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https://github.com/snakers4/silero-vad.git
synced 2026-02-04 17:39:22 +08:00
additional vad utils
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15
utils_vad.py
15
utils_vad.py
@@ -86,8 +86,11 @@ def get_speech_ts(wav: torch.Tensor,
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min_speech_samples: int = 10000, #samples
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min_silence_samples: int = 500,
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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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smoothed_prob_func='mean',
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device='cpu'):
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assert smoothed_prob_func in ['mean', 'max'], 'smoothed_prob_func not in ["max", "mean"]'
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num_samples = num_samples_per_window
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assert num_samples % num_steps == 0
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step = int(num_samples / num_steps) # stride / hop
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@@ -99,13 +102,13 @@ def get_speech_ts(wav: torch.Tensor,
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chunk = F.pad(chunk, (0, num_samples - len(chunk)))
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to_concat.append(chunk.unsqueeze(0))
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if len(to_concat) >= batch_size:
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chunks = torch.Tensor(torch.cat(to_concat, dim=0))
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chunks = torch.Tensor(torch.cat(to_concat, dim=0)).to(device)
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out = run_function(model, chunks)
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outs.append(out)
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to_concat = []
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if to_concat:
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chunks = torch.Tensor(torch.cat(to_concat, dim=0))
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chunks = torch.Tensor(torch.cat(to_concat, dim=0)).to(device)
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out = run_function(model, chunks)
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outs.append(out)
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@@ -123,7 +126,11 @@ def get_speech_ts(wav: torch.Tensor,
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temp_end = 0
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for i, predict in enumerate(speech_probs): # add name
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buffer.append(predict)
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smoothed_prob = (sum(buffer) / len(buffer))
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if smoothed_prob_func == 'mean':
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smoothed_prob = (sum(buffer) / len(buffer))
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elif smoothed_prob_func == 'max':
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smoothed_prob = max(buffer)
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if visualize_probs:
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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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56
utils_vad_additional.py
Normal file
56
utils_vad_additional.py
Normal file
@@ -0,0 +1,56 @@
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from utils_vad import *
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import sys
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import os
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from pathlib import Path
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sys.path.append('/home/keras/notebook/nvme_raid/adamnsandle/silero_mono/pipelines/align/bin/')
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from align_utils import load_audio_norm
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import torch
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import pandas as pd
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import numpy as np
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sys.path.append('/home/keras/notebook/nvme_raid/adamnsandle/silero_mono/utils/')
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from open_stt import soundfile_opus as sf
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def split_save_audio_chunks(audio_path, model_path, save_path=None, device='cpu', absolute=True, max_duration=10, adaptive=False, **kwargs):
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if not save_path:
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save_path = str(Path(audio_path).with_name('after_vad'))
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print(f'No save path specified! Using {save_path} to save audio chunks!')
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SAMPLE_RATE = 16000
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if type(model_path) == str:
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#print('Loading model...')
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model = init_jit_model(model_path, device)
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else:
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#print('Using loaded model')
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model = model_path
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save_name = Path(audio_path).stem
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audio, sr = load_audio_norm(audio_path)
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wav = torch.tensor(audio)
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if adaptive:
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speech_timestamps = get_speech_ts_adaptive(wav, model, device=device, **kwargs)
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else:
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speech_timestamps = get_speech_ts(wav, model, device=device, **kwargs)
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full_save_path = Path(save_path, save_name)
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if not os.path.exists(full_save_path):
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os.makedirs(full_save_path, exist_ok=True)
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chunks = []
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if not speech_timestamps:
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return pd.DataFrame()
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for ts in speech_timestamps:
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start_ts = int(ts['start'])
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end_ts = int(ts['end'])
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for i in range(start_ts, end_ts, max_duration * SAMPLE_RATE):
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new_start = i
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new_end = min(end_ts, i + max_duration * SAMPLE_RATE)
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duration = round((new_end - new_start) / SAMPLE_RATE, 2)
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chunk_path = Path(full_save_path, f'{save_name}_{new_start}-{new_end}.opus')
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chunk_path = chunk_path.absolute() if absolute else chunk_path
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sf.write(str(chunk_path), audio[new_start: new_end], 16000, format='OGG', subtype='OPUS')
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chunks.append({'audio_path': chunk_path,
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'text': '',
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'duration': duration,
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'domain': ''})
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return pd.DataFrame(chunks)
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