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basereal.py
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basereal.py
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import math
import torch
import numpy as np
import os
import time
import cv2
import glob
import pickle
import copy
import resampy
import queue
from queue import Queue
from threading import Thread, Event
from io import BytesIO
import soundfile as sf
import av
from fractions import Fraction
from ttsreal import EdgeTTS,VoitsTTS,XTTS,CosyVoiceTTS
from tqdm import tqdm
def read_imgs(img_list):
frames = []
print('reading images...')
for img_path in tqdm(img_list):
frame = cv2.imread(img_path)
frames.append(frame)
return frames
class BaseReal:
def __init__(self, opt):
self.opt = opt
self.sample_rate = 16000
self.chunk = self.sample_rate // opt.fps # 320 samples per chunk (20ms * 16000 / 1000)
if opt.tts == "edgetts":
self.tts = EdgeTTS(opt,self)
elif opt.tts == "gpt-sovits":
self.tts = VoitsTTS(opt,self)
elif opt.tts == "xtts":
self.tts = XTTS(opt,self)
elif opt.tts == "cosyvoice":
self.tts = CosyVoiceTTS(opt,self)
self.speaking = False
self.recording = False
self.recordq_video = Queue()
self.recordq_audio = Queue()
self.curr_state=0
self.custom_img_cycle = {}
self.custom_audio_cycle = {}
self.custom_audio_index = {}
self.custom_index = {}
self.custom_opt = {}
self.__loadcustom()
def put_msg_txt(self,msg):
self.tts.put_msg_txt(msg)
def put_audio_frame(self,audio_chunk): #16khz 20ms pcm
self.asr.put_audio_frame(audio_chunk)
def put_audio_file(self,filebyte):
input_stream = BytesIO(filebyte)
stream = self.__create_bytes_stream(input_stream)
streamlen = stream.shape[0]
idx=0
while streamlen >= self.chunk: #and self.state==State.RUNNING
self.put_audio_frame(stream[idx:idx+self.chunk])
streamlen -= self.chunk
idx += self.chunk
def __create_bytes_stream(self,byte_stream):
#byte_stream=BytesIO(buffer)
stream, sample_rate = sf.read(byte_stream) # [T*sample_rate,] float64
print(f'[INFO]put audio stream {sample_rate}: {stream.shape}')
stream = stream.astype(np.float32)
if stream.ndim > 1:
print(f'[WARN] audio has {stream.shape[1]} channels, only use the first.')
stream = stream[:, 0]
if sample_rate != self.sample_rate and stream.shape[0]>0:
print(f'[WARN] audio sample rate is {sample_rate}, resampling into {self.sample_rate}.')
stream = resampy.resample(x=stream, sr_orig=sample_rate, sr_new=self.sample_rate)
return stream
def pause_talk(self):
self.tts.pause_talk()
self.asr.pause_talk()
def is_speaking(self)->bool:
return self.speaking
def __loadcustom(self):
for item in self.opt.customopt:
print(item)
input_img_list = glob.glob(os.path.join(item['imgpath'], '*.[jpJP][pnPN]*[gG]'))
input_img_list = sorted(input_img_list, key=lambda x: int(os.path.splitext(os.path.basename(x))[0]))
self.custom_img_cycle[item['audiotype']] = read_imgs(input_img_list)
self.custom_audio_cycle[item['audiotype']], sample_rate = sf.read(item['audiopath'], dtype='float32')
self.custom_audio_index[item['audiotype']] = 0
self.custom_index[item['audiotype']] = 0
self.custom_opt[item['audiotype']] = item
def init_customindex(self):
self.curr_state=0
for key in self.custom_audio_index:
self.custom_audio_index[key]=0
for key in self.custom_index:
self.custom_index[key]=0
def start_recording(self,path):
"""开始录制视频"""
if self.recording:
return
self.recording = True
self.recordq_video.queue.clear()
self.recordq_audio.queue.clear()
self.container = av.open(path, mode="w")
process_thread = Thread(target=self.record_frame, args=())
process_thread.start()
def record_frame(self):
videostream = self.container.add_stream("libx264", rate=25)
videostream.codec_context.time_base = Fraction(1, 25)
audiostream = self.container.add_stream("aac")
audiostream.codec_context.time_base = Fraction(1, 16000)
init = True
framenum = 0
while self.recording:
try:
videoframe = self.recordq_video.get(block=True, timeout=1)
videoframe.pts = framenum #int(round(framenum*0.04 / videostream.codec_context.time_base))
videoframe.dts = videoframe.pts
if init:
videostream.width = videoframe.width
videostream.height = videoframe.height
init = False
for packet in videostream.encode(videoframe):
self.container.mux(packet)
for k in range(2):
audioframe = self.recordq_audio.get(block=True, timeout=1)
audioframe.pts = int(round((framenum*2+k)*0.02 / audiostream.codec_context.time_base))
audioframe.dts = audioframe.pts
for packet in audiostream.encode(audioframe):
self.container.mux(packet)
framenum += 1
except queue.Empty:
print('record queue empty,')
continue
except Exception as e:
print(e)
#break
for packet in videostream.encode(None):
self.container.mux(packet)
for packet in audiostream.encode(None):
self.container.mux(packet)
self.container.close()
self.recordq_video.queue.clear()
self.recordq_audio.queue.clear()
print('record thread stop')
def stop_recording(self):
"""停止录制视频"""
if not self.recording:
return
self.recording = False
def mirror_index(self,size, index):
#size = len(self.coord_list_cycle)
turn = index // size
res = index % size
if turn % 2 == 0:
return res
else:
return size - res - 1
def get_audio_stream(self,audiotype):
idx = self.custom_audio_index[audiotype]
stream = self.custom_audio_cycle[audiotype][idx:idx+self.chunk]
self.custom_audio_index[audiotype] += self.chunk
if self.custom_audio_index[audiotype]>=self.custom_audio_cycle[audiotype].shape[0]:
self.curr_state = 1 #当前视频不循环播放,切换到静音状态
return stream
def set_curr_state(self,audiotype, reinit):
print('set_curr_state:',audiotype)
self.curr_state = audiotype
if reinit:
self.custom_audio_index[audiotype] = 0
self.custom_index[audiotype] = 0
# def process_custom(self,audiotype:int,idx:int):
# if self.curr_state!=audiotype: #从推理切到口播
# if idx in self.switch_pos: #在卡点位置可以切换
# self.curr_state=audiotype
# self.custom_index=0
# else:
# self.custom_index+=1