huangmingming
2023-01-29 15fd6cdf1ac6737d92ae88a9147c0f6f7044e0ad
funasr/runtime/python/grpc/grpc_main_client_mic.py
@@ -10,15 +10,16 @@
import time
import asyncio
import datetime
import argparse
SPEAKING = False
stub = None
asr_user = None
language = None
#SPEAKING = False
#stub = None
#asr_user = None
#language = None
async def deal_chunk(sig_mic):
    
    global stub,SPEAKING,asr_user,language
    global stub,SPEAKING,asr_user,language,sample_rate
    sig = np.frombuffer(sig_mic, 'int16')
    if vad.is_speech(sig.tobytes(), sample_rate): #speaking
        SPEAKING = True
@@ -38,6 +39,7 @@
                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(resp.action)
                    print (json.loads(resp.sentence))
                    #print ("silence, end_time: %d " % end_time)
                    print ("delay in ms: %d " % (end_time - begin_time))
@@ -97,10 +99,12 @@
    args = parser.parse_args()
    
    global SPEAKING,asr_user,language
    SPEAKING = False
    asr_user = args.asr_user
    asr_user = args.user_allowed
    sample_rate = args.sample_rate
    language = 'zh-CN'  
    vad = webrtcvad.Vad()
    vad.set_mode(1)
@@ -116,7 +120,7 @@
                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.asr_user,args.language))
    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()