ++ III_psy_xmin *l3_xmin, /* the allowed distortion of the scalefactor */ ++ int l3_enc[576], /* vector of quantized values ix(0..575) */ ++ III_scalefac_t *scalefac, /* scalefactors */ ++ gr_info *cod_info, ++ FLOAT8 xfsf[4][SBPSY_l], ++ int ch) ++ { ++ III_scalefac_t scalefac_w; ++ gr_info save_cod_info; ++ int l3_enc_w[576]; ++ int i, iteration; ++ int status,bits_found=0; ++ int huff_bits; ++ FLOAT8 xrpow[576],temp; ++ int better; ++ int over=0; ++ FLOAT8 max_noise; ++ FLOAT8 over_noise; ++ FLOAT8 tot_noise; ++ int best_over=100; ++ FLOAT8 best_max_noise=0; ++ FLOAT8 best_over_noise=0; ++ FLOAT8 best_tot_noise=0; ++ FLOAT8 xfsf_w[4][SBPSY_l]; ++ FLOAT8 distort[4][SBPSY_l]; ++ ++ int compute_stepsize=1; ++ int notdone=1; ++ ++ /* BEGIN MAIN LOOP */ ++ iteration = 0; ++ while ( notdone ) { ++ static int OldValue[2] = {180, 180}; ++ int try_scale=0; ++ iteration ++; ++ ++ if (compute_stepsize) { ++ /* init and compute initial quantization step */ ++ compute_stepsize=0; ++ /* reset of iteration variables */ ++ memset(&scalefac_w, 0, sizeof(III_scalefac_t)); ++ for (i=0;i<576;i++) { ++ temp=fabs(xr[i]); ++ xrpow[i]=sqrt(sqrt(temp)*temp); ++ } ++ bits_found=bin_search_StepSize2(gfp,targ_bits,OldValue[ch], ++ l3_enc_w,xrpow,cod_info); ++ OldValue[ch] = cod_info->global_gain; ++ } ++ ++ ++ /* inner_loop starts with the initial quantization step computed above ++ * and slowly increases until the bits < huff_bits. ++ * Thus it is important not to start with too large of an inital ++ * quantization step. Too small is ok, but inner_loop will take longer ++ */ ++ huff_bits = targ_bits - cod_info->part2_length; ++ if (huff_bits < 0) { ++ assert(iteration != 1); ++ /* scale factors too large, not enough bits. use previous quantizaton */ ++ notdone=0; ++ } else { ++ /* if this is the first iteration, see if we can reuse the quantization ++ * computed in bin_search_StepSize above */ ++ int real_bits; ++ if (iteration==1) { ++ if(bits_found>huff_bits) { ++ cod_info->global_gain++; ++ real_bits = inner_loop(gfp,xrpow, l3_enc_w, huff_bits, cod_info); ++ } else real_bits=bits_found; ++ } ++ else ++ real_bits=inner_loop(gfp,xrpow, l3_enc_w, huff_bits, cod_info); ++ cod_info->part2_3_length = real_bits; ++ ++ /* compute the distortion in this quantization */ ++ if (gfp->noise_shaping==0) { ++ over=0; ++ }else{ ++ /* coefficients and thresholds both l/r (or both mid/side) */ ++ over=calc_noise1( xr, l3_enc_w, cod_info, ++ xfsf_w,distort, l3_xmin, &scalefac_w, &over_noise, ++ &tot_noise, &max_noise); ++ ++ } ++ ++ /* check if this quantization is better the our saved quantization */ ++ if (iteration == 1) better=1; ++ else ++ better=quant_compare(gfp->experimentalX, ++ best_over,best_tot_noise,best_over_noise,best_max_noise, ++ over, tot_noise, over_noise, max_noise); ++ ++ /* save data so we can restore this quantization later */ ++ if (better) { ++ best_over=over; ++ best_max_noise=max_noise; ++ best_over_noise=over_noise; ++ best_tot_noise=tot_noise; ++ ++ memcpy(scalefac, &scalefac_w, sizeof(III_scalefac_t)); ++ memcpy(l3_enc,l3_enc_w,sizeof(int)*576); ++ memcpy(&save_cod_info,cod_info,sizeof(save_cod_info)); ++ ++ #ifdef HAVEGTK ++ if (gfp->gtkflag) { ++ memcpy(xfsf, xfsf_w, sizeof(xfsf_w)); ++ } ++ #endif ++ } ++ } ++ ++ /* if no bands with distortion, we are done */ ++ if (gfp->noise_shaping_stop==0) ++ if (over==0) notdone=0; ++ ++ if (notdone) { ++ amp_scalefac_bands( xrpow, cod_info, &scalefac_w, distort); ++ /* check to make sure we have not amplified too much */ ++ /* loop_break returns 0 if there is an unamplified scalefac */ ++ /* scale_bitcount returns 0 if no scalefactors are too large */ ++ if ( (status = loop_break(&scalefac_w, cod_info)) == 0 ) { ++ if ( gfp->version == 1 ) { ++ status = scale_bitcount(&scalefac_w, cod_info); ++ }else{ ++ status = scale_bitcount_lsf(&scalefac_w, cod_info); ++ } ++ if (status && (cod_info->scalefac_scale==0)) try_scale=1; ++ } ++ notdone = !status; ++ } ++ ++ if (try_scale && gfp->experimentalY) { ++ init_outer_loop(gfp,xr, cod_info); ++ compute_stepsize=1; /* compute a new global gain */ ++ notdone=1; ++ cod_info->scalefac_scale=1; ++ } ++ } /* done with main iteration */ ++ ++ memcpy(cod_info,&save_cod_info,sizeof(save_cod_info)); ++ cod_info->part2_3_length += cod_info->part2_length; ++ ++ /* finish up */ ++ assert( cod_info->global_gain < 256 ); ++ ++ best_noise[0]=best_over; ++ best_noise[1]=best_max_noise; ++ best_noise[2]=best_over_noise; ++ best_noise[3]=best_tot_noise; ++ } ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ ++ /*************************************************************************/ ++ /* calc_noise */ ++ /*************************************************************************/ ++ /* mt 5/99: Function: Improved calc_noise for a single channel */ ++ int calc_noise1( FLOAT8 xr[576], int ix[576], gr_info *cod_info, ++ FLOAT8 xfsf[4][SBPSY_l], FLOAT8 distort[4][SBPSY_l], ++ III_psy_xmin *l3_xmin, III_scalefac_t *scalefac, ++ FLOAT8 *over_noise, ++ FLOAT8 *tot_noise, FLOAT8 *max_noise) ++ { ++ int start, end, l, i, over=0; ++ u_int sfb; ++ FLOAT8 sum,step,bw; ++ #ifdef RH_ATH ++ FLOAT8 ath_max; ++ #endif ++ ++ int count=0; ++ FLOAT8 noise; ++ *over_noise=0; ++ *tot_noise=0; ++ *max_noise = -999; ++ ++ for ( sfb = 0; sfb < cod_info->sfb_lmax; sfb++ ) { ++ FLOAT8 step; ++ int s = scalefac->l[sfb]; ++ ++ if (cod_info->preflag) ++ s += pretab[sfb]; ++ ++ s = cod_info->global_gain - (s << (cod_info->scalefac_scale + 1)); ++ assert(s<Q_MAX); ++ assert(s>=0); ++ step = POW20(s); ++ ++ start = scalefac_band.l[ sfb ]; ++ end = scalefac_band.l[ sfb+1 ]; ++ bw = end - start; ++ ++ #ifdef RH_ATH ++ ath_max = 0; ++ #endif ++ for ( sum = 0.0, l = start; l < end; l++ ) ++ { ++ FLOAT8 temp; ++ temp = fabs(xr[l]) - pow43[ix[l]] * step; ++ #ifdef MAXNOISE ++ temp = bw*temp*temp; ++ sum = Max(sum,temp); ++ #elif RH_ATH ++ temp = temp*temp; ++ sum += temp; ++ ath_max = Max( ath_max, temp/ATH_mdct_long[l] ); ++ #else ++ sum += temp * temp; ++ #endif ++ ++ } ++ xfsf[0][sfb] = sum / bw; ++ ++ /* max -30db noise below threshold */ ++ #ifdef RH_ATH ++ noise = 10*log10(Max(.001,Min(ath_max,xfsf[0][sfb]/l3_xmin->l[sfb]))); ++ #else ++ noise = 10*log10(Max(.001,xfsf[0][sfb] / l3_xmin->l[sfb])); ++ #endif ++ distort[0][sfb] = noise; ++ if (noise>0) { ++ over++; ++ *over_noise += noise; ++ } ++ *tot_noise += noise; ++ *max_noise=Max(*max_noise,noise); ++ count++; ++ ++ } ++ ++ ++ for ( i = 0; i < 3; i++ ) { ++ for ( sfb = cod_info->sfb_smax; sfb < SBPSY_s; sfb++ ) { ++ int s; ++ ++ s = (scalefac->s[sfb][i] << (cod_info->scalefac_scale + 1)) ++ + cod_info->subblock_gain[i] * 8; ++ s = cod_info->global_gain - s; ++ ++ assert(s<Q_MAX); ++ assert(s>=0); ++ step = POW20(s); ++ start = scalefac_band.s[ sfb ]; ++ end = scalefac_band.s[ sfb+1 ]; ++ bw = end - start; ++ #ifdef RH_ATH ++ ath_max = 0; ++ #endif ++ for ( sum = 0.0, l = start; l < end; l++ ) { ++ FLOAT8 temp; ++ temp = fabs(xr[l * 3 + i]) - pow43[ix[l * 3 + i]] * step; ++ #ifdef MAXNOISE ++ temp = bw*temp*temp; ++ sum = Max(sum,temp); ++ #elif RH_ATH ++ temp = temp*temp; ++ sum += temp; ++ ath_max = Max( ath_max, temp/ATH_mdct_short[l] ); ++ #else ++ sum += temp * temp; ++ #endif ++ } ++ xfsf[i+1][sfb] = sum / bw; ++ /* max -30db noise below threshold */ ++ #ifdef RH_ATH ++ noise = 10*log10(Max(.001,Min(ath_max,xfsf[i+1][sfb]/l3_xmin->s[sfb][i]))); ++ #else ++ noise = 10*log10(Max(.001,xfsf[i+1][sfb] / l3_xmin->s[sfb][i] )); ++ #endif ++ distort[i+1][sfb] = noise; ++ if (noise > 0) { ++ over++; ++ *over_noise += noise; ++ } ++ *tot_noise += noise; ++ *max_noise=Max(*max_noise,noise); ++ count++; ++ } ++ } ++ ++ if (count>1) *tot_noise /= count; ++ if (over>1) *over_noise /= over; ++ ++ return over; ++ } ++ ++ ++ ++ ++ ++ ++ ++ /*************************************************************************/ ++ /* amp_scalefac_bands */ ++ /*************************************************************************/ ++ ++ /* ++ Amplify the scalefactor bands that violate the masking threshold. ++ See ISO 11172-3 Section C.1.5.4.3.5 ++ */ ++ void amp_scalefac_bands(FLOAT8 xrpow[576], ++ gr_info *cod_info, ++ III_scalefac_t *scalefac, ++ FLOAT8 distort[4][SBPSY_l]) ++ { ++ int start, end, l,i; ++ u_int sfb; ++ FLOAT8 ifqstep34; ++ FLOAT8 distort_thresh; ++ ++ if ( cod_info->scalefac_scale == 0 ) ++ ifqstep34 = 1.29683955465100964055; ++ else ++ ifqstep34 = 1.68179283050742922612; ++ ++ /* distort_thresh = 0, unless all bands have distortion ++ * less than masking. In that case, just amplify bands with distortion ++ * within 95% of largest distortion/masking ratio */ ++ distort_thresh = -900; ++ for ( sfb = 0; sfb < cod_info->sfb_lmax; sfb++ ) { ++ distort_thresh = Max(distort[0][sfb],distort_thresh); ++ } ++ ++ for ( sfb = cod_info->sfb_smax; sfb < 12; sfb++ ) { ++ for ( i = 0; i < 3; i++ ) { ++ distort_thresh = Max(distort[i+1][sfb],distort_thresh); ++ } ++ } ++ distort_thresh=Min(distort_thresh * 1.05, 0.0); ++ ++ ++ ++ for ( sfb = 0; sfb < cod_info->sfb_lmax; sfb++ ) { ++ if ( distort[0][sfb]>distort_thresh ) { ++ scalefac->l[sfb]++; ++ start = scalefac_band.l[sfb]; ++ end = scalefac_band.l[sfb+1]; ++ for ( l = start; l < end; l++ ) ++ xrpow[l] *= ifqstep34; ++ } ++ } ++ ++ ++ for ( i = 0; i < 3; i++ ) { ++ for ( sfb = cod_info->sfb_smax; sfb < 12; sfb++ ) { ++ if ( distort[i+1][sfb]>distort_thresh) { ++ scalefac->s[sfb][i]++; ++ start = scalefac_band.s[sfb]; ++ end = scalefac_band.s[sfb+1]; ++ for (l = start; l < end; l++) ++ xrpow[l * 3 + i] *= ifqstep34; ++ } ++ } ++ } ++ } ++ ++ ++ ++ int quant_compare(int experimentalX, ++ int best_over,FLOAT8 best_tot_noise,FLOAT8 best_over_noise,FLOAT8 best_max_noise, ++ int over,FLOAT8 tot_noise, FLOAT8 over_noise, FLOAT8 max_noise) ++ { ++ /* ++ noise is given in decibals (db) relative to masking thesholds. ++ ++ over_noise: sum of quantization noise > masking ++ tot_noise: sum of all quantization noise ++ max_noise: max quantization noise ++ ++ */ ++ int better=0; ++ ++ if (experimentalX==0) { ++ better = ((over < best_over) || ++ ((over==best_over) && (over_noise<=best_over_noise)) ) ; ++ } ++ ++ if (experimentalX==1) ++ better = max_noise < best_max_noise; ++ ++ if (experimentalX==2) { ++ better = tot_noise < best_tot_noise; ++ } ++ if (experimentalX==3) { ++ better = (tot_noise < best_tot_noise) && ++ (max_noise < best_max_noise + 2); ++ } ++ if (experimentalX==4) { ++ better = ( ( (0>=max_noise) && (best_max_noise>2)) || ++ ( (0>=max_noise) && (best_max_noise<0) && ((best_max_noise+2)>max_noise) && (tot_noise<best_tot_noise) ) || ++ ( (0>=max_noise) && (best_max_noise>0) && ((best_max_noise+2)>max_noise) && (tot_noise<(best_tot_noise+best_over_noise)) ) || ++ ( (0<max_noise) && (best_max_noise>-0.5) && ((best_max_noise+1)>max_noise) && ((tot_noise+over_noise)<(best_tot_noise+best_over_noise)) ) || ++ ( (0<max_noise) && (best_max_noise>-1) && ((best_max_noise+1.5)>max_noise) && ((tot_noise+over_noise+over_noise)<(best_tot_noise+best_over_noise+best_over_noise)) ) ); ++ } ++ if (experimentalX==5) { ++ better = (over_noise < best_over_noise) ++ || ((over_noise == best_over_noise)&&(tot_noise < best_tot_noise)); ++ } ++ if (experimentalX==6) { ++ better = (over_noise < best_over_noise) ++ ||( (over_noise == best_over_noise) ++ &&( (max_noise < best_max_noise) ++ ||( (max_noise == best_max_noise) ++ &&(tot_noise <= best_tot_noise) ++ ) ++ ) ++ ); ++ } ++ ++ return better; ++ } ++ ++ ++ int VBR_compare( ++ int best_over,FLOAT8 best_tot_noise,FLOAT8 best_over_noise,FLOAT8 best_max_noise, ++ int over,FLOAT8 tot_noise, FLOAT8 over_noise, FLOAT8 max_noise) ++ { ++ /* ++ noise is given in decibals (db) relative to masking thesholds. ++ ++ over_noise: sum of quantization noise > masking ++ tot_noise: sum of all quantization noise ++ max_noise: max quantization noise ++ ++ */ ++ int better=0; ++ ++ better = ((over <= best_over) && ++ (over_noise<=best_over_noise) && ++ (tot_noise<=best_tot_noise) && ++ (max_noise<=best_max_noise)); ++ return better; ++ } ++ ++ ++ ++ ++ ++ ++ ++ +diff -r -c -N encoder/quantize.h lame3.70/quantize.h +*** encoder/quantize.h Wed Dec 31 17:00:00 1969 +--- lame3.70/quantize.h Thu Apr 6 12:50:51 2000 +*************** +*** 0 **** +--- 1,70 ---- ++ /********************************************************************** ++ * ISO MPEG Audio Subgroup Software Simulation Group (1996) ++ * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension ++ * ++ * $Id: quantize.h,v 1.5 2000/03/21 23:02:17 markt Exp $ ++ * ++ * $Log: quantize.h,v $ ++ * Revision 1.5 2000/03/21 23:02:17 markt ++ * replaced all "gf." by gfp-> ++ * ++ * Revision 1.4 2000/03/14 21:01:47 markt ++ * removed fr_ps struct ++ * ++ * Revision 1.3 2000/02/01 14:09:14 takehiro ++ * code clean up. changed definition of structure to optimize array index calculation ++ * ++ * Revision 1.2 2000/02/01 11:26:32 takehiro ++ * scalefactor's structure changed ++ * ++ * Revision 1.1.1.1 1999/11/24 08:43:45 markt ++ * initial checkin of LAME ++ * Starting with LAME 3.57beta with some modifications ++ * ++ * Revision 1.1 1996/02/14 04:04:23 rowlands ++ * Initial revision ++ * ++ * Received from Mike Coleman ++ **********************************************************************/ ++ ++ #ifndef LOOP_DOT_H ++ #define LOOP_DOT_H ++ #include "util.h" ++ #include "l3side.h" ++ ++ /********************************************************************** ++ * date programmers comment * ++ * 25. 6.92 Toshiyuki Ishino Ver 1.0 * ++ * 29.10.92 Masahiro Iwadare Ver 2.0 * ++ * 17. 4.93 Masahiro Iwadare Updated for IS Modification * ++ * * ++ *********************************************************************/ ++ ++ extern int cont_flag; ++ ++ ++ extern int pretab[]; ++ ++ void iteration_loop( lame_global_flags *gfp, ++ FLOAT8 pe[2][2], FLOAT8 ms_ratio[2], ++ FLOAT8 xr_org[2][2][576], III_psy_ratio ratio[2][2], ++ III_side_info_t *l3_side, int l3_enc[2][2][576], ++ III_scalefac_t scalefac[2][2]); ++ ++ void VBR_iteration_loop( lame_global_flags *gfp, ++ FLOAT8 pe[2][2], FLOAT8 ms_ratio[2], ++ FLOAT8 xr_org[2][2][576], III_psy_ratio ratio[2][2], ++ III_side_info_t *l3_side, int l3_enc[2][2][576], ++ III_scalefac_t scalefac[2][2]); ++ ++ ++ ++ ++ #define maximum(A,B) ( (A) > (B) ? (A) : (B) ) ++ #define minimum(A,B) ( (A) < (B) ? (A) : (B) ) ++ #define signum( A ) ( (A) > 0 ? 1 : -1 ) ++ ++ ++ extern int bit_buffer[50000]; ++ ++ #endif +diff -r -c -N encoder/reservoir.c lame3.70/reservoir.c +*** encoder/reservoir.c Wed Jan 22 02:43:18 1997 +--- lame3.70/reservoir.c Thu Apr 6 12:50:51 2000 +*************** +*** 2,14 **** + * ISO MPEG Audio Subgroup Software Simulation Group (1996) + * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension + * +- * $Id: reservoir.c,v 1.1 1996/02/14 04:04:23 rowlands Exp $ +- * +- * $Log: reservoir.c,v $ +- * Revision 1.1 1996/02/14 04:04:23 rowlands +- * Initial revision +- * +- * Received from Mike Coleman + **********************************************************************/ + /* + Revision History: +--- 2,7 ---- +*************** +*** 22,32 **** + #include <stdlib.h> + #include <math.h> + #include <assert.h> +! #include "l3side.h" +! #include "loop.h" +! #include "huffman.h" +! #include "l3bitstream.h" +! #include "reservoir.h" + + /* + Layer3 bit reservoir: +--- 15,25 ---- + #include <stdlib.h> + #include <math.h> + #include <assert.h> +! #include "util.h" +! #ifdef HAVEGTK +! #include "gtkanal.h" +! #endif +! + + /* + Layer3 bit reservoir: +*************** +*** 38,63 **** + + /* + ResvFrameBegin: +! Called at the beginning of a frame. Updates the maximum + size of the reservoir, and checks to make sure main_data_begin + was set properly by the formatter + */ +! void +! ResvFrameBegin( frame_params *fr_ps, III_side_info_t *l3_side, int mean_bits, int frameLength ) + { +! layer *info; +! int fullFrameBits, mode_gr; +! int expectedResvSize, resvLimit; + +! info = fr_ps->header; +! if ( info->version == 1 ) + { +- mode_gr = 2; + resvLimit = 4088; /* main_data_begin has 9 bits in MPEG 1 */ + } + else + { +- mode_gr = 1; + resvLimit = 2040; /* main_data_begin has 8 bits in MPEG 2 */ + } + +--- 31,57 ---- + + /* + ResvFrameBegin: +! Called (repeatedly) at the beginning of a frame. Updates the maximum + size of the reservoir, and checks to make sure main_data_begin + was set properly by the formatter + */ +! int +! ResvFrameBegin(lame_global_flags *gfp,III_side_info_t *l3_side, int mean_bits, int frameLength ) + { +! int fullFrameBits; +! int resvLimit; + +! if (gfp->frameNum==0) { +! ResvSize=0; +! } +! +! +! if ( gfp->version == 1 ) + { + resvLimit = 4088; /* main_data_begin has 9 bits in MPEG 1 */ + } + else + { + resvLimit = 2040; /* main_data_begin has 8 bits in MPEG 2 */ + } + +*************** +*** 66,78 **** + expected value for the next call -- this should + agree with our reservoir size + */ +! expectedResvSize = l3_side->main_data_begin * 8; + #ifdef DEBUG + fprintf( stderr, ">>> ResvSize = %d\n", ResvSize ); + #endif +! assert( expectedResvSize == ResvSize ); +! +! fullFrameBits = mean_bits * mode_gr; + + /* + determine maximum size of reservoir: +--- 60,72 ---- + expected value for the next call -- this should + agree with our reservoir size + */ +! + #ifdef DEBUG + fprintf( stderr, ">>> ResvSize = %d\n", ResvSize ); + #endif +! /* check expected resvsize */ +! assert( (l3_side->main_data_begin * 8) == ResvSize ); +! fullFrameBits = mean_bits * gfp->mode_gr + ResvSize; + + /* + determine maximum size of reservoir: +*************** +*** 82,87 **** +--- 76,83 ---- + ResvMax = 0; + else + ResvMax = 7680 - frameLength; ++ if (gfp->disable_reservoir) ResvMax=0; ++ + + /* + limit max size to resvLimit bits because +*************** +*** 90,136 **** + */ + if ( ResvMax > resvLimit ) + ResvMax = resvLimit; + } + + /* +! ResvMaxBits: +! Called at the beginning of each granule to get the max bit +! allowance for the current granule based on reservoir size +! and perceptual entropy. + */ +! int +! ResvMaxBits( frame_params *fr_ps, III_side_info_t *l3_side, double *pe, int mean_bits ) + { +! int more_bits, max_bits, add_bits, over_bits; + +! mean_bits /= fr_ps->stereo; +! max_bits = mean_bits; +! +! if ( max_bits > 4095 ) +! max_bits = 4095; +! +! if ( ResvMax == 0 ) +! return max_bits; +! +! more_bits = *pe * 3.1 - mean_bits; +! add_bits = 0; +! if ( more_bits > 100 ) +! { +! int frac = (ResvSize * 6) / 10; +! +! if ( frac < more_bits ) +! add_bits = frac; +! else +! add_bits = more_bits; +! } +! over_bits = ResvSize - ((ResvMax * 8) / 10) - add_bits; +! if ( over_bits > 0 ) +! add_bits += over_bits; +! +! max_bits += add_bits; +! if ( max_bits > 4095 ) +! max_bits = 4095; +! return max_bits; + } + + /* +--- 86,133 ---- + */ + if ( ResvMax > resvLimit ) + ResvMax = resvLimit; ++ ++ #ifdef HAVEGTK ++ if (gfp->gtkflag){ ++ pinfo->mean_bits=mean_bits/2; /* expected bits per channel per granule */ ++ pinfo->resvsize=ResvSize; ++ } ++ #endif ++ ++ return fullFrameBits; + } + ++ + /* +! ResvMaxBits2: +! As above, but now it *really* is bits per granule (both channels). +! Mark Taylor 4/99 + */ +! void ResvMaxBits(int mean_bits, int *targ_bits, int *extra_bits, int gr) + { +! int add_bits; +! *targ_bits = mean_bits ; +! /* extra bits if the reservoir is almost full */ +! if (ResvSize > ((ResvMax * 9) / 10)) { +! add_bits= ResvSize-((ResvMax * 9) / 10); +! *targ_bits += add_bits; +! }else { +! add_bits =0 ; +! /* build up reservoir. this builds the reservoir a little slower +! * than FhG. It could simple be mean_bits/15, but this was rigged +! * to always produce 100 (the old value) at 128kbs */ +! *targ_bits -= (int) (mean_bits/15.2); +! } +! +! +! /* amount from the reservoir we are allowed to use. ISO says 6/10 */ +! *extra_bits = +! (ResvSize < (ResvMax*6)/10 ? ResvSize : (ResvMax*6)/10); +! *extra_bits -= add_bits; +! +! if (*extra_bits < 0) *extra_bits=0; + +! + } + + /* +*************** +*** 139,149 **** + the reservoir to reflect the granule's usage. + */ + void +! ResvAdjust( frame_params *fr_ps, gr_info *gi, III_side_info_t *l3_side, int mean_bits ) + { +! ResvSize += (mean_bits / fr_ps->stereo) - gi->part2_3_length; + } + + /* + ResvFrameEnd: + Called after all granules in a frame have been allocated. Makes sure +--- 136,147 ---- + the reservoir to reflect the granule's usage. + */ + void +! ResvAdjust(lame_global_flags *gfp,gr_info *gi, III_side_info_t *l3_side, int mean_bits ) + { +! ResvSize += (mean_bits / gfp->stereo) - gi->part2_3_length; + } + ++ + /* + ResvFrameEnd: + Called after all granules in a frame have been allocated. Makes sure +*************** +*** 153,182 **** + appropriate stuffing bits to the bitstream. + */ + void +! ResvFrameEnd( frame_params *fr_ps, III_side_info_t *l3_side, int mean_bits ) + { +! layer *info; +! gr_info *gi; +! int mode_gr, gr, ch, stereo, ancillary_pad, stuffingBits; + int over_bits; + +- info = fr_ps->header; +- stereo = fr_ps->stereo; +- mode_gr = (info->version == 1) ? 