| from sklearn.metrics import roc_auc_score, accuracy_score |
| from torch.utils.data import Dataset |
| import torch.nn as nn |
| from tqdm import tqdm |
| import pandas as pd |
| import numpy as np |
| import torchaudio |
| import warnings |
| import random |
| import torch |
| import gc |
| warnings.filterwarnings('ignore') |
|
|
|
|
| def read_audio(path: str, |
| sampling_rate: int = 16000, |
| normalize=False): |
|
|
| wav, sr = torchaudio.load(path) |
|
|
| if wav.size(0) > 1: |
| wav = wav.mean(dim=0, keepdim=True) |
|
|
| if sampling_rate: |
| if sr != sampling_rate: |
| transform = torchaudio.transforms.Resample(orig_freq=sr, |
| new_freq=sampling_rate) |
| wav = transform(wav) |
| sr = sampling_rate |
|
|
| if normalize and wav.abs().max() != 0: |
| wav = wav / wav.abs().max() |
|
|
| return wav.squeeze(0) |
|
|
|
|
| def build_audiomentations_augs(p): |
| from audiomentations import SomeOf, AirAbsorption, BandPassFilter, BandStopFilter, ClippingDistortion, HighPassFilter, HighShelfFilter, \ |
| LowPassFilter, LowShelfFilter, Mp3Compression, PeakingFilter, PitchShift, RoomSimulator, SevenBandParametricEQ, \ |
| Aliasing, AddGaussianNoise |
| transforms = [Aliasing(p=1), |
| AddGaussianNoise(p=1), |
| AirAbsorption(p=1), |
| BandPassFilter(p=1), |
| BandStopFilter(p=1), |
| ClippingDistortion(p=1), |
| HighPassFilter(p=1), |
| HighShelfFilter(p=1), |
| LowPassFilter(p=1), |
| LowShelfFilter(p=1), |
| Mp3Compression(p=1), |
| PeakingFilter(p=1), |
| PitchShift(p=1), |
| RoomSimulator(p=1, leave_length_unchanged=True), |
| SevenBandParametricEQ(p=1)] |
| tr = SomeOf((1, 3), transforms=transforms, p=p) |
| return tr |
|
|
|
|
| class SileroVadDataset(Dataset): |
| def __init__(self, |
| config, |
| mode='train'): |
|
|
| self.num_samples = 512 |
| self.sr = 16000 |
|
|
| self.resample_to_8k = config.tune_8k |
| self.noise_loss = config.noise_loss |
| self.max_train_length_sec = config.max_train_length_sec |
| self.max_train_length_samples = config.max_train_length_sec * self.sr |
|
|
| assert self.max_train_length_samples % self.num_samples == 0 |
| assert mode in ['train', 'val'] |
|
|
| dataset_path = config.train_dataset_path if mode == 'train' else config.val_dataset_path |
| self.dataframe = pd.read_feather(dataset_path).reset_index(drop=True) |
| self.index_dict = self.dataframe.to_dict('index') |
| self.mode = mode |
| print(f'DATASET SIZE : {len(self.dataframe)}') |
|
|
| if mode == 'train': |
| self.augs = build_audiomentations_augs(p=config.aug_prob) |
| else: |
| self.augs = None |
|
|
| def __getitem__(self, idx): |
| idx = None if self.mode == 'train' else idx |
| wav, gt, mask = self.load_speech_sample(idx) |
|
|
| if self.mode == 'train': |
| wav = self.add_augs(wav) |
| if len(wav) > self.max_train_length_samples: |
| wav = wav[:self.max_train_length_samples] |
| gt = gt[:int(self.max_train_length_samples / self.num_samples)] |
| mask = mask[:int(self.max_train_length_samples / self.num_samples)] |
|
|
| wav = torch.FloatTensor(wav) |
| if self.resample_to_8k: |
| transform = torchaudio.transforms.Resample(orig_freq=self.sr, |
| new_freq=8000) |
| wav = transform(wav) |
| return wav, torch.FloatTensor(gt), torch.from_numpy(mask) |
|
|
| def __len__(self): |
| return len(self.index_dict) |
|
|
| def load_speech_sample(self, idx=None): |
| if idx is None: |
| idx = random.randint(0, len(self.index_dict) - 1) |
