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class DeepModel(nn.Module): def __init__(self, input_dim, output_dim, model_type, pooling, **kwargs): super(DeepModel, self).__init__() self.pooling = pooling self.model = eval(model_type)(input_dim=input_dim, output_class_num=output_dim, pooling=pooling, **kwargs) def forward(self, features, features_len): attention_mask = [torch.ones(math.ceil((l / self.pooling))) for l in features_len] attention_mask = pad_sequence(attention_mask, batch_first=True) attention_mask = ((1.0 - attention_mask) * (- 100000.0)) attention_mask = attention_mask.to(features.device) predicted = self.model(features, attention_mask) return (predicted, None)
class SeparationDataset(Dataset): def __init__(self, data_dir, rate=16000, src=['mix_clean'], tgt=['s1', 's2'], n_fft=512, hop_length=320, win_length=512, window='hann', center=True): super(SeparationDataset, self).__init__() "\n Args:\n data_dir (str):\n prepared data directory\n\n rate (int):\n audio sample rate\n\n src and tgt (list(str)):\n the input and desired output.\n LibriMix offeres different options for the users. For\n clean source separation, src=['mix_clean'] and tgt=['s1', 's2'].\n Please see https://github.com/JorisCos/LibriMix for details\n\n n_fft (int):\n length of the windowed signal after padding with zeros.\n\n hop_length (int):\n number of audio samples between adjacent STFT columns.\n\n win_length (int):\n length of window for each frame\n\n window (str):\n type of window function, only support Hann window now\n\n center (bool):\n whether to pad input on both sides so that the\n t-th frame is centered at time t * hop_length\n\n The STFT related parameters are the same as librosa.\n " self.data_dir = data_dir self.rate = rate self.src = src self.tgt = tgt self.n_fft = n_fft self.hop_length = hop_length self.win_length = win_length self.window = window self.center = center self.n_srcs = len(self.tgt) assert ((len(self.src) == 1) and (len(self.tgt) == 1)) cond_list = ['s1', 's2', 'noise', 'mix_clean', 'mix_both', 'mix_single', 'noisy', 'clean'] reco2path = {} for cond in (src + tgt): assert (cond in cond_list) assert os.path.exists('{}/{}/wav.scp'.format(self.data_dir, cond)) with open('{}/{}/wav.scp'.format(self.data_dir, cond), 'r') as fh: content = fh.readlines() for line in content: line = line.strip('\n') (uttname, path) = line.split() if (uttname not in reco2path): reco2path[uttname] = {} reco2path[uttname][cond] = path self.reco2path = reco2path self.recolist = list(self.reco2path.keys()) self.recolist.sort() def __len__(self): return len(self.recolist) def __getitem__(self, i): reco = self.recolist[i] src_path = self.reco2path[reco][self.src[0]] (src_samp, rate) = librosa.load(src_path, sr=SAMPLE_RATE) assert (rate == self.rate) src_feat = np.transpose(librosa.stft(src_samp, n_fft=self.n_fft, hop_length=self.hop_length, win_length=self.win_length, window=self.window, center=self.center)) (tgt_samp_list, tgt_feat_list) = ([], []) for j in range(self.n_srcs): tgt_path = self.reco2path[reco][self.tgt[j]] (tgt_samp, rate) = librosa.load(tgt_path, sr=SAMPLE_RATE) assert (rate == self.rate) tgt_feat = np.transpose(librosa.stft(tgt_samp, n_fft=self.n_fft, hop_length=self.hop_length, win_length=self.win_length, window=self.window, center=self.center)) tgt_samp_list.append(tgt_samp) tgt_feat_list.append(tgt_feat) '\n reco (str):\n name of the utterance\n\n src_sample (ndarray):\n audio samples for the source [T, ]\n\n src_feat (ndarray):\n the STFT feature map for the source with shape [T1, D]\n\n tgt_samp_list (list(ndarray)):\n list of audio samples for the targets\n\n tgt_feat_list (list(ndarray)):\n list of STFT feature map for the targets\n ' return (reco, src_samp, src_feat, tgt_samp_list, tgt_feat_list) def collate_fn(self, batch): sorted_batch = sorted(batch, key=(lambda x: (- x[1].shape[0]))) bs = len(sorted_batch) uttname_list = [sorted_batch[i][0] for i in range(bs)] source_attr = {} mix_magnitude_list = [torch.from_numpy(np.abs(sorted_batch[i][2])) for i in range(bs)] mix_phase_list = [torch.from_numpy(np.angle(sorted_batch[i][2])) for i in range(bs)] mix_stft_list = [torch.from_numpy(sorted_batch[i][2]) for i in range(bs)] mix_magnitude = pad_sequence(mix_magnitude_list, batch_first=True) mix_phase = pad_sequence(mix_phase_list, batch_first=True) mix_stft = pad_sequence(mix_stft_list, batch_first=True) source_attr['magnitude'] = mix_magnitude source_attr['phase'] = mix_phase source_attr['stft'] = mix_stft target_attr = {} target_attr['magnitude'] = [] target_attr['phase'] = [] for j in range(self.n_srcs): tgt_magnitude_list = [torch.from_numpy(np.abs(sorted_batch[i][4][j])) for i in range(bs)] tgt_phase_list = [torch.from_numpy(np.angle(sorted_batch[i][4][j])) for i in range(bs)] tgt_magnitude = pad_sequence(tgt_magnitude_list, batch_first=True) tgt_phase = pad_sequence(tgt_phase_list, batch_first=True) target_attr['magnitude'].append(tgt_magnitude) target_attr['phase'].append(tgt_phase) wav_length = torch.from_numpy(np.array([len(sorted_batch[i][1]) for i in range(bs)])) source_wav_list = [torch.from_numpy(sorted_batch[i][1]) for i in range(bs)] source_wav = pad_sequence(source_wav_list, batch_first=True) target_wav_list = [] for j in range(self.n_srcs): target_wav_list.append(pad_sequence([torch.from_numpy(sorted_batch[i][3][j]) for i in range(bs)], batch_first=True)) feat_length = torch.from_numpy(np.array([stft.size(0) for stft in mix_stft_list])) '\n source_wav_list (list(tensor)):\n list of audio samples for the source\n\n uttname_list (list(str)):\n list of utterance names\n\n source_attr (dict):\n dictionary containing magnitude and phase information for the sources\n\n source_wav (tensor):\n padded version of source_wav_list, with size [bs, max_T]\n\n target_attr (dict):\n dictionary containing magnitude and phase information for the targets\n\n feat_length (tensor):\n length of the STFT feature for each utterance\n\n wav_length (tensor):\n number of samples in each utterance\n ' return (source_wav_list, uttname_list, source_attr, source_wav, target_attr, target_wav_list, feat_length, wav_length)
class SepRNN(torch.nn.Module): def __init__(self, input_dim, num_bins, rnn='lstm', num_spks=2, num_layers=3, hidden_size=896, dropout=0.0, non_linear='relu', bidirectional=True): super(SepRNN, self).__init__() if (non_linear not in ['relu', 'sigmoid', 'tanh']): raise ValueError('Unsupported non-linear type:{}'.format(non_linear)) self.num_spks = num_spks rnn = rnn.upper() if (rnn not in ['RNN', 'LSTM', 'GRU']): raise ValueError('Unsupported rnn type: {}'.format(rnn)) self.rnn = getattr(torch.nn, rnn)(input_dim, hidden_size, num_layers, batch_first=True, dropout=dropout, bidirectional=bidirectional) self.drops = torch.nn.Dropout(p=dropout) self.linear = torch.nn.ModuleList([torch.nn.Linear(((hidden_size * 2) if bidirectional else hidden_size), num_bins) for _ in range(self.num_spks)]) self.non_linear = {'relu': torch.nn.functional.relu, 'sigmoid': torch.nn.functional.sigmoid, 'tanh': torch.nn.functional.tanh}[non_linear] self.num_bins = num_bins def forward(self, x, train=True): assert isinstance(x, PackedSequence) (x, _) = self.rnn(x) (x, len_x) = pad_packed_sequence(x, batch_first=True) x = self.drops(x) m = [] for linear in self.linear: y = linear(x) y = self.non_linear(y) if (not train): y = y.view((- 1), self.num_bins) m.append(y) return m