2 : 1; +- ancillary_pad = 0; +- +- #if 1 + /* just in case mean_bits is odd, this is necessary... */ +! if ( (stereo == 2) && (mean_bits & 1) ) + ResvSize += 1; +- #endif + + over_bits = ResvSize - ResvMax; + if ( over_bits < 0 ) + over_bits = 0; + + ResvSize -= over_bits; +! stuffingBits = over_bits + ancillary_pad; + + /* we must be byte aligned */ + if ( (over_bits = ResvSize % 8) ) +--- 151,171 ---- + appropriate stuffing bits to the bitstream. + */ + void +! ResvFrameEnd(lame_global_flags *gfp,III_side_info_t *l3_side, int mean_bits) + { +! int stuffingBits; + int over_bits; + + /* just in case mean_bits is odd, this is necessary... */ +! if ( gfp->stereo == 2 && mean_bits & 1) + ResvSize += 1; + + over_bits = ResvSize - ResvMax; + if ( over_bits < 0 ) + over_bits = 0; + + ResvSize -= over_bits; +! stuffingBits = over_bits; + + /* we must be byte aligned */ + if ( (over_bits = ResvSize % 8) ) +*************** +*** 185,228 **** + ResvSize -= over_bits; + } + +! if ( stuffingBits ) +! { +! /* +! plan a: put all into the first granule +! This was preferred by someone designing a +! real-time decoder... +! */ +! gi = (gr_info *) &(l3_side->gr[0].ch[0]); +! +! if ( gi->part2_3_length + stuffingBits < 4095 ) +! gi->part2_3_length += stuffingBits; +! else +! { +! /* plan b: distribute throughout the granules */ +! for (gr = 0; gr < mode_gr; gr++ ) +! for (ch = 0; ch < stereo; ch++ ) +! { +! int extraBits, bitsThisGr; +! gr_info *gi = (gr_info *) &(l3_side->gr[gr].ch[ch]); +! if ( stuffingBits == 0 ) +! break; +! extraBits = 4095 - gi->part2_3_length; +! bitsThisGr = extraBits < stuffingBits ? extraBits : stuffingBits; +! gi->part2_3_length += bitsThisGr; +! stuffingBits -= bitsThisGr; +! } +! /* +! If any stuffing bits remain, we elect to spill them +! into ancillary data. The bitstream formatter will do this if +! l3side->resvDrain is set +! */ +! #ifdef DEBUG +! if ( stuffingBits ) +! fprintf( stderr, "spilling %d stuffing bits into ancillary data\n", stuffingBits ); +! #endif +! l3_side->resvDrain = stuffingBits; +! } +! } + } + + +--- 174,183 ---- + ResvSize -= over_bits; + } + +! +! l3_side->resvDrain = stuffingBits; +! return; +! + } + + +diff -r -c -N encoder/reservoir.h lame3.70/reservoir.h +*** encoder/reservoir.h Wed Jan 22 02:43:18 1997 +--- lame3.70/reservoir.h Tue Mar 21 16:02:17 2000 +*************** +*** 2,10 **** + * ISO MPEG Audio Subgroup Software Simulation Group (1996) + * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension + * +! * $Id: reservoir.h,v 1.1 1996/02/14 04:04:23 rowlands Exp $ + * + * $Log: reservoir.h,v $ + * Revision 1.1 1996/02/14 04:04:23 rowlands + * Initial revision + * +--- 2,23 ---- + * ISO MPEG Audio Subgroup Software Simulation Group (1996) + * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension + * +! * $Id: reservoir.h,v 1.4 2000/03/21 23:02:17 markt Exp $ + * + * $Log: reservoir.h,v $ ++ * Revision 1.4 2000/03/21 23:02:17 markt ++ * replaced all "gf." by gfp-> ++ * ++ * Revision 1.3 2000/03/14 21:01:47 markt ++ * removed fr_ps struct ++ * ++ * Revision 1.2 2000/01/13 16:26:50 takehiro ++ * moved info.stereo into gf.stereo ++ * ++ * Revision 1.1.1.1 1999/11/24 08:43:40 markt ++ * initial checkin of LAME ++ * Starting with LAME 3.57beta with some modifications ++ * + * Revision 1.1 1996/02/14 04:04:23 rowlands + * Initial revision + * +*************** +*** 22,30 **** + #ifndef RESERVOIR_H + #define RESERVOIR_H + +! void ResvFrameBegin( frame_params *fr_ps, III_side_info_t *l3_side, int mean_bits, int frameLength ); +! int ResvMaxBits( frame_params *fr_ps, III_side_info_t *l3_side, double *pe, int mean_bits ); +! void ResvAdjust( frame_params *fr_ps, gr_info *gi, III_side_info_t *l3_side, int mean_bits ); +! void ResvFrameEnd( frame_params *fr_ps, III_side_info_t *l3_side, int mean_bits ); + + #endif +--- 35,43 ---- + #ifndef RESERVOIR_H + #define RESERVOIR_H + +! int ResvFrameBegin( lame_global_flags *gfp,III_side_info_t *l3_side, int mean_bits, int frameLength ); +! void ResvMaxBits( int mean_bits, int *targ_bits, int *max_bits, int gr); +! void ResvAdjust(lame_global_flags *gfp,gr_info *gi, III_side_info_t *l3_side, int mean_bits ); +! void ResvFrameEnd(lame_global_flags *gfp,III_side_info_t *l3_side, int mean_bits ); + + #endif +diff -r -c -N encoder/rtp.c lame3.70/rtp.c +*** encoder/rtp.c Wed Dec 31 17:00:00 1969 +--- lame3.70/rtp.c Wed Jan 26 21:35:48 2000 +*************** +*** 0 **** +--- 1,102 ---- ++ #include <stdlib.h> ++ #include <string.h> ++ #include <netinet/in.h> ++ #include <unistd.h> ++ #include <stdlib.h> ++ #include <stdio.h> ++ #include <sys/types.h> ++ #include <sys/socket.h> ++ #include <arpa/inet.h> ++ ++ struct rtpbits { ++ int sequence:16; /* sequence number: random */ ++ int pt:7; /* payload type: 14 for MPEG audio */ ++ int m:1; /* marker: 0 */ ++ int cc:4; /* number of CSRC identifiers: 0 */ ++ int x:1; /* number of extension headers: 0 */ ++ int p:1; /* is there padding appended: 0 */ ++ int v:2; /* version: 2 */ ++ }; ++ ++ struct rtpheader { /* in network byte order */ ++ struct rtpbits b; ++ int timestamp; /* start: random */ ++ int ssrc; /* random */ ++ int iAudioHeader; /* =0?! */ ++ }; ++ ++ void initrtp(struct rtpheader *foo) { ++ foo->b.v=2; ++ foo->b.p=0; ++ foo->b.x=0; ++ foo->b.cc=0; ++ foo->b.m=0; ++ foo->b.pt=14; /* MPEG Audio */ ++ #ifdef FEFE ++ foo->b.sequence=42; ++ foo->timestamp=0; ++ #else ++ foo->b.sequence=rand() & 65535; ++ foo->timestamp=rand(); ++ #endif ++ foo->ssrc=rand(); ++ foo->iAudioHeader=0; ++ } ++ ++ int sendrtp(int fd, struct sockaddr_in *sSockAddr, struct rtpheader *foo, void *data, int len) { ++ char *buf=alloca(len+sizeof(struct rtpheader)); ++ int *cast=(int *)foo; ++ int *outcast=(int *)buf; ++ outcast[0]=htonl(cast[0]); ++ outcast[1]=htonl(cast[1]); ++ outcast[2]=htonl(cast[2]); ++ outcast[3]=htonl(cast[3]); ++ memmove(buf+sizeof(struct rtpheader),data,len); ++ return sendto(fd,buf,len+sizeof(*foo),0,(struct sockaddr *)sSockAddr,sizeof(*sSockAddr)); ++ /* return write(fd,buf,len+sizeof(*foo))==len+sizeof(*foo); */ ++ } ++ ++ /* create a sender socket. */ ++ int makesocket(char *szAddr,unsigned short port,int TTL,struct sockaddr_in *sSockAddr) { ++ int iRet, iLoop = 1; ++ struct sockaddr_in sin; ++ char cTtl = (char)TTL; ++ char cLoop=0; ++ unsigned int tempaddr; ++ ++ int iSocket = socket( AF_INET, SOCK_DGRAM, 0 ); ++ if (iSocket < 0) { ++ fprintf(stderr,"socket() failed.\n"); ++ exit(1); ++ } ++ ++ tempaddr=inet_addr(szAddr); ++ sSockAddr->sin_family = sin.sin_family = AF_INET; ++ sSockAddr->sin_port = sin.sin_port = htons(port); ++ sSockAddr->sin_addr.s_addr = tempaddr; ++ ++ iRet = setsockopt(iSocket, SOL_SOCKET, SO_REUSEADDR, &iLoop, sizeof(int)); ++ if (iRet < 0) { ++ fprintf(stderr,"setsockopt SO_REUSEADDR failed\n"); ++ exit(1); ++ } ++ ++ if ((ntohl(tempaddr) >> 28) == 0xe) { ++ /* only set multicast parameters for multicast destination IPs */ ++ iRet = setsockopt(iSocket, IPPROTO_IP, IP_MULTICAST_TTL, &cTtl, sizeof(char)); ++ if (iRet < 0) { ++ fprintf(stderr,"setsockopt IP_MULTICAST_TTL failed. multicast in kernel?\n"); ++ exit(1); ++ } ++ ++ cLoop = 1; /* !? */ ++ iRet = setsockopt(iSocket, IPPROTO_IP, IP_MULTICAST_LOOP, ++ &cLoop, sizeof(char)); ++ if (iRet < 0) { ++ fprintf(stderr,"setsockopt IP_MULTICAST_LOOP failed. multicast in kernel?\n"); ++ exit(1); ++ } ++ } ++ ++ return iSocket; ++ } +diff -r -c -N encoder/rtp.h lame3.70/rtp.h +*** encoder/rtp.h Wed Dec 31 17:00:00 1969 +--- lame3.70/rtp.h Tue Jan 25 10:48:23 2000 +*************** +*** 0 **** +--- 1,24 ---- ++ #include <sys/socket.h> ++ #include <netinet/in.h> ++ ++ struct rtpbits { ++ int sequence:16; /* sequence number: random */ ++ int pt:7; /* payload type: 14 for MPEG audio */ ++ int m:1; /* marker: 0 */ ++ int cc:4; /* number of CSRC identifiers: 0 */ ++ int x:1; /* number of extension headers: 0 */ ++ int p:1; /* is there padding appended: 0 */ ++ int v:2; /* version: 2 */ ++ }; ++ ++ struct rtpheader { /* in network byte order */ ++ struct rtpbits b; ++ int timestamp; /* start: random */ ++ int ssrc; /* random */ ++ int iAudioHeader; /* =0?! */ ++ }; ++ ++ void initrtp(struct rtpheader *foo); ++ int sendrtp(int fd, struct sockaddr_in *sSockAddr, struct rtpheader *foo, void *data, int len); ++ int makesocket(char *szAddr,unsigned short port,int TTL,struct sockaddr_in *sSockAddr); ++ void rtp_output(char *mp3buffer,int mp3size); +diff -r -c -N encoder/subs.c lame3.70/subs.c +*** encoder/subs.c Wed Jan 22 02:43:18 1997 +--- lame3.70/subs.c Wed Dec 31 17:00:00 1969 +*************** +*** 1,163 **** +- /********************************************************************** +- * ISO MPEG Audio Subgroup Software Simulation Group (1996) +- * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- * +- * $Id: subs.c,v 1.1 1996/02/14 04:04:23 rowlands Exp $ +- * +- * $Log: subs.c,v $ +- * Revision 1.1 1996/02/14 04:04:23 rowlands +- * Initial revision +- * +- * Received from Mike Coleman +- **********************************************************************/ +- /********************************************************************** +- * date programmers comment * +- * 2/25/91 Davis Pan start of version 1.0 records * +- * 5/10/91 W. Joseph Carter Ported to Macintosh and Unix. * +- * 7/10/91 Earle Jennings Ported to MsDos from Macintosh * +- * Replacement of one float with FLOAT * +- * 2/11/92 W. Joseph Carter Added type casting to memset() args. * +- * 4/27/92 Masahiro Iwadare Added 256 point version for Layer III * +- **********************************************************************/ +- +- #include "common.h" +- #include "encoder.h" +- +- /***************************************************************************** +- ************************** Start of Subroutines ***************************** +- *****************************************************************************/ +- +- /***************************************************************************** +- * FFT computes fast fourier transform of BLKSIZE samples of data * +- * uses decimation-in-frequency algorithm described in "Digital * +- * Signal Processing" by Oppenheim and Schafer, refer to pages 304 * +- * (flow graph) and 330-332 (Fortran program in problem 5) * +- * to get the inverse fft, change line 20 from * +- * w_imag[L] = -sin(PI/le1); * +- * to * +- * w_imag[L] = sin(PI/le1); * +- * * +- * required constants: * +- * #define PI 3.14159265358979 * +- * #define BLKSIZE 1024 * +- * #define LOGBLKSIZE 10 * +- * #define BLKSIZE_S 256 * +- * #define LOGBLKSIZE_S 8 * +- * * +- *****************************************************************************/ +- #define BLKSIZE_S 256 +- #define LOGBLKSIZE_S 8 +- +- void fft(x_real,x_imag, energy, phi, N) +- FLOAT x_real[BLKSIZE], x_imag[BLKSIZE], energy[BLKSIZE], phi[BLKSIZE]; +- int N; +- { +- int M,MM1; +- static int init=0; +- int NV2, NM1, MP; +- static double w_real[2][LOGBLKSIZE], w_imag[2][LOGBLKSIZE]; +- int i,j,k,L; +- int ip, le,le1; +- double t_real, t_imag, u_real, u_imag; +- +- if(init==0) { +- memset((char *) w_real, 0, sizeof(w_real)); /* preset statics to 0 */ +- memset((char *) w_imag, 0, sizeof(w_imag)); /* preset statics to 0 */ +- M = LOGBLKSIZE; +- for(L=0; L<M; L++){ +- le = 1 << (M-L); +- le1 = le >> 1; +- w_real[0][L] = cos(PI/le1); +- w_imag[0][L] = -sin(PI/le1); +- } +- M = LOGBLKSIZE_S; +- for(L=0; L<M; L++){ +- le = 1 << (M-L); +- le1 = le >> 1; +- w_real[1][L] = cos(PI/le1); +- w_imag[1][L] = -sin(PI/le1); +- } +- init++; +- } +- switch(N) { +- case BLKSIZE: +- M = LOGBLKSIZE; +- MP = 0; +- break; +- case BLKSIZE_S: +- M = LOGBLKSIZE_S; +- MP = 1; +- break; +- default: printf("Error: Bad FFT Size in subs.c\n"); +- exit(-1); +- } +- MM1 = M-1; +- NV2 = N >> 1; +- NM1 = N - 1; +- for(L=0; L<MM1; L++){ +- le = 1 << (M-L); +- le1 = le >> 1; +- u_real = 1; +- u_imag = 0; +- for(j=0; j<le1; j++){ +- for(i=j; i<N; i+=le){ +- ip = i + le1; +- t_real = x_real[i] + x_real[ip]; +- t_imag = x_imag[i] + x_imag[ip]; +- x_real[ip] = x_real[i] - x_real[ip]; +- x_imag[ip] = x_imag[i] - x_imag[ip]; +- x_real[i] = t_real; +- x_imag[i] = t_imag; +- t_real = x_real[ip]; +- x_real[ip] = x_real[ip]*u_real - x_imag[ip]*u_imag; +- x_imag[ip] = x_imag[ip]*u_real + t_real*u_imag; +- } +- t_real = u_real; +- u_real = u_real*w_real[MP][L] - u_imag*w_imag[MP][L]; +- u_imag = u_imag*w_real[MP][L] + t_real*w_imag[MP][L]; +- } +- } +- /* special case: L = M-1; all Wn = 1 */ +- for(i=0; i<N; i+=2){ +- ip = i + 1; +- t_real = x_real[i] + x_real[ip]; +- t_imag = x_imag[i] + x_imag[ip]; +- x_real[ip] = x_real[i] - x_real[ip]; +- x_imag[ip] = x_imag[i] - x_imag[ip]; +- x_real[i] = t_real; +- x_imag[i] = t_imag; +- energy[i] = x_real[i]*x_real[i] + x_imag[i]*x_imag[i]; +- if(energy[i] <= 0.0005){phi[i] = 0;energy[i] = 0.0005;} +- else phi[i] = atan2((double) x_imag[i],(double) x_real[i]); +- energy[ip] = x_real[ip]*x_real[ip] + x_imag[ip]*x_imag[ip]; +- if(energy[ip] == 0)phi[ip] = 0; +- else phi[ip] = atan2((double) x_imag[ip],(double) x_real[ip]); +- } +- /* this section reorders the data to the correct ordering */ +- j = 0; +- for(i=0; i<NM1; i++){ +- if(i<j){ +- /* use this section only if you need the FFT in complex number form * +- * (and in the correct ordering) */ +- t_real = x_real[j]; +- t_imag = x_imag[j]; +- x_real[j] = x_real[i]; +- x_imag[j] = x_imag[i]; +- x_real[i] = t_real; +- x_imag[i] = t_imag; +- /* reorder the energy and phase, phi */ +- t_real = energy[j]; +- energy[j] = energy[i]; +- energy[i] = t_real; +- t_real = phi[j]; +- phi[j] = phi[i]; +- phi[i] = t_real; +- } +- k=NV2; +- while(k<=j){ +- j = j-k; +- k = k >> 1; +- } +- j = j+k; +- } +- } +--- 0 ---- +diff -r -c -N encoder/tables.c lame3.70/tables.c +*** encoder/tables.c Wed Dec 31 17:00:00 1969 +--- lame3.70/tables.c Thu Apr 6 12:50:52 2000 +*************** +*** 0 **** +--- 1,1215 ---- ++ #include "util.h" ++ #include "tables.h" ++ ++ /* ++ Here are MPEG1 Table B.8 and MPEG2 Table B.1 ++ -- Layer III scalefactor bands. ++ Index into this using a method such as: ++ idx = fr_ps->header->sampling_frequency ++ + (fr_ps->header->version * 3) ++ */ ++ ++ ++ ++ ++ unsigned int hs = sizeof(HUFFBITS)*8; ++ ++ static HUFFBITS t1HB[] = { ++ 1, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 1, 0}; ++ ++ static HUFFBITS t2HB[] = { ++ 1, 2, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 3, 1, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 3, 2, 0}; ++ ++ static HUFFBITS t3HB[] = { ++ 3, 2, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 1, 1, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 3, 2, 0}; ++ ++ static HUFFBITS t5HB[] = { ++ 1, 2, 6, 5, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 3, 1, 4, 4, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 7, 5, 7, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 6, 1, 1, 0}; ++ ++ static HUFFBITS t6HB[] = { ++ 7, 3, 5, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 6, 2, 3, 2, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 5, 4, 4, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 3, 3, 2, 0}; ++ ++ static HUFFBITS t7HB[] = { ++ 1, 2,10,19,16,10, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 3, 3, 7,10, 5, 3, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 11, 4,13,17, 8, 4, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 12,11,18,15,11, 2, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 7, 6, 9,14, 3, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 6, 4, 5, 3, 2, 0}; ++ ++ static HUFFBITS t8HB[] = { ++ 3, 4, 6, 18,12, 5, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 5, 1, 2, 16, 9, 3, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 7, 3, 5, 14, 7, 3, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 19,17,15, 13,10, 4, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 13, 5, 8, 11, 5, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 12, 4, 4, 1, 1, 0}; ++ ++ static HUFFBITS t9HB[] = { ++ 7, 5, 9, 14, 15, 7, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 6, 4, 5, 5, 6, 7, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 7, 6, 8, 8, 8, 5, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 15, 6, 9, 10, 5, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 11, 7, 9, 6, 4, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 14, 4, 6, 2, 6, 0}; ++ ++ static HUFFBITS t10HB[] = { ++ 1, 2, 10, 23, 35, 30, 12, 17, 0, 0, 0, 0, 0, 0, 0, 0, ++ 3, 3, 8, 12, 18, 21, 12, 7, 0, 0, 0, 0, 0, 0, 0, 0, ++ 11, 9, 15, 21, 32, 40, 19, 6, 0, 0, 0, 0, 0, 0, 0, 0, ++ 14,13, 22, 34, 46, 23, 18, 7, 0, 0, 0, 0, 0, 0, 0, 0, ++ 20,19, 33, 47, 27, 22, 9, 3, 0, 0, 0, 0, 0, 0, 0, 0, ++ 31,22, 41, 26, 21, 20, 5, 3, 0, 0, 0, 0, 0, 0, 0, 0, ++ 14,13, 10, 11, 16, 6, 5, 1, 0, 0, 0, 0, 0, 0, 0, 0, ++ 9, 8, 7, 8, 4, 4, 2, 0}; ++ ++ static HUFFBITS t11HB[] = { ++ 3, 4, 10, 24, 34, 33, 21, 15, 0, 0, 0, 0, 0, 0, 0, 0, ++ 5, 3, 4, 10, 32, 17, 11, 10, 0, 0, 0, 0, 0, 0, 0, 0, ++ 11, 7, 13, 18, 30, 31, 20, 5, 0, 0, 0, 0, 0, 0, 0, 0, ++ 25,11, 19, 59, 27, 18, 12, 5, 0, 0, 0, 0, 0, 0, 0, 0, ++ 35,33, 31, 58, 30, 16, 7, 5, 0, 0, 0, 0, 0, 0, 0, 0, ++ 28,26, 32, 19, 17, 15, 8, 14, 0, 0, 0, 0, 0, 0, 0, 0, ++ 14,12, 9, 13, 14, 9, 4, 1, 0, 0, 0, 0, 0, 0, 0, 0, ++ 11, 4, 6, 6, 6, 3, 2, 0}; ++ ++ static HUFFBITS t12HB[] = { ++ 9, 6, 16, 33, 41, 39, 38,26, 0, 0, 0, 0, 0, 0, 0, 0, ++ 7, 5, 6, 9, 23, 16, 26,11, 0, 0, 0, 0, 0, 0, 0, 0, ++ 17, 7, 11, 14, 21, 30, 10, 7, 0, 0, 0, 0, 0, 0, 0, 0, ++ 17, 10, 15, 12, 18, 28, 14, 5, 0, 0, 0, 0, 0, 0, 0, 0, ++ 32, 13, 22, 19, 18, 16, 9, 5, 0, 0, 0, 0, 0, 0, 0, 0, ++ 40, 17, 31, 29, 17, 13, 4, 2, 0, 0, 0, 0, 0, 0, 0, 0, ++ 27, 12, 11, 15, 10, 7, 4, 1, 0, 0, 0, 0, 0, 0, 0, 0, ++ 27, 12, 8, 12, 6, 3, 1, 0}; ++ ++ static HUFFBITS t13HB[] = { ++ 1, 5, 14, 21, 34, 51, 46, 71, 42, 52, 68, 52, 67, 44, 43, 19, ++ 3, 4, 12, 19, 31, 26, 44, 33, 31, 24, 32, 24, 31, 35, 22, 14, ++ 15, 13, 23, 36, 59, 49, 77, 65, 29, 40, 30, 40, 27, 33, 42, 16, ++ 22, 20, 37, 61, 56, 79, 73, 64, 43, 76, 56, 37, 26, 31, 25, 14, ++ 35, 16, 60, 57, 97, 75,114, 91, 54, 73, 55, 41, 48, 53, 23, 24, ++ 58, 27, 50, 96, 76, 70, 93, 84, 77, 58, 79, 29, 74, 49, 41, 17, ++ 47, 45, 78, 74,115, 94, 90, 79, 69, 83, 71, 50, 59, 38, 36, 15, ++ 72, 34, 56, 95, 92, 85, 91, 90, 86, 73, 77, 65, 51, 44, 43, 42, ++ 43, 20, 30, 44, 55, 78, 72, 87, 78, 61, 46, 54, 37, 30, 20, 16, ++ 53, 25, 41, 37, 44, 59, 54, 81, 66, 76, 57, 54, 37, 18, 39, 11, ++ 35, 33, 31, 57, 42, 82, 72, 80, 47, 58, 55, 21, 22, 26, 38, 22, ++ 53, 25, 23, 38, 70, 60, 51, 36, 55, 26, 34, 23, 27, 14, 9, 7, ++ 34, 32, 28, 39, 49, 75, 30, 52, 48, 40, 52, 28, 18, 17, 9, 5, ++ 45, 21, 34, 64, 56, 50, 49, 45, 31, 19, 12, 15, 10, 7, 6, 3, ++ 48, 23, 20, 39, 36, 35, 53, 21, 16, 23, 13, 10, 6, 1, 4, 2, ++ 16, 15, 17, 27, 25, 20, 29, 11, 17, 12, 16, 8, 1, 1, 0, 1}; ++ ++ static HUFFBITS t15HB[] = { ++ 7, 12, 18, 53, 47, 76,124,108, 89,123,108,119,107, 81,122, 63, ++ 13, 5, 16, 27, 46, 36, 61, 51, 42, 70, 52, 83, 65, 41, 59, 36, ++ 19, 17, 15, 24, 41, 34, 59, 48, 40, 64, 50, 78, 62, 80, 56, 33, ++ 29, 28, 25, 43, 39, 63, 55, 93, 76, 59, 93, 72, 54, 75, 50, 29, ++ 52, 22, 42, 40, 67, 57, 95, 79, 72, 57, 89, 69, 49, 66, 46, 27, ++ 77, 37, 35, 66, 58, 52, 91, 74, 62, 48, 79, 63, 90, 62, 40, 38, ++ 125, 32, 60, 56, 50, 92, 78, 65, 55, 87, 71, 51, 73, 51, 70, 30, ++ 109, 53, 49, 94, 88, 75, 66,122, 91, 73, 56, 42, 64, 44, 21, 25, ++ 90, 43, 41, 77, 73, 63, 56, 92, 77, 66, 47, 67, 48, 53, 36, 20, ++ 71, 34, 67, 60, 58, 49, 88, 76, 67,106, 71, 54, 38, 39, 23, 15, ++ 109, 53, 51, 47, 90, 82, 58, 57, 48, 72, 57, 41, 23, 27, 62, 9, ++ 86, 42, 40, 37, 70, 64, 52, 43, 70, 55, 42, 25, 29, 18, 11, 11, ++ 118, 68, 30, 55, 50, 46, 74, 65, 49, 39, 24, 16, 22, 13, 14, 7, ++ 91, 44, 39, 38, 34, 63, 52, 45, 31, 52, 28, 19, 14, 8, 9, 3, ++ 123, 60, 58, 53, 47, 43, 32, 22, 37, 24, 17, 12, 15, 10, 2, 1, ++ 71, 37, 34, 30, 28, 20, 17, 26, 21, 16, 10, 6, 8, 6, 2, 0}; ++ ++ static HUFFBITS t16HB[] = { ++ 1, 5, 14, 44, 74, 63, 110, 93, 172, 149, 138, 242, 225, 195, 376, 17, ++ 3, 4, 12, 20, 35, 62, 53, 47, 83, 75, 68, 119, 201, 107, 207, 9, ++ 15, 13, 23, 38, 67, 58, 103, 90, 161, 72, 127, 117, 110, 209, 206, 16, ++ 45, 21, 39, 69, 64,114, 99, 87, 158, 140, 252, 212, 199, 387, 365, 26, ++ 75, 36, 68, 65,115,101, 179,164, 155, 264, 246, 226, 395, 382, 362, 9, ++ 66, 30, 59, 56,102,185, 173,265, 142, 253, 232, 400, 388, 378, 445, 16, ++ 111, 54, 52,100,184,178, 160,133, 257, 244, 228, 217, 385, 366, 715, 10, ++ 98, 48, 91, 88,165,157, 148,261, 248, 407, 397, 372, 380, 889, 884, 8, ++ 85, 84, 81,159,156,143, 260,249, 427, 401, 392, 383, 727, 713, 708, 7, ++ 154, 76, 73,141,131,256, 245,426, 406, 394, 384, 735, 359, 710, 352, 11, ++ 139, 129, 67,125,247,233, 229,219, 393, 743, 737, 720, 885, 882, 439, 4, ++ 243, 120,118,115,227,223, 396,746, 742, 736, 721, 712, 706, 223, 436, 6, ++ 202, 224,222,218,216,389, 386,381, 364, 888, 443, 707, 440, 437,1728, 4, ++ 747, 211,210,208,370, 379,734,723, 714,1735, 883, 877, 876,3459, 865, 2, ++ 377, 369,102,187, 726,722,358,711, 709, 866,1734, 871,3458, 870, 434, 0, ++ 12, 10, 7, 11, 10, 17, 11, 9, 13, 12, 10, 7, 5, 3, 1, 3}; ++ ++ static HUFFBITS t24HB[] = { ++ 15, 13, 46, 80, 146, 262, 248, 434, 426, 669, 653, 649, 621, 517, 1032, 88, ++ 14, 12, 21, 38, 71, 130, 122, 216, 209, 198, 327, 345, 319, 297, 279, 42, ++ 47, 22, 41, 74, 68, 128, 120, 221, 207, 194, 182, 340, 315, 295, 541, 18, ++ 81, 39, 75, 70, 134, 125, 116, 220, 204, 190, 178, 325, 311, 293, 271, 16, ++ 147, 72, 69,135, 127, 118, 112, 210, 200, 188, 352, 323, 306, 285, 540, 14, ++ 263, 66,129,126, 119, 114, 214, 202, 192, 180, 341, 317, 301, 281, 262, 12, ++ 249,123,121,117, 113, 215, 206, 195, 185, 347, 330, 308, 291, 272, 520, 10, ++ 435,115,111,109, 211, 203, 196, 187, 353, 332, 313, 298, 283, 531, 381, 17, ++ 427,212,208,205, 201, 193, 186, 177, 169, 320, 303, 286, 268, 514, 377, 16, ++ 335,199,197,191, 189, 181, 174, 333, 321, 305, 289, 275, 521, 379, 371, 11, ++ 668,184,183,179, 175, 344, 331, 314, 304, 290, 277, 530, 383, 373, 366, 10, ++ 652,346,171,168, 164, 318, 309, 299, 287, 276, 263, 513, 375, 368, 362, 6, ++ 648,322,316,312, 307, 302, 292, 284, 269, 261, 512, 376, 370, 364, 359, 4, ++ 620,300,296,294, 288, 282, 273, 266, 515, 380, 374, 369, 365, 361, 357, 2, ++ 1033,280,278,274, 267, 264, 259, 382, 378, 372, 367, 363, 360, 358, 356, 0, ++ 43, 20, 19, 17, 15, 13, 11, 9, 7, 6, 4, 7, 5, 3, 1, 3}; ++ ++ static HUFFBITS t32HB[] = { ++ 1, 5, 4, 5, 6, 5, 4, 4, 7, 3, 6, 0, 7, 2, 3, 1}; ++ static HUFFBITS t33HB[] = { ++ 15, 14, 13, 12, 11, 10, 9, 8, 7, 6, 5, 4, 3, 2, 1, 0}; ++ ++ static unsigned char t1l[] = { ++ 1, 3, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 2, 3}; ++ ++ static unsigned char t2l[] = { ++ 1, 3, 6, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 3, 3, 5, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 5, 5, 6}; ++ ++ static unsigned char t3l[] = { ++ 2, 2, 6, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 3, 2, 5, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 5, 5, 6}; ++ ++ static unsigned char t5l[] = { ++ 1, 3, 6, 7, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 3, 3, 6, 7, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 6, 6, 7, 8, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 7, 6, 7, 8}; ++ ++ static unsigned char t6l[] = { ++ 3, 3, 5, 7, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 3, 2, 4, 5, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 4, 4, 5, 6, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 6, 5, 6, 7}; ++ ++ static unsigned char t7l[] = { ++ 1, 3, 6, 8, 8, 9, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 3, 4, 6, 7, 7, 8, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 6, 5, 7, 8, 8, 9, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 7, 7, 8, 9, 9, 9, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 7, 7, 8, 9, 9,10, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 8, 8, 9,10,10,10}; ++ ++ static unsigned char t8l[] = { ++ 2, 3, 6, 8, 8, 9, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 3, 2, 4, 8, 8, 8, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 6, 4, 6, 8, 8, 9, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 8, 8, 8, 9, 9,10, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 8, 7, 8, 9,10,10, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 9, 8, 9, 9,11,11}; ++ ++ static unsigned char t9l[] = { ++ 3, 3, 5, 6, 8, 9, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 3, 3, 4, 5, 6, 8, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 4, 4, 5, 6, 7, 8, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 6, 5, 6, 7, 7, 8, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 7, 6, 7, 7, 8, 9, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, ++ 8, 7, 8, 8, 9, 9}; ++ ++ static unsigned char t10l[] = { ++ 1, 3, 6, 8, 9, 9, 9, 10, 0, 0, 0, 0, 0, 0, 0, 0, ++ 3, 4, 6, 7, 8, 9, 8, 8, 0, 0, 0, 0, 0, 0, 0, 0, ++ 6, 6, 7, 8, 9,10, 9, 9, 0, 0, 0, 0, 0, 0, 0, 0, ++ 7, 7, 8, 9,10,10, 9, 10, 0, 0, 0, 0, 0, 0, 0, 0, ++ 8, 8, 9,10,10,10,10, 10, 0, 0, 0, 0, 0, 0, 0, 0, ++ 9, 9,10,10,11,11,10, 11, 0, 0, 0, 0, 0, 0, 0, 0, ++ 8, 8, 9,10,10,10,11, 11, 0, 0, 0, 0, 0, 0, 0, 0, ++ 9, 8, 9,10,10,11,11, 11}; ++ ++ static unsigned char t11l[] = { ++ 2, 3, 5, 7, 8, 9, 8, 9, 0, 0, 0, 0, 0, 0, 0, 0, ++ 3, 3, 4, 6, 8, 8, 7, 8, 0, 0, 0, 0, 0, 0, 0, 0, ++ 5, 5, 6, 7, 8, 9, 8, 8, 0, 0, 0, 0, 0, 0, 0, 0, ++ 7, 6, 7, 9, 8, 10, 8, 9, 0, 0, 0, 0, 0, 0, 0, 0, ++ 8, 8, 8, 9, 9, 10, 9, 10, 0, 0, 0, 0, 0, 0, 0, 0, ++ 8, 8, 9,10,10, 11, 10, 11, 0, 0, 0, 0, 0, 0, 0, 0, ++ 8, 7, 7, 8, 9, 10, 10, 10, 0, 0, 0, 0, 0, 0, 0, 0, ++ 8, 7, 8, 9,10, 10, 10, 10}; ++ ++ static unsigned char t12l[] = { ++ 4, 3, 5, 7, 8, 9, 9, 9, 0, 0, 0, 0, 0, 0, 0, 0, ++ 3, 3, 4, 5, 7, 7, 8, 8, 0, 0, 0, 0, 0, 0, 0, 0, ++ 5, 4, 5, 6, 7, 8, 7, 8, 0, 0, 0, 0, 0, 0, 0, 0, ++ 6, 5, 6, 6, 7, 8, 8, 8, 0, 0, 0, 0, 0, 0, 0, 0, ++ 7, 6, 7, 7, 8, 8, 8, 9, 0, 0, 0, 0, 0, 0, 0, 0, ++ 8, 7, 8, 8, 8, 9, 8, 9, 0, 0, 0, 0, 0, 0, 0, 0, ++ 8, 7, 7, 8, 8, 9, 9,10, 0, 0, 0, 0, 0, 0, 0, 0, ++ 9, 8, 8, 9, 9, 9, 9,10}; ++ ++ static unsigned char t13l[] = { ++ 1, 4, 6, 7, 8, 9, 9, 10, 9, 10, 11, 11, 12, 12, 13, 13, ++ 3, 4, 6, 7, 8, 8, 9, 9, 9, 9, 10, 10, 11, 12, 12, 12, ++ 6, 6, 7, 8, 9, 9, 10, 10, 9, 10, 10, 11, 11, 12, 13, 13, ++ 7, 7, 8, 9, 9, 10, 10, 10, 10, 11, 11, 11, 11, 12, 13, 13, ++ 8, 7, 9, 9, 10, 10, 11, 11, 10, 11, 11, 12, 12, 13, 13, 14, ++ 9, 8, 9, 10, 10, 10, 11, 11, 11, 11, 12, 11, 13, 13, 14, 14, ++ 9, 9, 10, 10, 11, 11, 11, 11, 11, 12, 12, 12, 13, 13, 14, 14, ++ 10, 9, 10, 11, 11, 11, 12, 12, 12, 12, 13, 13, 13, 14, 16, 16, ++ 9, 8, 9, 10, 10, 11, 11, 12, 12, 12, 12, 13, 13, 14, 15, 15, ++ 10, 9, 10, 10, 11, 11, 11, 13, 12, 13, 13, 14, 14, 14, 16, 15, ++ 10, 10, 10, 11, 11, 12, 12, 13, 12, 13, 14, 13, 14, 15, 16, 17, ++ 11, 10, 10, 11, 12, 12, 12, 12, 13, 13, 13, 14, 15, 15, 15, 16, ++ 11, 11, 11, 12, 12, 13, 12, 13, 14, 14, 15, 15, 15, 16, 16, 16, ++ 12, 11, 12, 13, 13, 13, 14, 14, 14, 14, 14, 15, 16, 15, 16, 16, ++ 13, 12, 12, 13, 13, 13, 15, 14, 14, 17, 15, 15, 15, 17, 16, 16, ++ 12, 12, 13, 14, 14, 14, 15, 14, 15, 15, 16, 16, 19, 18, 19, 16}; ++ ++ static unsigned char t15l[] = { ++ 3, 4, 5, 7, 7, 8, 9, 9, 9, 10, 10, 11, 11, 11, 12, 13, ++ 4, 3, 5, 6, 7, 7, 8, 8, 8, 9, 9, 10, 10, 10, 11, 11, ++ 5, 5, 5, 6, 7, 7, 8, 8, 8, 9, 9, 10, 10, 11, 11, 11, ++ 6, 6, 6, 7, 7, 8, 8, 9, 9, 9, 10, 10, 10, 11, 11, 11, ++ 7, 6, 7, 7, 8, 8, 9, 9, 9, 9, 10, 10, 10, 11, 11, 11, ++ 8, 7, 7, 8, 8, 8, 9, 9, 9, 9, 10, 10, 11, 11, 11, 12, ++ 9, 7, 8, 8, 8, 9, 9, 9, 9, 10, 10, 10, 11, 11, 12, 12, ++ 9, 8, 8, 9, 9, 9, 9, 10, 10, 10, 10, 10, 11, 11, 11, 12, ++ 9, 8, 8, 9, 9, 9, 9, 10, 10, 10, 10, 11, 11, 12, 12, 12, ++ 9, 8, 9, 9, 9, 9, 10, 10, 10, 11, 11, 11, 11, 12, 12, 12, ++ 10, 9, 9, 9, 10, 10, 10, 10, 10, 11, 11, 11, 11, 12, 13, 12, ++ 10, 9, 9, 9, 10, 10, 10, 10, 11, 11, 11, 11, 12, 12, 12, 13, ++ 11, 10, 9, 10, 10, 10, 11, 11, 11, 11, 11, 11, 12, 12, 13, 13, ++ 11, 10, 10, 10, 10, 11, 11, 11, 11, 12, 12, 12, 12, 12, 13, 13, ++ 12, 11, 11, 11, 11, 11, 11, 11, 12, 12, 12, 12, 13, 13, 12, 13, ++ 12, 11, 11, 11, 11, 11, 11, 12, 12, 12, 12, 12, 13, 13, 13, 13}; ++ ++ static unsigned char t16l[] = { ++ 1, 4, 6, 8, 9, 9, 10, 10, 11, 11, 11, 12, 12, 12, 13, 9, ++ 3, 4, 6, 7, 8, 9, 9, 9, 10, 10, 10, 11, 12, 11, 12, 8, ++ 6, 6, 7, 8, 9, 9, 10, 10, 11, 10, 11, 11, 11, 12, 12, 9, ++ 8, 7, 8, 9, 9, 10, 10, 10, 11, 11, 12, 12, 12, 13, 13, 10, ++ 9, 8, 9, 9, 10, 10, 11, 11, 11, 12, 12, 12, 13, 13, 13, 9, ++ 9, 8, 9, 9, 10, 11, 11, 12, 11, 12, 12, 13, 13, 13, 14, 10, ++ 10, 9, 9, 10, 11, 11, 11, 11, 12, 12, 12, 12, 13, 13, 14, 10, ++ 10, 9, 10, 10, 11, 11, 11, 12, 12, 13, 13, 13, 13, 15, 15, 10, ++ 10, 10, 10, 11, 11, 11, 12, 12, 13, 13, 13, 13, 14, 14, 14, 10, ++ 11, 10, 10, 11, 11, 12, 12, 13, 13, 13, 13, 14, 13, 14, 13, 11, ++ 11, 11, 10, 11, 12, 12, 12, 12, 13, 14, 14, 14, 15, 15, 14, 10, ++ 12, 11, 11, 11, 12, 12, 13, 14, 14, 14, 14, 14, 14, 13, 14, 11, ++ 12, 12, 12, 12, 12, 13, 13, 13, 13, 15, 14, 14, 14, 14, 16, 11, ++ 14, 12, 12, 12, 13, 13, 14, 14, 14, 16, 15, 15, 15, 17, 15, 11, ++ 13, 13, 11, 12, 14, 14, 13, 14, 14, 15, 16, 15, 17, 15, 14, 11, ++ 9, 8, 8, 9, 9, 10, 10, 10, 11, 11, 11, 11, 11, 11, 11, 8}; ++ ++ static unsigned char t24l[] = { ++ 4, 4, 6, 7, 8, 9, 9, 10, 10, 11, 11, 11, 11, 11, 12, 9, ++ 4, 4, 5, 6, 7, 8, 8, 9, 9, 9, 10, 10, 10, 10, 10, 8, ++ 6, 5, 6, 7, 7, 8, 8, 9, 9, 9, 9, 10, 10, 10, 11, 7, ++ 7, 6, 7, 7, 8, 8, 8, 9, 9, 9, 9, 10, 10, 10, 10, 7, ++ 8, 7, 7, 8, 8, 8, 8, 9, 9, 9, 10, 10, 10, 10, 11, 7, ++ 9, 7, 8, 8, 8, 8, 9, 9, 9, 9, 10, 10, 10, 10, 10, 7, ++ 9, 8, 8, 8, 8, 9, 9, 9, 9, 10, 10, 10, 10, 10, 11, 7, ++ 10, 8, 8, 8, 9, 9, 9, 9, 10, 10, 10, 10, 10, 11, 11, 8, ++ 10, 9, 9, 9, 9, 9, 9, 9, 9, 10, 10, 10, 10, 11, 11, 8, ++ 10, 9, 9, 9, 9, 9, 9, 10, 10, 10, 10, 10, 11, 11, 11, 8, ++ 11, 9, 9, 9, 9, 10, 10, 10, 10, 10, 10, 11, 11, 11, 11, 8, ++ 11, 10, 9, 9, 9, 10, 10, 10, 10, 10, 10, 11, 11, 11, 11, 8, ++ 11, 10, 10, 10, 10, 10, 10, 10, 10, 10, 11, 11, 11, 11, 11, 8, ++ 11, 10, 10, 10, 10, 10, 10, 10, 11, 11, 11, 11, 11, 11, 11, 8, ++ 12, 10, 10, 10, 10, 10, 10, 11, 11, 11, 11, 11, 11, 11, 11, 8, ++ 8, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 8, 8, 8, 8, 4}; ++ ++ static unsigned char t32l[] = { ++ 1, 4, 4, 5, 4, 6, 5, 6, 4, 5, 5, 6, 5, 6, 6, 6}; ++ static unsigned char t33l[] = { ++ 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4}; ++ ++ struct huffcodetab ht[HTN] = ++ { ++ /* xlen, linmax, table, hlen */ ++ { 0, 0,NULL,NULL}, ++ { 2, 0,t1HB, t1l}, ++ { 3, 0,t2HB, t2l}, ++ { 3, 0,t3HB, t3l}, ++ { 0, 0,NULL,NULL},/* Apparently not used */ ++ { 4, 0,t5HB, t5l}, ++ { 4, 0,t6HB, t6l}, ++ { 6, 0,t7HB, t7l}, ++ { 6, 0,t8HB, t8l}, ++ { 6, 0,t9HB, t9l}, ++ { 8, 0,t10HB, t10l}, ++ { 8, 0,t11HB, t11l}, ++ { 8, 0,t12HB, t12l}, ++ {16, 0,t13HB, t13l}, ++ { 0, 0,NULL,NULL},/* Apparently not used */ ++ {16, 0,t15HB, t15l}, ++ ++ { 1, 1,t16HB, t16l}, ++ { 2, 3,t16HB, t16l}, ++ { 3, 7,t16HB, t16l}, ++ { 4, 15,t16HB, t16l}, ++ { 6, 63,t16HB, t16l}, ++ { 8, 255,t16HB, t16l}, ++ {10,1023,t16HB, t16l}, ++ {13,8191,t16HB, t16l}, ++ ++ { 4, 15,t24HB, t24l}, ++ { 5, 31,t24HB, t24l}, ++ { 6, 63,t24HB, t24l}, ++ { 7, 127,t24HB, t24l}, ++ { 8, 255,t24HB, t24l}, ++ { 9, 511,t24HB, t24l}, ++ {11,2047,t24HB, t24l}, ++ {13,8191,t24HB, t24l}, ++ ++ { 0, 0,t32HB, t32l}, ++ { 0, 0,t33HB, t33l}, ++ }; ++ ++ ++ ++ ++ ++ FLOAT8 psy_data[] = ++ {48000.0, 61, ++ 0, 1, 24.5, 4.532, 0.970, 0.000, ++ 1, 1, 24.5, 4.532, 0.755, 0.469, ++ 2, 1, 24.5, 4.532, 0.738, 0.938, ++ 3, 1, 24.5, 0.904, 0.730, 1.406, ++ 4, 1, 24.5, 0.904, 0.724, 1.875, ++ 5, 1, 20.0, 0.090, 0.723, 2.344, ++ 6, 1, 20.0, 0.090, 0.723, 2.813, ++ 7, 1, 20.0, 0.029, 0.723, 3.281, ++ 8, 1, 20.0, 0.029, 0.718, 3.750, ++ 9, 1, 20.0, 0.009, 0.690, 4.199, ++ 10, 1, 20.0, 0.009, 0.660, 4.625, ++ 11, 1, 18.0, 0.009, 0.641, 5.047, ++ 12, 1, 18.0, 0.009, 0.600, 5.438, ++ 13, 1, 18.0, 0.009, 0.584, 5.828, ++ 14, 1, 12.0, 0.009, 0.532, 6.188, ++ 15, 1, 12.0, 0.009, 0.537, 6.522, ++ 16, 2, 6.0, 0.018, 0.857, 7.174, ++ 17, 2, 6.0, 0.018, 0.858, 7.801, ++ 18, 2, 3.0, 0.018, 0.853, 8.402, ++ 19, 2, 3.0, 0.018, 0.824, 8.966, ++ 20, 2, 3.0, 0.018, 0.778, 9.484, ++ 21, 2, 3.0, 0.018, 0.740, 9.966, ++ 22, 2, 0.0, 0.018, 0.709, 10.426, ++ 23, 2, 0.0, 0.018, 0.676, 10.866, ++ 24, 2, 0.0, 0.018, 0.632, 11.279, ++ 25, 2, 0.0, 0.018, 0.592, 11.669, ++ 26, 2, 0.0, 0.018, 0.553, 12.042, ++ 27, 2, 0.0, 0.018, 0.510, 12.386, ++ 28, 2, 0.0, 0.018, 0.513, 12.721, ++ 29, 3, 0.0, 0.027, 0.608, 13.115, ++ 30, 3, 0.0, 0.027, 0.673, 13.562, ++ 31, 3, 0.0, 0.027, 0.637, 13.984, ++ 32, 3, 0.0, 0.027, 0.586, 14.371, ++ 33, 3, 0.0, 0.027, 0.571, 14.741, ++ 34, 4, 0.0, 0.036, 0.616, 15.140, ++ 35, 4, 0.0, 0.036, 0.640, 15.563, ++ 36, 4, 0.0, 0.036, 0.598, 15.962, ++ 37, 4, 0.0, 0.036, 0.538, 16.324, ++ 38, 4, 0.0, 0.036, 0.512, 16.665, ++ 39, 5, 0.0, 0.045, 0.528, 17.020, ++ 40, 5, 0.0, 0.045, 0.517, 17.373, ++ 41, 5, 0.0, 0.045, 0.493, 17.708, ++ 42, 6, 0.0, 0.054, 0.499, 18.045, ++ 43, 7, 0.0, 0.063, 0.525, 18.398, ++ 44, 7, 0.0, 0.063, 0.541, 18.762, ++ 45, 8, 0.0, 0.072, 0.528, 19.120, ++ 46, 8, 0.0, 0.072, 0.510, 19.466, ++ 47, 8, 0.0, 0.072, 0.506, 19.807, ++ 48, 10, 0.0, 0.180, 0.525, 20.159, ++ 49, 10, 0.0, 0.180, 0.536, 20.522, ++ 50, 10, 0.0, 0.180, 0.518, 20.874, ++ 51, 13, 0.0, 0.372, 0.501, 21.214, ++ 52, 13, 0.0, 0.372, 0.497, 21.553, ++ 53, 14, 0.0, 0.400, 0.497, 21.892, ++ 54, 18, 0.0, 1.627, 0.495, 22.231, ++ 55, 18, 0.0, 1.627, 0.494, 22.569, ++ 56, 20, 0.0, 1.808, 0.497, 22.909, ++ 57, 25, 0.0, 22.607, 0.494, 23.248, ++ 58, 25, 0.0, 22.607, 0.487, 23.583, ++ 59, 35, 0.0, 31.650, 0.483, 23.915, ++ 60, 67, 0.0, 605.867, 0.482, 24.246, ++ 61, 67, 0.0, 605.867, 0.524, 24.576, ++ 44100.0, 62, ++ 0, 1, 24.5, 4.532, 0.951, 0.000, ++ 1, 1, 24.5, 4.532, 0.700, 0.431, ++ 2, 1, 24.5, 4.532, 0.681, 0.861, ++ 3, 1, 24.5, 0.904, 0.675, 1.292, ++ 4, 1, 24.5, 0.904, 0.667, 1.723, ++ 5, 1, 20.0, 0.090, 0.665, 2.153, ++ 6, 1, 20.0, 0.090, 0.664, 2.584, ++ 7, 1, 20.0, 0.029, 0.664, 3.015, ++ 8, 1, 20.0, 0.029, 0.664, 3.445, ++ 9, 1, 20.0, 0.029, 0.655, 3.876, ++ 10, 1, 20.0, 0.009, 0.616, 4.279, ++ 11, 1, 20.0, 0.009, 0.597, 4.670, ++ 12, 1, 18.0, 0.009, 0.578, 5.057, ++ 13, 1, 18.0, 0.009, 0.541, 5.416, ++ 14, 1, 18.0, 0.009, 0.575, 5.774, ++ 15, 2, 12.0, 0.018, 0.856, 6.422, ++ 16, 2, 6.0, 0.018, 0.846, 7.026, ++ 17, 2, 6.0, 0.018, 0.840, 7.609, ++ 18, 2, 3.0, 0.018, 0.822, 8.168, ++ 19, 2, 3.0, 0.018, 0.800, 8.710, ++ 20, 2, 3.0, 0.018, 0.753, 9.207, ++ 21, 2, 3.0, 0.018, 0.704, 9.662, ++ 22, 2, 0.0, 0.018, 0.674, 10.099, ++ 23, 2, 0.0, 0.018, 0.640, 10.515, ++ 24, 2, 0.0, 0.018, 0.609, 10.917, ++ 25, 2, 0.0, 0.018, 0.566, 11.293, ++ 26, 2, 0.0, 0.018, 0.535, 11.652, ++ 27, 2, 0.0, 0.018, 0.531, 11.997, ++ 28, 3, 0.0, 0.027, 0.615, 12.394, ++ 29, 3, 0.0, 0.027, 0.686, 12.850, ++ 30, 3, 0.0, 0.027, 0.650, 13.277, ++ 31, 3, 0.0, 0.027, 0.612, 13.681, ++ 32, 3, 0.0, 0.027, 0.567, 14.062, ++ 33, 3, 0.0, 0.027, 0.520, 14.411, ++ 34, 3, 0.0, 0.027, 0.513, 14.751, ++ 35, 4, 0.0, 0.036, 0.557, 15.119, ++ 36, 4, 0.0, 0.036, 0.584, 15.508, ++ 37, 4, 0.0, 0.036, 0.570, 15.883, ++ 38, 5, 0.0, 0.045, 0.579, 16.263, ++ 39, 5, 0.0, 0.045, 0.585, 16.654, ++ 40, 5, 0.0, 0.045, 0.548, 17.020, ++ 41, 6, 0.0, 0.054, 0.536, 17.374, ++ 42, 6, 0.0, 0.054, 0.550, 17.744, ++ 43, 7, 0.0, 0.063, 0.532, 18.104, ++ 44, 7, 0.0, 0.063, 0.504, 18.447, ++ 45, 7, 0.0, 0.063, 0.496, 18.782, ++ 46, 9, 0.0, 0.081, 0.517, 19.130, ++ 47, 9, 0.0, 0.081, 0.527, 19.487, ++ 48, 9, 0.0, 0.081, 0.516, 19.838, ++ 49, 10, 0.0, 0.180, 0.497, 20.179, ++ 50, 10, 0.0, 0.180, 0.489, 20.510, ++ 51, 11, 0.0, 0.198, 0.502, 20.852, ++ 52, 14, 0.0, 0.400, 0.501, 21.196, ++ 53, 14, 0.0, 0.400, 0.491, 21.531, ++ 54, 15, 0.0, 0.429, 0.497, 21.870, ++ 55, 20, 0.0, 1.808, 0.504, 22.214, ++ 56, 20, 0.0, 1.808, 0.504, 22.558, ++ 57, 21, 0.0, 1.898, 0.495, 22.898, ++ 58, 27, 0.0, 24.416, 0.486, 23.232, ++ 59, 27, 0.0, 24.416, 0.484, 23.564, ++ 60, 36, 0.0, 32.554, 0.483, 23.897, ++ 61, 73, 0.0, 660.124, 0.475, 24.229, ++ 62, 18, 0.0, 162.770, 0.515, 24.442, ++ 32000.0, 58, ++ 0, 2, 24.5, 4.532, 0.997, 0.313, ++ 1, 2, 24.5, 4.532, 0.893, 0.938, ++ 2, 2, 24.5, 1.809, 0.881, 1.563, ++ 3, 2, 20.0, 0.181, 0.873, 2.188, ++ 4, 2, 20.0, 0.181, 0.872, 2.813, ++ 5, 2, 20.0, 0.057, 0.871, 3.438, ++ 6, 2, 20.0, 0.018, 0.860, 4.045, ++ 7, 2, 20.0, 0.018, 0.839, 4.625, ++ 8, 2, 18.0, 0.018, 0.812, 5.173, ++ 9, 2, 18.0, 0.018, 0.784, 5.698, ++ 10, 2, 12.0, 0.018, 0.741, 6.185, ++ 11, 2, 12.0, 0.018, 0.697, 6.634, ++ 12, 2, 6.0, 0.018, 0.674, 7.070, ++ 13, 2, 6.0, 0.018, 0.651, 7.492, ++ 14, 2, 6.0, 0.018, 0.633, 7.905, ++ 15, 2, 3.0, 0.018, 0.611, 8.305, ++ 16, 2, 3.0, 0.018, 0.589, 8.695, ++ 17, 2, 3.0, 0.018, 0.575, 9.064, ++ 18, 3, 3.0, 0.027, 0.654, 9.484, ++ 19, 3, 3.0, 0.027, 0.724, 9.966, ++ 20, 3, 0.0, 0.027, 0.701, 10.426, ++ 21, 3, 0.0, 0.027, 0.673, 10.866, ++ 22, 3, 0.0, 0.027, 0.631, 11.279, ++ 23, 3, 0.0, 0.027, 0.592, 11.669, ++ 24, 3, 0.0, 0.027, 0.553, 12.042, ++ 25, 3, 0.0, 0.027, 0.510, 12.386, ++ 26, 3, 0.0, 0.027, 0.506, 12.721, ++ 27, 4, 0.0, 0.036, 0.562, 13.091, ++ 28, 4, 0.0, 0.036, 0.598, 13.488, ++ 29, 4, 0.0, 0.036, 0.589, 13.873, ++ 30, 5, 0.0, 0.045, 0.607, 14.268, ++ 31, 5, 0.0, 0.045, 0.620, 14.679, ++ 32, 5, 0.0, 0.045, 0.580, 15.067, ++ 33, 5, 0.0, 0.045, 0.532, 15.424, ++ 34, 5, 0.0, 0.045, 0.517, 15.771, ++ 35, 6, 0.0, 0.054, 0.517, 16.120, ++ 36, 6, 0.0, 0.054, 0.509, 16.466, ++ 37, 6, 0.0, 0.054, 0.506, 16.807, ++ 38, 8, 0.0, 0.072, 0.522, 17.158, ++ 39, 8, 0.0, 0.072, 0.531, 17.518, ++ 40, 8, 0.0, 0.072, 0.519, 17.869, ++ 41, 10, 0.0, 0.090, 0.512, 18.215, ++ 42, 10, 0.0, 0.090, 0.509, 18.563, ++ 43, 10, 0.0, 0.090, 0.498, 18.902, ++ 44, 12, 0.0, 0.109, 0.494, 19.239, ++ 45, 12, 0.0, 0.109, 0.501, 19.580, ++ 46, 13, 0.0, 0.118, 0.508, 19.925, ++ 47, 14, 0.0, 0.252, 0.502, 20.269, ++ 48, 14, 0.0, 0.252, 0.493, 20.606, ++ 49, 16, 0.0, 0.288, 0.497, 20.944, ++ 50, 20, 0.0, 0.572, 0.506, 21.288, ++ 51, 20, 0.0, 0.572, 0.510, 21.635, ++ 52, 23, 0.0, 0.658, 0.504, 21.980, ++ 53, 27, 0.0, 2.441, 0.496, 22.319, ++ 54, 27, 0.0, 2.441, 0.493, 22.656, ++ 55, 32, 0.0, 2.893, 0.490, 22.993, ++ 56, 37, 0.0, 33.458, 0.482, 23.326, ++ 57, 37, 0.0, 33.458, 0.458, 23.656, ++ 58, 12, 0.0, 10.851, 0.500, 23.937, ++ 24000, 59, ++ 0, 2, 15, 17.8250179, 0.697374165, 0.236874461, ++ 1, 2, 15, 17.8250179, 0.455024809, 0.71016103, ++ 2, 2, 15, 1.78250182, 0.431440443, 1.18193281, ++ 3, 2, 15, 1.78250182, 0.42391625, 1.65102732, ++ 4, 2, 13, 0.178250194, 0.418206781, 2.11632562, ++ 5, 2, 13, 0.178250194, 0.41158545, 2.57676744, ++ 6, 2, 13, 0.0563676581, 0.405409157, 3.03136396, ++ 7, 2, 13, 0.0563676581, 0.399695486, 3.47920918, ++ 8, 2, 13, 0.0563676581, 0.393753231, 3.91948748, ++ 9, 2, 12, 0.0178250186, 0.387357473, 4.35147953, ++ 10, 2, 12, 0.0178250186, 0.38045457, 4.77456427, ++ 11, 2, 10, 0.0178250186, 0.373053908, 5.18822002, ++ 12, 2, 10, 0.0178250186, 0.365188122, 5.59202194, ++ 13, 2, 10, 0.0178250186, 0.356897771, 5.98564005, ++ 14, 2, 9, 0.0178250186, 0.348700613, 6.36883163, ++ 15, 2, 9, 0.0178250186, 0.340260029, 6.74143791, ++ 16, 2, 6, 0.0178250186, 0.332341045, 7.10337448, ++ 17, 2, 6, 0.0178250186, 0.330462843, 7.45462418, ++ 18, 2, 6, 0.0178250186, 0.345568359, 7.79523182, ++ 19, 3, 3, 0.0267375279, 0.377859652, 8.20455742, ++ 20, 3, 3, 0.0267375279, 0.396689415, 8.67640114, ++ 21, 3, 3, 0.0267375279, 0.391237885, 9.12561035, ++ 22, 3, 3, 0.0267375279, 0.37761277, 9.55298138, ++ 23, 3, 3, 0.0267375279, 0.362836808, 9.95940971, ++ 24, 3, 0, 0.0267375279, 0.349010617, 10.3458519, ++ 25, 3, 0, 0.0267375279, 0.339673489, 10.7132998, ++ 26, 3, 0, 0.0267375279, 0.343845725, 11.0627575, ++ 27, 4, 0, 0.0356500372, 0.355822682, 11.447506, ++ 28, 4, 0, 0.0356500372, 0.358104348, 11.8627586, ++ 29, 4, 0, 0.0356500372, 0.34745428, 12.2520256, ++ 30, 4, 0, 0.0356500372, 0.334927917, 12.6173973, ++ 31, 4, 0, 0.0356500372, 0.331643254, 12.9608269, ++ 32, 5, 0, 0.0445625484, 0.333368897, 13.3219252, ++ 33, 5, 0, 0.0445625484, 0.332313001, 13.6976833, ++ 34, 5, 0, 0.0445625484, 0.3314417, 14.047802, ++ 35, 6, 0, 0.0534750558, 0.330947191, 14.405302, ++ 36, 6, 0, 0.0534750558, 0.332477689, 14.7684803, ++ 37, 7, 0, 0.062387567, 0.332647532, 15.1315956, ++ 38, 7, 0, 0.062387567, 0.330841452, 15.4940481, ++ 39, 8, 0, 0.0713000745, 0.327769846, 15.8516159, ++ 40, 8, 0, 0.0713000745, 0.324572712, 16.204628, ++ 41, 9, 0, 0.0802125856, 0.323825002, 16.5502281, ++ 42, 10, 0, 0.0891250968, 0.321414798, 16.9067478, ++ 43, 10, 0, 0.0891250968, 0.318189293, 17.2537231, ++ 44, 11, 0, 0.0980376005, 0.315934151, 17.5901108, ++ 45, 12, 0, 0.106950112, 0.315639287, 17.931406, ++ 46, 13, 0, 0.115862623, 0.316569835, 18.2750721, ++ 47, 14, 0, 0.124775134, 0.31656, 18.6191597, ++ 48, 15, 0, 0.133687645, 0.315465957, 18.9621754, ++ 49, 16, 0, 0.142600149, 0.313576341, 19.3029613, ++ 50, 17, 0, 0.151512653, 0.311635971, 19.6405869, ++ 51, 18, 0, 0.160425171, 0.311066717, 19.9742699, ++ 52, 20, 0, 0.355655879, 0.311465651, 20.3115921, ++ 53, 21, 0, 0.373438686, 0.311872005, 20.6507797, ++ 54, 23, 0, 0.409004271, 0.311015964, 20.9890823, ++ 55, 24, 0, 0.676411927, 0.309207708, 21.3251152, ++ 56, 26, 0, 0.732779562, 0.3081128, 21.6565971, ++ 57, 28, 0, 0.789147198, 0.310006589, 21.9881554, ++ 58, 31, 0, 2.76287794, 0.327113092, 22.3222847, ++ 59, 34, 0, 3.03025317, 0.416082352, 22.6605186, ++ 22050, 59, ++ 0, 2, 15, 17.8250179, 0.658683598, 0.217637643, ++ 1, 2, 15, 17.8250179, 0.432554901, 0.652563453, ++ 2, 2, 15, 1.78250182, 0.405113578, 1.08633137, ++ 3, 2, 15, 1.78250182, 0.397231787, 1.51803517, ++ 4, 2, 15, 1.78250182, 0.392088681, 1.94679713, ++ 5, 2, 13, 0.178250194, 0.386788279, 2.37177849, ++ 6, 2, 13, 0.178250194, 0.380574644, 2.79218864, ++ 7, 2, 13, 0.0563676581, 0.375309765, 3.20729256, ++ 8, 2, 13, 0.0563676581, 0.370087624, 3.61641645, ++ 9, 2, 12, 0.0178250186, 0.364568561, 4.01895428, ++ 10, 2, 12, 0.0178250186, 0.358959526, 4.4143672, ++ 11, 2, 12, 0.0178250186, 0.352938265, 4.80218887, ++ 12, 2, 10, 0.0178250186, 0.3465029, 5.18202305, ++ 13, 2, 10, 0.0178250186, 0.33968094, 5.55354261, ++ 14, 2, 10, 0.0178250186, 0.332571507, 5.91648674, ++ 15, 2, 9, 0.0178250186, 0.326015651, 6.27065945, ++ 16, 2, 9, 0.0178250186, 0.325442046, 6.61592293, ++ 17, 2, 9, 0.0178250186, 0.341315031, 6.95219517, ++ 18, 3, 6, 0.0267375279, 0.374984443, 7.3584404, ++ 19, 3, 6, 0.0267375279, 0.396138102, 7.8290925, ++ 20, 3, 3, 0.0267375279, 0.39271906, 8.27975655, ++ 21, 3, 3, 0.0267375279, 0.380755007, 8.71083069, ++ 22, 3, 3, 0.0267375279, 0.367386311, 9.12284088, ++ 23, 3, 3, 0.0267375279, 0.354351997, 9.51640987, ++ 24, 3, 3, 0.0267375279, 0.341508389, 9.89222908, ++ 25, 3, 0, 0.0267375279, 0.333577901, 10.2510386, ++ 26, 3, 0, 0.0267375279, 0.338108748, 10.5936022, ++ 27, 4, 0, 0.0356500372, 0.350744486, 10.9723492, ++ 28, 4, 0, 0.0356500372, 0.354519457, 11.38272, ++ 29, 4, 0, 0.0356500372, 0.345274031, 11.7689981, ++ 30, 4, 0, 0.0356500372, 0.333828837, 12.1329184, ++ 31, 4, 0, 0.0356500372, 0.331436664, 12.4761295, ++ 32, 5, 0, 0.0445625484, 0.334172577, 12.8381901, ++ 33, 5, 0, 0.0445625484, 0.334024847, 13.2160273, ++ 34, 5, 0, 0.0445625484, 0.33392629, 13.5690479, ++ 35, 6, 0, 0.0534750558, 0.334218502, 13.9303951, ++ 36, 6, 0, 0.0534750558, 0.336405039, 14.298193, ++ 37, 7, 0, 0.062387567, 0.337080389, 14.666563, ++ 38, 7, 0, 0.062387567, 0.335603535, 15.0346909, ++ 39, 8, 0, 0.0713000745, 0.332515866, 15.398139, ++ 40, 8, 0, 0.0713000745, 0.327727586, 15.7570457, ++ 41, 9, 0, 0.0802125856, 0.322346836, 16.1083431, ++ 42, 9, 0, 0.0802125856, 0.317575186, 16.4528522, ++ 43, 10, 0, 0.0891250968, 0.31632933, 16.7886105, ++ 44, 11, 0, 0.0980376005, 0.317602783, 17.132, ++ 45, 12, 0, 0.106950112, 0.319945186, 17.4796028, ++ 46, 13, 0, 0.115862623, 0.320881754, 17.8287659, ++ 47, 14, 0, 0.124775134, 0.320346534, 18.1774921, ++ 48, 15, 0, 0.133687645, 0.318628669, 18.5243168, ++ 49, 16, 0, 0.142600149, 0.316125751, 18.8681736, ++ 50, 17, 0, 0.151512653, 0.313746184, 19.2082729, ++ 51, 18, 0, 0.160425171, 0.312971771, 19.5440025, ++ 52, 20, 0, 0.178250194, 0.313278913, 19.8831882, ++ 53, 21, 0, 0.373438686, 0.313735574, 20.224247, ++ 54, 23, 0, 0.409004271, 0.31308493, 20.5646286, ++ 55, 24, 0, 0.426787049, 0.31156227, 20.903141, ++ 56, 26, 0, 0.732779562, 0.310435742, 21.2376747, ++ 57, 28, 0, 0.789147198, 0.31132248, 21.5730591, ++ 58, 30, 