Update README.md

zhifu gao

2023-10-16 1d7bbbffb6a024a33859b48a7a656d0455dc0be1

 funasr/runtime/python/websocket/funasr_wss_server.py

@@ -10,7 +10,6 @@
from modelscope.pipelines import pipeline
from modelscope.utils.constant import Tasks
from modelscope.utils.logger import get_logger
from funasr.runtime.python.onnxruntime.funasr_onnx.utils.frontend import load_bytes

tracemalloc.start()

@@ -104,8 +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!!!!")
@@ -160,6 +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):
@@ -233,8 +236,6 @@
async def async_asr(websocket, audio_in):
            if len(audio_in) > 0:
                # print(len(audio_in))
                audio_in = load_bytes(audio_in)
                
                rec_result = inference_pipeline_asr(audio_in=audio_in,
                                                    param_dict=websocket.param_dict_asr)
                # print(rec_result)
@@ -243,13 +244,13 @@
                                                         param_dict=websocket.param_dict_punc)
                    # print("offline", rec_result)
                if 'text' in rec_result:
                    message = json.dumps({"mode": "2pass-offline", "text": rec_result["text"], "wav_name": websocket.wav_name})
                    mode = "2pass-offline" if "2pass" in websocket.mode else websocket.mode
                    message = json.dumps({"mode": mode, "text": rec_result["text"], "wav_name": websocket.wav_name,"is_final":websocket.is_speaking})
                    await websocket.send(message)


async def async_asr_online(websocket, audio_in):
    if len(audio_in) > 0:
        audio_in = load_bytes(audio_in)
        # print(websocket.param_dict_asr_online.get("is_final", False))
        rec_result = inference_pipeline_asr_online(audio_in=audio_in,
                                                   param_dict=websocket.param_dict_asr_online)
@@ -260,7 +261,8 @@
        if "text" in rec_result:
            if rec_result["text"] != "sil" and rec_result["text"] != "waiting_for_more_voice":
                # print("online", rec_result)
                message = json.dumps({"mode": "2pass-online", "text": rec_result["text"], "wav_name": websocket.wav_name})
                mode = "2pass-online" if "2pass" in websocket.mode else websocket.mode
                message = json.dumps({"mode": mode, "text": rec_result["text"], "wav_name": websocket.wav_name,"is_final":websocket.is_speaking})
                await websocket.send(message)

if len(args.certfile)>0: