| | |
|---|
| | | websocket.param_dict_punc = {'cache': list()} |
|---|
| | | websocket.vad_pre_idx = 0 |
|---|
| | | speech_start = 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) |
|---|
| | | if len(frames_asr_online) % websocket.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(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: |
|---|
| | | 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: |
|---|
| | | if not websocket.is_speaking: |
|---|
| | | websocket.vad_pre_idx = 0 |
|---|
| | | frames = [] |
|---|
| | | websocket.param_dict_vad = {'in_cache': dict()} |
|---|
| | |
|---|
| | | audio_in = load_bytes(audio_in) |
|---|
| | | 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"]: |
|---|
| | | if websocket.param_dict_asr_online.get("is_final", False): |
|---|
| | | 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": |
|---|
