fatmop
commented
1 year ago import grpc
import json
import time
import asyncio
import soundfile as sf
import argparse
from grpc_client import transcribe_audio_bytes
from paraformer_pb2_grpc import ASRStub
# send the audio data once
async def grpc_rec(wav_scp, grpc_uri, asr_user, language):
with grpc.insecure_channel(grpc_uri) as channel:
stub = ASRStub(channel)
for line in wav_scp:
wav_file = line.split()[1]
wav, _ = sf.read(wav_file, dtype='int16')
speech_length = wav.shape[0]
sample_offset = 0
chunk_size = [8, 8, 4] # [5, 10, 5] 600ms, [8, 8, 4] 480ms
stride_size = chunk_size[1] * 960
for sample_offset in range(0, speech_length, min(stride_size, speech_length - sample_offset)):
if sample_offset + stride_size >= speech_length - 1:
stride_size = speech_length - sample_offset
b = time.time()
response = transcribe_audio_bytes(stub, wav[sample_offset: sample_offset + stride_size].tobytes(), user=asr_user, language=language, speaking=True, isEnd=False)
response = transcribe_audio_bytes(stub, None,
user=asr_user, language=language, speaking=False, isEnd=False)
resp = response.next()
text = ''
if 'decoding' == resp.action:
resp = response.next()
if 'finish' == resp.action:
text = json.loads(resp.sentence)['text']
res= {'text': text, 'time': time.time() - b}
print(res)
time.sleep(0.48)
response = transcribe_audio_bytes(stub, None, user=asr_user, language=language, speaking=False, isEnd=True)
async def test(args):
wav_scp = open(args.wav_scp, "r").readlines()
uri = '{}:{}'.format(args.host, args.port)
res = await grpc_rec(wav_scp, uri, args.user_allowed, language = 'zh-CN')
if __name__ == '__main__':
parser = argparse.ArgumentParser()
parser.add_argument("--host",
type=str,
default="127.0.0.1",
required=False,
help="grpc server host ip")
parser.add_argument("--port",
type=int,
default=10108,
required=False,
help="grpc server port")
parser.add_argument("--user_allowed",
type=str,
default="project1_user1",
help="allowed user for grpc client")
parser.add_argument("--sample_rate",
type=int,
default=16000,
help="audio sample_rate from client")
parser.add_argument("--wav_scp",
type=str,
required=True,
help="audio wav scp")
args = parser.parse_args()
asyncio.run(test(args))
onfireAI
PangziZhang523
icowan
Tongjilibo
wuhongsheng
Nancis1130
做项目遇到并发报错，跑到FunAsr Demo里也去执行了下，发现了同样的错误，如下：
OS: [e.g. linux] ubuntu20.04 和 Macbook都测试了 Python Version：py37-torch1.13.1 Package Version：modelscope 1.4.1 funasr 0.3.1
Model：damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch vad_model：damo/speech_fsmn_vad_zh-cn-16k-common-pytorch punc_model：''
Demo目录：funasr/runtime/python/grpc
Command：同时执行 grpc_main_client_mic.py和grpc_main_client.py
问题描述： 并发执行asr_vad_pnuc 和单独执行vad，都会报错，单线程执行任何一个脚本都没有问题 [后述]。 为了还原同时录音的场景，对grpc_main_client_mac.py做了简单变更，把一次上传一个文件改成截断上传，每480ms一个包 在最后我贴了修改后的脚本 Error log：