LiveTalking/museasr.py

import time
import torch
import numpy as np
import soundfile as sf
import resampy

import queue
from queue import Queue
from io import BytesIO

from musetalk.whisper.audio2feature import Audio2Feature

class MuseASR:
    def __init__(self, opt, audio_processor:Audio2Feature):
        self.opt = opt

        self.fps = opt.fps # 20 ms per frame
        self.sample_rate = 16000
        self.chunk = self.sample_rate // self.fps # 320 samples per chunk (20ms * 16000 / 1000)
        self.queue = Queue()
        # self.input_stream = BytesIO()
        self.output_queue = Queue()

        self.audio_processor = audio_processor
        self.batch_size = opt.batch_size

        self.stride_left_size = self.stride_right_size = 6
        self.audio_feats = []

        self.warm_up()

    def put_audio_frame(self,audio_chunk): #16khz 20ms pcm
        self.queue.put(audio_chunk)

    def __get_audio_frame(self):        
        try:
            frame = self.queue.get(block=True,timeout=0.02)
            type = 0
            #print(f'[INFO] get frame {frame.shape}')
        except queue.Empty:
            frame = np.zeros(self.chunk, dtype=np.float32)
            type = 1

        return frame,type 

    def get_audio_out(self):  #get origin audio pcm to nerf
        return self.output_queue.get()
    
    def warm_up(self):
        frames = []
        for _ in range(self.stride_left_size + self.stride_right_size):
            audio_frame,type=self.__get_audio_frame()
            frames.append(audio_frame)
            self.output_queue.put((audio_frame,type))
        inputs = np.concatenate(frames) # [N * chunk]
        whisper_feature = self.audio_processor.audio2feat(inputs)
        for feature in whisper_feature:
            self.audio_feats.append(feature)

        for _ in range(self.stride_left_size):
            self.output_queue.get()

    def run_step(self):
        ############################################## extract audio feature ##############################################
        start_time = time.time()
        frames = []
        for _ in range(self.batch_size*2):
            audio_frame,type=self.__get_audio_frame()
            frames.append(audio_frame)
            self.output_queue.put((audio_frame,type))
        inputs = np.concatenate(frames) # [N * chunk]
        whisper_feature = self.audio_processor.audio2feat(inputs)
        for feature in whisper_feature:
            self.audio_feats.append(feature)
        
        #print(f"processing audio costs {(time.time() - start_time) * 1000}ms, inputs shape:{inputs.shape} whisper_feature len:{len(whisper_feature)}")

    def get_next_feat(self):
        whisper_chunks = self.audio_processor.feature2chunks(feature_array=self.audio_feats,fps=self.fps/2,batch_size=self.batch_size,start=self.stride_left_size/2 )
        #print(f"whisper_chunks len:{len(whisper_chunks)},self.audio_feats len:{len(self.audio_feats)},self.output_queue len:{self.output_queue.qsize()}")
        self.audio_feats = self.audio_feats[-(self.stride_left_size + self.stride_right_size):]
        return whisper_chunks