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@@ -95,7 +95,7 @@ async def ws_serve(websocket, path):
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websocket.param_dict_punc = {'cache': list()}
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websocket.vad_pre_idx = 0
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speech_start = False
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- speech_end_i = False
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+ speech_end_i = -1
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websocket.wav_name = "microphone"
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websocket.mode = "2pass"
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print("new user connected", flush=True)
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@@ -124,7 +124,7 @@ async def ws_serve(websocket, path):
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# asr online
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frames_asr_online.append(message)
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- websocket.param_dict_asr_online["is_final"] = speech_end_i
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+ websocket.param_dict_asr_online["is_final"] = speech_end_i != -1
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if len(frames_asr_online) % websocket.chunk_interval == 0 or websocket.param_dict_asr_online["is_final"]:
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if websocket.mode == "2pass" or websocket.mode == "online":
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audio_in = b"".join(frames_asr_online)
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@@ -134,14 +134,14 @@ async def ws_serve(websocket, path):
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frames_asr.append(message)
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# vad online
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speech_start_i, speech_end_i = await async_vad(websocket, message)
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- if speech_start_i:
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+ if speech_start_i != -1:
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speech_start = True
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beg_bias = (websocket.vad_pre_idx-speech_start_i)//duration_ms
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frames_pre = frames[-beg_bias:]
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frames_asr = []
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frames_asr.extend(frames_pre)
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# asr punc offline
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- if speech_end_i or not websocket.is_speaking:
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+ if speech_end_i != -1 or not websocket.is_speaking:
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# print("vad end point")
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if websocket.mode == "2pass" or websocket.mode == "offline":
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audio_in = b"".join(frames_asr)
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@@ -172,15 +172,15 @@ async def async_vad(websocket, audio_in):
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segments_result = inference_pipeline_vad(audio_in=audio_in, param_dict=websocket.param_dict_vad)
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- speech_start = False
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- speech_end = False
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+ speech_start = -1
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+ speech_end = -1
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if len(segments_result) == 0 or len(segments_result["text"]) > 1:
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return speech_start, speech_end
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if segments_result["text"][0][0] != -1:
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speech_start = segments_result["text"][0][0]
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if segments_result["text"][0][1] != -1:
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- speech_end = True
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+ speech_end = segments_result["text"][0][1]
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return speech_start, speech_end
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游雁