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https://github.com/modelscope/FunASR
synced 2025-09-15 14:48:36 +08:00
commit
ba4256ecab
@ -30,14 +30,7 @@ from funasr.models.frontend.wav_frontend import WavFrontendOnline
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from funasr.models.frontend.wav_frontend import WavFrontend
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from funasr.bin.vad_inference import Speech2VadSegment
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header_colors = '\033[95m'
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end_colors = '\033[0m'
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global_asr_language: str = 'zh-cn'
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global_sample_rate: Union[int, Dict[Any, int]] = {
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'audio_fs': 16000,
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'model_fs': 16000
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}
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class Speech2VadSegmentOnline(Speech2VadSegment):
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@ -38,6 +38,10 @@ parser.add_argument("--punc_model",
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type=str,
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default="",
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help="model from modelscope")
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parser.add_argument("--ngpu",
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type=int,
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default=1,
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help="0 for cpu, 1 for gpu")
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args = parser.parse_args()
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@ -52,7 +56,9 @@ vad_pipline = pipeline(
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model_revision="v1.2.0",
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output_dir=None,
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batch_size=1,
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mode='online'
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)
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param_dict_vad = {'in_cache': dict(), "is_final": False}
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# 创建一个ASR对象
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param_dict = dict()
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@ -81,14 +87,22 @@ start_server = websockets.serve(ws_serve, args.host, args.port, subprotocols=["b
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def vad(data): # 推理
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global vad_pipline
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global vad_pipline, param_dict_vad
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#print(type(data))
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segments_result = vad_pipline(audio_in=data)
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#print(segments_result)
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if len(segments_result) == 0:
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return False
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else:
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return True
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# print(param_dict_vad)
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segments_result = vad_pipline(audio_in=data, param_dict=param_dict_vad)
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# print(segments_result)
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# print(param_dict_vad)
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speech_start = False
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speech_end = False
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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 = True
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if segments_result["text"][0][1] != -1:
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speech_end = True
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return speech_start, speech_end
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def asr(): # 推理
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global inference_pipeline2
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@ -106,11 +120,12 @@ def asr(): # 推理
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def main(): # 推理
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frames = [] # 存储所有的帧数据
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buffer = [] # 存储缓存中的帧数据(最多两个片段)
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silence_count = 0 # 统计连续静音的次数
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speech_detected = False # 标记是否检测到语音
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# silence_count = 0 # 统计连续静音的次数
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# speech_detected = False # 标记是否检测到语音
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RECORD_NUM = 0
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global voices
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global speek
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speech_start, speech_end = False, False
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while True:
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while not voices.empty():
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@ -121,32 +136,35 @@ def main(): # 推理
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if len(buffer) > 2:
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buffer.pop(0) # 如果缓存超过两个片段,则删除最早的一个
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if speech_detected:
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if speech_start:
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frames.append(data)
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RECORD_NUM += 1
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if vad(data):
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if not speech_detected:
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print("检测到人声...")
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speech_detected = True # 标记为检测到语音
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frames = []
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frames.extend(buffer) # 把之前2个语音数据快加入
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silence_count = 0 # 重置静音次数
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else:
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silence_count += 1 # 增加静音次数
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if speech_detected and (silence_count > 4 or RECORD_NUM > 50): #这里 50 可根据需求改为合适的数据快数量
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print("说话结束或者超过设置最长时间...")
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audio_in = b"".join(frames)
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#asrt = threading.Thread(target=asr,args=(audio_in,))
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#asrt.start()
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speek.put(audio_in)
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#rec_result = inference_pipeline2(audio_in=audio_in) # ASR 模型里跑一跑
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frames = [] # 清空所有的帧数据
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buffer = [] # 清空缓存中的帧数据(最多两个片段)
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silence_count = 0 # 统计连续静音的次数清零
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speech_detected = False # 标记是否检测到语音
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RECORD_NUM = 0
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RECORD_NUM += 1
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speech_start_i, speech_end_i = vad(data)
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# print(speech_start_i, speech_end_i)
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if speech_start_i:
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speech_start = speech_start_i
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# if not speech_detected:
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# print("检测到人声...")
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# speech_detected = True # 标记为检测到语音
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frames = []
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frames.extend(buffer) # 把之前2个语音数据快加入
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# silence_count = 0 # 重置静音次数
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if speech_end_i or RECORD_NUM > 300:
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# silence_count += 1 # 增加静音次数
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# speech_end = speech_end_i
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speech_start = False
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# if RECORD_NUM > 300: #这里 50 可根据需求改为合适的数据快数量
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# print("说话结束或者超过设置最长时间...")
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audio_in = b"".join(frames)
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#asrt = threading.Thread(target=asr,args=(audio_in,))
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#asrt.start()
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speek.put(audio_in)
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#rec_result = inference_pipeline2(audio_in=audio_in) # ASR 模型里跑一跑
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frames = [] # 清空所有的帧数据
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buffer = [] # 清空缓存中的帧数据(最多两个片段)
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# silence_count = 0 # 统计连续静音的次数清零
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# speech_detected = False # 标记是否检测到语音
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RECORD_NUM = 0
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time.sleep(0.01)
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time.sleep(0.01)
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@ -158,16 +176,4 @@ s = threading.Thread(target=asr)
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s.start()
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asyncio.get_event_loop().run_until_complete(start_server)
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asyncio.get_event_loop().run_forever()
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asyncio.get_event_loop().run_forever()
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