def main(): output_dir = '{}/wav{}/{}/{}'.format(args.tgt_dir, args.sample_rate, args.mode, args.part) if os.path.exists(output_dir): raise ValueError('Warning: {} already exists, please check!') else: os.makedirs(output_dir) wav_dir = '{}/wav{}/{}/{}'.format(args.src_dir, args.sample_rate, args.mode, args.part) assert os.path.exists(wav_dir) for cond in ['s1', 's2', 'mix_clean', 'mix_both', 'mix_single', 'noise']: filelist = [f for f in os.listdir('{}/{}'.format(wav_dir, cond)) if f.endswith('.wav')] filelist.sort() cond_dir = '{}/{}'.format(output_dir, cond) if (not os.path.exists(cond_dir)): os.makedirs(cond_dir) wav_scp_file = open('{}/wav.scp'.format(cond_dir), 'w') utt2spk_file = open('{}/utt2spk'.format(cond_dir), 'w') for f in filelist: uttname = f.strip('.wav') wav_scp_file.write('{} {}/{}/{}\n'.format(uttname, wav_dir, cond, f)) utt2spk_file.write('{} {}\n'.format(uttname, uttname)) wav_scp_file.close() utt2spk_file.close() shutil.copyfile('{}/utt2spk'.format(cond_dir), '{}/spk2utt'.format(cond_dir)) return 0
def main(): output_dir = '{}/wav{}/{}'.format(args.tgt_dir, args.sample_rate, args.part) if os.path.exists(output_dir): raise ValueError('Warning: {} already exists, please check!') else: os.makedirs(output_dir) if ((args.part == 'train') or (args.part == 'dev')): dset = 'trainset_28spk_wav_16k' elif (args.part == 'test'): dset = 'testset_wav_16k' for cond in ['clean', 'noisy']: wav_dir = '{}/{}_{}'.format(args.src_dir, cond, dset) filelist = [f for f in os.listdir(wav_dir) if f.endswith('.wav')] filelist.sort() cond_dir = '{}/{}'.format(output_dir, cond) if (not os.path.exists(cond_dir)): os.makedirs(cond_dir) wav_scp_file = open('{}/wav.scp'.format(cond_dir), 'w') utt2spk_file = open('{}/utt2spk'.format(cond_dir), 'w') for f in filelist: uttname = f.strip('.wav') if (uttname.startswith('p226') or uttname.startswith('p287')): if (args.part == 'train'): continue elif (args.part == 'dev'): continue wav_scp_file.write('{} {}/{}\n'.format(uttname, wav_dir, f)) utt2spk_file.write('{} {}\n'.format(uttname, uttname)) wav_scp_file.close() utt2spk_file.close() shutil.copyfile('{}/utt2spk'.format(cond_dir), '{}/spk2utt'.format(cond_dir)) return 0
class SeparationDataset(Dataset): def __init__(self, data_dir, rate=16000, src=['noisy'], tgt=['clean'], n_fft=512, hop_length=320, win_length=512, window='hann', center=True): super(SeparationDataset, self).__init__() '\n Args:\n data_dir (str):\n prepared data directory\n\n rate (int):\n audio sample rate\n\n src and tgt (list(str)):\n the input and desired output.\n\n n_fft (int):\n length of the windowed signal after padding with zeros.\n\n hop_length (int):\n number of audio samples between adjacent STFT columns.\n\n win_length (int):\n length of window for each frame\n\n window (str):\n type of window function, only support Hann window now\n\n center (bool):\n whether to pad input on both sides so that the\n t-th frame is centered at time t * hop_length\n\n The STFT related parameters are the same as librosa.\n ' self.data_dir = data_dir self.rate = rate self.src = src self.tgt = tgt self.n_fft = n_fft self.hop_length = hop_length self.win_length = win_length self.window = window self.center = center self.n_srcs = len(self.tgt) assert ((len(self.src) == 1) and (len(self.tgt) == 1)) cond_list = ['noisy', 'clean'] reco2path = {} for cond in (src + tgt): assert (cond in cond_list) assert os.path.exists('{}/{}/wav.scp'.format(self.data_dir, cond)) with open('{}/{}/wav.scp'.format(self.data_dir, cond), 'r') as fh: content = fh.readlines() for line in content: line = line.strip('\n') (uttname, path) = line.split() if (uttname not in reco2path): reco2path[uttname] = {} reco2path[uttname][cond] = path self.reco2path = reco2path