0, 0.845514894, 0.32730341, 21.9066811, ++ 59, 33, 0, 2.94112802, 0.414659739, 22.2411156, ++ 16000, 55, ++ 0, 3, 15, 26.7375278, 0.697374165, 0.236874461, ++ 1, 3, 15, 26.7375278, 0.455024809, 0.71016103, ++ 2, 3, 15, 2.67375278, 0.431440443, 1.18193281, ++ 3, 3, 15, 2.67375278, 0.42391625, 1.65102732, ++ 4, 3, 13, 0.26737529, 0.418206781, 2.11632562, ++ 5, 3, 13, 0.26737529, 0.41158545, 2.57676744, ++ 6, 3, 13, 0.0845514908, 0.405409157, 3.03136396, ++ 7, 3, 13, 0.0845514908, 0.399695486, 3.47920918, ++ 8, 3, 13, 0.0845514908, 0.393753231, 3.91948748, ++ 9, 3, 12, 0.0267375279, 0.387357473, 4.35147953, ++ 10, 3, 12, 0.0267375279, 0.38045457, 4.77456427, ++ 11, 3, 10, 0.0267375279, 0.373053908, 5.18822002, ++ 12, 3, 10, 0.0267375279, 0.365188122, 5.59202194, ++ 13, 3, 10, 0.0267375279, 0.356897742, 5.98564005, ++ 14, 3, 9, 0.0267375279, 0.34869957, 6.36883163, ++ 15, 3, 9, 0.0267375279, 0.340241522, 6.74143791, ++ 16, 3, 6, 0.0267375279, 0.332089454, 7.10337448, ++ 17, 3, 6, 0.0267375279, 0.328292668, 7.45462418, ++ 18, 3, 6, 0.0267375279, 0.336574793, 7.79523182, ++ 19, 4, 3, 0.0356500372, 0.354600489, 8.17827797, ++ 20, 4, 3, 0.0356500372, 0.364343345, 8.59994984, ++ 21, 4, 3, 0.0356500372, 0.359369367, 9.00363636, ++ 22, 4, 3, 0.0356500372, 0.347775847, 9.38988018, ++ 23, 4, 3, 0.0356500372, 0.335562587, 9.7592926, ++ 24, 4, 0, 0.0356500372, 0.326988578, 10.1125278, ++ 25, 4, 0, 0.0356500372, 0.327966213, 10.4502735, ++ 26, 5, 0, 0.0445625484, 0.334450752, 10.811614, ++ 27, 5, 0, 0.0445625484, 0.335228145, 11.1935263, ++ 28, 5, 0, 0.0445625484, 0.329595625, 11.5549288, ++ 29, 5, 0, 0.0445625484, 0.326683223, 11.8971443, ++ 30, 6, 0, 0.0534750558, 0.326986551, 12.2520256, ++ 31, 6, 0, 0.0534750558, 0.325072199, 12.6173973, ++ 32, 6, 0, 0.0534750558, 0.323560268, 12.9608269, ++ 33, 7, 0, 0.062387567, 0.322494298, 13.3093863, ++ 34, 7, 0, 0.062387567, 0.323403448, 13.6617231, ++ 35, 8, 0, 0.0713000745, 0.323232353, 14.0134668, ++ 36, 8, 0, 0.0713000745, 0.322662383, 14.3639784, ++ 37, 9, 0, 0.0802125856, 0.324054241, 14.7098465, ++ 38, 10, 0, 0.0891250968, 0.323228806, 15.0686541, ++ 39, 10, 0, 0.0891250968, 0.320751846, 15.4191036, ++ 40, 11, 0, 0.0980376005, 0.318823338, 15.7594051, ++ 41, 12, 0, 0.106950112, 0.318418682, 16.104557, ++ 42, 13, 0, 0.115862623, 0.318762124, 16.451416, ++ 43, 14, 0, 0.124775134, 0.317806393, 16.7975388, ++ 44, 15, 0, 0.133687645, 0.315653771, 17.1411018, ++ 45, 16, 0, 0.142600149, 0.313369036, 17.4808159, ++ 46, 17, 0, 0.151512653, 0.312513858, 17.8158207, ++ 47, 19, 0, 0.169337675, 0.312785119, 18.1543369, ++ 48, 20, 0, 0.178250194, 0.31343773, 18.4948578, ++ 49, 22, 0, 0.196075201, 0.313258767, 18.8350143, ++ 50, 23, 0, 0.20498772, 0.312570423, 19.1740704, ++ 51, 25, 0, 0.222812727, 0.312572777, 19.5104179, ++ 52, 27, 0, 0.240637749, 0.313047856, 19.8497677, ++ 53, 29, 0, 0.515701056, 0.315029174, 20.1900635, ++ 54, 31, 0, 0.551266611, 0.330613613, 20.5294952, ++ 55, 33, 0, 0.586832225, 0.41819948, 20.8664398, ++ 48000.0, 37, ++ 0, 1, 4.532, 1.000, -8.240, 0.000, ++ 1, 1, 0.904, 0.989, -8.240, 1.875, ++ 2, 1, 0.029, 0.989, -8.240, 3.750, ++ 3, 1, 0.009, 0.981, -8.240, 5.438, ++ 4, 1, 0.009, 0.985, -8.240, 6.857, ++ 5, 1, 0.009, 0.984, -8.240, 8.109, ++ 6, 1, 0.009, 0.980, -8.240, 9.237, ++ 7, 1, 0.009, 0.968, -8.240, 10.202, ++ 8, 1, 0.009, 0.954, -8.240, 11.083, ++ 9, 1, 0.009, 0.929, -8.240, 11.865, ++ 10, 1, 0.009, 0.906, -7.447, 12.554, ++ 11, 1, 0.009, 0.883, -7.447, 13.195, ++ 12, 1, 0.009, 0.844, -7.447, 13.781, ++ 13, 1, 0.009, 0.792, -7.447, 14.309, ++ 14, 1, 0.009, 0.747, -7.447, 14.803, ++ 15, 1, 0.009, 0.689, -7.447, 15.250, ++ 16, 1, 0.009, 0.644, -7.447, 15.667, ++ 17, 1, 0.009, 0.592, -7.447, 16.068, ++ 18, 1, 0.009, 0.553, -7.447, 16.409, ++ 19, 2, 0.018, 0.850, -7.447, 17.045, ++ 20, 2, 0.018, 0.811, -6.990, 17.607, ++ 21, 2, 0.018, 0.736, -6.990, 18.097, ++ 22, 2, 0.018, 0.665, -6.990, 18.528, ++ 23, 2, 0.018, 0.610, -6.990, 18.931, ++ 24, 2, 0.018, 0.544, -6.990, 19.295, ++ 25, 2, 0.018, 0.528, -6.990, 19.636, ++ 26, 3, 0.054, 0.621, -6.990, 20.038, ++ 27, 3, 0.054, 0.673, -6.990, 20.486, ++ 28, 3, 0.054, 0.635, -6.990, 20.900, ++ 29, 4, 0.114, 0.626, -6.990, 21.306, ++ 30, 4, 0.114, 0.636, -6.020, 21.722, ++ 31, 5, 0.452, 0.615, -6.020, 22.128, ++ 32, 5, 0.452, 0.579, -6.020, 22.513, ++ 33, 5, 0.452, 0.551, -6.020, 22.877, ++ 34, 7, 6.330, 0.552, -5.229, 23.241, ++ 35, 7, 6.330, 0.559, -5.229, 23.616, ++ 36, 11, 9.947, 0.528, -5.229, 23.974, ++ 37, 17, 153.727, 0.479, -5.229, 24.313, ++ 44100.0, 38, ++ 0, 1, 4.532, 1.000, -8.240, 0.000, ++ 1, 1, 0.904, 0.983, -8.240, 1.723, ++ 2, 1, 0.029, 0.983, -8.240, 3.445, ++ 3, 1, 0.009, 0.982, -8.240, 5.057, ++ 4, 1, 0.009, 0.985, -8.240, 6.422, ++ 5, 1, 0.009, 0.983, -8.240, 7.609, ++ 6, 1, 0.009, 0.978, -8.240, 8.710, ++ 7, 1, 0.009, 0.967, -8.240, 9.662, ++ 8, 1, 0.009, 0.948, -8.240, 10.515, ++ 9, 1, 0.009, 0.930, -8.240, 11.293, ++ 10, 1, 0.009, 0.914, -7.447, 12.009, ++ 11, 1, 0.009, 0.870, -7.447, 12.625, ++ 12, 1, 0.009, 0.845, -7.447, 13.210, ++ 13, 1, 0.009, 0.800, -7.447, 13.748, ++ 14, 1, 0.009, 0.749, -7.447, 14.241, ++ 15, 1, 0.009, 0.701, -7.447, 14.695, ++ 16, 1, 0.009, 0.653, -7.447, 15.125, ++ 17, 1, 0.009, 0.590, -7.447, 15.508, ++ 18, 1, 0.009, 0.616, -7.447, 15.891, ++ 19, 2, 0.018, 0.860, -7.447, 16.537, ++ 20, 2, 0.018, 0.823, -6.990, 17.112, ++ 21, 2, 0.018, 0.762, -6.990, 17.621, ++ 22, 2, 0.018, 0.688, -6.990, 18.073, ++ 23, 2, 0.018, 0.612, -6.990, 18.470, ++ 24, 2, 0.018, 0.594, -6.990, 18.849, ++ 25, 3, 0.027, 0.658, -6.990, 19.271, ++ 26, 3, 0.027, 0.706, -6.990, 19.741, ++ 27, 3, 0.054, 0.660, -6.990, 20.177, ++ 28, 3, 0.054, 0.606, -6.990, 20.576, ++ 29, 3, 0.054, 0.565, -6.990, 20.950, ++ 30, 4, 0.114, 0.560, -6.020, 21.316, ++ 31, 4, 0.114, 0.579, -6.020, 21.699, ++ 32, 5, 0.452, 0.567, -6.020, 22.078, ++ 33, 5, 0.452, 0.534, -6.020, 22.438, ++ 34, 5, 0.452, 0.514, -5.229, 22.782, ++ 35, 7, 6.330, 0.520, -5.229, 23.133, ++ 36, 7, 6.330, 0.518, -5.229, 23.484, ++ 37, 7, 6.330, 0.507, -5.229, 23.828, ++ 38, 19, 171.813, 0.447, -4.559, 24.173, ++ 32000.0, 41, ++ 0, 1, 4.532, 1.000, -8.240, 0.000, ++ 1, 1, 0.904, 0.985, -8.240, 1.250, ++ 2, 1, 0.090, 0.983, -8.240, 2.500, ++ 3, 1, 0.029, 0.983, -8.240, 3.750, ++ 4, 1, 0.009, 0.981, -8.240, 4.909, ++ 5, 1, 0.009, 0.975, -8.240, 5.958, ++ 6, 1, 0.009, 0.959, -8.240, 6.857, ++ 7, 1, 0.009, 0.944, -8.240, 7.700, ++ 8, 1, 0.009, 0.933, -8.240, 8.500, ++ 9, 1, 0.009, 0.920, -8.240, 9.237, ++ 10, 1, 0.009, 0.892, -7.447, 9.895, ++ 11, 1, 0.009, 0.863, -7.447, 10.500, ++ 12, 1, 0.009, 0.839, -7.447, 11.083, ++ 13, 1, 0.009, 0.786, -7.447, 11.604, ++ 14, 1, 0.009, 0.755, -7.447, 12.107, ++ 15, 1, 0.009, 0.698, -7.447, 12.554, ++ 16, 1, 0.009, 0.673, -7.447, 13.000, ++ 17, 1, 0.009, 0.605, -7.447, 13.391, ++ 18, 1, 0.009, 0.629, -7.447, 13.781, ++ 19, 2, 0.018, 0.883, -7.447, 14.474, ++ 20, 2, 0.018, 0.858, -6.990, 15.096, ++ 21, 2, 0.018, 0.829, -6.990, 15.667, ++ 22, 2, 0.018, 0.767, -6.990, 16.177, ++ 23, 2, 0.018, 0.705, -6.990, 16.636, ++ 24, 2, 0.018, 0.637, -6.990, 17.057, ++ 25, 2, 0.018, 0.564, -6.990, 17.429, ++ 26, 2, 0.018, 0.550, -6.990, 17.786, ++ 27, 3, 0.027, 0.603, -6.990, 18.177, ++ 28, 3, 0.027, 0.635, -6.990, 18.597, ++ 29, 3, 0.027, 0.592, -6.990, 18.994, ++ 30, 3, 0.027, 0.533, -6.020, 19.352, ++ 31, 3, 0.027, 0.518, -6.020, 19.693, ++ 32, 4, 0.072, 0.568, -6.020, 20.066, ++ 33, 4, 0.072, 0.594, -6.020, 20.462, ++ 34, 4, 0.072, 0.568, -5.229, 20.841, ++ 35, 5, 0.143, 0.536, -5.229, 21.201, ++ 36, 5, 0.143, 0.522, -5.229, 21.549, ++ 37, 6, 0.172, 0.542, -5.229, 21.911, ++ 38, 7, 0.633, 0.539, -4.559, 22.275, ++ 39, 7, 0.633, 0.519, -4.559, 22.625, ++ 40, 8, 0.723, 0.514, -3.980, 22.971, ++ 41, 10, 9.043, 0.518, -3.980, 23.321, ++ 24000, 44, ++ 0, 1, 8.91250896, 0.971850038, 0.150000006, 0, ++ 1, 1, 8.91250896, 0.874727964, 0.150000006, 0.946573138, ++ 2, 1, 0.891250908, 0.85779953, 0.150000006, 1.88476217, ++ 3, 1, 0.0891250968, 0.839743853, 0.150000006, 2.8056457, ++ 4, 1, 0.028183829, 0.82260257, 0.150000006, 3.70133615, ++ 5, 1, 0.00891250931, 0.80018574, 0.150000006, 4.56532001, ++ 6, 1, 0.00891250931, 0.771475196, 0.150000006, 5.39263105, ++ 7, 1, 0.00891250931, 0.737389982, 0.150000006, 6.17986727, ++ 8, 1, 0.00891250931, 0.701111019, 0.150000006, 6.92507982, ++ 9, 1, 0.00891250931, 0.65977633, 0.150000006, 7.62757969, ++ 10, 1, 0.00891250931, 0.615037441, 0.150000006, 8.28770351, ++ 11, 1, 0.00891250931, 0.568658054, 0.150000006, 8.90657234, ++ 12, 1, 0.00891250931, 0.522260666, 0.180000007, 9.48587132, ++ 13, 1, 0.00891250931, 0.478903115, 0.180000007, 10.0276566, ++ 14, 1, 0.00891250931, 0.43808648, 0.180000007, 10.5341988, ++ 15, 1, 0.00891250931, 0.412505627, 0.180000007, 11.0078659, ++ 16, 1, 0.00891250931, 0.39070797, 0.180000007, 11.4510288, ++ 17, 1, 0.00891250931, 0.371887118, 0.180000007, 11.866004, ++ 18, 1, 0.00891250931, 0.367617637, 0.180000007, 12.2550087, ++ 19, 1, 0.00891250931, 0.422220588, 0.180000007, 12.6201363, ++ 20, 2, 0.0178250186, 0.564990044, 0.180000007, 13.2772083, ++ 21, 2, 0.0178250186, 0.519700944, 0.180000007, 13.871047, ++ 22, 2, 0.0178250186, 0.455360681, 0.200000003, 14.4024391, ++ 23, 2, 0.0178250186, 0.408867925, 0.200000003, 14.8811684, ++ 24, 2, 0.0178250186, 0.381538749, 0.200000003, 15.3153324, ++ 25, 2, 0.0178250186, 0.362357527, 0.200000003, 15.7116165, ++ 26, 2, 0.0178250186, 0.365735918, 0.200000003, 16.0755405, ++ 27, 3, 0.0267375279, 0.38064, 0.200000003, 16.4882088, ++ 28, 3, 0.0267375279, 0.379183382, 0.200000003, 16.9410992, ++ 29, 3, 0.0267375279, 0.360672712, 0.200000003, 17.3513336, ++ 30, 3, 0.0267375279, 0.343065977, 0.200000003, 17.7264423, ++ 31, 3, 0.0267375279, 0.339290261, 0.200000003, 18.0722466, ++ 32, 4, 0.0356500372, 0.342963994, 0.200000003, 18.4426575, ++ 33, 4, 0.0356500372, 0.343128443, 0.200000003, 18.8344078, ++ 34, 4, 0.0356500372, 0.343988508, 0.25, 19.1955795, ++ 35, 5, 0.0445625484, 0.343928397, 0.25, 19.5697021, ++ 36, 5, 0.0445625484, 0.339527696, 0.25, 19.9551182, ++ 37, 5, 0.0889139697, 0.336541563, 0.280000001, 20.3115921, ++ 38, 6, 0.106696762, 0.334955156, 0.280000001, 20.6737747, ++ 39, 6, 0.169102982, 0.335601568, 0.300000012, 21.0404968, ++ 40, 7, 0.1972868, 0.334716886, 0.300000012, 21.4060211, ++ 41, 7, 0.1972868, 0.331676662, 0.300000012, 21.7696877, ++ 42, 8, 0.713000774, 0.328550965, 0.400000006, 22.1267223, ++ 43, 8, 0.713000774, 0.339241952, 0.400000006, 22.4769249, ++ 44, 9, 0.802125871, 0.425207615, 0.400000006, 22.8164864, ++ 22050, 44, ++ 0, 1, 8.91250896, 0.954045713, 0.150000006, 0, ++ 1, 1, 8.91250896, 0.833381653, 0.150000006, 0.869851649, ++ 2, 1, 0.891250908, 0.815945923, 0.150000006, 1.73325908, ++ 3, 1, 0.0891250968, 0.794244766, 0.150000006, 2.58322191, ++ 4, 1, 0.028183829, 0.776486695, 0.150000006, 3.4134295, ++ 5, 1, 0.00891250931, 0.755260408, 0.150000006, 4.21850443, ++ 6, 1, 0.00891250931, 0.731070817, 0.150000006, 4.99414825, ++ 7, 1, 0.00891250931, 0.701775849, 0.150000006, 5.73718691, ++ 8, 1, 0.00891250931, 0.667876124, 0.150000006, 6.44553185, ++ 9, 1, 0.00891250931, 0.630284071, 0.150000006, 7.11807632, ++ 10, 1, 0.00891250931, 0.590170324, 0.150000006, 7.75455618, ++ 11, 1, 0.00891250931, 0.548788548, 0.150000006, 8.3553915, ++ 12, 1, 0.00891250931, 0.507795513, 0.150000006, 8.92152882, ++ 13, 1, 0.00891250931, 0.469515711, 0.180000007, 9.45430183, ++ 14, 1, 0.00891250931, 0.432291716, 0.180000007, 9.95530319, ++ 15, 1, 0.00891250931, 0.411131173, 0.180000007, 10.4262848, ++ 16, 1, 0.00891250931, 0.390771538, 0.180000007, 10.8690758, ++ 17, 1, 0.00891250931, 0.373318017, 0.180000007, 11.2855215, ++ 18, 1, 0.00891250931, 0.36956048, 0.180000007, 11.6774378, ++ 19, 1, 0.00891250931, 0.42595759, 0.180000007, 12.0465794, ++ 20, 2, 0.0178250186, 0.576900065, 0.180000007, 12.7141209, ++ 21, 2, 0.0178250186, 0.533114731, 0.180000007, 13.3197365, ++ 22, 2, 0.0178250186, 0.469967514, 0.180000007, 13.8634901, ++ 23, 2, 0.0178250186, 0.417268544, 0.200000003, 14.3544445, ++ 24, 2, 0.0178250186, 0.389299124, 0.200000003, 14.8002586, ++ 25, 2, 0.0178250186, 0.362824857, 0.200000003, 15.2073727, ++ 26, 2, 0.0178250186, 0.346801281, 0.200000003, 15.5811834, ++ 27, 2, 0.0178250186, 0.349400043, 0.200000003, 15.926218, ++ 28, 3, 0.0267375279, 0.364026934, 0.200000003, 16.3194923, ++ 29, 3, 0.0267375279, 0.36560446, 0.200000003, 16.752903, ++ 30, 3, 0.0267375279, 0.354275256, 0.200000003, 17.1470814, ++ 31, 3, 0.0267375279, 0.351219416, 0.200000003, 17.5086212, ++ 32, 4, 0.0356500372, 0.354364097, 0.200000003, 17.8938141, ++ 33, 4, 0.0356500372, 0.348915905, 0.200000003, 18.2992878, ++ 34, 4, 0.0356500372, 0.337649345, 0.200000003, 18.6713982, ++ 35, 4, 0.0356500372, 0.332076877, 0.25, 19.015646, ++ 36, 5, 0.0445625484, 0.330793113, 0.25, 19.3734016, ++ 37, 5, 0.0445625484, 0.327528268, 0.25, 19.7430382, ++ 38, 5, 0.0889139697, 0.32551071, 0.280000001, 20.0859604, ++ 39, 6, 0.106696762, 0.324436843, 0.280000001, 20.4354992, ++ 40, 6, 0.106696762, 0.325835049, 0.280000001, 20.7905579, ++ 41, 7, 0.1972868, 0.326221824, 0.300000012, 21.1458054, ++ 42, 7, 0.1972868, 0.325960994, 0.300000012, 21.5005951, ++ 43, 8, 0.225470632, 0.339019388, 0.300000012, 21.8504524, ++ 44, 8, 0.713000774, 0.426850349, 0.400000006, 22.1951065, ++ 16000, 45, ++ 0, 1, 8.91250896, 0.834739447, 0.150000006, 0, ++ 1, 1, 8.91250896, 0.623757005, 0.150000006, 0.631518543, ++ 2, 1, 0.891250908, 0.60420388, 0.150000006, 1.2606914, ++ 3, 1, 0.891250908, 0.591974258, 0.150000006, 1.88476217, ++ 4, 1, 0.0891250968, 0.575301588, 0.150000006, 2.50111985, ++ 5, 1, 0.028183829, 0.561547697, 0.150000006, 3.1073606, ++ 6, 1, 0.028183829, 0.546665847, 0.150000006, 3.70133615, ++ 7, 1, 0.00891250931, 0.52986443, 0.150000006, 4.28118753, ++ 8, 1, 0.00891250931, 0.511183441, 0.150000006, 4.84536505, ++ 9, 1, 0.00891250931, 0.490902334, 0.150000006, 5.39263105, ++ 10, 1, 0.00891250931, 0.46938166, 0.150000006, 5.92205667, ++ 11, 1, 0.00891250931, 0.447003782, 0.150000006, 6.43299866, ++ 12, 1, 0.00891250931, 0.428170592, 0.150000006, 6.92507982, ++ 13, 1, 0.00891250931, 0.414536625, 0.150000006, 7.39815664, ++ 14, 1, 0.00891250931, 0.401033074, 0.150000006, 7.85228777, ++ 15, 1, 0.00891250931, 0.38779071, 0.150000006, 8.28770351, ++ 16, 1, 0.00891250931, 0.374230444, 0.150000006, 8.704772, ++ 17, 1, 0.00891250931, 0.360547513, 0.180000007, 9.10397339, ++ 18, 1, 0.00891250931, 0.348256677, 0.180000007, 9.48587132, ++ 19, 1, 0.00891250931, 0.350327015, 0.180000007, 9.85109234, ++ 20, 1, 0.00891250931, 0.406330824, 0.180000007, 10.200304, ++ 21, 2, 0.0178250186, 0.554098248, 0.180000007, 10.846529, ++ 22, 2, 0.0178250186, 0.528312504, 0.180000007, 11.4447651, ++ 23, 2, 0.0178250186, 0.476527005, 0.180000007, 11.9928398, ++ 24, 2, 0.0178250186, 0.428205669, 0.180000007, 12.495945, ++ 25, 2, 0.0178250186, 0.402271926, 0.180000007, 12.9588718, ++ 26, 2, 0.0178250186, 0.378024429, 0.180000007, 13.3859692, ++ 27, 2, 0.0178250186, 0.36254698, 0.180000007, 13.7811394, ++ 28, 2, 0.0178250186, 0.368058592, 0.200000003, 14.1478529, ++ 29, 3, 0.0267375279, 0.385963261, 0.200000003, 14.5674343, ++ 30, 3, 0.0267375279, 0.38640517, 0.200000003, 15.0304852, ++ 31, 3, 0.0267375279, 0.367834061, 0.200000003, 15.4513416, ++ 32, 3, 0.0267375279, 0.349686563, 0.200000003, 15.836277, ++ 33, 3, 0.0267375279, 0.345709383, 0.200000003, 16.1904697, ++ 34, 4, 0.0356500372, 0.34871915, 0.200000003, 16.5683517, ++ 35, 4, 0.0356500372, 0.347054332, 0.200000003, 16.9660263, ++ 36, 4, 0.0356500372, 0.346329987, 0.200000003, 17.3304482, ++ 37, 5, 0.0445625484, 0.344658494, 0.200000003, 17.7055588, ++ 38, 5, 0.0445625484, 0.338779271, 0.200000003, 18.0899811, ++ 39, 5, 0.0445625484, 0.334878683, 0.200000003, 18.4440536, ++ 40, 6, 0.0534750558, 0.332811534, 0.200000003, 18.8030052, ++ 41, 6, 0.0534750558, 0.333717585, 0.25, 19.1665268, ++ 42, 7, 0.062387567, 0.333986402, 0.25, 19.5299358, ++ 43, 7, 0.062387567, 0.334142625, 0.25, 19.8934898, ++ 44, 8, 0.142262354, 0.34677428, 0.280000001, 20.2535706, ++ 45, 8, 0.142262354, 0.436254472, 0.280000001, 20.610569, ++ 48000.0, 20, ++ 0, 3, 0, 4, 1.000, 0.056, ++ 1, 3, 4, 7, 0.944, 0.611, ++ 2, 4, 7, 11, 0.389, 0.167, ++ 3, 3, 11, 14, 0.833, 0.722, ++ 4, 3, 14, 17, 0.278, 0.639, ++ 5, 2, 17, 19, 0.361, 0.417, ++ 6, 3, 19, 22, 0.583, 0.083, ++ 7, 2, 22, 24, 0.917, 0.750, ++ 8, 3, 24, 27, 0.250, 0.417, ++ 9, 3, 27, 30, 0.583, 0.648, ++ 10, 3, 30, 33, 0.352, 0.611, ++ 11, 3, 33, 36, 0.389, 0.625, ++ 12, 4, 36, 40, 0.375, 0.144, ++ 13, 3, 40, 43, 0.856, 0.389, ++ 14, 3, 43, 46, 0.611, 0.160, ++ 15, 3, 46, 49, 0.840, 0.217, ++ 16, 3, 49, 52, 0.783, 0.184, ++ 17, 2, 52, 54, 0.816, 0.886, ++ 18, 3, 54, 57, 0.114, 0.313, ++ 19, 2, 57, 59, 0.687, 0.452, ++ 20, 1, 59, 60, 0.548, 0.908, ++ 44100.0, 20, ++ 0, 3, 0, 4, 1.000, 0.056, ++ 1, 3, 4, 7, 0.944, 0.611, ++ 2, 4, 7, 11, 0.389, 0.167, ++ 3, 3, 11, 14, 0.833, 0.722, ++ 4, 3, 14, 17, 0.278, 0.139, ++ 5, 1, 17, 18, 0.861, 0.917, ++ 6, 3, 18, 21, 0.083, 0.583, ++ 7, 3, 21, 24, 0.417, 0.250, ++ 8, 3, 24, 27, 0.750, 0.805, ++ 9, 3, 27, 30, 0.194, 0.574, ++ 10, 3, 30, 33, 0.426, 0.537, ++ 11, 3, 33, 36, 0.463, 0.819, ++ 12, 4, 36, 40, 0.180, 0.100, ++ 13, 3, 40, 43, 0.900, 0.468, ++ 14, 3, 43, 46, 0.532, 0.623, ++ 15, 3, 46, 49, 0.376, 0.450, ++ 16, 3, 49, 52, 0.550, 0.552, ++ 17, 3, 52, 55, 0.448, 0.403, ++ 18, 2, 55, 57, 0.597, 0.643, ++ 19, 2, 57, 59, 0.357, 0.722, ++ 20, 2, 59, 61, 0.278, 0.960, ++ 32000, 20, ++ 0, 1, 0, 2, 1.000, 0.528, ++ 1, 2, 2, 4, 0.472, 0.305, ++ 2, 2, 4, 6, 0.694, 0.083, ++ 3, 1, 6, 7, 0.917, 0.861, ++ 4, 2, 7, 9, 0.139, 0.639, ++ 5, 2, 9, 11, 0.361, 0.417, ++ 6, 3, 11, 14, 0.583, 0.083, ++ 7, 2, 14, 16, 0.917, 0.750, ++ 8, 3, 16, 19, 0.250, 0.870, ++ 9, 3, 19, 22, 0.130, 0.833, ++ 10, 4, 22, 26, 0.167, 0.389, ++ 11, 4, 26, 30, 0.611, 0.478, ++ 12, 4, 30, 34, 0.522, 0.033, ++ 13, 3, 34, 37, 0.967, 0.917, ++ 14, 4, 37, 41, 0.083, 0.617, ++ 15, 3, 41, 44, 0.383, 0.995, ++ 16, 4, 44, 48, 0.005, 0.274, ++ 17, 3, 48, 51, 0.726, 0.480, ++ 18, 3, 51, 54, 0.519, 0.261, ++ 19, 2, 54, 56, 0.739, 0.884, ++ 20, 2, 56, 58, 0.116, 1.000, ++ 24000, 20, ++ 0, 2, 0, 3, 1, 0.916666746, ++ 1, 3, 3, 6, 0.0833332539, 0.583333492, ++ 2, 3, 6, 9, 0.416666508, 0.25, ++ 3, 2, 9, 11, 0.75, 0.916666985, ++ 4, 3, 11, 14, 0.0833330154, 0.583333969, ++ 5, 3, 14, 17, 0.416666031, 0.25, ++ 6, 3, 17, 20, 0.75, 0.537036896, ++ 7, 3, 20, 23, 0.462963104, 0.5, ++ 8, 4, 23, 27, 0.5, 0.0555559993, ++ 9, 3, 27, 30, 0.944444001, 0.402778625, ++ 10, 3, 30, 33, 0.597221375, 0.766667187, ++ 11, 3, 33, 36, 0.233332828, 0.805555999, ++ 12, 3, 36, 39, 0.194444016, 0.769841909, ++ 13, 3, 39, 42, 0.23015812, 0.611111104, ++ 14, 3, 42, 45, 0.388888896, 0.449494779, ++ 15, 3, 45, 48, 0.550505221, 0.194444954, ++ 16, 2, 48, 50, 0.805555046, 0.913194656, ++ 17, 3, 50, 53, 0.0868053436, 0.580555737, ++ 18, 3, 53, 56, 0.419444263, 0.113426208, ++ 19, 2, 56, 58, 0.886573792, 0.533730626, ++ 20, 2, 58, 60, 0.466269344, 0.691176474, ++ 22050, 20, ++ 0, 2, 0, 3, 1, 0.916666746, ++ 1, 3, 3, 6, 0.0833332539, 0.583333492, ++ 2, 3, 6, 9, 0.416666508, 0.25, ++ 3, 2, 9, 11, 0.75, 0.916666985, ++ 4, 3, 11, 14, 0.0833330154, 0.583333969, ++ 5, 3, 14, 17, 0.416666031, 0.25, ++ 6, 3, 17, 20, 0.75, 0.203703582, ++ 7, 3, 20, 23, 0.796296418, 0.166666687, ++ 8, 3, 23, 26, 0.833333313, 0.722222686, ++ 9, 4, 26, 30, 0.277777344, 0.152778625, ++ 10, 3, 30, 33, 0.847221375, 0.566667199, ++ 11, 3, 33, 36, 0.433332831, 0.93518573, ++ 12, 4, 36, 40, 0.0648142472, 0.118056297, ++ 13, 3, 40, 43, 0.881943703, 0.0925937295, ++ 14, 2, 43, 45, 0.907406271, 0.934344172, ++ 15, 3, 45, 48, 0.0656557977, 0.575398028, ++ 16, 3, 48, 51, 0.424601972, 0.232026935, ++ 17, 2, 51, 53, 0.767973065, 0.758334339, ++ 18, 3, 53, 56, 0.241665646, 0.187501252, ++ 19, 2, 56, 58, 0.812498748, 0.533731699, ++ 20, 2, 58, 60, 0.466268271, 0.257577598, ++ 16000, 20, ++ 0, 1, 0, 2, 1, 0.944444478, ++ 1, 2, 2, 4, 0.0555555038, 0.722222328, ++ 2, 2, 4, 6, 0.277777672, 0.5, ++ 3, 2, 6, 8, 0.5, 0.27777797, ++ 4, 2, 8, 10, 0.72222203, 0.0555559993, ++ 5, 1, 10, 11, 0.944444001, 0.833333313, ++ 6, 3, 11, 14, 0.166666672, 0.203703582, ++ 7, 3, 14, 17, 0.796296418, 0.166666687, ++ 8, 3, 17, 20, 0.833333313, 0.54166698, ++ 9, 3, 20, 23, 0.458333015, 0.652778625, ++ 10, 4, 23, 27, 0.347221375, 0.166667163, ++ 11, 3, 27, 30, 0.833332837, 0.722222924, ++ 12, 4, 30, 34, 0.277777106, 0.277778625, ++ 13, 3, 34, 37, 0.722221375, 0.604167938, ++ 14, 3, 37, 40, 0.395832062, 0.627778649, ++ 15, 3, 40, 43, 0.37222138, 0.542736351, ++ 16, 3, 43, 46, 0.457263649, 0.371528625, ++ 17, 3, 46, 49, 0.628471375, 0.00833433867, ++ 18, 2, 49, 51, 0.991665661, 0.500001311, ++ 19, 2, 51, 53, 0.499998659, 0.886832893, ++ 20, 2, 53, 55, 0.113167092, 0.629034221, ++ 48000.0, 11, ++ 0, 2, 0, 2, 1.000, 0.167, ++ 1, 2, 3, 5, 0.833, 0.833, ++ 2, 3, 5, 8, 0.167, 0.500, ++ 3, 3, 8, 11, 0.500, 0.167, ++ 4, 4, 11, 15, 0.833, 0.167, ++ 5, 4, 15, 19, 0.833, 0.583, ++ 6, 3, 19, 22, 0.417, 0.917, ++ 7, 4, 22, 26, 0.083, 0.944, ++ 8, 4, 26, 30, 0.055, 0.042, ++ 9, 2, 30, 32, 0.958, 0.567, ++ 10, 3, 32, 35, 0.433, 0.167, ++ 11, 2, 35, 37, 0.833, 0.618, ++ 44100.0, 11, ++ 0, 2, 0, 2, 1.000, 0.167, ++ 1, 2, 3, 5, 0.833, 0.833, ++ 2, 3, 5, 8, 0.167, 0.500, ++ 3, 3, 8, 11, 0.500, 0.167, ++ 4, 4, 11, 15, 0.833, 0.167, ++ 5, 5, 15, 20, 0.833, 0.250, ++ 6, 3, 20, 23, 0.750, 0.583, ++ 7, 4, 23, 27, 0.417, 0.055, ++ 8, 3, 27, 30, 0.944, 0.375, ++ 9, 3, 30, 33, 0.625, 0.300, ++ 10, 3, 33, 36, 0.700, 0.167, ++ 11, 2, 36, 38, 0.833, 1.000, ++ 32000, 11, ++ 0, 2, 0, 2, 1.000, 0.167, ++ 1, 2, 3, 5, 0.833, 0.833, ++ 2, 3, 5, 8, 0.167, 0.500, ++ 3, 3, 8, 11, 0.500, 0.167, ++ 4, 4, 11, 15, 0.833, 0.167, ++ 5, 5, 15, 20, 0.833, 0.250, ++ 6, 4, 20, 24, 0.750, 0.250, ++ 7, 5, 24, 29, 0.750, 0.055, ++ 8, 4, 29, 33, 0.944, 0.375, ++ 9, 4, 33, 37, 0.625, 0.472, ++ 10, 3, 37, 40, 0.528, 0.937, ++ 11, 1, 40, 41, 0.062, 1.000, ++ 24000, 11, ++ 0, 3, 0, 4, 1, 0.166666746, ++ 1, 2, 4, 6, 0.833333254, 0.833333492, ++ 2, 3, 6, 9, 0.166666508, 0.5, ++ 3, 4, 9, 13, 0.5, 0.5, ++ 4, 5, 13, 18, 0.5, 0.833333969, ++ 5, 5, 18, 23, 0.166666031, 0.25, ++ 6, 4, 23, 27, 0.75, 0.25, ++ 7, 3, 27, 30, 0.75, 0.611111999, ++ 8, 4, 30, 34, 0.388888031, 0.208333969, ++ 9, 3, 34, 37, 0.791666031, 0.766667187, ++ 10, 4, 37, 41, 0.233332828, 0.45238167, ++ 11, 4, 41, 45, 0.54761833, 0.277778625, ++ 22050, 11, ++ 0, 3, 0, 4, 1, 0.166666746, ++ 1, 2, 4, 6, 0.833333254, 0.833333492, ++ 2, 3, 6, 9, 0.166666508, 0.5, ++ 3, 4, 9, 13, 0.5, 0.5, ++ 4, 4, 13, 17, 0.5, 0.5, ++ 5, 4, 17, 21, 0.5, 0.916666985, ++ 6, 4, 21, 25, 0.0833330154, 0.25, ++ 7, 4, 25, 29, 0.75, 0.611111999, ++ 8, 4, 29, 33, 0.388888031, 0.458333969, ++ 9, 4, 33, 37, 0.541666031, 0.633334339, ++ 10, 4, 37, 41, 0.366665661, 0.583334565, ++ 11, 4, 41, 45, 0.416665405, 0.437500954, ++ 16000, 11, ++ 0, 0, 0, 4, 1, 0.166666746, ++ 1, 0, 4, 6, 0.833333254, 0.833333492, ++ 2, 0, 6, 9, 0.166666508, 0.5, ++ 3, 0, 9, 13, 0.5, 0.5, ++ 4, 0, 13, 18, 0.5, 0.833333969, ++ 5, 0, 18, 23, 0.166666031, 0.75, ++ 6, 0, 23, 27, 0.25, 0.75, ++ 7, 0, 27, 31, 0.25, 0.611111999, ++ 8, 0, 31, 35, 0.388888031, 0.458333969, ++ 9, 0, 35, 39, 0.541666031, 0.166667163, ++ 10, 0, 39, 42, 0.833332837, 0.805555999, ++ 11, 0, 42, 46, 0.194444016, 0.4375}; ++ +diff -r -c -N encoder/tables.h lame3.70/tables.h +*** encoder/tables.h Wed Dec 31 17:00:00 1969 +--- lame3.70/tables.h Thu Apr 6 12:50:52 2000 +*************** +*** 0 **** +--- 1,43 ---- ++ /* ++ ** Copyright (C) 1999 Albert L. Faber ++ ** ++ * This library is free software; you can redistribute it and/or ++ * modify it under the terms of the GNU Library General Public ++ * License as published by the Free Software Foundation; either ++ * version 2 of the License, or (at your option) any later version. ++ * ++ * This library is distributed in the hope that it will be useful, ++ * but WITHOUT ANY WARRANTY; without even the implied warranty of ++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU ++ * Library General Public License for more details. ++ * ++ * You should have received a copy of the GNU Library General Public ++ * License along with this library; if not, write to the ++ * Free Software Foundation, Inc., 59 Temple Place - Suite 330, ++ * Boston, MA 02111-1307, USA. ++ */ ++ ++ #ifndef TABLES_H_INCLUDED ++ #define TABLES_H_INCLUDED ++ #include "encoder.h" ++ extern FLOAT8 psy_data[]; ++ ++ #define HUFFBITS unsigned long int ++ #define HTN 34 ++ ++ struct huffcodetab { ++ unsigned int xlen; /*max. x-index+ */ ++ unsigned int linmax; /*max number to be stored in linbits */ ++ HUFFBITS *table; /*pointer to array[xlen][ylen] */ ++ unsigned char *hlen; /*pointer to array[xlen][ylen] */ ++ }; ++ ++ extern struct huffcodetab ht[HTN];/* global memory block */ ++ /* array of all huffcodtable headers */ ++ /* 0..31 Huffman code table 0..31 */ ++ /* 32,33 count1-tables */ ++ ++ ++ ++ ++ #endif +diff -r -c -N encoder/takehiro.c lame3.70/takehiro.c +*** encoder/takehiro.c Wed Dec 31 17:00:00 1969 +--- lame3.70/takehiro.c Thu Apr 6 12:50:52 2000 +*************** +*** 0 **** +--- 1,765 ---- ++ /* ++ * MP3 huffman table selecting and bit counting ++ * ++ * Copyright (c) 1999 Takehiro TOMINAGA ++ * ++ * This library is free software; you can redistribute it and/or ++ * modify it under the terms of the GNU Library General Public ++ * License as published by the Free Software Foundation; either ++ * version 2 of the License, or (at your option) any later version. ++ * ++ * This library is distributed in the hope that it will be useful, ++ * but WITHOUT ANY WARRANTY; without even the implied warranty of ++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU ++ * Library General Public License for more details. ++ * ++ * You should have received a copy of the GNU Library General Public ++ * License along with this library; if not, write to the ++ * Free Software Foundation, Inc., 59 Temple Place - Suite 330, ++ * Boston, MA 02111-1307, USA. ++ */ ++ ++ ++ #include "util.h" ++ #include "l3side.h" ++ #include "tables.h" ++ #include "quantize-pvt.h" ++ ++ ++ ++ ++ struct ++ { ++ unsigned region0_count; ++ unsigned region1_count; ++ } subdv_table[ 23 ] = ++ { ++ {0, 0}, /* 0 bands */ ++ {0, 0}, /* 1 bands */ ++ {0, 0}, /* 2 bands */ ++ {0, 0}, /* 3 bands */ ++ {0, 0}, /* 4 bands */ ++ {0, 1}, /* 5 bands */ ++ {1, 1}, /* 6 bands */ ++ {1, 1}, /* 7 bands */ ++ {1, 2}, /* 8 bands */ ++ {2, 2}, /* 9 bands */ ++ {2, 3}, /* 10 bands */ ++ {2, 3}, /* 11 bands */ ++ {3, 4}, /* 12 bands */ ++ {3, 4}, /* 13 bands */ ++ {3, 4}, /* 14 bands */ ++ {4, 5}, /* 15 bands */ ++ {4, 5}, /* 16 bands */ ++ {4, 6}, /* 17 bands */ ++ {5, 6}, /* 18 bands */ ++ {5, 6}, /* 19 bands */ ++ {5, 7}, /* 20 bands */ ++ {6, 7}, /* 21 bands */ ++ {6, 7}, /* 22 bands */ ++ }; ++ ++ ++ /*************************************************************************/ ++ /* ix_max */ ++ /*************************************************************************/ ++ ++ static int ix_max(int *ix, int *end) ++ { ++ int max = 0; ++ ++ while (ix < end) { ++ int x = *ix++; ++ if (max < x) ++ max = x; ++ ++ x = *ix++; ++ if (max < x) ++ max = x; ++ } ++ return max; ++ } ++ ++ ++ /*************************************************************************/ ++ /* count_bit */ ++ /*************************************************************************/ ++ ++ /* ++ Function: Count the number of bits necessary to code the subregion. ++ */ ++ ++ static int cb_esc_buf[288]; ++ static int cb_esc_sign; ++ static int *cb_esc_end; ++ static const int huf_tbl_noESC[15] = { ++ 1, 2, 5, 7, 7,10,10,13,13,13,13,13,13,13,13 ++ }; ++ ++ static int ++ count_bit_ESC(int *ix, int *end, int t1, int t2, int *s) ++ { ++ /* ESC-table is used */ ++ int linbits1 = ht[t1].xlen; ++ int linbits2 = ht[t2].xlen; ++ int sum = 0; ++ int sum1 = 0; ++ int sum2 = 0; ++ ++ while (ix < end) { ++ int x = *ix++; ++ int y = *ix++; ++ ++ if (x != 0) { ++ sum++; ++ if (x > 14) { ++ x = 15; ++ sum1 += linbits1; ++ sum2 += linbits2; ++ } ++ x *= 16; ++ } ++ ++ if (y != 0) { ++ sum++; ++ if (y > 14) { ++ y = 15; ++ sum1 += linbits1; ++ sum2 += linbits2; ++ } ++ x += y; ++ } ++ ++ sum1 += ht[16].hlen[x]; ++ sum2 += ht[24].hlen[x]; ++ } ++ ++ if (sum1 > sum2) { ++ sum1 = sum2; ++ t1 = t2; ++ } ++ ++ *s += sum + sum1; ++ return t1; ++ } ++ ++ static int ++ count_bit_noESC(int *ix, int *end, unsigned int table) ++ { ++ /* No ESC-words */ ++ int sum = 0, sign = 0; ++ unsigned char *hlen = ht[table].hlen; ++ int *p = cb_esc_buf; ++ ++ do { ++ int x = *ix++; ++ int y = *ix++; ++ if (x != 0) { ++ sign++; ++ x *= 16; ++ } ++ ++ if (y != 0) { ++ sign++; ++ x += y; ++ } ++ ++ *p++ = x; ++ sum += hlen[x]; ++ } while (ix < end); ++ ++ cb_esc_sign = sign; ++ cb_esc_end = p; ++ return sum + sign; ++ } ++ ++ ++ ++ static int ++ count_bit_noESC2(unsigned int table) ++ { ++ /* No ESC-words */ ++ int sum = cb_esc_sign; ++ int *p = cb_esc_buf; ++ ++ do { ++ sum += ht[table].hlen[*p++]; ++ } while (p < cb_esc_end); ++ ++ return sum; ++ } ++ ++ ++ ++ static int ++ count_bit_short_ESC(int *ix, int *end, int t1, int t2, int *s) ++ { ++ /* ESC-table is used */ ++ int linbits1 = ht[t1].xlen; ++ int linbits2 = ht[t2].xlen; ++ int sum = 0; ++ int sum1 = 0; ++ int sum2 = 0; ++ ++ do { ++ int i; ++ for (i = 0; i < 3; i++) { ++ int y = *(ix + 3); ++ int x = *ix++; ++ ++ if (x != 0) { ++ sum++; ++ if (x > 14) { ++ x = 15; ++ sum1 += linbits1; ++ sum2 += linbits2; ++ } ++ x *= 16; ++ } ++ ++ if (y != 0) { ++ sum++; ++ if (y > 14) { ++ y = 15; ++ sum1 += linbits1; ++ sum2 += linbits2; ++ } ++ x += y; ++ } ++ ++ sum1 += ht[16].hlen[x]; ++ sum2 += ht[24].hlen[x]; ++ } ++ ix += 3; ++ } while (ix < end); ++ ++ if (sum1 > sum2) { ++ sum1 = sum2; ++ t1 = t2; ++ } ++ ++ *s += sum + sum1; ++ return t1; ++ } ++ ++ ++ ++ static int ++ count_bit_short_noESC(int *ix, int *end, unsigned int table) ++ { ++ /* No ESC-words */ ++ int sum = 0, sign = 0; ++ unsigned char *hlen = ht[table].hlen; ++ int *p = cb_esc_buf; ++ ++ do { ++ int i; ++ for (i = 0; i < 3; i++) { ++ int y = *(ix + 3); ++ int x = *ix++; ++ if (x != 0) { ++ sign++; ++ x *= 16; ++ } ++ ++ if (y != 0) { ++ sign++; ++ x += y; ++ } ++ ++ *p++ = x; ++ sum += hlen[x]; ++ } ++ ix += 3; ++ } while (ix < end); ++ ++ cb_esc_sign = sign; ++ cb_esc_end = p; ++ return sum + sign; ++ } ++ ++ ++ ++ /*************************************************************************/ ++ /* new_choose table */ ++ /*************************************************************************/ ++ ++ /* ++ Choose the Huffman table that will encode ix[begin..end] with ++ the fewest bits. ++ ++ Note: This code contains knowledge about the sizes and characteristics ++ of the Huffman tables as defined in the IS (Table B.7), and will not work ++ with any arbitrary tables. ++ */ ++ ++ static int choose_table(int *ix, int *end, int *s) ++ { ++ int max; ++ int choice0, sum0; ++ int choice1, sum1; ++ ++ max = ix_max(ix, end); ++ ++ if (max > IXMAX_VAL) { ++ *s = 100000; ++ return -1; ++ } ++ ++ if (max <= 15) { ++ if (max == 0) { ++ return 0; ++ } ++ /* try tables with no linbits */ ++ choice0 = huf_tbl_noESC[max - 1]; ++ sum0 = count_bit_noESC(ix, end, choice0); ++ choice1 = choice0; ++ ++ switch (choice0) { ++ case 7: ++ case 10: ++ choice1++; ++ sum1 = count_bit_noESC2(choice1); ++ if (sum0 > sum1) { ++ sum0 = sum1; ++ choice0 = choice1; ++ } ++ /*fall*/ ++ case 2: ++ case 5: ++ choice1++; ++ sum1 = count_bit_noESC2(choice1); ++ if (sum0 > sum1) { ++ sum0 = sum1; ++ choice0 = choice1; ++ } ++ break; ++ ++ case 13: ++ choice1 += 2; ++ sum1 = count_bit_noESC2(choice1); ++ if (sum0 > sum1) { ++ sum0 = sum1; ++ choice0 = choice1; ++ } ++ break; ++ ++ default: ++ break; ++ } ++ *s += sum0; ++ } else { ++ /* try tables with linbits */ ++ max -= 15; ++ ++ for (choice1 = 24; choice1 < 32; choice1++) { ++ if ((int)ht[choice1].linmax >= max) { ++ break; ++ } ++ } ++ ++ for (choice0 = choice1 - 8; choice0 < 24; choice0++) { ++ if ((int)ht[choice0].linmax >= max) { ++ break; ++ } ++ } ++ ++ choice0 = count_bit_ESC(ix, end, choice0, choice1, s); ++ } ++ ++ return choice0; ++ } ++ ++ static int choose_table_short(int *ix, int *end, int * s) ++ { ++ int max; ++ int choice0, sum0; ++ int choice1, sum1; ++ ++ max = ix_max(ix, end); ++ ++ if (max > IXMAX_VAL) { ++ *s = 100000; ++ return -1; ++ } ++ ++ if (max <= 15) { ++ if (max == 0) { ++ return 0; ++ } ++ /* try tables with no linbits */ ++ choice0 = huf_tbl_noESC[max - 1]; ++ sum0 = count_bit_short_noESC(ix, end, choice0); ++ choice1 = choice0; ++ ++ switch (choice0) { ++ case 7: ++ case 10: ++ choice1++; ++ sum1 = count_bit_noESC2(choice1); ++ if (sum0 > sum1) { ++ sum0 = sum1; ++ choice0 = choice1; ++ } ++ /*fall*/ ++ case 2: ++ case 5: ++ choice1++; ++ sum1 = count_bit_noESC2(choice1); ++ if (sum0 > sum1) { ++ sum0 = sum1; ++ choice0 = choice1; ++ } ++ break; ++ ++ case 13: ++ choice1 += 2; ++ sum1 = count_bit_noESC2(choice1); ++ if (sum0 > sum1) { ++ sum0 = sum1; ++ choice0 = choice1; ++ } ++ break; ++ ++ default: ++ break; ++ } ++ *s += sum0; ++ } else { ++ /* try tables with linbits */ ++ max -= 15; ++ for (choice1 = 24; choice1 < 32; choice1++) { ++ if ((int)ht[choice1].linmax >= max) { ++ break; ++ } ++ } ++ ++ for (choice0 = choice1 - 8; choice0 < 24; choice0++) { ++ if ((int)ht[choice0].linmax >= max) { ++ break; ++ } ++ } ++ choice0 = count_bit_short_ESC(ix, end, choice0, choice1, s); ++ } ++ ++ return choice0; ++ } ++ ++ ++ ++ static int count_bits_long(int ix[576], gr_info *gi) ++ { ++ int i, a1, a2; ++ int bits = 0; ++ ++ i=576; ++ for (; i > 1; i -= 2) ++ if (ix[i - 1] | ix[i - 2]) ++ break; ++ ++ /* Determines the number of bits to encode the quadruples. */ ++ gi->count1 = i; ++ a1 = 0; ++ for (; i > 3; i -= 4) { ++ int p, v; ++ if ((unsigned int)(ix[i-1] | ix[i-2] | ix[i-3] | ix[i-4]) > 1) ++ break; ++ ++ v = ix[i-1]; ++ p = v; ++ bits += v; ++ ++ v = ix[i-2]; ++ if (v != 0) { ++ p += 2; ++ bits++; ++ } ++ ++ v = ix[i-3]; ++ if (v != 0) { ++ p += 4; ++ bits++; ++ } ++ ++ v = ix[i-4]; ++ if (v != 0) { ++ p += 8; ++ bits++; ++ } ++ ++ a1 += ht[32].hlen[p]; ++ } ++ a2 = gi->count1 - i; ++ if (a1 < a2) { ++ bits += a1; ++ gi->count1table_select = 0; ++ } else { ++ bits += a2; ++ gi->count1table_select = 1; ++ } ++ ++ gi->count1bits = bits; ++ gi->big_values = i; ++ if (i == 0) ++ return bits; ++ ++ if (gi->block_type == NORM_TYPE) { ++ int index; ++ int scfb_anz = 0; ++ ++ while (scalefac_band.l[++scfb_anz] < i) ++ ; ++ index = subdv_table[scfb_anz].region0_count; ++ while (scalefac_band.l[index + 1] > i) ++ index--; ++ gi->region0_count = index; ++ ++ index = subdv_table[scfb_anz].region1_count; ++ while (scalefac_band.l[index + gi->region0_count + 2] > i) ++ index--; ++ gi->region1_count = index; ++ ++ a1 = scalefac_band.l[gi->region0_count + 1]; ++ a2 = scalefac_band.l[index + gi->region0_count + 2]; ++ gi->table_select[2] = choose_table(ix + a2, ix + i, &bits); ++ ++ } else { ++ gi->region0_count = 7; ++ /*gi->region1_count = SBPSY_l - 7 - 1;*/ ++ gi->region1_count = SBMAX_l -1 - 7 - 1; ++ a1 = scalefac_band.l[7 + 1]; ++ a2 = i; ++ if (a1 > a2) { ++ a1 = a2; ++ } ++ } ++ ++ /* Count the number of bits necessary to code the bigvalues region. */ ++ gi->table_select[0] = choose_table(ix, ix + a1, &bits); ++ gi->table_select[1] = choose_table(ix + a1, ix + a2, &bits); ++ return bits; ++ } ++ ++ ++ ++ ++ int count_bits(lame_global_flags *gfp,int *ix, FLOAT8 *xr, gr_info *cod_info) ++ { ++ int bits=0,i; ++ /* since quantize_xrpow uses table lookup, we need to check this first: */ ++ FLOAT8 w = (IXMAX_VAL) / IPOW20(cod_info->global_gain); ++ for ( i = 0; i < 576; i++ ) { ++ if (xr[i] > w) ++ return 100000; ++ } ++ if (gfp->quantization) ++ quantize_xrpow(xr, ix, cod_info); ++ else ++ quantize_xrpow_ISO(xr, ix, cod_info); ++ ++ ++ ++ if (cod_info->block_type==SHORT_TYPE) { ++ cod_info->table_select[0] = choose_table_short(ix, ix + 36, &bits); ++ cod_info->table_select[1] = choose_table_short(ix + 36, ix + 576, &bits); ++ cod_info->big_values = 288; ++ }else{ ++ bits=count_bits_long(ix, cod_info); ++ cod_info->count1 = (cod_info->count1 - cod_info->big_values) / 4; ++ cod_info->big_values /= 2; ++ } ++ return bits; ++ ++ } ++ ++ void best_huffman_divide(int gr, int ch, gr_info *gi, int *ix) ++ { ++ int *bits, r0, r1, a1, a2, bigv; ++ int r1_bits; ++ int r3_bits[7 + 15 + 2 + 1]; ++ int r3_tbl[7 + 15 + 2 + 1]; ++ gr_info cod_info; ++ ++ memcpy(&cod_info, gi, sizeof(gr_info)); ++ bigv = cod_info.big_values * 2; ++ bits = (int *) &cod_info.part2_3_length; ++ ++ for (r0 = 2; r0 < SBMAX_l + 1; r0++) { ++ a2 = scalefac_band.l[r0]; ++ if (a2 > bigv) ++ break; ++ ++ r3_bits[r0] = cod_info.count1bits + cod_info.part2_length; ++ r3_tbl[r0] = choose_table(ix + a2, ix + bigv, &r3_bits[r0]); ++ } ++ for (; r0 <= 7 + 15 + 2; r0++) { ++ r3_bits[r0] = 100000; ++ } ++ ++ for (r0 = 0; r0 < 16; r0++) { ++ a1 = scalefac_band.l[r0 + 1]; ++ if (a1 > bigv) ++ break; ++ cod_info.region0_count = r0; ++ r1_bits = 0; ++ cod_info.table_select[0] = choose_table(ix, ix + a1, &r1_bits); ++ if ((int)gi->part2_3_length < r1_bits) ++ break; ++ ++ for (r1 = 0; r1 < 8; r1++) { ++ *bits = r1_bits + r3_bits[r0 + r1 + 2]; ++ if ((int)gi->part2_3_length < *bits) ++ continue; ++ ++ a2 = scalefac_band.l[r0 + r1 + 2]; ++ ++ cod_info.table_select[1] = choose_table(ix + a1, ix + a2, bits); ++ if ((int)gi->part2_3_length < *bits) ++ continue; ++ ++ cod_info.region1_count = r1; ++ cod_info.table_select[2] = r3_tbl[r0 + r1 + 2]; ++ memcpy(gi, &cod_info, sizeof(gr_info)); ++ } ++ } ++ } ++ ++ static void ++ scfsi_calc(int ch, ++ III_side_info_t *l3_side, ++ III_scalefac_t scalefac[2][2]) ++ { ++ int i, s1, s2, c1, c2; ++ int sfb; ++ gr_info *gi = &l3_side->gr[1].ch[ch].tt; ++ ++ static const int scfsi_band[5] = { 0, 6, 11, 16, 21 }; ++ ++ static const int slen1_n[16] = { 0, 1, 1, 1, 8, 2, 2, 2, 4, 4, 4, 8, 8, 8,16,16 }; ++ static const int slen2_n[16] = { 0, 2, 4, 8, 1, 2, 4, 8, 2, 4, 8, 2, 4, 8, 4, 8 }; ++ ++ static const int slen1_tab[16] = { 0, 0, 0, 0, 3, 1, 1, 1, 2, 2, 2, 3, 3, 3, 4, 4 }; ++ static const int slen2_tab[16] = { 0, 1, 2, 3, 0, 1, 2, 3, 1, 2, 3, 1, 2, 3, 2, 3 }; ++ ++ for (i = 0; i < 4; i++) ++ l3_side->scfsi[ch][i] = 0; ++ ++ for (i = 0; i < (int)(sizeof(scfsi_band) / sizeof(int)) - 1; i++) { ++ for (sfb = scfsi_band[i]; sfb < scfsi_band[i + 1]; sfb++) { ++ if (scalefac[0][ch].l[sfb] != scalefac[1][ch].l[sfb]) ++ break; ++ } ++ if (sfb == scfsi_band[i + 1]) { ++ for (sfb = scfsi_band[i]; sfb < scfsi_band[i + 1]; sfb++) { ++ scalefac[1][ch].l[sfb] = -1; ++ } ++ l3_side->scfsi[ch][i] = 1; ++ } ++ } ++ ++ s1 = c1 = 0; ++ for (sfb = 0; sfb < 11; sfb++) { ++ if (scalefac[1][ch].l[sfb] < 0) ++ continue; ++ c1++; ++ if (s1 < scalefac[1][ch].l[sfb]) ++ s1 = scalefac[1][ch].l[sfb]; ++ } ++ ++ s2 = c2 = 0; ++ for (; sfb < SBPSY_l; sfb++) { ++ if (scalefac[1][ch].l[sfb] < 0) ++ continue; ++ c2++; ++ if (s2 < scalefac[1][ch].l[sfb]) ++ s2 = scalefac[1][ch].l[sfb]; ++ } ++ for (i = 0; i < 16; i++) { ++ if (s1 < slen1_n[i] && s2 < slen2_n[i]) { ++ int c = slen1_tab[i] * c1 + slen2_tab[i] * c2; ++ if ((int)gi->part2_length > c) { ++ gi->part2_length = c; ++ gi->scalefac_compress = i; ++ } ++ } ++ } ++ } ++ ++ void best_scalefac_store(lame_global_flags *gfp,int gr, int ch, ++ int l3_enc[2][2][576], ++ III_side_info_t *l3_side, ++ III_scalefac_t scalefac[2][2]) ++ { ++ /* use scalefac_scale if we can */ ++ gr_info *gi = &l3_side->gr[gr].ch[ch].tt; ++ ++ /* remove scalefacs from bands with ix=0. This idea comes ++ * from the AAC ISO docs. added mt 3/00 */ ++ int sfb,i,l,start,end; ++ /* check if l3_enc=0 */ ++ for ( sfb = 0; sfb < gi->sfb_lmax; sfb++ ) { ++ if (scalefac[gr][ch].l[sfb]>0) { ++ start = scalefac_band.l[ sfb ]; ++ end = scalefac_band.l[ sfb+1 ]; ++ for ( l = start; l < end; l++ ) if (l3_enc[gr][ch][l]!=0) break; ++ if (l==end) scalefac[gr][ch].l[sfb]=0; ++ } ++ } ++ for ( i = 0; i < 3; i++ ) { ++ for ( sfb = gi->sfb_smax; sfb < SBPSY_s; sfb++ ) { ++ if (scalefac[gr][ch].s[sfb][i]>0) { ++ start = scalefac_band.s[ sfb ]; ++ end = scalefac_band.s[ sfb+1 ]; ++ for ( l = start; l < end; l++ ) ++ if (l3_enc[gr][ch][3*l+i]!=0) break; ++ if (l==end) scalefac[gr][ch].s[sfb][i]=0; ++ } ++ } ++ } ++ ++ ++ gi->part2_3_length -= gi->part2_length; ++ if (!gi->scalefac_scale && !gi->preflag) { ++ u_int sfb; ++ int b, s = 0; ++ for (sfb = 0; sfb < gi->sfb_lmax; sfb++) { ++ s |= scalefac[gr][ch].l[sfb]; ++ } ++ ++ for (sfb = gi->sfb_smax; sfb < SBPSY_s; sfb++) { ++ for (b = 0; b < 3; b++) { ++ s |= scalefac[gr][ch].s[sfb][b]; ++ } ++ } ++ ++ if (!