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- import asyncio
- import json
- import websockets
- import time
- import logging
- import tracemalloc
- import numpy as np
- from parse_args import args
- 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()
- logger = get_logger(log_level=logging.CRITICAL)
- logger.setLevel(logging.CRITICAL)
- websocket_users = set()
- print("model loading")
- # asr
- inference_pipeline_asr = pipeline(
- task=Tasks.auto_speech_recognition,
- model=args.asr_model,
- ngpu=args.ngpu,
- ncpu=args.ncpu,
- model_revision=None)
- # vad
- inference_pipeline_vad = pipeline(
- task=Tasks.voice_activity_detection,
- model=args.vad_model,
- model_revision=None,
- output_dir=None,
- batch_size=1,
- mode='online',
- ngpu=args.ngpu,
- ncpu=args.ncpu,
- )
- if args.punc_model != "":
- inference_pipeline_punc = pipeline(
- task=Tasks.punctuation,
- model=args.punc_model,
- model_revision=None,
- ngpu=args.ngpu,
- ncpu=args.ncpu,
- )
- else:
- inference_pipeline_punc = None
- print("model loaded")
- async def ws_serve(websocket, path):
- frames = []
- frames_asr = []
- global websocket_users
- websocket_users.add(websocket)
- websocket.param_dict_asr = {}
- websocket.param_dict_vad = {'in_cache': dict(), "is_final": False}
- websocket.param_dict_punc = {'cache': list()}
- websocket.vad_pre_idx = 0
- speech_start = False
- try:
- async for message in websocket:
- message = json.loads(message)
- is_finished = message["is_finished"]
- if not is_finished:
- audio = bytes(message['audio'], 'ISO-8859-1')
- frames.append(audio)
- duration_ms = len(audio)//32
- websocket.vad_pre_idx += duration_ms
- is_speaking = message["is_speaking"]
- websocket.param_dict_vad["is_final"] = not is_speaking
- websocket.wav_name = message.get("wav_name", "demo")
- if speech_start:
- frames_asr.append(audio)
- speech_start_i, speech_end_i = await async_vad(websocket, audio)
- if speech_start_i:
- speech_start = True
- beg_bias = (websocket.vad_pre_idx-speech_start_i)//duration_ms
- frames_pre = frames[-beg_bias:]
- frames_asr = []
- frames_asr.extend(frames_pre)
- if speech_end_i or not is_speaking:
- audio_in = b"".join(frames_asr)
- await async_asr(websocket, audio_in)
- frames_asr = []
- speech_start = False
- if not is_speaking:
- websocket.vad_pre_idx = 0
- frames = []
- else:
- frames = frames[-10:]
-
- except websockets.ConnectionClosed:
- print("ConnectionClosed...", websocket_users)
- websocket_users.remove(websocket)
- except websockets.InvalidState:
- print("InvalidState...")
- except Exception as e:
- print("Exception:", e)
- async def async_vad(websocket, audio_in):
- segments_result = inference_pipeline_vad(audio_in=audio_in, param_dict=websocket.param_dict_vad)
- speech_start = False
- speech_end = False
-
- if len(segments_result) == 0 or len(segments_result["text"]) > 1:
- return speech_start, speech_end
- if segments_result["text"][0][0] != -1:
- speech_start = segments_result["text"][0][0]
- if segments_result["text"][0][1] != -1:
- speech_end = True
- return speech_start, speech_end
- 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)
- if inference_pipeline_punc is not None and 'text' in rec_result and len(rec_result["text"])>0:
- rec_result = inference_pipeline_punc(text_in=rec_result['text'],
- param_dict=websocket.param_dict_punc)
- # print(rec_result)
- message = json.dumps({"mode": "offline", "text": [rec_result["text"]], "wav_name": websocket.wav_name})
- await websocket.send(message)
-
-
-
- start_server = websockets.serve(ws_serve, args.host, args.port, subprotocols=["binary"], ping_interval=None)
- asyncio.get_event_loop().run_until_complete(start_server)
- asyncio.get_event_loop().run_forever()
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