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Replace deprecated FFmpeg APIs to support FFmpeg 5.x
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@Greedysky
Greedysky committed Jan 11, 2023
1 parent dbe0129 commit 1db0c89
Showing 1 changed file with 64 additions and 51 deletions.
115 changes: 64 additions & 51 deletions spek-audio.cc
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Original file line number Diff line number Diff line change
Expand Up @@ -55,8 +55,7 @@ class AudioFileImpl : public AudioFile

int channel;

AVPacket packet;
int offset;
AVPacket *packet;
AVFrame *frame;
int buffer_len;
float *buffer;
Expand All @@ -69,13 +68,17 @@ class AudioFileImpl : public AudioFile

Audio::Audio()
{
#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(58, 10, 100) //ffmpeg-3.5
av_register_all();
#endif
}

Audio::~Audio()
{
#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(58, 10, 100) //ffmpeg-3.5
// This prevents a memory leak.
av_lockmgr_register(nullptr);
#endif
}

std::unique_ptr<AudioFile> Audio::open(const std::string& file_name, int stream)
Expand Down Expand Up @@ -113,7 +116,7 @@ std::unique_ptr<AudioFile> Audio::open(const std::string& file_name, int stream)

AVStream *avstream = nullptr;
AVCodecParameters *codecpar = nullptr;
AVCodec *codec = nullptr;
const AVCodec *codec = nullptr;
if (!error) {
avstream = format_context->streams[audio_stream];
codecpar = avstream->codecpar;
Expand Down Expand Up @@ -153,7 +156,11 @@ std::unique_ptr<AudioFile> Audio::open(const std::string& file_name, int stream)
if (bits_per_sample) {
bit_rate = 0;
}
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(59, 37, 100) //ffmpeg-5.1
channels = codecpar->ch_layout.nb_channels;
#else
channels = codecpar->channels;
#endif

if (avstream->duration != AV_NOPTS_VALUE) {
duration = avstream->duration * av_q2d(avstream->time_base);
Expand Down Expand Up @@ -211,10 +218,9 @@ AudioFileImpl::AudioFileImpl(
sample_rate(sample_rate),
bits_per_sample(bits_per_sample), streams(streams), channels(channels), duration(duration)
{
av_init_packet(&this->packet);
this->packet.data = nullptr;
this->packet.size = 0;
this->offset = 0;
this->packet = av_packet_alloc();
this->packet->data = nullptr;
this->packet->size = 0;
this->frame = av_frame_alloc();
this->buffer_len = 0;
this->buffer = nullptr;
Expand All @@ -231,11 +237,8 @@ AudioFileImpl::~AudioFileImpl()
if (this->frame) {
av_frame_free(&this->frame);
}
if (this->packet.data) {
this->packet.data -= this->offset;
this->packet.size += this->offset;
this->offset = 0;
av_packet_unref(&this->packet);
if (this->packet) {
av_packet_free(&this->packet);
}
if (this->codec_context) {
avcodec_free_context(&codec_context);
Expand Down Expand Up @@ -264,41 +267,63 @@ void AudioFileImpl::start(int channel, int samples)
int AudioFileImpl::read()
{
if (!!this->error) {
// Stop on error.
return -1;
}

// This runs a state machine to allow incremental calls to decode and return the next chunk of samples.
// FFmpeg docs: https://ffmpeg.org/doxygen/5.1/group__lavc__encdec.html
for (;;) {
while (this->packet.size > 0) {
av_frame_unref(this->frame);
int got_frame = 0;
int len = avcodec_decode_audio4(this->codec_context, this->frame, &got_frame, &this->packet
);
if (len < 0) {
// Error, skip the frame.
if (!this->packet) {
// Finished decoding.
return 0;
}

// Read the next packet.
int ret = 0;
while ((ret = av_read_frame(this->format_context, this->packet)) >= 0) {
if (this->packet->stream_index == this->audio_stream) {
break;
}
this->packet.data += len;
this->packet.size -= len;
this->offset += len;
if (!got_frame) {
// No data yet, get more frames.
continue;
}
const int samples = m_frame->nb_samples;
// Occasionally the frame has no samples, move on to the next one.
av_packet_unref(this->packet);
}

if (ret < 0) {
// End of file or error, empty the packet to flush the decoder.
av_packet_unref(this->packet);
av_packet_free(&this->packet);
this->packet = nullptr;
}

ret = avcodec_send_packet(this->codec_context, this->packet);
if (this->packet) {
av_packet_unref(this->packet);
}

if (ret < 0) {
// Skip the packet.
continue;
}

// The packet can contain multiple frames, read all of them.
int total_samples = 0;
while ((ret = avcodec_receive_frame(this->codec_context, this->frame)) >= 0) {
const int samples = this->frame->nb_samples;
if (samples == 0) {
// Occasionally the frame has no samples, move on to the next one.
av_frame_unref(this->frame);
continue;
}
// We have data, return it and come back for more later.
if (samples > this->buffer_len) {
// We have the data, normalise and write to the buffer.
if ((total_samples + samples) > this->buffer_len) {
this->buffer = static_cast<float*>(
av_realloc(this->buffer, samples * sizeof(float))
av_realloc(this->buffer, (total_samples + samples) * sizeof(float))
);
this->buffer_len = samples;
this->buffer_len = total_samples + samples;
}

AVSampleFormat format = static_cast<AVSampleFormat>(this->frame->format);
int is_planar = av_sample_fmt_is_planar(format);
const int is_planar = av_sample_fmt_is_planar(format);
for (int sample = 0; sample < samples; ++sample) {
uint8_t *data;
int offset;
Expand Down Expand Up @@ -333,27 +358,15 @@ int AudioFileImpl::read()
value = 0.0f;
break;
}
this->buffer[sample] = value;
this->buffer[total_samples + sample] = value;
}
return samples;
}
if (this->packet.data) {
this->packet.data -= this->offset;
this->packet.size += this->offset;
this->offset = 0;
av_packet_unref(&this->packet);
total_samples += samples;
av_frame_unref(this->frame);
}

int res = 0;
while ((res = av_read_frame(this->format_context, &this->packet)) >= 0) {
if (this->packet.stream_index == this->audio_stream) {
break;
}
av_packet_unref(&this->packet);
}
if (res < 0) {
// End of file or error.
return 0;
if (total_samples > 0) {
return total_samples;
}
// Error during decoding or EOF, skip the packet.
}
}

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