(s & 1) && s != 0) { ++ for (sfb = 0; sfb < gi->sfb_lmax; sfb++) { ++ scalefac[gr][ch].l[sfb] /= 2; ++ } ++ for (sfb = gi->sfb_smax; sfb < SBPSY_s; sfb++) { ++ for (b = 0; b < 3; b++) { ++ scalefac[gr][ch].s[sfb][b] /= 2; ++ } ++ } ++ ++ gi->scalefac_scale = 1; ++ gi->part2_length = 99999999; ++ if (gfp->mode_gr == 2) { ++ scale_bitcount(&scalefac[gr][ch], gi); ++ } else { ++ scale_bitcount_lsf(&scalefac[gr][ch], gi); ++ } ++ } ++ } ++ ++ if (gfp->mode_gr == 2 && gr == 1 ++ && l3_side->gr[0].ch[ch].tt.block_type != SHORT_TYPE ++ && l3_side->gr[1].ch[ch].tt.block_type != SHORT_TYPE ++ && l3_side->gr[0].ch[ch].tt.scalefac_scale ++ == l3_side->gr[1].ch[ch].tt.scalefac_scale ++ && l3_side->gr[0].ch[ch].tt.preflag ++ == l3_side->gr[1].ch[ch].tt.preflag) { ++ scfsi_calc(ch, l3_side, scalefac); ++ } ++ gi->part2_3_length += gi->part2_length; ++ } +Binary files encoder/testcase.mp3 and lame3.70/testcase.mp3 differ +Binary files encoder/testcase.wav and lame3.70/testcase.wav differ +diff -r -c -N encoder/timestatus.c lame3.70/timestatus.c +*** encoder/timestatus.c Wed Dec 31 17:00:00 1969 +--- lame3.70/timestatus.c Thu Mar 16 09:44:22 2000 +*************** +*** 0 **** +--- 1,146 ---- ++ #include "timestatus.h" ++ #include "util.h" ++ #include <time.h> ++ ++ #if defined(CLOCKS_PER_SEC) ++ /* ANSI/ISO systems */ ++ # define TS_CLOCKS_PER_SEC CLOCKS_PER_SEC ++ #elif defined(CLK_TCK) ++ /* Non-standard systems */ ++ # define TS_CLOCKS_PER_SEC CLK_TCK ++ #elif defined(HZ) ++ /* Older BSD systems */ ++ # define TS_CLOCKS_PER_SEC HZ ++ #else ++ # error no suitable value for TS_CLOCKS_PER_SEC ++ #endif ++ ++ /*********************************************************/ ++ /* ts_real_time: real time elapsed in seconds */ ++ /*********************************************************/ ++ FLOAT ts_real_time(long frame) { ++ ++ static time_t initial_time; ++ time_t current_time; ++ ++ time(¤t_time); ++ ++ if (frame==0) { ++ initial_time = current_time; ++ } ++ ++ return (FLOAT) difftime(current_time, initial_time); ++ } ++ ++ /*********************************************************/ ++ /* ts_process_time: process time elapsed in seconds */ ++ /*********************************************************/ ++ FLOAT ts_process_time(long frame) { ++ static clock_t initial_time; ++ clock_t current_time; ++ ++ #if ( defined(_MSC_VER) || defined(__BORLANDC__) ) ++ ++ { static HANDLE hProcess; ++ FILETIME Ignored1, Ignored2, KernelTime, UserTime; ++ ++ if ( frame==0 ) { ++ hProcess = GetCurrentProcess(); ++ } ++ ++ /* GetProcessTimes() always fails under Win9x */ ++ if (GetProcessTimes(hProcess, &Ignored1, &Ignored2, &KernelTime, &UserTime)) { ++ LARGE_INTEGER Kernel = { KernelTime.dwLowDateTime, KernelTime.dwHighDateTime }; ++ LARGE_INTEGER User = { UserTime.dwLowDateTime, UserTime.dwHighDateTime }; ++ ++ current_time = (clock_t)((FLOAT)(Kernel.QuadPart + User.QuadPart) * TS_CLOCKS_PER_SEC / 10000000); ++ } else { ++ current_time = clock(); ++ } ++ } ++ #else ++ current_time = clock(); ++ #endif ++ ++ if (frame==0) { ++ initial_time = current_time; ++ } ++ ++ return (FLOAT)(current_time - initial_time) / TS_CLOCKS_PER_SEC; ++ } ++ ++ #undef TS_CLOCKS_PER_SEC ++ ++ typedef struct ts_times { ++ FLOAT so_far; ++ FLOAT estimated; ++ FLOAT speed; ++ FLOAT eta; ++ } ts_times; ++ ++ /*********************************************************/ ++ /* ts_calc_times: calculate time info (eta, speed, etc.) */ ++ /*********************************************************/ ++ void ts_calc_times(ts_times *time, int samp_rate, long frame, long frames,int framesize) ++ { ++ if (frame > 0) { ++ time->estimated = time->so_far * frames / frame; ++ if (samp_rate * time->estimated > 0) { ++ time->speed = frames * framesize / (samp_rate * time->estimated); ++ } else { ++ time->speed = 0; ++ } ++ time->eta = time->estimated - time->so_far; ++ } else { ++ time->estimated = 0; ++ time->speed = 0; ++ time->eta = 0; ++ } ++ } ++ ++ /*********************************************************/ ++ /* timestatus: display encoding process time information */ ++ /*********************************************************/ ++ void timestatus(int samp_rate,long frameNum,long totalframes,int framesize) ++ { ++ ts_times real_time, process_time; ++ int percent; ++ ++ real_time.so_far = ts_real_time(frameNum); ++ process_time.so_far = ts_process_time(frameNum); ++ ++ if (frameNum == 0) { ++ fprintf(stderr, " Frame | CPU/estimated | time/estimated | play/CPU | ETA\n"); ++ return; ++ } ++ ++ ts_calc_times(&real_time, samp_rate, frameNum, totalframes, framesize); ++ ts_calc_times(&process_time, samp_rate, frameNum, totalframes, framesize); ++ ++ if (totalframes > 1) { ++ percent = (int)(100.0 * frameNum / (totalframes - 1)); ++ } else { ++ percent = 100; ++ } ++ ++ # define TS_TIME_DECOMPOSE(time) \ ++ (int)((long)(time+.5) / 3600), \ ++ (int)((long)((time+.5) / 60) % 60), \ ++ (int)((long)(time+.5) % 60) ++ ++ fprintf(stderr, ++ "\r%6ld/%6ld(%3d%%)|%2d:%02d:%02d/%2d:%02d:%02d|%2d:%02d:%02d/%2d:%02d:%02d|%10.4f|%2d:%02d:%02d ", ++ frameNum, ++ totalframes - 1, ++ percent, ++ TS_TIME_DECOMPOSE(process_time.so_far), ++ TS_TIME_DECOMPOSE(process_time.estimated), ++ TS_TIME_DECOMPOSE(real_time.so_far), ++ TS_TIME_DECOMPOSE(real_time.estimated), ++ process_time.speed, ++ TS_TIME_DECOMPOSE(real_time.eta) ++ ); ++ ++ fflush(stderr); ++ } ++ +diff -r -c -N encoder/timestatus.h lame3.70/timestatus.h +*** encoder/timestatus.h Wed Dec 31 17:00:00 1969 +--- lame3.70/timestatus.h Tue Mar 14 13:45:04 2000 +*************** +*** 0 **** +--- 1,8 ---- ++ #ifndef TIMESTATUS_H_INCLUDED ++ #define TIMESTATUS_H_INCLUDED ++ ++ ++ void timestatus(int samp_rate,long frameNum,long totalframes, int framesize); ++ ++ ++ #endif +diff -r -c -N encoder/tonal.c lame3.70/tonal.c +*** encoder/tonal.c Wed Jan 22 02:43:23 1997 +--- lame3.70/tonal.c Wed Dec 31 17:00:00 1969 +*************** +*** 1,1051 **** +- /********************************************************************** +- * ISO MPEG Audio Subgroup Software Simulation Group (1996) +- * ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension +- * +- * $Id: tonal.c,v 1.1 1996/02/14 04:04:23 rowlands Exp $ +- * +- * $Log: tonal.c,v $ +- * Revision 1.1 1996/02/14 04:04:23 rowlands +- * Initial revision +- * +- * Received from Mike Coleman +- **********************************************************************/ +- /********************************************************************** +- * date programmers comment * +- * 2/25/91 Douglas Wong start of version 1.1 records * +- * 3/06/91 Douglas Wong rename: setup.h to endef.h * +- * updated I_psycho_one and II_psycho_one* +- * 3/11/91 W. J. Carter Added Douglas Wong's updates dated * +- * 3/9/91 for I_Psycho_One() and for * +- * II_Psycho_One(). * +- * 5/10/91 W. Joseph Carter Ported to Macintosh and Unix. * +- * Located and fixed numerous software * +- * bugs and table data errors. * +- * 6/11/91 Davis Pan corrected several bugs * +- * based on comments from H. Fuchs * +- * 01jul91 dpwe (Aware Inc.) Made pow() args float * +- * Removed logical bug in I_tonal_label: * +- * Sometimes *tone returned == STOP * +- * 7/10/91 Earle Jennings no change necessary in port to MsDos * +- * 11sep91 dpwe@aware.com Subtracted 90.3dB from II_f_f_t peaks * +- * 10/1/91 Peter W. Farrett Updated II_Psycho_One(),I_Psycho_One()* +- * to include comments. * +- *11/29/91 Masahiro Iwadare Bug fix regarding POWERNORM * +- * fixed several other miscellaneous bugs* +- * 2/11/92 W. Joseph Carter Ported new code to Macintosh. Most * +- * important fixes involved changing * +- * 16-bit ints to long or unsigned in * +- * bit alloc routines for quant of 65535 * +- * and passing proper function args. * +- * Removed "Other Joint Stereo" option * +- * and made bitrate be total channel * +- * bitrate, irrespective of the mode. * +- * Fixed many small bugs & reorganized. * +- * 2/12/92 Masahiro Iwadare Fixed some potential bugs in * +- * Davis Pan subsampling() * +- * 2/25/92 Masahiro Iwadare Fixed some more potential bugs * +- * 6/24/92 Tan Ah Peng Modified window for FFT * +- * (denominator N-1 to N) * +- * Updated all critical band rate & * +- * absolute threshold tables and critical* +- * boundaries for use with Layer I & II * +- * Corrected boundary limits for tonal * +- * component computation * +- * Placement of non-tonal component at * +- * geometric mean of critical band * +- * (previous placement method commented * +- * out - can be used if desired) * +- * 3/01/93 Mike Li Infinite looping fix in noise_label() * +- * 3/19/93 Jens Spille fixed integer overflow problem in * +- * psychoacoutic model 1 * +- * 3/19/93 Giorgio Dimino modifications to better account for * +- * tonal and non-tonal components * +- * 5/28/93 Sriram Jayasimha "London" mod. to psychoacoustic model1* +- * 8/05/93 Masahiro Iwadare noise_label modification "option" * +- * 1/21/94 Seymore Shlien fixed another infinite looping problem* +- * 7/12/95 Soeren H. Nielsen Changes for LSF, new tables * +- **********************************************************************/ +- +- #include "common.h" +- #include "encoder.h" +- #define LONDON /* enable "LONDON" modification */ +- #define MAKE_SENSE /* enable "MAKE_SENSE" modification */ +- #define MI_OPTION /* enable "MI_OPTION" modification */ +- /********************************************************************** +- * +- * This module implements the psychoacoustic model I for the +- * MPEG encoder layer II. It uses simplified tonal and noise masking +- * threshold analysis to generate SMR for the encoder bit allocation +- * routine. +- * +- **********************************************************************/ +- +- int crit_band; +- int FAR *cbound; +- int sub_size; +- +- void read_cbound(lay,freq) /* this function reads in critical */ +- int lay, freq; /* band boundaries */ +- { +- int i,j,k; +- FILE *fp; +- char r[16], t[80]; +- +- strcpy(r, "2cb1"); +- r[0] = (char) lay + '0'; +- r[3] = (char) freq + '0'; +- if( !(fp = OpenTableFile(r)) ){ /* check boundary values */ +- printf("Please check %s boundary table\n",r); +- exit(1); +- } +- fgets(t,80,fp); /* read input for critical bands */ +- sscanf(t,"%d\n",&crit_band); +- cbound = (int FAR *) mem_alloc(sizeof(int) * crit_band, "cbound"); +- for(i=0;i<crit_band;i++){ /* continue to read input for */ +- fgets(t,80,fp); /* critical band boundaries */ +- sscanf(t,"%d %d\n",&j, &k); +- if(i==j) cbound[j] = k; +- else { /* error */ +- printf("Please check index %d in cbound table %s\n",i,r); +- exit(1); +- } +- } +- fclose(fp); +- } +- +- void read_freq_band(ltg,lay,freq) /* this function reads in */ +- int lay, freq; /* frequency bands and bark */ +- g_ptr FAR *ltg; /* values */ +- { +- int i,j, k; +- double b,c; +- FILE *fp; +- char r[16], t[80]; +- +- strcpy(r, "2th1"); +- r[0] = (char) lay + '0'; +- r[3] = (char) freq + '0'; +- if( !(fp = OpenTableFile(r)) ){ /* check freq. values */ +- printf("Please check frequency and cband table %s\n",r); +- exit(1); +- } +- fgets(t,80,fp); /* read input for freq. subbands */ +- sscanf(t,"%d\n",&sub_size); +- *ltg = (g_ptr FAR ) mem_alloc(sizeof(g_thres) * sub_size, "ltg"); +- (*ltg)[0].line = 0; /* initialize global masking threshold */ +- (*ltg)[0].bark = 0; +- (*ltg)[0].hear = 0; +- for(i=1;i<sub_size;i++){ /* continue to read freq. subband */ +- fgets(t,80,fp); /* and assign */ +- sscanf(t,"%d %d %lf %lf\n",&j, &k, &b, &c); +- if(i == j){ +- (*ltg)[j].line = k; +- (*ltg)[j].bark = b; +- (*ltg)[j].hear = c; +- } +- else { /* error */ +- printf("Please check index %d in freq-cb table %s\n",i,r); +- exit(1); +- } +- } +- fclose(fp); +- } +- +- void make_map(power, ltg) /* this function calculates the */ +- mask FAR power[HAN_SIZE]; /* global masking threshold */ +- g_thres FAR *ltg; +- { +- int i,j; +- +- for(i=1;i<sub_size;i++) for(j=ltg[i-1].line;j<=ltg[i].line;j++) +- power[j].map = i; +- } +- +- double add_db(a,b) +- double a,b; +- { +- a = pow(10.0,a/10.0); +- b = pow(10.0,b/10.0); +- return 10 * log10(a+b); +- } +- +- /**************************************************************** +- * +- * Fast Fourier transform of the input samples. +- * +- ****************************************************************/ +- +- void II_f_f_t(sample, power) /* this function calculates an */ +- double FAR sample[FFT_SIZE]; /* FFT analysis for the freq. */ +- mask FAR power[HAN_SIZE]; /* domain */ +- { +- int i,j,k,L,l=0; +- int ip, le, le1; +- double t_r, t_i, u_r, u_i; +- static int M, MM1, init = 0, N; +- double *x_r, *x_i, *energy; +- static int *rev; +- static double *w_r, *w_i; +- +- x_r = (double *) mem_alloc(sizeof(DFFT), "x_r"); +- x_i = (double *) mem_alloc(sizeof(DFFT), "x_i"); +- energy = (double *) mem_alloc(sizeof(DFFT), "energy"); +- for(i=0;i<FFT_SIZE;i++) x_r[i] = x_i[i] = energy[i] = 0; +- if(!init){ +- rev = (int *) mem_alloc(sizeof(IFFT), "rev"); +- w_r = (double *) mem_alloc(sizeof(D10), "w_r"); +- w_i = (double *) mem_alloc(sizeof(D10), "w_i"); +- M = 10; +- MM1 = 9; +- N = FFT_SIZE; +- for(L=0;L<M;L++){ +- le = 1 << (M-L); +- le1 = le >> 1; +- w_r[L] = cos(PI/le1); +- w_i[L] = -sin(PI/le1); +- } +- for(i=0;i<FFT_SIZE;rev[i] = l,i++) for(j=0,l=0;j<10;j++){ +- k=(i>>j) & 1; +- l |= (k<<(9-j)); +- } +- init = 1; +- } +- memcpy( (char *) x_r, (char *) sample, sizeof(double) * FFT_SIZE); +- for(L=0;L<MM1;L++){ +- le = 1 << (M-L); +- le1 = le >> 1; +- u_r = 1; +- u_i = 0; +- for(j=0;j<le1;j++){ +- for(i=j;i<N;i+=le){ +- ip = i + le1; +- t_r = x_r[i] + x_r[ip]; +- t_i = x_i[i] + x_i[ip]; +- x_r[ip] = x_r[i] - x_r[ip]; +- x_i[ip] = x_i[i] - x_i[ip]; +- x_r[i] = t_r; +- x_i[i] = t_i; +- t_r = x_r[ip]; +- x_r[ip] = x_r[ip] * u_r - x_i[ip] * u_i; +- x_i[ip] = x_i[ip] * u_r + t_r * u_i; +- } +- t_r = u_r; +- u_r = u_r * w_r[L] - u_i * w_i[L]; +- u_i = u_i * w_r[L] + t_r * w_i[L]; +- } +- } +- for(i=0;i<N;i+=2){ +- ip = i + 1; +- t_r = x_r[i] + x_r[ip]; +- t_i = x_i[i] + x_i[ip]; +- x_r[ip] = x_r[i] - x_r[ip]; +- x_i[ip] = x_i[i] - x_i[ip]; +- x_r[i] = t_r; +- x_i[i] = t_i; +- energy[i] = x_r[i] * x_r[i] + x_i[i] * x_i[i]; +- } +- for(i=0;i<FFT_SIZE;i++) if(i<rev[i]){ +- t_r = energy[i]; +- energy[i] = energy[rev[i]]; +- energy[rev[i]] = t_r; +- } +- for(i=0;i<HAN_SIZE;i++){ /* calculate power density spectrum */ +- if (energy[i] < 1E-20) energy[i] = 1E-20; +- power[i].x = 10 * log10(energy[i]) + POWERNORM; +- power[i].next = STOP; +- power[i].type = FALSE; +- } +- mem_free((void **) &x_r); +- mem_free((void **) &x_i); +- mem_free((void **) &energy); +- } +- +- /**************************************************************** +- * +- * Window the incoming audio signal. +- * +- ****************************************************************/ +- +- void II_hann_win(sample) /* this function calculates a */ +- double FAR sample[FFT_SIZE]; /* Hann window for PCM (input) */ +- { /* samples for a 1024-pt. FFT */ +- register int i; +- register double sqrt_8_over_3; +- static int init = 0; +- static double FAR *window; +- +- if(!init){ /* calculate window function for the Fourier transform */ +- window = (double FAR *) mem_alloc(sizeof(DFFT), "window"); +- sqrt_8_over_3 = pow(8.0/3.0, 0.5); +- for(i=0;i<FFT_SIZE;i++){ +- /* Hann window formula */ +- window[i]=sqrt_8_over_3*0.5*(1-cos(2.0*PI*i/(FFT_SIZE)))/FFT_SIZE; +- } +- init = 1; +- } +- for(i=0;i<FFT_SIZE;i++) sample[i] *= window[i]; +- } +- +- /******************************************************************* +- * +- * This function finds the maximum spectral component in each +- * subband and return them to the encoder for time-domain threshold +- * determination. +- * +- *******************************************************************/ +- #ifndef LONDON +- void II_pick_max(power, spike) +- double FAR spike[SBLIMIT]; +- mask FAR power[HAN_SIZE]; +- { +- double max; +- int i,j; +- +- for(i=0;i<HAN_SIZE;spike[i>>4] = max, i+=16) /* calculate the */ +- for(j=0, max = DBMIN;j<16;j++) /* maximum spectral */ +- max = (max>power[i+j].x) ? max : power[i+j].x; /* component in each */ +- } /* subband from bound */ +- /* 4-16 */ +- #else +- void II_pick_max(power, spike) +- double FAR spike[SBLIMIT]; +- mask FAR power[HAN_SIZE]; +- { +- double sum; +- int i,j; +- +- for(i=0;i<HAN_SIZE;spike[i>>4] = 10.0*log10(sum), i+=16) +- /* calculate the */ +- for(j=0, sum = pow(10.0,0.1*DBMIN);j<16;j++) /* sum of spectral */ +- sum += pow(10.0,0.1*power[i+j].x); /* component in each */ +- } /* subband from bound */ +- /* 4-16 */ +- #endif +- +- /**************************************************************** +- * +- * This function labels the tonal component in the power +- * spectrum. +- * +- ****************************************************************/ +- +- void II_tonal_label(power, tone) /* this function extracts (tonal) */ +- mask FAR power[HAN_SIZE]; /* sinusoidals from the spectrum */ +- int *tone; +- { +- int i,j, last = LAST, first, run, last_but_one = LAST; /* dpwe */ +- double max; +- +- *tone = LAST; +- for(i=2;i<HAN_SIZE-12;i++){ +- if(power[i].x>power[i-1].x && power[i].x>=power[i+1].x){ +- power[i].type = TONE; +- power[i].next = LAST; +- if(last != LAST) power[last].next = i; +- else first = *tone = i; +- last = i; +- } +- } +- last = LAST; +- first = *tone; +- *tone = LAST; +- while(first != LAST){ /* the conditions for the tonal */ +- if(first<3 || first>500) run = 0;/* otherwise k+/-j will be out of bounds */ +- else if(first<63) run = 2; /* components in layer II, which */ +- else if(first<127) run = 3; /* are the boundaries for calc. */ +- else if(first<255) run = 6; /* the tonal components */ +- else run = 12; +- max = power[first].x - 7; /* after calculation of tonal */ +- for(j=2;j<=run;j++) /* components, set to local max */ +- if(max < power[first-j].x || max < power[first+j].x){ +- power[first].type = FALSE; +- break; +- } +- if(power[first].type == TONE){ /* extract tonal components */ +- int help=first; +- if(*tone==LAST) *tone = first; +- while((power[help].next!=LAST)&&(power[help].next-first)<=run) +- help=power[help].next; +- help=power[help].next; +- power[first].next=help; +- if((first-last)<=run){ +- if(last_but_one != LAST) power[last_but_one].next=first; +- } +- if(first>1 && first<500){ /* calculate the sum of the */ +- double tmp; /* powers of the components */ +- tmp = add_db(power[first-1].x, power[first+1].x); +- power[first].x = add_db(power[first].x, tmp); +- } +- for(j=1;j<=run;j++){ +- power[first-j].x = power[first+j].x = DBMIN; +- power[first-j].next = power[first+j].next = STOP; +- power[first-j].type = power[first+j].type = FALSE; +- } +- last_but_one=last; +- last = first; +- first = power[first].next; +- } +- else { +- int ll; +- if(last == LAST); /* *tone = power[first].next; dpwe */ +- else power[last].next = power[first].next; +- ll = first; +- first = power[first].next; +- power[ll].next = STOP; +- } +- } +- } +- +- /**************************************************************** +- * +- * This function groups all the remaining non-tonal +- * spectral lines into critical band where they are replaced by +- * one single line. +- * +- ****************************************************************/ +- +- void noise_label(power, noise, ltg) +- g_thres FAR *ltg; +- mask FAR *power; +- int *noise; +- { +- int i,j, centre, last = LAST; +- double index, weight, sum; +- /* calculate the remaining spectral */ +- for(i=0;i<crit_band-1;i++){ /* lines for non-tonal components */ +- for(j=cbound[i],weight = 0.0,sum = DBMIN;j<cbound[i+1];j++){ +- if(power[j].type != TONE){ +- if(power[j].x != DBMIN){ +- sum = add_db(power[j].x,sum); +- /* the line below and others under the "MAKE_SENSE" condition are an alternate +- interpretation of "geometric mean". This approach may make more sense but +- it has not been tested with hardware. */ +- #ifdef MAKE_SENSE +- /* weight += pow(10.0, power[j].x/10.0) * (ltg[power[j].map].bark-i); +- bad code [SS] 21-1-93 +- */ +- weight += pow(10.0,power[j].x/10.0) * (double) (j-cbound[i]) / +- (double) (cbound[i+1]-cbound[i]); /* correction */ +- #endif +- power[j].x = DBMIN; +- } +- } /* check to see if the spectral line is low dB, and if */ +- } /* so replace the center of the critical band, which is */ +- /* the center freq. of the noise component */ +- +- #ifdef MAKE_SENSE +- if(sum <= DBMIN) centre = (cbound[i+1]+cbound[i]) /2; +- else { +- index = weight/pow(10.0,sum/10.0); +- centre = cbound[i] + (int) (index * (double) (cbound[i+1]-cbound[i]) ); +- } +- #else +- index = (double)( ((double)cbound[i]) * ((double)(cbound[i+1]-1)) ); +- centre = (int)(pow(index,0.5)+0.5); +- #endif +- +- /* locate next non-tonal component until finished; */ +- /* add to list of non-tonal components */ +- #ifdef MI_OPTION +- /* Masahiro Iwadare's fix for infinite looping problem? */ +- if(power[centre].type == TONE) +- if (power[centre+1].type == TONE) centre++; else centre--; +- #else +- /* Mike Li's fix for infinite looping problem */ +- if(power[centre].type == FALSE) centre++; +- +- if(power[centre].type == NOISE){ +- if(power[centre].x >= ltg[power[i].map].hear){ +- if(sum >= ltg[power[i].map].hear) sum = add_db(power[j].x,sum); +- else +- sum = power[centre].x; +- } +- } +- #endif +- if(last == LAST) *noise = centre; +- else { +- power[centre].next = LAST; +- power[last].next = centre; +- } +- power[centre].x = sum; +- power[centre].type = NOISE; +- last = centre; +- } +- } +- +- /**************************************************************** +- * +- * This function reduces the number of noise and tonal +- * component for further threshold analysis. +- * +- ****************************************************************/ +- +- void subsampling(power, ltg, tone, noise) +- mask FAR power[HAN_SIZE]; +- g_thres FAR *ltg; +- int *tone, *noise; +- { +- int i, old; +- +- i = *tone; old = STOP; /* calculate tonal components for */ +- while(i!=LAST){ /* reduction of spectral lines */ +- if(power[i].x < ltg[power[i].map].hear){ +- power[i].type = FALSE; +- power[i].x = DBMIN; +- if(old == STOP) *tone = power[i].next; +- else power[old].next = power[i].next; +- } +- else old = i; +- i = power[i].next; +- } +- i = *noise; old = STOP; /* calculate non-tonal components for */ +- while(i!