import pyaudio
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import scipy.io.wavfile as wav
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import grpc_client
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import grpc
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import json
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from grpc_client import transcribe_audio_bytes
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from paraformer_pb2_grpc import ASRStub
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import webrtcvad
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import numpy as np
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import time
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import asyncio
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import datetime
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import argparse
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async def deal_chunk(sig_mic):
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global stub,SPEAKING,asr_user,language,sample_rate
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sig = np.frombuffer(sig_mic, 'int16')
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if vad.is_speech(sig.tobytes(), sample_rate): #speaking
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SPEAKING = True
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response = transcribe_audio_bytes(stub, sig, user=asr_user, language=language, speaking = True, isEnd = False) #speaking, send audio to server.
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else: #silence
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begin_time = 0
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if SPEAKING: #means we have some audio recorded, send recognize order to server.
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SPEAKING = False
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begin_time = int(round(time.time() * 1000))
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response = transcribe_audio_bytes(stub, None, user=asr_user, language=language, speaking = False, isEnd = False) #speak end, call server for recognize one sentence
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resp = response.next()
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if "decoding" == resp.action:
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resp = response.next() #TODO, blocking operation may leads to miss some audio clips. C++ multi-threading is preferred.
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if "finish" == resp.action:
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end_time = int(round(time.time() * 1000))
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print (json.loads(resp.sentence))
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print ("delay in ms: %d " % (end_time - begin_time))
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else:
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pass
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async def record(host,port,sample_rate,mic_chunk,record_seconds,asr_user,language):
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with grpc.insecure_channel('{}:{}'.format(host, port)) as channel:
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global stub
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stub = ASRStub(channel)
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for i in range(0, int(sample_rate / mic_chunk * record_seconds)):
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sig_mic = stream.read(mic_chunk,exception_on_overflow = False)
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await asyncio.create_task(deal_chunk(sig_mic))
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#end grpc
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response = transcribe_audio_bytes(stub, None, user=asr_user, language=language, speaking = False, isEnd = True)
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print (response.next().action)
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if __name__ == '__main__':
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parser = argparse.ArgumentParser()
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parser.add_argument("--host",
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type=str,
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default="127.0.0.1",
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required=True,
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help="grpc server host ip")
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parser.add_argument("--port",
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type=int,
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default=10095,
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required=True,
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help="grpc server port")
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parser.add_argument("--user_allowed",
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type=str,
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default="project1_user1",
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help="allowed user for grpc client")
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parser.add_argument("--sample_rate",
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type=int,
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default=16000,
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help="audio sample_rate from client")
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parser.add_argument("--mic_chunk",
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type=int,
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default=160,
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help="chunk size for mic")
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parser.add_argument("--record_seconds",
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type=int,
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default=120,
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help="run specified seconds then exit ")
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args = parser.parse_args()
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SPEAKING = False
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asr_user = args.user_allowed
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sample_rate = args.sample_rate
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language = 'zh-CN'
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vad = webrtcvad.Vad()
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vad.set_mode(1)
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FORMAT = pyaudio.paInt16
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CHANNELS = 1
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p = pyaudio.PyAudio()
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stream = p.open(format=FORMAT,
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channels=CHANNELS,
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rate=args.sample_rate,
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input=True,
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frames_per_buffer=args.mic_chunk)
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print("* recording")
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asyncio.run(record(args.host,args.port,args.sample_rate,args.mic_chunk,args.record_seconds,args.user_allowed,language))
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stream.stop_stream()
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stream.close()
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p.terminate()
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print("recording stop")
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