| wav = read_audio(self.index_dict[idx]['audio_path'], self.sr).numpy() |
|
|
| if len(wav) % self.num_samples != 0: |
| pad_num = self.num_samples - (len(wav) % (self.num_samples)) |
| wav = np.pad(wav, (0, pad_num), 'constant', constant_values=0) |
|
|
| gt, mask = self.get_ground_truth_annotated(self.index_dict[idx]['speech_ts'], len(wav)) |
|
|
| assert len(gt) == len(wav) / self.num_samples |
|
|
| return wav, gt, mask |
|
|
| def get_ground_truth_annotated(self, annotation, audio_length_samples): |
| gt = np.zeros(audio_length_samples) |
|
|
| for i in annotation: |
| gt[int(i['start'] * self.sr): int(i['end'] * self.sr)] = 1 |
|
|
| squeezed_predicts = np.average(gt.reshape(-1, self.num_samples), axis=1) |
| squeezed_predicts = (squeezed_predicts > 0.5).astype(int) |
| mask = np.ones(len(squeezed_predicts)) |
| mask[squeezed_predicts == 0] = self.noise_loss |
| return squeezed_predicts, mask |
|
|
| def add_augs(self, wav): |
| while True: |
| try: |
| wav_aug = self.augs(wav, self.sr) |
| if np.isnan(wav_aug.max()) or np.isnan(wav_aug.min()): |
| return wav |
| return wav_aug |
| except Exception as e: |
| continue |
|
|
|
|
| def SileroVadPadder(batch): |
| wavs = [batch[i][0] for i in range(len(batch))] |
| labels = [batch[i][1] for i in range(len(batch))] |
| masks = [batch[i][2] for i in range(len(batch))] |
|
|
| wavs = torch.nn.utils.rnn.pad_sequence( |
| wavs, batch_first=True, padding_value=0) |
|
|
| labels = torch.nn.utils.rnn.pad_sequence( |
| labels, batch_first=True, padding_value=0) |
|
|
| masks = torch.nn.utils.rnn.pad_sequence( |
| masks, batch_first=True, padding_value=0) |
|
|
| return wavs, labels, masks |
|
|
|
|
| class VADDecoderRNNJIT(nn.Module): |
|
|
| def __init__(self): |
| super(VADDecoderRNNJIT, self).__init__() |
|
|
| self.rnn = nn.LSTMCell(128, 128) |
| self.decoder = nn.Sequential(nn.Dropout(0.1), |
| nn.ReLU(), |
| nn.Conv1d(128, 1, kernel_size=1), |
| nn.Sigmoid()) |
|
|
| def forward(self, x, state=torch.zeros(0)): |
| x = x.squeeze(-1) |
| if len(state): |
| h, c = self.rnn(x, (state[0], state[1])) |
| else: |
| h, c = self.rnn(x) |
|
|
| x = h.unsqueeze(-1).float() |
| state = torch.stack([h, c]) |
| x = self.decoder(x) |
| return x, state |
|
|
|
|
| class AverageMeter(object): |
| """Computes and stores the average and current value""" |
|
|
| def __init__(self): |
| self.reset() |
|
|
| def reset(self): |
| self.val = 0 |
| self.avg = 0 |
| self.sum = 0 |
| self.count = 0 |
|
|
| def update(self, val, n=1): |
| self.val = val |
| self.sum += val * n |
| self.count += n |
| self.avg = self.sum / self.count |
|
|
|
|
| def train(config, |
| loader, |
| jit_model, |
| decoder, |
| criterion, |
| optimizer, |
| device): |
|
|
| losses = AverageMeter() |
| decoder.train() |
|
|
| context_size = 32 if config.tune_8k else 64 |
| num_samples = 256 if config.tune_8k else 512 |
| stft_layer = jit_model._model_8k.stft if config.tune_8k else jit_model._model.stft |
| encoder_layer = jit_model._model_8k.encoder if config.tune_8k else jit_model._model.encoder |
|
|
| with torch.enable_grad(): |
| for _, (x, targets, masks) in tqdm(enumerate(loader), total=len(loader)): |
| targets = targets.to(device) |
| x = x.to(device) |
| masks = masks.to(device) |
| x = torch.nn.functional.pad(x, (context_size, 0)) |
|
|
| outs = [] |
| state = torch.zeros(0) |
| for i in range(context_size, x.shape[1], num_samples): |
| input_ = x[:, i-context_size:i+num_samples] |