=LAST){ /* reduction of spectral lines */ +- if(power[i].x < ltg[power[i].map].hear){ +- power[i].type = FALSE; +- power[i].x = DBMIN; +- if(old == STOP) *noise = power[i].next; +- else power[old].next = power[i].next; +- } +- else old = i; +- i = power[i].next; +- } +- i = *tone; old = STOP; +- while(i != LAST){ /* if more than one */ +- if(power[i].next == LAST)break; /* tonal component */ +- if(ltg[power[power[i].next].map].bark - /* is less than .5 */ +- ltg[power[i].map].bark < 0.5) { /* bark, take the */ +- if(power[power[i].next].x > power[i].x ){/* maximum */ +- if(old == STOP) *tone = power[i].next; +- else power[old].next = power[i].next; +- power[i].type = FALSE; +- power[i].x = DBMIN; +- i = power[i].next; +- } +- else { +- power[power[i].next].type = FALSE; +- power[power[i].next].x = DBMIN; +- power[i].next = power[power[i].next].next; +- old = i; +- } +- } +- else { +- old = i; +- i = power[i].next; +- } +- } +- } +- +- /**************************************************************** +- * +- * This function calculates the individual threshold and +- * sum with the quiet threshold to find the global threshold. +- * +- ****************************************************************/ +- +- void threshold(power, ltg, tone, noise, bit_rate) +- mask FAR power[HAN_SIZE]; +- g_thres FAR *ltg; +- int *tone, *noise, bit_rate; +- { +- int k, t; +- double dz, tmps, vf; +- +- for(k=1;k<sub_size;k++){ +- ltg[k].x = DBMIN; +- t = *tone; /* calculate individual masking threshold for */ +- while(t != LAST){ /* components in order to find the global */ +- if(ltg[k].bark-ltg[power[t].map].bark >= -3.0 && /*threshold (LTG)*/ +- ltg[k].bark-ltg[power[t].map].bark <8.0){ +- dz = ltg[k].bark-ltg[power[t].map].bark; /* distance of bark value*/ +- tmps = -1.525-0.275*ltg[power[t].map].bark - 4.5 + power[t].x; +- /* masking function for lower & upper slopes */ +- if(-3<=dz && dz<-1) vf = 17*(dz+1)-(0.4*power[t].x +6); +- else if(-1<=dz && dz<0) vf = (0.4 *power[t].x + 6) * dz; +- else if(0<=dz && dz<1) vf = (-17*dz); +- else if(1<=dz && dz<8) vf = -(dz-1) * (17-0.15 *power[t].x) - 17; +- tmps += vf; +- ltg[k].x = add_db(ltg[k].x, tmps); +- } +- t = power[t].next; +- } +- +- t = *noise; /* calculate individual masking threshold */ +- while(t != LAST){ /* for non-tonal components to find LTG */ +- if(ltg[k].bark-ltg[power[t].map].bark >= -3.0 && +- ltg[k].bark-ltg[power[t].map].bark <8.0){ +- dz = ltg[k].bark-ltg[power[t].map].bark; /* distance of bark value */ +- tmps = -1.525-0.175*ltg[power[t].map].bark -0.5 + power[t].x; +- /* masking function for lower & upper slopes */ +- if(-3<=dz && dz<-1) vf = 17*(dz+1)-(0.4*power[t].x +6); +- else if(-1<=dz && dz<0) vf = (0.4 *power[t].x + 6) * dz; +- else if(0<=dz && dz<1) vf = (-17*dz); +- else if(1<=dz && dz<8) vf = -(dz-1) * (17-0.15 *power[t].x) - 17; +- tmps += vf; +- ltg[k].x = add_db(ltg[k].x, tmps); +- } +- t = power[t].next; +- } +- if(bit_rate<96)ltg[k].x = add_db(ltg[k].hear, ltg[k].x); +- else ltg[k].x = add_db(ltg[k].hear-12.0, ltg[k].x); +- } +- } +- +- /**************************************************************** +- * +- * This function finds the minimum masking threshold and +- * return the value to the encoder. +- * +- ****************************************************************/ +- +- void II_minimum_mask(ltg,ltmin,sblimit) +- g_thres FAR *ltg; +- double FAR ltmin[SBLIMIT]; +- int sblimit; +- { +- double min; +- int i,j; +- +- j=1; +- for(i=0;i<sblimit;i++) +- if(j>=sub_size-1) /* check subband limit, and */ +- ltmin[i] = ltg[sub_size-1].hear; /* calculate the minimum masking */ +- else { /* level of LTMIN for each subband*/ +- min = ltg[j].x; +- while(ltg[j].line>>4 == i && j < sub_size){ +- if(min>ltg[j].x) min = ltg[j].x; +- j++; +- } +- ltmin[i] = min; +- } +- } +- +- /***************************************************************** +- * +- * This procedure is called in musicin to pick out the +- * smaller of the scalefactor or threshold. +- * +- *****************************************************************/ +- +- void II_smr(ltmin, spike, scale, sblimit) +- double FAR spike[SBLIMIT], scale[SBLIMIT], ltmin[SBLIMIT]; +- int sblimit; +- { +- int i; +- double max; +- +- for(i=0;i<sblimit;i++){ /* determine the signal */ +- max = 20 * log10(scale[i] * 32768) - 10; /* level for each subband */ +- if(spike[i]>max) max = spike[i]; /* for the maximum scale */ +- max -= ltmin[i]; /* factors */ +- ltmin[i] = max; +- } +- } +- +- /**************************************************************** +- * +- * This procedure calls all the necessary functions to +- * complete the psychoacoustic analysis. +- * +- ****************************************************************/ +- +- void II_Psycho_One(buffer, scale, ltmin, fr_ps) +- short FAR buffer[2][1152]; +- double FAR scale[2][SBLIMIT], ltmin[2][SBLIMIT]; +- frame_params *fr_ps; +- { +- layer *info = fr_ps->header; +- int stereo = fr_ps->stereo; +- int sblimit = fr_ps->sblimit; +- int k,i, tone=0, noise=0; +- static char init = 0; +- static int off[2] = {256,256}; +- double *sample; +- DSBL *spike; +- static D1408 *fft_buf; +- static mask_ptr FAR power; +- static g_ptr FAR ltg; +- +- sample = (double *) mem_alloc(sizeof(DFFT), "sample"); +- spike = (DSBL *) mem_alloc(sizeof(D2SBL), "spike"); +- /* call functions for critical boundaries, freq. */ +- if(!init){ /* bands, bark values, and mapping */ +- fft_buf = (D1408 *) mem_alloc((long) sizeof(D1408) * 2, "fft_buf"); +- power = (mask_ptr FAR ) mem_alloc(sizeof(mask) * HAN_SIZE, "power"); +- if (info->version == MPEG_AUDIO_ID) { +- read_cbound(info->lay, info->sampling_frequency); +- read_freq_band(<g, info->lay, info->sampling_frequency); +- } else { +- read_cbound(info->lay, info->sampling_frequency + 4); +- read_freq_band(<g, info->lay, info->sampling_frequency + 4); +- } +- make_map(power,ltg); +- for (i=0;i<1408;i++) fft_buf[0][i] = fft_buf[1][i] = 0; +- init = 1; +- } +- for(k=0;k<stereo;k++){ /* check pcm input for 3 blocks of 384 samples */ +- for(i=0;i<1152;i++) fft_buf[k][(i+off[k])%1408]= (double)buffer[k][i]/SCALE; +- for(i=0;i<FFT_SIZE;i++) sample[i] = fft_buf[k][(i+1216+off[k])%1408]; +- off[k] += 1152; +- off[k] %= 1408; +- /* call functions for windowing PCM samples,*/ +- II_hann_win(sample); /* location of spectral components in each */ +- for(i=0;i<HAN_SIZE;i++) power[i].x = DBMIN; /*subband with labeling*/ +- II_f_f_t(sample, power); /*locate remaining non-*/ +- II_pick_max(power, &spike[k][0]); /*tonal sinusoidals, */ +- II_tonal_label(power, &tone); /*reduce noise & tonal */ +- noise_label(power, &noise, ltg); /*components, find */ +- subsampling(power, ltg, &tone, &noise); /*global & minimal */ +- threshold(power, ltg, &tone, &noise, /*threshold, and sgnl- */ +- bitrate[info->version][info->lay-1][info->bitrate_index]/stereo); /*to-mask ratio*/ +- II_minimum_mask(ltg, <min[k][0], sblimit); +- II_smr(<min[k][0], &spike[k][0], &scale[k][0], sblimit); +- } +- mem_free((void **) &sample); +- mem_free((void **) &spike); +- } +- +- /********************************************************************** +- * +- * This module implements the psychoacoustic model I for the +- * MPEG encoder layer I. It uses simplified tonal and noise masking +- * threshold analysis to generate SMR for the encoder bit allocation +- * routine. +- * +- **********************************************************************/ +- +- /**************************************************************** +- * +- * Fast Fourier transform of the input samples. +- * +- ****************************************************************/ +- +- void I_f_f_t(sample, power) /* this function calculates */ +- double FAR sample[FFT_SIZE/2]; /* an FFT analysis for the */ +- mask FAR power[HAN_SIZE/2]; /* freq. domain */ +- { +- int i,j,k,L,l=0; +- int ip, le, le1; +- double t_r, t_i, u_r, u_i; +- static int M, MM1, init = 0, N; +- double *x_r, *x_i, *energy; +- static int *rev; +- static double *w_r, *w_i; +- +- x_r = (double *) mem_alloc(sizeof(DFFT2), "x_r"); +- x_i = (double *) mem_alloc(sizeof(DFFT2), "x_i"); +- energy = (double *) mem_alloc(sizeof(DFFT2), "energy"); +- for(i=0;i<FFT_SIZE/2;i++) x_r[i] = x_i[i] = energy[i] = 0; +- if(!init){ +- rev = (int *) mem_alloc(sizeof(IFFT2), "rev"); +- w_r = (double *) mem_alloc(sizeof(D9), "w_r"); +- w_i = (double *) mem_alloc(sizeof(D9), "w_i"); +- M = 9; +- MM1 = 8; +- N = FFT_SIZE/2; +- for(L=0;L<M;L++){ +- le = 1 << (M-L); +- le1 = le >> 1; +- w_r[L] = cos(PI/le1); +- w_i[L] = -sin(PI/le1); +- } +- for(i=0;i<FFT_SIZE/2;rev[i] = l,i++) for(j=0,l=0;j<9;j++){ +- k=(i>>j) & 1; +- l |= (k<<(8-j)); +- } +- init = 1; +- } +- memcpy( (char *) x_r, (char *) sample, sizeof(double) * FFT_SIZE/2); +- for(L=0;L<MM1;L++){ +- le = 1 << (M-L); +- le1 = le >> 1; +- u_r = 1; +- u_i = 0; +- for(j=0;j<le1;j++){ +- for(i=j;i<N;i+=le){ +- ip = i + le1; +- t_r = x_r[i] + x_r[ip]; +- t_i = x_i[i] + x_i[ip]; +- x_r[ip] = x_r[i] - x_r[ip]; +- x_i[ip] = x_i[i] - x_i[ip]; +- x_r[i] = t_r; +- x_i[i] = t_i; +- t_r = x_r[ip]; +- x_r[ip] = x_r[ip] * u_r - x_i[ip] * u_i; +- x_i[ip] = x_i[ip] * u_r + t_r * u_i; +- } +- t_r = u_r; +- u_r = u_r * w_r[L] - u_i * w_i[L]; +- u_i = u_i * w_r[L] + t_r * w_i[L]; +- } +- } +- for(i=0;i<N;i+=2){ +- ip = i + 1; +- t_r = x_r[i] + x_r[ip]; +- t_i = x_i[i] + x_i[ip]; +- x_r[ip] = x_r[i] - x_r[ip]; +- x_i[ip] = x_i[i] - x_i[ip]; +- x_r[i] = t_r; +- x_i[i] = t_i; +- energy[i] = x_r[i] * x_r[i] + x_i[i] * x_i[i]; +- } +- for(i=0;i<FFT_SIZE/2;i++) if(i<rev[i]){ +- t_r = energy[i]; +- energy[i] = energy[rev[i]]; +- energy[rev[i]] = t_r; +- } +- for(i=0;i<HAN_SIZE/2;i++){ /* calculate power */ +- if(energy[i] < 1E-20) energy[i] = 1E-20; /* density spectrum */ +- power[i].x = 10 * log10(energy[i]) + POWERNORM; +- power[i].next = STOP; +- power[i].type = FALSE; +- } +- mem_free((void **) &x_r); +- mem_free((void **) &x_i); +- mem_free((void **) &energy); +- } +- +- /**************************************************************** +- * +- * Window the incoming audio signal. +- * +- ****************************************************************/ +- +- void I_hann_win(sample) /* this function calculates a */ +- double FAR sample[FFT_SIZE/2]; /* Hann window for PCM (input) */ +- { /* samples for a 512-pt. FFT */ +- register int i; +- register double sqrt_8_over_3; +- static int init = 0; +- static double FAR *window; +- +- if(!init){ /* calculate window function for the Fourier transform */ +- window = (double FAR *) mem_alloc(sizeof(DFFT2), "window"); +- sqrt_8_over_3 = pow(8.0/3.0, 0.5); +- for(i=0;i<FFT_SIZE/2;i++){ +- /* Hann window formula */ +- window[i]=sqrt_8_over_3*0.5*(1-cos(2.0*PI*i/(FFT_SIZE/2)))/(FFT_SIZE/2); +- } +- init = 1; +- } +- for(i=0;i<FFT_SIZE/2;i++) sample[i] *= window[i]; +- } +- +- /******************************************************************* +- * +- * This function finds the maximum spectral component in each +- * subband and return them to the encoder for time-domain threshold +- * determination. +- * +- *******************************************************************/ +- #ifndef LONDON +- void I_pick_max(power, spike) +- double FAR spike[SBLIMIT]; +- mask FAR power[HAN_SIZE/2]; +- { +- double max; +- int i,j; +- +- /* calculate the spectral component in each subband */ +- for(i=0;i<HAN_SIZE/2;spike[i>>3] = max, i+=8) +- for(j=0, max = DBMIN;j<8;j++) max = (max>power[i+j].x) ? max : power[i+j].x; +- } +- #else +- void I_pick_max(power, spike) +- double FAR spike[SBLIMIT]; +- mask FAR power[HAN_SIZE]; +- { +- double sum; +- int i,j; +- +- for(i=0;i<HAN_SIZE/2;spike[i>>3] = 10.0*log10(sum), i+=8) +- /* calculate the */ +- for(j=0, sum = pow(10.0,0.1*DBMIN);j<8;j++) /* sum of spectral */ +- sum += pow(10.0,0.1*power[i+j].x); /* component in each */ +- } /* subband from bound */ +- #endif +- /**************************************************************** +- * +- * This function labels the tonal component in the power +- * spectrum. +- * +- ****************************************************************/ +- +- void I_tonal_label(power, tone) /* this function extracts */ +- mask FAR power[HAN_SIZE/2]; /* (tonal) sinusoidals from */ +- int *tone; /* the spectrum */ +- { +- int i,j, last = LAST, first, run; +- double max; +- int last_but_one= LAST; +- +- *tone = LAST; +- for(i=2;i<HAN_SIZE/2-6;i++){ +- if(power[i].x>power[i-1].x && power[i].x>=power[i+1].x){ +- power[i].type = TONE; +- power[i].next = LAST; +- if(last != LAST) power[last].next = i; +- else first = *tone = i; +- last = i; +- } +- } +- last = LAST; +- first = *tone; +- *tone = LAST; +- while(first != LAST){ /* conditions for the tonal */ +- if(first<3 || first>250) run = 0; /* otherwise k+/-j will be out of bounds*/ +- else if(first<63) run = 2; /* components in layer I, which */ +- else if(first<127) run = 3; /* are the boundaries for calc. */ +- else run = 6; /* the tonal components */ +- max = power[first].x - 7; +- for(j=2;j<=run;j++) /* after calc. of tonal components, set to loc.*/ +- if(max < power[first-j].x || max < power[first+j].x){ /* max */ +- power[first].type = FALSE; +- break; +- } +- if(power[first].type == TONE){ /* extract tonal components */ +- int help=first; +- if(*tone == LAST) *tone = first; +- while((power[help].next!=LAST)&&(power[help].next-first)<=run) +- help=power[help].next; +- help=power[help].next; +- power[first].next=help; +- if((first-last)<=run){ +- if(last_but_one != LAST) power[last_but_one].next=first; +- } +- if(first>1 && first<255){ /* calculate the sum of the */ +- double tmp; /* powers of the components */ +- tmp = add_db(power[first-1].x, power[first+1].x); +- power[first].x = add_db(power[first].x, tmp); +- } +- for(j=1;j<=run;j++){ +- power[first-j].x = power[first+j].x = DBMIN; +- power[first-j].next = power[first+j].next = STOP; /*dpwe: 2nd was .x*/ +- power[first-j].type = power[first+j].type = FALSE; +- } +- last_but_one=last; +- last = first; +- first = power[first].next; +- } +- else { +- int ll; +- if(last == LAST) ; /* *tone = power[first].next; dpwe */ +- else power[last].next = power[first].next; +- ll = first; +- first = power[first].next; +- power[ll].next = STOP; +- } +- } +- } +- +- /**************************************************************** +- * +- * This function finds the minimum masking threshold and +- * return the value to the encoder. +- * +- ****************************************************************/ +- +- void I_minimum_mask(ltg,ltmin) +- g_thres FAR *ltg; +- double FAR ltmin[SBLIMIT]; +- { +- double min; +- int i,j; +- +- j=1; +- for(i=0;i<SBLIMIT;i++) +- if(j>=sub_size-1) /* check subband limit, and */ +- ltmin[i] = ltg[sub_size-1].hear; /* calculate the minimum masking */ +- else { /* level of LTMIN for each subband*/ +- min = ltg[j].x; +- while(ltg[j].line>>3 == i && j < sub_size){ +- if (min>ltg[j].x) min = ltg[j].x; +- j++; +- } +- ltmin[i] = min; +- } +- } +- +- /***************************************************************** +- * +- * This procedure is called in musicin to pick out the +- * smaller of the scalefactor or threshold. +- * +- *****************************************************************/ +- +- void I_smr(ltmin, spike, scale) +- double FAR spike[SBLIMIT], scale[SBLIMIT], ltmin[SBLIMIT]; +- { +- int i; +- double max; +- +- for(i=0;i<SBLIMIT;i++){ /* determine the signal */ +- max = 20 * log10(scale[i] * 32768) - 10; /* level for each subband */ +- if(spike[i]>max) max = spike[i]; /* for the scalefactor */ +- max -= ltmin[i]; +- ltmin[i] = max; +- } +- } +- +- /**************************************************************** +- * +- * This procedure calls all the necessary functions to +- * complete the psychoacoustic analysis. +- * +- ****************************************************************/ +- +- void I_Psycho_One(buffer, scale, ltmin, fr_ps) +- short FAR buffer[2][1152]; +- double FAR scale[2][SBLIMIT], ltmin[2][SBLIMIT]; +- frame_params *fr_ps; +- { +- int stereo = fr_ps->stereo; +- the_layer info = fr_ps->header; +- int k,i, tone=0, noise=0; +- static char init = 0; +- static int off[2] = {256,256}; +- double *sample; +- DSBL *spike; +- static D640 *fft_buf; +- static mask_ptr FAR power; +- static g_ptr FAR ltg; +- +- sample = (double *) mem_alloc(sizeof(DFFT2), "sample"); +- spike = (DSBL *) mem_alloc(sizeof(D2SBL), "spike"); +- /* call functions for critical boundaries, freq. */ +- if(!init){ /* bands, bark values, and mapping */ +- fft_buf = (D640 *) mem_alloc(sizeof(D640) * 2, "fft_buf"); +- power = (mask_ptr FAR ) mem_alloc(sizeof(mask) * HAN_SIZE/2, "power"); +- if (info->version == MPEG_AUDIO_ID) { +- read_cbound(info->lay, info->sampling_frequency); +- read_freq_band(<g, info->lay, info->sampling_frequency); +- } else { +- read_cbound(info->lay, info->sampling_frequency + 4); +- read_freq_band(<g, info->lay, info->sampling_frequency + 4); +- } +- make_map(power,ltg); +- for(i=0;i<640;i++) fft_buf[0][i] = fft_buf[1][i] = 0; +- init = 1; +- } +- for(k=0;k<stereo;k++){ /* check PCM input for a block of */ +- for(i=0;i<384;i++) /* 384 samples for a 512-pt. FFT */ +- fft_buf[k][(i+off[k])%640]= (double) buffer[k][i]/SCALE; +- for(i=0;i<FFT_SIZE/2;i++) +- sample[i] = fft_buf[k][(i+448+off[k])%640]; +- off[k] += 384; +- off[k] %= 640; +- /* call functions for windowing PCM samples, */ +- I_hann_win(sample); /* location of spectral components in each */ +- for(i=0;i<HAN_SIZE/2;i++) power[i].x = DBMIN; /* subband with */ +- I_f_f_t(sample, power); /* labeling, locate remaining */ +- I_pick_max(power, &spike[k][0]); /* non-tonal sinusoidals, */ +- I_tonal_label(power, &tone); /* reduce noise & tonal com., */ +- noise_label(power, &noise, ltg); /* find global & minimal */ +- subsampling(power, ltg, &tone, &noise); /* threshold, and sgnl- */ +- threshold(power, ltg, &tone, &noise, /* to-mask ratio */ +- bitrate[info->version][info->lay-1][info->bitrate_index]/stereo); +- I_minimum_mask(ltg, <min[k][0]); +- I_smr(<min[k][0], &spike[k][0], &scale[k][0]); +- } +- mem_free((void **) &sample); +- mem_free((void **) &spike); +- } +--- 0 ---- +diff -r -c -N encoder/util.c lame3.70/util.c +*** encoder/util.c Wed Dec 31 17:00:00 1969 +--- lame3.70/util.c Thu Apr 6 12:50:52 2000 +*************** +*** 0 **** +--- 1,342 ---- ++ #include "util.h" ++ #include <assert.h> ++ ++ /*********************************************************************** ++ * ++ * Global Variable Definitions ++ * ++ ***********************************************************************/ ++ ++ ++ /* 1: MPEG-1, 0: MPEG-2 LSF, 1995-07-11 shn */ ++ FLOAT8 s_freq_table[2][4] = {{22.05, 24, 16, 0}, {44.1, 48, 32, 0}}; ++ ++ /* 1: MPEG-1, 0: MPEG-2 LSF, 1995-07-11 shn */ ++ int bitrate_table[2][15] = { ++ {0,8,16,24,32,40,48,56,64,80,96,112,128,144,160}, ++ {0,32,40,48,56,64,80,96,112,128,160,192,224,256,320}}; ++ ++ ++ enum byte_order NativeByteOrder = order_unknown; ++ ++ /*********************************************************************** ++ * ++ * Global Function Definitions ++ * ++ ***********************************************************************/ ++ ++ ++ /*********************************************************************** ++ * compute bitsperframe and mean_bits for a layer III frame ++ **********************************************************************/ ++ void getframebits(lame_global_flags *gfp,int *bitsPerFrame, int *mean_bits) { ++ int whole_SpF; ++ FLOAT8 bit_rate,samp; ++ int bitsPerSlot; ++ int sideinfo_len; ++ ++ samp = gfp->out_samplerate/1000.0; ++ bit_rate = bitrate_table[gfp->version][gfp->bitrate_index]; ++ bitsPerSlot = 8; ++ ++ /* determine the mean bitrate for main data */ ++ sideinfo_len = 32; ++ if ( gfp->version == 1 ) ++ { /* MPEG 1 */ ++ if ( gfp->stereo == 1 ) ++ sideinfo_len += 136; ++ else ++ sideinfo_len += 256; ++ } ++ else ++ { /* MPEG 2 */ ++ if ( gfp->stereo == 1 ) ++ sideinfo_len += 72; ++ else ++ sideinfo_len += 136; ++ } ++ ++ if (gfp->error_protection) sideinfo_len += 16; ++ ++ /* -f fast-math option causes some strange rounding here, be carefull: */ ++ whole_SpF = floor( (gfp->framesize /samp)*(bit_rate / (FLOAT8)bitsPerSlot) + 1e-9); ++ *bitsPerFrame = 8 * whole_SpF + (gfp->padding * 8); ++ *mean_bits = (*bitsPerFrame - sideinfo_len) / gfp->mode_gr; ++ } ++ ++ ++ ++ ++ void display_bitrates(FILE *out_fh) ++ { ++ int index,version; ++ ++ version = 1; ++ fprintf(out_fh,"\n"); ++ fprintf(out_fh,"MPEG1 samplerates(kHz): 32 44.1 48 \n"); ++ ++ fprintf(out_fh,"bitrates(kbs): "); ++ for (index=1;index<15;index++) { ++ fprintf(out_fh,"%i ",bitrate_table[version][index]); ++ } ++ fprintf(out_fh,"\n"); ++ ++ ++ version = 0; ++ fprintf(out_fh,"\n"); ++ fprintf(out_fh,"MPEG2 samplerates(kHz): 16 22.05 24 \n"); ++ fprintf(out_fh,"bitrates(kbs): "); ++ for (index=1;index<15;index++) { ++ fprintf(out_fh,"%i ",bitrate_table[version][index]); ++ } ++ fprintf(out_fh,"\n"); ++ } ++ ++ ++ int BitrateIndex( ++ int bRate, /* legal rates from 32 to 448 */ ++ int version, /* MPEG-1 or MPEG-2 LSF */ ++ int samplerate) /* convert bitrate in kbps to index */ ++ { ++ int index = 0; ++ int found = 0; ++ ++ while(!found && index<15) { ++ if(bitrate_table[version][index] == bRate) ++ found = 1; ++ else ++ ++index; ++ } ++ if(found) ++ return(index); ++ else { ++ fprintf(stderr,"Bitrate %dkbs not legal for %iHz output sampling.\n", ++ bRate, samplerate); ++ return(-1); /* Error! */ ++ } ++ } ++ ++ int SmpFrqIndex( /* convert samp frq in Hz to index */ ++ long sRate, /* legal rates 16000, 22050, 24000, 32000, 44100, 48000 */ ++ int *version) ++ { ++ /* Assign default value */ ++ *version=0; ++ ++ if (sRate == 44100L) { ++ *version = 1; return(0); ++ } ++ else if (sRate == 48000L) { ++ *version = 1; return(1); ++ } ++ else if (sRate == 32000L) { ++ *version = 1; return(2); ++ } ++ else if (sRate == 24000L) { ++ *version = 0; return(1); ++ } ++ else if (sRate == 22050L) { ++ *version = 0; return(0); ++ } ++ else if (sRate == 16000L) { ++ *version = 0; return(2); ++ } ++ else { ++ fprintf(stderr, "SmpFrqIndex: %ldHz is not a legal sample rate\n", sRate); ++ return(-1); /* Error! */ ++ } ++ } ++ ++ /******************************************************************************* ++ * ++ * Allocate number of bytes of memory equal to "block". ++ * ++ *******************************************************************************/ ++ /* exit(0) changed to exit(1) on memory allocation ++ * error -- 1999/06 Alvaro Martinez Echevarria */ ++ ++ void *mem_alloc(unsigned long block, char *item) ++ { ++ ++ void *ptr; ++ ++ /* what kind of shit does ISO put out? */ ++ ptr = (void *) malloc((size_t) block /* <<1 */ ); /* allocate twice as much memory as needed. fixes dodgy ++ memory problem on most systems */ ++ ++ ++ if (ptr != NULL) { ++ memset(ptr, 0, (size_t) block); ++ } else { ++ fprintf(stderr,"Unable to allocate %s\n", item); ++ exit(1); ++ } ++ return(ptr); ++ } ++ ++ ++ ++ /***************************************************************************** ++ * ++ * Routines to determine byte order and swap bytes ++ * ++ *****************************************************************************/ ++ ++ enum byte_order DetermineByteOrder(void) ++ { ++ char s[ sizeof(long) + 1 ]; ++ union ++ { ++ long longval; ++ char charval[ sizeof(long) ]; ++ } probe; ++ probe.longval = 0x41424344L; /* ABCD in ASCII */ ++ strncpy( s, probe.charval, sizeof(long) ); ++ s[ sizeof(long) ] = '\0'; ++ /* fprintf( stderr, "byte order is %s\n", s ); */ ++ if ( strcmp(s, "ABCD") == 0 ) ++ return order_bigEndian; ++ else ++ if ( strcmp(s, "DCBA") == 0 ) ++ return order_littleEndian; ++ else ++ return order_unknown; ++ } ++ ++ void SwapBytesInWords( short *loc, int words ) ++ { ++ int i; ++ short thisval; ++ char *dst, *src; ++ src = (char *) &thisval; ++ for ( i = 0; i < words; i++ ) ++ { ++ thisval = *loc; ++ dst = (char *) loc++; ++ dst[0] = src[1]; ++ dst[1] = src[0]; ++ } ++ } ++ ++ ++ ++ ++ ++ ++ /***************************************************************************** ++ * ++ * bit_stream.c package ++ * Author: Jean-Georges Fritsch, C-Cube Microsystems ++ * ++ *****************************************************************************/ ++ ++ /******************************************************************** ++ This package provides functions to write (exclusive or read) ++ information from (exclusive or to) the bit stream. ++ ++ If the bit stream is opened in read mode only the get functions are ++ available. If the bit stream is opened in write mode only the put ++ functions are available. ++ ********************************************************************/ ++ ++ /*alloc_buffer(); open and initialize the buffer; */ ++ /*desalloc_buffer(); empty and close the buffer */ ++ /*back_track_buffer(); goes back N bits in the buffer */ ++ /*unsigned int get1bit(); read 1 bit from the bit stream */ ++ /*unsigned long look_ahead(); grep the next N bits in the bit stream without*/ ++ /* changing the buffer pointer */ ++ /*putbits(); write N bits from the bit stream */ ++ /*int seek_sync(); return 1 if a sync word was found in the bit stream */ ++ /* otherwise returns 0 */ ++ ++ ++ ++ void empty_buffer(Bit_stream_struc *bs) ++ { ++ int minimum=1+bs->buf_byte_idx; /* end of the buffer to empty */ ++ if (bs->buf_size-minimum <= 0) return; ++ bs->buf_byte_idx = bs->buf_size -1; ++ bs->buf_bit_idx = 8; ++ ++ bs->buf[bs->buf_byte_idx] = 0; /* what does this do? */ ++ ++ } ++ int copy_buffer(char *buffer,int size,Bit_stream_struc *bs) ++ { ++ int i,j=0; ++ if (size!