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https://github.com/snakers4/silero-vad.git
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Merge branch 'adamnsandle' of github.com:snakers4/silero-vad into adamnsandle
This commit is contained in:
@@ -346,6 +346,7 @@ Since our VAD (only VAD, other networks are more flexible) was trained on chunks
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- `num_steps` - nubmer of overlapping windows to split audio chunk into (we recommend 4 or 8)
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- `num_steps` - nubmer of overlapping windows to split audio chunk into (we recommend 4 or 8)
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- `num_samples_per_window` - number of samples in each window, our models were trained using `4000` samples (250 ms) per window, so this is preferable value (lesser values reduce [quality](https://github.com/snakers4/silero-vad/issues/2#issuecomment-750840434));
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- `num_samples_per_window` - number of samples in each window, our models were trained using `4000` samples (250 ms) per window, so this is preferable value (lesser values reduce [quality](https://github.com/snakers4/silero-vad/issues/2#issuecomment-750840434));
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- `min_speech_samples` - minimum speech chunk duration in samples
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- `min_speech_samples` - minimum speech chunk duration in samples
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- `min_silence_samples` - minimum silence duration in samples between to separate speech chunks
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Optimal parameters may vary per domain, but we provided a tiny tool to learn the best parameters. You can invoke `speech_timestamps` with visualize_probs=True (`pandas` required):
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Optimal parameters may vary per domain, but we provided a tiny tool to learn the best parameters. You can invoke `speech_timestamps` with visualize_probs=True (`pandas` required):
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22
utils_vad.py
22
utils_vad.py
@@ -60,6 +60,7 @@ def get_speech_ts(wav: torch.Tensor,
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batch_size: int = 200,
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batch_size: int = 200,
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num_samples_per_window: int = 4000,
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num_samples_per_window: int = 4000,
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min_speech_samples: int = 10000, #samples
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min_speech_samples: int = 10000, #samples
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min_silence_samples: int = 8000,
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run_function=validate,
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run_function=validate,
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visualize_probs=False):
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visualize_probs=False):
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@@ -95,20 +96,31 @@ def get_speech_ts(wav: torch.Tensor,
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smoothed_probs = []
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smoothed_probs = []
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speech_probs = outs[:, 1] # this is very misleading
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speech_probs = outs[:, 1] # this is very misleading
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temp_end = 0
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for i, predict in enumerate(speech_probs): # add name
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for i, predict in enumerate(speech_probs): # add name
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buffer.append(predict)
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buffer.append(predict)
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smoothed_prob = (sum(buffer) / len(buffer))
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smoothed_prob = (sum(buffer) / len(buffer))
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if visualize_probs:
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if visualize_probs:
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smoothed_probs.append(float(smoothed_prob))
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smoothed_probs.append(float(smoothed_prob))
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if (smoothed_prob >= trig_sum) and temp_end:
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temp_end=0
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if (smoothed_prob >= trig_sum) and not triggered:
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if (smoothed_prob >= trig_sum) and not triggered:
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triggered = True
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triggered = True
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current_speech['start'] = step * max(0, i-num_steps)
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current_speech['start'] = step * max(0, i-num_steps)
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continue
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if (smoothed_prob < neg_trig_sum) and triggered:
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if (smoothed_prob < neg_trig_sum) and triggered:
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current_speech['end'] = step * i
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if not temp_end:
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if (current_speech['end'] - current_speech['start']) > min_speech_samples:
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temp_end = step * i
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speeches.append(current_speech)
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if step * i - temp_end < min_silence_samples:
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current_speech = {}
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continue
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triggered = False
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else:
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current_speech['end'] = temp_end
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if (current_speech['end'] - current_speech['start']) > min_speech_samples:
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speeches.append(current_speech)
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temp_end = 0
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current_speech = {}
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triggered = False
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continue
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if current_speech:
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if current_speech:
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current_speech['end'] = len(wav)
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current_speech['end'] = len(wav)
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speeches.append(current_speech)
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speeches.append(current_speech)
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