Merge pull request #979 from alibaba-damo-academy/dev_lhn

hnluo

2023-09-25 5f21828d613c3e5e976cf0512c69bf9892300dc5

Merge pull request #979 from alibaba-damo-academy/dev_lhn

update paraformer online python websocket code

 funasr/runtime/python/websocket/funasr_client_api.py

@@ -51,7 +51,8 @@
        stride = int(60 *  chunk_size[1]/  chunk_interval / 1000 * 16000 * 2)

        chunk_num = (len(audio_bytes) - 1) // stride + 1

       

        message = json.dumps({"mode":  mode, "chunk_size":  chunk_size, "chunk_interval":  chunk_interval,

        message = json.dumps({"mode": args.mode, "chunk_size": args.chunk_size, "encoder_chunk_look_back": 4,

                              "decoder_chunk_look_back": 1, "chunk_interval": args.chunk_interval, 
                              "wav_name": wav_name, "is_speaking": True})

 

        self.websocket.send(message)

@@ -131,4 +132,4 @@
    print("text",text)

 

    

            
            

 funasr/runtime/python/websocket/funasr_wss_client.py

@@ -29,6 +29,14 @@
                    type=str,
                    default="5, 10, 5",
                    help="chunk")
parser.add_argument("--encoder_chunk_look_back",
                    type=int,
                    default=4,
                    help="number of chunks to lookback for encoder self-attention")
parser.add_argument("--decoder_chunk_look_back",
                    type=int,
                    default=1,
                    help="number of encoder chunks to lookback for decoder cross-attention")
parser.add_argument("--chunk_interval",
                    type=int,
                    default=10,
@@ -99,7 +107,8 @@
                    input=True,
                    frames_per_buffer=CHUNK)

    message = json.dumps({"mode": args.mode, "chunk_size": args.chunk_size, "chunk_interval": args.chunk_interval,
    message = json.dumps({"mode": args.mode, "chunk_size": args.chunk_size, "encoder_chunk_look_back": args.encoder_chunk_look_back,
                          "decoder_chunk_look_back": args.decoder_chunk_look_back, "chunk_interval": args.chunk_interval, 
                          "wav_name": "microphone", "is_speaking": True})
    #voices.put(message)
    await websocket.send(message)

 funasr/runtime/python/websocket/funasr_wss_server.py

@@ -103,8 +103,8 @@
    model=args.asr_model_online,
    ngpu=args.ngpu,
    ncpu=args.ncpu,
    model_revision='v1.0.4',
    update_model='v1.0.4',
    model_revision='v1.0.7',
    update_model='v1.0.7',
    mode='paraformer_streaming')

print("model loaded! only support one client at the same time now!!!!")
@@ -159,6 +159,10 @@
                    websocket.wav_name = messagejson.get("wav_name")
                if "chunk_size" in messagejson:
                    websocket.param_dict_asr_online["chunk_size"] = messagejson["chunk_size"]
                if "encoder_chunk_look_back" in messagejson:
                    websocket.param_dict_asr_online["encoder_chunk_look_back"] = messagejson["encoder_chunk_look_back"]
                if "decoder_chunk_look_back" in messagejson:
                    websocket.param_dict_asr_online["decoder_chunk_look_back"] = messagejson["decoder_chunk_look_back"]
                if "mode" in messagejson:
                    websocket.mode = messagejson["mode"]
            if len(frames_asr_online) > 0 or len(frames_asr) > 0 or not isinstance(message, str):