| out = stft_layer(input_) |
| out = encoder_layer(out) |
| out, state = decoder(out, state) |
| outs.append(out) |
| stacked = torch.cat(outs, dim=2).squeeze(1) |
|
|
| loss = criterion(stacked, targets) |
| loss = (loss * masks).mean() |
| optimizer.zero_grad() |
| loss.backward() |
| optimizer.step() |
| losses.update(loss.item(), masks.numel()) |
|
|
| torch.cuda.empty_cache() |
| gc.collect() |
|
|
| return losses.avg |
|
|
|
|
| def validate(config, |
| loader, |
| jit_model, |
| decoder, |
| criterion, |
| device): |
|
|
| losses = AverageMeter() |
| decoder.eval() |
|
|
| predicts = [] |
| gts = [] |
|
|
| context_size = 32 if config.tune_8k else 64 |
| num_samples = 256 if config.tune_8k else 512 |
| stft_layer = jit_model._model_8k.stft if config.tune_8k else jit_model._model.stft |
| encoder_layer = jit_model._model_8k.encoder if config.tune_8k else jit_model._model.encoder |
|
|
| with torch.no_grad(): |
| for _, (x, targets, masks) in tqdm(enumerate(loader), total=len(loader)): |
| targets = targets.to(device) |
| x = x.to(device) |
| masks = masks.to(device) |
| x = torch.nn.functional.pad(x, (context_size, 0)) |
|
|
| outs = [] |
| state = torch.zeros(0) |
| for i in range(context_size, x.shape[1], num_samples): |
| input_ = x[:, i-context_size:i+num_samples] |
| out = stft_layer(input_) |
| out = encoder_layer(out) |
| out, state = decoder(out, state) |
| outs.append(out) |
| stacked = torch.cat(outs, dim=2).squeeze(1) |
|
|
| predicts.extend(stacked[masks != 0].tolist()) |
| gts.extend(targets[masks != 0].tolist()) |
|
|
| loss = criterion(stacked, targets) |
| loss = (loss * masks).mean() |
| losses.update(loss.item(), masks.numel()) |
| score = roc_auc_score(gts, predicts) |
|
|
| torch.cuda.empty_cache() |
| gc.collect() |
|
|
| return losses.avg, round(score, 3) |
|
|
|
|
| def init_jit_model(model_path: str, |
| device=torch.device('cpu')): |
| torch.set_grad_enabled(False) |
| model = torch.jit.load(model_path, map_location=device) |
| model.eval() |
| return model |
|
|
|
|
| def predict(model, loader, device, sr): |
| with torch.no_grad(): |
| all_predicts = [] |
| all_gts = [] |
| for _, (x, targets, masks) in tqdm(enumerate(loader), total=len(loader)): |
| x = x.to(device) |
| out = model.audio_forward(x, sr=sr) |
|
|
| for i, out_chunk in enumerate(out): |
| predict = out_chunk[masks[i] != 0].cpu().tolist() |
| gt = targets[i, masks[i] != 0].cpu().tolist() |
|
|
| all_predicts.append(predict) |
| all_gts.append(gt) |
| return all_predicts, all_gts |
|
|
|
|
| def calculate_best_thresholds(all_predicts, all_gts): |
| best_acc = 0 |
| for ths_enter in tqdm(np.linspace(0, 1, 20)): |
| for ths_exit in np.linspace(0, 1, 20): |
| if ths_exit >= ths_enter: |
| continue |
|
|
| accs = [] |
| for j, predict in enumerate(all_predicts): |
| predict_bool = [] |
| is_speech = False |
| for i in predict: |
| if i >= ths_enter: |
| is_speech = True |
| predict_bool.append(1) |
| elif i <= ths_exit: |
| is_speech = False |
| predict_bool.append(0) |
| else: |
| val = 1 if is_speech else 0 |
| predict_bool.append(val) |
|
|
| score = round(accuracy_score(all_gts[j], predict_bool), 4) |
| accs.append(score) |
|
|
| mean_acc = round(np.mean(accs), 3) |
| if mean_acc > best_acc: |
| best_acc = mean_acc |
| best_ths_enter = round(ths_enter, 2) |
| best_ths_exit = round(ths_exit, 2) |
| return best_ths_enter, best_ths_exit, best_acc |
|
|