add files

funasr_local/runtime/python/grpc/grpc_main_client_mic.py

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+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")
+
+    
+