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- import asyncio
- import json
- import websockets
- import time
- import logging
- import tracemalloc
- import numpy as np
- import argparse
- import ssl
- parser = argparse.ArgumentParser()
- parser.add_argument("--host",
- type=str,
- default="0.0.0.0",
- required=False,
- help="host ip, localhost, 0.0.0.0")
- parser.add_argument("--port",
- type=int,
- default=10095,
- required=False,
- help="grpc server port")
- parser.add_argument("--asr_model",
- type=str,
- default="iic/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch",
- help="model from modelscope")
- parser.add_argument("--asr_model_revision",
- type=str,
- default="v2.0.4",
- help="")
- parser.add_argument("--asr_model_online",
- type=str,
- default="iic/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online",
- help="model from modelscope")
- parser.add_argument("--asr_model_online_revision",
- type=str,
- default="v2.0.4",
- help="")
- parser.add_argument("--vad_model",
- type=str,
- default="iic/speech_fsmn_vad_zh-cn-16k-common-pytorch",
- help="model from modelscope")
- parser.add_argument("--vad_model_revision",
- type=str,
- default="v2.0.4",
- help="")
- parser.add_argument("--punc_model",
- type=str,
- default="iic/punc_ct-transformer_zh-cn-common-vad_realtime-vocab272727",
- help="model from modelscope")
- parser.add_argument("--punc_model_revision",
- type=str,
- default="v2.0.4",
- help="")
- parser.add_argument("--ngpu",
- type=int,
- default=1,
- help="0 for cpu, 1 for gpu")
- parser.add_argument("--device",
- type=str,
- default="cuda",
- help="cuda, cpu")
- parser.add_argument("--ncpu",
- type=int,
- default=4,
- help="cpu cores")
- parser.add_argument("--certfile",
- type=str,
- default="../../ssl_key/server.crt",
- required=False,
- help="certfile for ssl")
- parser.add_argument("--keyfile",
- type=str,
- default="../../ssl_key/server.key",
- required=False,
- help="keyfile for ssl")
- args = parser.parse_args()
- websocket_users = set()
- print("model loading")
- from funasr import AutoModel
- # asr
- model_asr = AutoModel(model=args.asr_model,
- model_revision=args.asr_model_revision,
- ngpu=args.ngpu,
- ncpu=args.ncpu,
- device=args.device,
- disable_pbar=True,
- disable_log=True,
- )
- # asr
- model_asr_streaming = AutoModel(model=args.asr_model_online,
- model_revision=args.asr_model_online_revision,
- ngpu=args.ngpu,
- ncpu=args.ncpu,
- device=args.device,
- disable_pbar=True,
- disable_log=True,
- )
- # vad
- model_vad = AutoModel(model=args.vad_model,
- model_revision=args.vad_model_revision,
- ngpu=args.ngpu,
- ncpu=args.ncpu,
- device=args.device,
- disable_pbar=True,
- disable_log=True,
- # chunk_size=60,
- )
- if args.punc_model != "":
- model_punc = AutoModel(model=args.punc_model,
- model_revision=args.punc_model_revision,
- ngpu=args.ngpu,
- ncpu=args.ncpu,
- device=args.device,
- disable_pbar=True,
- disable_log=True,
- )
- else:
- model_punc = None
- print("model loaded! only support one client at the same time now!!!!")
- async def ws_reset(websocket):
- print("ws reset now, total num is ",len(websocket_users))
- websocket.status_dict_asr_online["cache"] = {}
- websocket.status_dict_asr_online["is_final"] = True
- websocket.status_dict_vad["cache"] = {}
- websocket.status_dict_vad["is_final"] = True
- websocket.status_dict_punc["cache"] = {}
-
- await websocket.close()
- async def clear_websocket():
- for websocket in websocket_users:
- await ws_reset(websocket)
- websocket_users.clear()
- async def ws_serve(websocket, path):
- frames = []
- frames_asr = []
- frames_asr_online = []
- global websocket_users
- # await clear_websocket()
- websocket_users.add(websocket)
- websocket.status_dict_asr = {}
- websocket.status_dict_asr_online = {"cache": {}, "is_final": False}
- websocket.status_dict_vad = {'cache': {}, "is_final": False}
- websocket.status_dict_punc = {'cache': {}}
- websocket.chunk_interval = 10
- websocket.vad_pre_idx = 0
- speech_start = False
- speech_end_i = -1
- websocket.wav_name = "microphone"
- websocket.mode = "2pass"
- print("new user connected", flush=True)
-
- try:
- async for message in websocket:
- if isinstance(message, str):
- messagejson = json.loads(message)
-
- if "is_speaking" in messagejson:
- websocket.is_speaking = messagejson["is_speaking"]
- websocket.status_dict_asr_online["is_final"] = not websocket.is_speaking
- if "chunk_interval" in messagejson:
- websocket.chunk_interval = messagejson["chunk_interval"]
- if "wav_name" in messagejson:
