python/FunASR-XL.git

			@@ -5,7 +5,7 @@
			import logging
			import tracemalloc
			import numpy as np

			import ssl
			from parse_args import args
			from modelscope.pipelines import pipeline
			from modelscope.utils.constant import Tasks
			@@ -46,7 +46,7 @@
			inference_pipeline_punc = pipeline(
			task=Tasks.punctuation,
			model=args.punc_model,
			model_revision=None,
			model_revision="v1.0.2",
			ngpu=args.ngpu,
			ncpu=args.ncpu,
			)
			@@ -74,47 +74,58 @@
			websocket.param_dict_punc = {'cache': list()}
			websocket.vad_pre_idx = 0
			speech_start = False
			speech_end_i = False
			websocket.wav_name = "microphone"
			print("new user connected", flush=True)

			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.param_dict_asr_online["is_final"] = not is_speaking
			websocket.param_dict_asr_online["chunk_size"] = message["chunk_size"]
			websocket.wav_name = message.get("wav_name", "demo")
			# asr online
			frames_asr_online.append(audio)
			if len(frames_asr_online) % message["chunk_interval"] == 0:
			audio_in = b"".join(frames_asr_online)
			await async_asr_online(websocket, audio_in)
			frames_asr_online = []
			if speech_start:
			frames_asr.append(audio)
			# vad online
			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 isinstance(message, str):
			messagejson = json.loads(message)

			if "is_speaking" in messagejson:
			websocket.is_speaking = messagejson["is_speaking"]
			websocket.param_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:
			websocket.param_dict_asr_online["chunk_size"] = messagejson["chunk_size"]
			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.param_dict_asr_online["is_final"] = speech_end_i
			if len(frames_asr_online) % websocket.chunk_interval == 0 or websocket.param_dict_asr_online["is_final"]:

			audio_in = b"".join(frames_asr_online)
			await async_asr_online(websocket, audio_in)
			frames_asr_online = []
			if speech_start:
			frames_asr.append(message)
			# vad online
			speech_start_i, speech_end_i = await async_vad(websocket, message)
			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)
			# asr punc offline
			if speech_end_i or not is_speaking:
			if speech_end_i or not websocket.is_speaking:
			# print("vad end point")
			audio_in = b"".join(frames_asr)
			await async_asr(websocket, audio_in)
			frames_asr = []
			speech_start = False
			frames_asr_online = []
			websocket.param_dict_asr_online = {"cache": dict()}
			if not is_speaking:
			# frames_asr_online = []
			# websocket.param_dict_asr_online = {"cache": dict()}
			if not websocket.is_speaking:
			websocket.vad_pre_idx = 0
			frames = []
			websocket.param_dict_vad = {'in_cache': dict()}
			@@ -159,24 +170,37 @@
			rec_result = inference_pipeline_punc(text_in=rec_result['text'],
			param_dict=websocket.param_dict_punc)
			# print("offline", rec_result)
			message = json.dumps({"mode": "2pass-offline", "text": rec_result["text"], "wav_name": websocket.wav_name})
			await websocket.send(message)
			if 'text' in rec_result:
			message = json.dumps({"mode": "2pass-offline", "text": rec_result["text"], "wav_name": websocket.wav_name})
			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 websocket.param_dict_asr_online["is_final"]:
			websocket.param_dict_asr_online["cache"] = dict()
			# print(rec_result)
			if websocket.param_dict_asr_online.get("is_final", False):
			return
			# websocket.param_dict_asr_online["cache"] = dict()
			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})
			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

			start_server = websockets.serve(ws_serve, args.host, args.port, subprotocols=["binary"], ping_interval=None)
			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()