=0 && (bs->buf_size-1 - bs->buf_byte_idx) > size ) return -1; ++ for (i=bs->buf_size-1 ; i > bs->buf_byte_idx ; (i-- )) ++ buffer[j++]=bs->buf[i]; ++ assert(j == (bs->buf_size-1 - bs->buf_byte_idx)); ++ empty_buffer(bs); /* empty buffer, (changes bs->buf_size) */ ++ return j; ++ } ++ ++ ++ ++ ++ ++ void init_bit_stream_w(Bit_stream_struc* bs) ++ { ++ alloc_buffer(bs, BUFFER_SIZE); ++ bs->buf_byte_idx = BUFFER_SIZE-1; ++ bs->buf_bit_idx=8; ++ bs->totbit=0; ++ } ++ ++ ++ /*open and initialize the buffer; */ ++ void alloc_buffer( ++ Bit_stream_struc *bs, /* bit stream structure */ ++ int size) ++ { ++ bs->buf = (unsigned char *) ++ mem_alloc((unsigned long) (size * sizeof(unsigned char)), "buffer"); ++ bs->buf_size = size; ++ } ++ ++ /*empty and close the buffer */ ++ void desalloc_buffer(Bit_stream_struc *bs) /* bit stream structure */ ++ { ++ free(bs->buf); ++ } ++ ++ int putmask[9]={0x0, 0x1, 0x3, 0x7, 0xf, 0x1f, 0x3f, 0x7f, 0xff}; ++ ++ ++ /*write N bits into the bit stream */ ++ void putbits( ++ Bit_stream_struc *bs, /* bit stream structure */ ++ unsigned int val, /* val to write into the buffer */ ++ int N) /* number of bits of val */ ++ { ++ register int j = N; ++ register int k, tmp; ++ ++ if (N > MAX_LENGTH) ++ fprintf(stderr,"Cannot read or write more than %d bits at a time.\n", MAX_LENGTH); ++ ++ bs->totbit += N; ++ while (j > 0) { ++ k = Min(j, bs->buf_bit_idx); ++ tmp = val >> (j-k); ++ bs->buf[bs->buf_byte_idx] |= (tmp&putmask[k]) << (bs->buf_bit_idx-k); ++ bs->buf_bit_idx -= k; ++ if (!bs->buf_bit_idx) { ++ bs->buf_bit_idx = 8; ++ bs->buf_byte_idx--; ++ assert(bs->buf_byte_idx >= 0); ++ bs->buf[bs->buf_byte_idx] = 0; ++ } ++ j -= k; ++ } ++ } ++ ++ ++ ++ /***************************************************************************** ++ * ++ * End of bit_stream.c package ++ * ++ *****************************************************************************/ ++ +diff -r -c -N encoder/util.h lame3.70/util.h +*** encoder/util.h Wed Dec 31 17:00:00 1969 +--- lame3.70/util.h Thu Apr 6 12:50:52 2000 +*************** +*** 0 **** +--- 1,158 ---- ++ #ifndef UTIL_DOT_H ++ #define UTIL_DOT_H ++ /*********************************************************************** ++ * ++ * Global Include Files ++ * ++ ***********************************************************************/ ++ #include "machine.h" ++ #include "encoder.h" ++ #include "lame.h" ++ ++ /*********************************************************************** ++ * ++ * Global Definitions ++ * ++ ***********************************************************************/ ++ ++ /* General Definitions */ ++ #ifndef FALSE ++ #define FALSE 0 ++ #endif ++ ++ #ifndef TRUE ++ #define TRUE 1 ++ #endif ++ ++ #define MAX_U_32_NUM 0xFFFFFFFF ++ ++ #ifndef PI ++ # ifdef M_PI ++ # define PI M_PI ++ # else ++ # define PI 3.14159265358979323846 ++ # endif ++ #endif ++ ++ ++ #ifdef M_LN2 ++ # define LOG2 M_LN2 ++ #else ++ # define LOG2 0.69314718055994530942 ++ #endif ++ ++ #ifdef M_LN10 ++ # define LOG10 M_LN10 ++ #else ++ # define LOG10 2.30258509299404568402 ++ #endif ++ ++ ++ #ifdef M_SQRT2 ++ # define SQRT2 M_SQRT2 ++ #else ++ # define SQRT2 1.41421356237309504880 ++ #endif ++ ++ ++ #define BITS_IN_A_BYTE 8 ++ #define HAN_SIZE 512 ++ #define CRC16_POLYNOMIAL 0x8005 ++ ++ /* MPEG Header Definitions - Mode Values */ ++ ++ #define MPG_MD_STEREO 0 ++ #define MPG_MD_JOINT_STEREO 1 ++ #define MPG_MD_DUAL_CHANNEL 2 ++ #define MPG_MD_MONO 3 ++ ++ /* Mode Extention */ ++ ++ #define MPG_MD_LR_LR 0 ++ #define MPG_MD_LR_I 1 ++ #define MPG_MD_MS_LR 2 ++ #define MPG_MD_MS_I 3 ++ ++ ++ /* "bit_stream.h" Definitions */ ++ ++ #define MINIMUM 4 /* Minimum size of the buffer in bytes */ ++ #define MAX_LENGTH 32 /* Maximum length of word written or ++ read from bit stream */ ++ #define BUFFER_SIZE LAME_MAXMP3BUFFER ++ ++ #define Min(A, B) ((A) < (B) ? (A) : (B)) ++ #define Max(A, B) ((A) > (B) ? (A) : (B)) ++ ++ /*********************************************************************** ++ * ++ * Global Type Definitions ++ * ++ ***********************************************************************/ ++ ++ /* Structure for Reading Layer II Allocation Tables from File */ ++ ++ typedef struct { ++ unsigned int steps; ++ unsigned int bits; ++ unsigned int group; ++ unsigned int quant; ++ } sb_alloc, *alloc_ptr; ++ ++ typedef sb_alloc al_table[SBLIMIT][16]; ++ ++ /* Header Information Structure */ ++ ++ ++ ++ enum byte_order { order_unknown, order_bigEndian, order_littleEndian }; ++ extern enum byte_order NativeByteOrder; ++ ++ /* "bit_stream.h" Type Definitions */ ++ ++ typedef struct bit_stream_struc { ++ unsigned char* pbtOutBuf; /* for .DLL code */ ++ int nOutBufPos; /* for .DLL code */ ++ FILE *pt; /* pointer to bit stream device */ ++ unsigned char *buf; /* bit stream buffer */ ++ int buf_size; /* size of buffer (in number of bytes) */ ++ unsigned long totbit; /* bit counter of bit stream */ ++ int buf_byte_idx; /* pointer to top byte in buffer */ ++ int buf_bit_idx; /* pointer to top bit of top byte in buffer */ ++ ++ /* format of file in rd mode (BINARY/ASCII) */ ++ } Bit_stream_struc; ++ ++ #include "l3side.h" ++ ++ /*********************************************************************** ++ * ++ * Global Variable External Declarations ++ * ++ ***********************************************************************/ ++ ++ extern int bitrate_table[2][15]; ++ ++ /*********************************************************************** ++ * ++ * Global Function Prototype Declarations ++ * ++ ***********************************************************************/ ++ ++ extern void display_bitrates(FILE *out_fh); ++ extern int BitrateIndex(int, int,int); ++ extern int SmpFrqIndex(long, int*); ++ extern void *mem_alloc(unsigned long, char*); ++ extern int copy_buffer(char *buffer,int buffer_size,Bit_stream_struc *bs); ++ extern void init_bit_stream_w(Bit_stream_struc*); ++ extern void alloc_buffer(Bit_stream_struc*, int); ++ extern void desalloc_buffer(Bit_stream_struc*); ++ extern void putbits(Bit_stream_struc*, unsigned int, int); ++ ++ extern enum byte_order DetermineByteOrder(void); ++ extern void SwapBytesInWords( short *loc, int words ); ++ ++ extern void ++ getframebits(lame_global_flags *gfp,int *bitsPerFrame, int *mean_bits); ++ ++ #endif +diff -r -c -N encoder/vbrquantize.c lame3.70/vbrquantize.c +*** encoder/vbrquantize.c Wed Dec 31 17:00:00 1969 +--- lame3.70/vbrquantize.c Thu Apr 6 12:50:52 2000 +*************** +*** 0 **** +--- 1,342 ---- ++ /* ++ * MP3 quantization ++ * ++ * Copyright (c) 1999 Mark Taylor ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License as published by ++ * the Free Software Foundation; either version 2, or (at your option) ++ * any later version. ++ * ++ * This program is distributed in the hope that it will be useful, ++ * but WITHOUT ANY WARRANTY; without even the implied warranty of ++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the ++ * GNU General Public License for more details. ++ * ++ * You should have received a copy of the GNU General Public License ++ * along with this program; see the file COPYING. If not, write to ++ * the Free Software Foundation, 675 Mass Ave, Cambridge, MA 02139, USA. ++ */ ++ #include <assert.h> ++ #include "util.h" ++ #include "l3side.h" ++ #include "quantize.h" ++ #include "l3bitstream.h" ++ #include "reservoir.h" ++ #include "quantize-pvt.h" ++ #ifdef HAVEGTK ++ #include "gtkanal.h" ++ #endif ++ ++ ++ ++ #define DEBUGXX ++ FLOAT8 calc_sfb_ave_noise(FLOAT8 *xr, FLOAT8 *xr34, int stride, int bw, FLOAT8 sfpow) ++ { ++ int j; ++ FLOAT8 xfsf=0; ++ FLOAT8 sfpow34 = pow(sfpow,3.0/4.0); ++ ++ for ( j=0; j < stride*bw ; j += stride) { ++ int ix; ++ FLOAT8 temp,temp2; ++ ++ /* ix=(int)( xr34[j]/sfpow34 + 0.4054);*/ ++ ix=floor( xr34[j]/sfpow34); ++ if (ix > IXMAX_VAL) return -1.0; ++ ++ temp = fabs(xr[j])- pow43[ix]*sfpow; ++ if (ix < IXMAX_VAL) { ++ temp2 = fabs(xr[j])- pow43[ix+1]*sfpow; ++ if (fabs(temp2)<fabs(temp)) temp=temp2; ++ } ++ #ifdef MAXQUANTERROR ++ temp *= temp; ++ xfsf = bw*Max(xfsf,temp); ++ #else ++ xfsf += temp * temp; ++ #endif ++ } ++ return xfsf/bw; ++ } ++ ++ ++ ++ FLOAT8 find_scalefac(FLOAT8 *xr,FLOAT8 *xr34,int stride,int sfb, ++ FLOAT8 l3_xmin,int bw) ++ { ++ FLOAT8 xfsf,sfpow,sf,sf_ok,delsf; ++ int sf4,sf_ok4,delsf4; ++ int i; ++ ++ /* search will range from sf: -52.25 -> 11.25 */ ++ /* search will range from sf4: -209 -> 45 */ ++ sf = -20.5; ++ sf4 = -82; ++ delsf = 32; ++ delsf4 = 128; ++ ++ sf_ok =10000; ++ sf_ok4=10000; ++ for (i=0; i<7; i++) { ++ delsf /= 2; ++ delsf4 /= 2; ++ sfpow = pow(2.0,sf); ++ /* sfpow = pow(2.0,sf4/4.0); */ ++ xfsf = calc_sfb_ave_noise(xr,xr34,stride,bw,sfpow); ++ ++ if (xfsf < 0) { ++ /* scalefactors too small */ ++ sf += delsf; ++ sf4 += delsf4; ++ }else{ ++ if (sf_ok==10000) sf_ok=sf; ++ if (sf_ok4==10000) sf_ok4=sf4; ++ if (xfsf > l3_xmin) { ++ /* distortion. try a smaller scalefactor */ ++ sf -= delsf; ++ sf4 -= delsf4; ++ }else{ ++ sf_ok=sf; ++ sf_ok4 = sf4; ++ sf += delsf; ++ sf4 += delsf4; ++ } ++ } ++ } ++ /* sf_ok accurate to within +/- 2*final_value_of_delsf */ ++ assert(sf_ok!=10000); ++ ++ /* NOTE: noise is not a monotone function of the sf, even though ++ * the number of bits used is! do a brute force search in the ++ * neighborhood of sf_ok: ++ * ++ * sf = sf_ok + 1.75 works 1% of the time ++ * sf = sf_ok + 1.50 works 1% of the time ++ * sf = sf_ok + 1.25 works 2% of the time ++ * sf = sf_ok + 1.00 works 3% of the time ++ * sf = sf_ok + 0.75 works 9% of the time ++ * sf = sf_ok + 0.50 0 % (because it was tried above) ++ * sf = sf_ok + 0.25 works 39% of the time ++ * sf = sf_ok + 0.00 works the rest of the time ++ */ ++ ++ sf = sf_ok + 0.75; ++ sf4 = sf_ok4 + 3; ++ ++ while (sf>(sf_ok+.01)) { ++ /* sf = sf_ok + 2*delsf was tried above, skip it: */ ++ if (fabs(sf-(sf_ok+2*delsf)) < .01) sf -=.25; ++ if (sf4 == sf_ok4+2*delsf4) sf4 -=1; ++ ++ sfpow = pow(2.0,sf); ++ /* sfpow = pow(2.0,sf4/4.0) */ ++ xfsf = calc_sfb_ave_noise(xr,xr34,stride,bw,sfpow); ++ if (xfsf > 0) { ++ if (xfsf <= l3_xmin) return sf; ++ } ++ sf -= .25; ++ sf4 -= 1; ++ } ++ return sf_ok; ++ } ++ ++ ++ ++ /* ++ sfb=0..5 scalefac < 16 ++ sfb>5 scalefac < 8 ++ ++ ifqstep = ( cod_info->scalefac_scale == 0 ) ? .5 : 1.0; ++ ol_sf = (cod_info->global_gain-210.0)/4.0; ++ ol_sf -= 2*cod_info->subblock_gain[i]; ++ ol_sf -= ifqstep*scalefac[gr][ch].s[sfb][i]; ++ */ ++ FLOAT8 compute_scalefacs_short(FLOAT8 vbrsf[SBPSY_s][3],gr_info *cod_info,int scalefac[SBPSY_s][3]) ++ { ++ FLOAT8 maxrange,maxover; ++ FLOAT8 sf[SBPSY_s][3]; ++ int sfb,i; ++ int ifqstep_inv = ( cod_info->scalefac_scale == 0 ) ? 2 : 1; ++ ++ /* make a working copy of the desired scalefacs */ ++ memcpy(sf,vbrsf,SBPSY_s*3*sizeof(FLOAT8)); ++ ++ /* see if we should use subblock gain */ ++ ++ ++ maxover=0; ++ for ( sfb = 0; sfb < SBPSY_s; sfb++ ) { ++ for (i=0; i<3; ++i) { ++ /* ifqstep*scalefac + 2*subblock_gain >= -sf[sfb] */ ++ scalefac[sfb][i]=floor( -sf[sfb][i]*ifqstep_inv +.75 + .0001) ; ++ ++ if (sfb < 6) maxrange = 15.0/ifqstep_inv; ++ else maxrange = 7.0/ifqstep_inv; ++ ++ if (maxrange + sf[sfb][i] > maxover) maxover = maxrange+sf[sfb][i]; ++ } ++ } ++ return maxover; ++ } ++ ++ ++ ++ ++ /* ++ sfb=0..10 scalefac < 16 ++ sfb>10 scalefac < 8 ++ ++ ifqstep = ( cod_info->scalefac_scale == 0 ) ? .5 : 1.0; ++ ol_sf = (cod_info->global_gain-210.0)/4.0; ++ ol_sf -= ifqstep*scalefac[gr][ch].l[sfb]; ++ if (cod_info->preflag && sfb>=11) ++ ol_sf -= ifqstep*pretab[sfb]; ++ */ ++ FLOAT8 compute_scalefacs_long(FLOAT8 vbrsf[SBPSY_l],gr_info *cod_info,int scalefac[SBPSY_l]) ++ { ++ int sfb; ++ FLOAT8 sf[SBPSY_l]; ++ FLOAT8 maxrange,maxover; ++ int ifqstep_inv = ( cod_info->scalefac_scale == 0 ) ? 2 : 1; ++ ++ /* make a working copy of the desired scalefacs */ ++ memcpy(sf,vbrsf,SBPSY_l*sizeof(FLOAT8)); ++ ++ cod_info->preflag=0; ++ for ( sfb = 11; sfb < SBPSY_l; sfb++ ) { ++ if (sf[sfb] + pretab[sfb]/ifqstep_inv > 0) break; ++ } ++ if (sfb==SBPSY_l) { ++ cod_info->preflag=1; ++ for ( sfb = 11; sfb < SBPSY_l; sfb++ ) ++ sf[sfb] += pretab[sfb]/ifqstep_inv; ++ } ++ ++ maxover=0; ++ for ( sfb = 0; sfb < SBPSY_l; sfb++ ) { ++ /* ifqstep*scalefac >= -sf[sfb] */ ++ scalefac[sfb]=floor( -sf[sfb]*ifqstep_inv +.75 + .0001) ; ++ ++ if (sfb < 11) maxrange = 15.0/ifqstep_inv; ++ else maxrange = 7.0/ifqstep_inv; ++ ++ if (maxrange + sf[sfb] > maxover) maxover = maxrange+sf[sfb]; ++ } ++ return maxover; ++ } ++ ++ ++ ++ ++ ++ /************************************************************************ ++ * ++ * VBR_iteration_loop() ++ * ++ * ++ ************************************************************************/ ++ void ++ VBR_iteration_loop_new (lame_global_flags *gfp, ++ FLOAT8 pe[2][2], FLOAT8 ms_ener_ratio[2], ++ FLOAT8 xr[2][2][576], III_psy_ratio ratio[2][2], ++ III_side_info_t * l3_side, int l3_enc[2][2][576], ++ III_scalefac_t scalefac[2][2]) ++ { ++ III_psy_xmin l3_xmin[2][2]; ++ FLOAT8 masking_lower_db; ++ FLOAT8 ifqstep; ++ int start,end,bw,sfb, i,ch, gr, over; ++ III_psy_xmin vbrsf; ++ FLOAT8 vbrmax; ++ ++ ++ iteration_init(gfp,l3_side,l3_enc); ++ ++ /* Adjust allowed masking based on quality setting */ ++ /* db_lower varies from -10 to +8 db */ ++ masking_lower_db = -10 + 2*gfp->VBR_q; ++ /* adjust by -6(min)..0(max) depending on bitrate */ ++ masking_lower = pow(10.0,masking_lower_db/10); ++ masking_lower = 1; ++ ++ ++ for (gr = 0; gr < gfp->mode_gr; gr++) { ++ if (convert_mdct) ++ ms_convert(xr[gr],xr[gr]); ++ for (ch = 0; ch < gfp->stereo; ch++) { ++ FLOAT8 xr34[576]; ++ gr_info *cod_info = &l3_side->gr[gr].ch[ch].tt; ++ int shortblock; ++ over = 0; ++ shortblock = (cod_info->block_type == SHORT_TYPE); ++ ++ for(i=0;i<576;i++) { ++ FLOAT8 temp=fabs(xr[gr][ch][i]); ++ xr34[i]=sqrt(sqrt(temp)*temp); ++ } ++ ++ calc_xmin( gfp,xr[gr][ch], &ratio[gr][ch], cod_info, &l3_xmin[gr][ch]); ++ ++ vbrmax=0; ++ if (shortblock) { ++ for ( sfb = 0; sfb < SBPSY_s; sfb++ ) { ++ for ( i = 0; i < 3; i++ ) { ++ start = scalefac_band.s[ sfb ]; ++ end = scalefac_band.s[ sfb+1 ]; ++ bw = end - start; ++ vbrsf.s[sfb][i] = find_scalefac(&xr[gr][ch][3*start+i],&xr34[3*start+i],3,sfb, ++ masking_lower*l3_xmin[gr][ch].s[sfb][i],bw); ++ if (vbrsf.s[sfb][i]>vbrmax) vbrmax=vbrsf.s[sfb][i]; ++ } ++ } ++ }else{ ++ for ( sfb = 0; sfb < SBPSY_l; sfb++ ) { ++ start = scalefac_band.l[ sfb ]; ++ end = scalefac_band.l[ sfb+1 ]; ++ bw = end - start; ++ vbrsf.l[sfb] = find_scalefac(&xr[gr][ch][start],&xr34[start],1,sfb, ++ masking_lower*l3_xmin[gr][ch].l[sfb],bw); ++ if (vbrsf.l[sfb]>vbrmax) vbrmax = vbrsf.l[sfb]; ++ } ++ ++ } /* compute scalefactors */ ++ ++ /* sf = (cod_info->global_gain-210.0)/4.0; */ ++ cod_info->global_gain = floor(4*vbrmax +210 + .5); ++ ++ ++ if (shortblock) { ++ for ( sfb = 0; sfb < SBPSY_s; sfb++ ) { ++ for ( i = 0; i < 3; i++ ) { ++ vbrsf.s[sfb][i] -= vbrmax; ++ } ++ } ++ cod_info->scalefac_scale = 0; ++ if (compute_scalefacs_short(vbrsf.s,cod_info,scalefac[gr][ch].s) > 0) { ++ cod_info->scalefac_scale = 1; ++ if (compute_scalefacs_short(vbrsf.s,cod_info,scalefac[gr][ch].s) >0) { ++ /* what do we do now? */ ++ exit(32); ++ } ++ } ++ }else{ ++ for ( sfb = 0; sfb < SBPSY_l; sfb++ ) ++ vbrsf.l[sfb] -= vbrmax; ++ ++ /* can we get away with scalefac_scale=0? */ ++ cod_info->scalefac_scale = 0; ++ if (compute_scalefacs_long(vbrsf.l,cod_info,scalefac[gr][ch].l) > 0) { ++ cod_info->scalefac_scale = 1; ++ if (compute_scalefacs_long(vbrsf.l,cod_info,scalefac[gr][ch].l) >0) { ++ /* what do we do now? */ ++ exit(32); ++ } ++ } ++ } ++ } /* ch */ ++ } /* gr */ ++ } ++ ++ ++ +diff -r -c -N encoder/version.c lame3.70/version.c +*** encoder/version.c Wed Dec 31 17:00:00 1969 +--- lame3.70/version.c Fri Mar 10 13:56:35 2000 +*************** +*** 0 **** +--- 1,69 ---- ++ /* ++ * Version numbering for LAME. ++ * ++ * Copyright (c) 1999 A.L. Faber ++ * ++ * This library is free software; you can redistribute it and/or ++ * modify it under the terms of the GNU Library General Public ++ * License as published by the Free Software Foundation; either ++ * version 2 of the License, or (at your option) any later version. ++ * ++ * This library is distributed in the hope that it will be useful, ++ * but WITHOUT ANY WARRANTY; without even the implied warranty of ++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU ++ * Library General Public License for more details. ++ * ++ * You should have received a copy of the GNU Library General Public ++ * License along with this library; if not, write to the ++ * Free Software Foundation, Inc., 59 Temple Place - Suite 330, ++ * Boston, MA 02111-1307, USA. ++ */ ++ ++ ++ #include "version.h" ++ #include "lame.h" ++ #include <stdio.h> ++ ++ static char lpszVersion[80]; ++ ++ void lame_print_version(FILE *ofile) { ++ fprintf(ofile,"LAME version %s (www.sulaco.org/mp3) \n",get_lame_version()); ++ fprintf(ofile,"GPSYCHO: GPL psycho-acoustic and noise shaping model version %s. \n",get_psy_version()); ++ #ifdef LIBSNDFILE ++ fprintf(ofile,"Input handled by libsndfile (www.zip.com.au/~erikd/libsndfile)\n"); ++ #endif ++ } ++ ++ ++ char* get_lame_version(void) ++ { ++ if (LAME_ALPHAVERSION>0) ++ sprintf(lpszVersion,"%d.%02d (alpha %d)",LAME_MAJOR_VERSION,LAME_MINOR_VERSION,LAME_ALPHAVERSION); ++ else if (LAME_BETAVERSION>0) ++ sprintf(lpszVersion,"%d.%02d (beta %d)",LAME_MAJOR_VERSION,LAME_MINOR_VERSION,LAME_BETAVERSION); ++ else ++ sprintf(lpszVersion,"%d.%02d",LAME_MAJOR_VERSION,LAME_MINOR_VERSION); ++ return lpszVersion; ++ } ++ ++ char* get_psy_version(void) ++ { ++ if (PSY_ALPHAVERSION>0) ++ sprintf(lpszVersion,"%d.%02d (alpha %d)",PSY_MAJOR_VERSION,PSY_MINOR_VERSION,PSY_ALPHAVERSION); ++ else if (PSY_BETAVERSION>0) ++ sprintf(lpszVersion,"%d.%02d (beta %d)",PSY_MAJOR_VERSION,PSY_MINOR_VERSION,PSY_BETAVERSION); ++ else ++ sprintf(lpszVersion,"%d.%02d",PSY_MAJOR_VERSION,PSY_MINOR_VERSION); ++ return lpszVersion; ++ } ++ ++ char* get_mp3x_version(void) ++ { ++ if (MP3X_ALPHAVERSION>0) ++ sprintf(lpszVersion,"%d:%02d (alpha %d)",MP3X_MAJOR_VERSION,MP3X_MINOR_VERSION,MP3X_ALPHAVERSION); ++ else if (MP3X_BETAVERSION>0) ++ sprintf(lpszVersion,"%d:%02d (beta %d)",MP3X_MAJOR_VERSION,MP3X_MINOR_VERSION,MP3X_BETAVERSION); ++ else ++ sprintf(lpszVersion,"%d:%02d",MP3X_MAJOR_VERSION,MP3X_MINOR_VERSION); ++ return lpszVersion; ++ } +diff -r -c -N encoder/version.h lame3.70/version.h +*** encoder/version.h Wed Dec 31 17:00:00 1969 +--- lame3.70/version.h Thu Apr 6 12:50:52 2000 +*************** +*** 0 **** +--- 1,46 ---- ++ /* ++ * Version numbering for LAME. ++ * ++ * Copyright (c) 1999 A.L. Faber ++ * ++ * This library is free software; you can redistribute it and/or ++ * modify it under the terms of the GNU Library General Public ++ * License as published by the Free Software Foundation; either ++ * version 2 of the License, or (at your option) any later version. ++ * ++ * This library is distributed in the hope that it will be useful, ++ * but WITHOUT ANY WARRANTY; without even the implied warranty of ++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU ++ * Library General Public License for more details. ++ * ++ * You should have received a copy of the GNU Library General Public ++ * License along with this library; if not, write to the ++ * Free Software Foundation, Inc., 59 Temple Place - Suite 330, ++ * Boston, MA 02111-1307, USA. ++ */ ++ ++ #ifndef LAME_H_INCLUDED ++ #define LAME_H_INCLUDED ++ ++ #define LAME_MAJOR_VERSION 3 /* Major version number */ ++ #define LAME_MINOR_VERSION 70 /* Minor version number */ ++ #define LAME_ALPHAVERSION 0 /* Set number if this is an alpha version, otherwise zero */ ++ #define LAME_BETAVERSION 0 /* Set number if this is a beta version, otherwise zero */ ++ ++ #define PSY_MAJOR_VERSION 0 /* Major version number */ ++ #define PSY_MINOR_VERSION 77 /* Minor version number */ ++ #define PSY_ALPHAVERSION 0 /* Set number if this is an alpha version, otherwise zero */ ++ #define PSY_BETAVERSION 0 /* Set number if this is a beta version, otherwise zero */ ++ ++ #define MP3X_MAJOR_VERSION 0 /* Major version number */ ++ #define MP3X_MINOR_VERSION 82 /* Minor version number */ ++ #define MP3X_ALPHAVERSION 0 /* Set number if this is an alpha version, otherwise zero */ ++ #define MP3X_BETAVERSION 0 /* Set number if this is a beta version, otherwise zero */ ++ ++ #include "machine.h" ++ void lame_print_version(FILE *); ++ char* get_lame_version(void); /* returns lame version number string */ ++ char* get_psy_version(void); /* returns psy model version number string */ ++ char* get_mp3x_version(void); /* returns mp3x version number string */ ++ ++ #endif