From 7bdcefb198484d067419fde3eb02ab38b6d7884c Mon Sep 17 00:00:00 2001
From: zhifu gao <zhifu.gzf@alibaba-inc.com>
Date: 星期一, 30 一月 2023 18:25:30 +0800
Subject: [PATCH] Merge pull request #48 from dyyzhmm/main
---
funasr/runtime/python/grpc/grpc_main_client_mic.py | 112 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++
1 files changed, 112 insertions(+), 0 deletions(-)
diff --git a/funasr/runtime/python/grpc/grpc_main_client_mic.py b/funasr/runtime/python/grpc/grpc_main_client_mic.py
new file mode 100644
index 0000000..acbe90b
--- /dev/null
+++ b/funasr/runtime/python/grpc/grpc_main_client_mic.py
@@ -0,0 +1,112 @@
+import pyaudio
+import grpc
+import json
+import webrtcvad
+import time
+import asyncio
+import argparse
+
+from grpc_client import transcribe_audio_bytes
+from paraformer_pb2_grpc import ASRStub
+
+async def deal_chunk(sig_mic):
+ global stub,SPEAKING,asr_user,language,sample_rate
+ if vad.is_speech(sig_mic, sample_rate): #speaking
+ SPEAKING = True
+ response = transcribe_audio_bytes(stub, sig_mic, user=asr_user, language=language, speaking = True, isEnd = False) #speaking, send audio to server.
+ else: #silence
+ begin_time = 0
+ if SPEAKING: #means we have some audio recorded, send recognize order to server.
+ SPEAKING = False
+ begin_time = int(round(time.time() * 1000))
+ response = transcribe_audio_bytes(stub, None, user=asr_user, language=language, speaking = False, isEnd = False) #speak end, call server for recognize one sentence
+ resp = response.next()
+ if "decoding" == resp.action:
+ resp = response.next() #TODO, blocking operation may leads to miss some audio clips. C++ multi-threading is preferred.
+ if "finish" == resp.action:
+ end_time = int(round(time.time() * 1000))
+ print (json.loads(resp.sentence))
+ print ("delay in ms: %d " % (end_time - begin_time))
+ else:
+ pass
+
+
+async def record(host,port,sample_rate,mic_chunk,record_seconds,asr_user,language):
+ with grpc.insecure_channel('{}:{}'.format(host, port)) as channel:
+ global stub
+ stub = ASRStub(channel)
+ for i in range(0, int(sample_rate / mic_chunk * record_seconds)):
+
+ sig_mic = stream.read(mic_chunk,exception_on_overflow = False)
+ await asyncio.create_task(deal_chunk(sig_mic))
+
+ #end grpc
+ response = transcribe_audio_bytes(stub, None, user=asr_user, language=language, speaking = False, isEnd = True)
+ print (response.next().action)
+
+
+if __name__ == '__main__':
+ parser = argparse.ArgumentParser()
+ parser.add_argument("--host",
+ type=str,
+ default="127.0.0.1",
+ required=True,
+ help="grpc server host ip")
+
+ parser.add_argument("--port",
+ type=int,
+ default=10095,
+ required=True,
+ 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("--mic_chunk",
+ type=int,
+ default=160,
+ help="chunk size for mic")
+
+ parser.add_argument("--record_seconds",
+ type=int,
+ default=120,
+ help="run specified seconds then exit ")
+
+ args = parser.parse_args()
+
+
+ SPEAKING = False
+ asr_user = args.user_allowed
+ sample_rate = args.sample_rate
+ language = 'zh-CN'
+
+
+ vad = webrtcvad.Vad()
+ vad.set_mode(1)
+
+ FORMAT = pyaudio.paInt16
+ CHANNELS = 1
+ p = pyaudio.PyAudio()
+
+ stream = p.open(format=FORMAT,
+ channels=CHANNELS,
+ rate=args.sample_rate,
+ input=True,
+ frames_per_buffer=args.mic_chunk)
+
+ print("* recording")
+ asyncio.run(record(args.host,args.port,args.sample_rate,args.mic_chunk,args.record_seconds,args.user_allowed,language))
+ stream.stop_stream()
+ stream.close()
+ p.terminate()
+ print("recording stop")
+
+
+
--
Gitblit v1.9.1