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https://github.com/modelscope/FunASR
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funasr1.0 streaming
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@ -4,7 +4,8 @@
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# MIT License (https://opensource.org/licenses/MIT)
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# MIT License (https://opensource.org/licenses/MIT)
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from funasr import AutoModel
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from funasr import AutoModel
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wav_file = "/Users/zhifu/funasr_github/test_local/asr_example.wav"
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wav_file = "https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/vad_example.wav"
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chunk_size = 60000 # ms
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chunk_size = 60000 # ms
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model = AutoModel(model="/Users/zhifu/Downloads/modelscope_models/speech_fsmn_vad_zh-cn-16k-common-streaming", model_revision="v2.0.0")
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model = AutoModel(model="/Users/zhifu/Downloads/modelscope_models/speech_fsmn_vad_zh-cn-16k-common-streaming", model_revision="v2.0.0")
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@ -14,23 +15,23 @@ res = model(input=wav_file,
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print(res)
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print(res)
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#
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# import soundfile
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import soundfile
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# import os
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import os
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#
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# # wav_file = os.path.join(model.model_path, "example/vad_example.wav")
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wav_file = os.path.join(model.model_path, "example/vad_example.wav")
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# speech, sample_rate = soundfile.read(wav_file)
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speech, sample_rate = soundfile.read(wav_file)
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#
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# chunk_stride = int(chunk_size * 16000 / 1000)
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chunk_stride = int(chunk_size * 16000 / 1000)
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#
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# cache = {}
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cache = {}
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#
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# for i in range(int(len((speech)-1)/chunk_stride+1)):
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for i in range(int(len((speech)-1)/chunk_stride+1)):
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# speech_chunk = speech[i*chunk_stride:(i+1)*chunk_stride]
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speech_chunk = speech[i*chunk_stride:(i+1)*chunk_stride]
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# is_final = i == int(len((speech)-1)/chunk_stride+1)
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is_final = i == int(len((speech)-1)/chunk_stride+1) - 1
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# res = model(input=speech_chunk,
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res = model(input=speech_chunk,
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# cache=cache,
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cache=cache,
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# is_final=is_final,
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is_final=is_final,
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# chunk_size=chunk_size,
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chunk_size=chunk_size,
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# )
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)
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# print(res)
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print(res)
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@ -31,7 +31,7 @@ cache = {}
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for i in range(int(len((speech)-1)/chunk_stride+1)):
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for i in range(int(len((speech)-1)/chunk_stride+1)):
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speech_chunk = speech[i*chunk_stride:(i+1)*chunk_stride]
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speech_chunk = speech[i*chunk_stride:(i+1)*chunk_stride]
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is_final = i == int(len((speech)-1)/chunk_stride+1)
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is_final = i == int(len((speech)-1)/chunk_stride+1) - 1
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res = model(input=speech_chunk,
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res = model(input=speech_chunk,
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cache=cache,
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cache=cache,
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is_final=is_final,
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is_final=is_final,
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@ -448,8 +448,8 @@ class WavFrontendOnline(nn.Module):
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feats = torch.stack(cache["lfr_splice_cache"])
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feats = torch.stack(cache["lfr_splice_cache"])
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feats_lengths = torch.zeros(batch_size, dtype=torch.int) + feats.shape[1]
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feats_lengths = torch.zeros(batch_size, dtype=torch.int) + feats.shape[1]
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feats, feats_lengths, _ = self.forward_lfr_cmvn(feats, feats_lengths, is_final, cache=cache)
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feats, feats_lengths, _ = self.forward_lfr_cmvn(feats, feats_lengths, is_final, cache=cache)
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if is_final:
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# if is_final:
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self.init_cache(cache)
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# self.init_cache(cache)
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return feats, feats_lengths
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return feats, feats_lengths
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@ -12,6 +12,7 @@ from funasr.register import tables
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from funasr.utils.load_utils import load_audio_text_image_video,extract_fbank
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from funasr.utils.load_utils import load_audio_text_image_video,extract_fbank
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from funasr.utils.datadir_writer import DatadirWriter
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from funasr.utils.datadir_writer import DatadirWriter
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from torch.nn.utils.rnn import pad_sequence
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from torch.nn.utils.rnn import pad_sequence
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from funasr.train_utils.device_funcs import to_device
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class VadStateMachine(Enum):
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class VadStateMachine(Enum):
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kVadInStateStartPointNotDetected = 1
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kVadInStateStartPointNotDetected = 1
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@ -579,7 +580,8 @@ class FsmnVAD(nn.Module):
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"cache": cache
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"cache": cache
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}
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}
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batch = to_device(batch, device=kwargs["device"])
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segments_part, cache = self.forward(**batch)
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segments_part, cache = self.forward(**batch)
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if segments_part:
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if segments_part:
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for batch_num in range(0, batch_size):
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for batch_num in range(0, batch_size):
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@ -587,7 +587,6 @@ class FsmnVADStreaming(nn.Module):
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"cache": cache["encoder"]
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"cache": cache["encoder"]
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}
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}
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segments_i = self.forward(**batch)
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segments_i = self.forward(**batch)
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print(segments_i)
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segments.extend(segments_i)
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segments.extend(segments_i)
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