| | |
| | | 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() |
| | | |
| | |
| | | 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!!!!") |
| | |
| | | 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): |
| | |
| | | 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) |
| | |
| | | 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) |
| | |
| | | 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: |