- websocket.wav_name = messagejson.get("wav_name")
- if "chunk_size" in messagejson:
- chunk_size = messagejson["chunk_size"]
- if isinstance(chunk_size, str):
- chunk_size = chunk_size.split(',')
- websocket.status_dict_asr_online["chunk_size"] = [int(x) for x in chunk_size]
- if "encoder_chunk_look_back" in messagejson:
- websocket.status_dict_asr_online["encoder_chunk_look_back"] = messagejson["encoder_chunk_look_back"]
- if "decoder_chunk_look_back" in messagejson:
- websocket.status_dict_asr_online["decoder_chunk_look_back"] = messagejson["decoder_chunk_look_back"]
- if "hotword" in messagejson:
- websocket.status_dict_asr["hotword"] = messagejson["hotword"]
- if "mode" in messagejson:
- websocket.mode = messagejson["mode"]
-
- websocket.status_dict_vad["chunk_size"] = int(websocket.status_dict_asr_online["chunk_size"][1]*60/websocket.chunk_interval)
- if len(frames_asr_online) > 0 or len(frames_asr) > 0 or not isinstance(message, str):
- if not isinstance(message, str):
- frames.append(message)
- duration_ms = len(message)//32
- websocket.vad_pre_idx += duration_ms
-
- # asr online
- frames_asr_online.append(message)
- websocket.status_dict_asr_online["is_final"] = speech_end_i != -1
- if len(frames_asr_online) % websocket.chunk_interval == 0 or websocket.status_dict_asr_online["is_final"]:
- if websocket.mode == "2pass" or websocket.mode == "online":
- audio_in = b"".join(frames_asr_online)
- try:
- await async_asr_online(websocket, audio_in)
- except:
- print(f"error in asr streaming, {websocket.status_dict_asr_online}")
- frames_asr_online = []
- if speech_start:
- frames_asr.append(message)
- # vad online
- try:
- speech_start_i, speech_end_i = await async_vad(websocket, message)
- except:
- print("error in vad")
- if speech_start_i != -1:
- 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)
- # asr punc offline
- if speech_end_i != -1 or not websocket.is_speaking:
- # print("vad end point")
- if websocket.mode == "2pass" or websocket.mode == "offline":
- audio_in = b"".join(frames_asr)
- try:
- await async_asr(websocket, audio_in)
- except:
- print("error in asr offline")
- frames_asr = []
- speech_start = False
- frames_asr_online = []
- websocket.status_dict_asr_online["cache"] = {}
- if not websocket.is_speaking:
- websocket.vad_pre_idx = 0
- frames = []
- websocket.status_dict_vad["cache"] = {}
- else:
- frames = frames[-20:]
-
-
- except websockets.ConnectionClosed:
- print("ConnectionClosed...", websocket_users,flush=True)
- await ws_reset(websocket)
- 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 = model_vad.generate(input=audio_in, **websocket.status_dict_vad)[0]["value"]
- # print(segments_result)
-
- speech_start = -1
- speech_end = -1
-
- if len(segments_result) == 0 or len(segments_result) > 1:
- return speech_start, speech_end
- if segments_result[0][0] != -1:
- speech_start = segments_result[0][0]
- if segments_result[0][1] != -1:
- speech_end = segments_result[0][1]
- return speech_start, speech_end
- async def async_asr(websocket, audio_in):
- if len(audio_in) > 0:
- # print(len(audio_in))
- rec_result = model_asr.generate(input=audio_in, **websocket.status_dict_asr)[0]
- # print("offline_asr, ", rec_result)
- if model_punc is not None and len(rec_result["text"])>0:
- # print("offline, before punc", rec_result, "cache", websocket.status_dict_punc)
- rec_result = model_punc.generate(input=rec_result['text'], **websocket.status_dict_punc)[0]
- # print("offline, after punc", rec_result)
- if len(rec_result["text"])>0:
- # print("offline", rec_result)
- 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:
- # print(websocket.status_dict_asr_online.get("is_final", False))
- rec_result = model_asr_streaming.generate(input=audio_in, **websocket.status_dict_asr_online)[0]
- # print("online, ", rec_result)
- if websocket.mode == "2pass" and websocket.status_dict_asr_online.get("is_final", False):
- return
- # websocket.status_dict_asr_online["cache"] = dict()
- if len(rec_result["text"]):
- 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:
- ssl_context = ssl.SSLContext(ssl.PROTOCOL_TLS_SERVER)
-
- # Generate with Lets Encrypt, copied to this location, chown to current user and 400 permissions
- ssl_cert = args.certfile
- ssl_key = args.keyfile
-
- ssl_context.load_cert_chain(ssl_cert, keyfile=ssl_key)
- start_server = websockets.serve(ws_serve, args.host, args.port, subprotocols=["binary"], ping_interval=None,ssl=ssl_context)
- else:
- 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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