self.recolist = list(self.reco2path.keys()) self.recolist.sort() def __len__(self): return len(self.recolist) def __getitem__(self, i): reco = self.recolist[i] src_path = self.reco2path[reco][self.src[0]] (src_samp, rate) = librosa.load(src_path, sr=SAMPLE_RATE) assert (rate == self.rate) src_feat = np.transpose(librosa.stft(src_samp, n_fft=self.n_fft, hop_length=self.hop_length, win_length=self.win_length, window=self.window, center=self.center)) (tgt_samp_list, tgt_feat_list) = ([], []) for j in range(self.n_srcs): tgt_path = self.reco2path[reco][self.tgt[j]] (tgt_samp, rate) = librosa.load(tgt_path, sr=SAMPLE_RATE) assert (rate == self.rate) tgt_feat = np.transpose(librosa.stft(tgt_samp, n_fft=self.n_fft, hop_length=self.hop_length, win_length=self.win_length, window=self.window, center=self.center)) tgt_samp_list.append(tgt_samp) tgt_feat_list.append(tgt_feat) '\n reco (str):\n name of the utterance\n\n src_sample (ndarray):\n audio samples for the source [T, ]\n\n src_feat (ndarray):\n the STFT feature map for the source with shape [T1, D]\n\n tgt_samp_list (list(ndarray)):\n list of audio samples for the targets\n\n tgt_feat_list (list(ndarray)):\n list of STFT feature map for the targets\n ' return (reco, src_samp, src_feat, tgt_samp_list, tgt_feat_list) def collate_fn(self, batch): sorted_batch = sorted(batch, key=(lambda x: (- x[1].shape[0]))) bs = len(sorted_batch) uttname_list = [sorted_batch[i][0] for i in range(bs)] source_attr = {} mix_magnitude_list = [torch.from_numpy(np.abs(sorted_batch[i][2])) for i in range(bs)] mix_phase_list = [torch.from_numpy(np.angle(sorted_batch[i][2])) for i in range(bs)] mix_stft_list = [torch.from_numpy(sorted_batch[i][2]) for i in range(bs)] mix_magnitude = pad_sequence(mix_magnitude_list, batch_first=True) mix_phase = pad_sequence(mix_phase_list, batch_first=True) mix_stft = pad_sequence(mix_stft_list, batch_first=True) source_attr['magnitude'] = mix_magnitude source_attr['phase'] = mix_phase source_attr['stft'] = mix_stft target_attr = {} target_attr['magnitude'] = [] target_attr['phase'] = [] for j in range(self.n_srcs): tgt_magnitude_list = [torch.from_numpy(np.abs(sorted_batch[i][4][j])) for i in range(bs)] tgt_phase_list = [torch.from_numpy(np.angle(sorted_batch[i][4][j])) for i in range(bs)] tgt_magnitude = pad_sequence(tgt_magnitude_list, batch_first=True) tgt_phase = pad_sequence(tgt_phase_list, batch_first=True) target_attr['magnitude'].append(tgt_magnitude) target_attr['phase'].append(tgt_phase) wav_length = torch.from_numpy(np.array([len(sorted_batch[i][1]) for i in range(bs)])) source_wav_list = [torch.from_numpy(sorted_batch[i][1]) for i in range(bs)] source_wav = pad_sequence(source_wav_list, batch_first=True) target_wav_list = [] for j in range(self.n_srcs): target_wav_list.append(pad_sequence([torch.from_numpy(sorted_batch[i][3][j]) for i in range(bs)], batch_first=True)) feat_length = torch.from_numpy(np.array([stft.size(0) for stft in mix_stft_list])) '\n source_wav_list (list(tensor)):\n list of audio samples for the source\n\n uttname_list (list(str)):\n list of utterance names\n\n source_attr (dict):\n dictionary containing magnitude and phase information for the sources\n\n source_wav (tensor):\n padded version of source_wav_list, with size [bs, max_T]\n\n target_attr (dict):\n dictionary containing magnitude and phase information for the targets\n\n feat_length (tensor):\n length of the STFT feature for each utterance\n\n wav_length (tensor):\n number of samples in each utterance\n ' return (source_wav_list, uttname_list, source_attr, source_wav, target_attr, target_wav_list, feat_length, wav_length)
class SepRNN(torch.nn.Module): def __init__(self, input_dim, num_bins, rnn='lstm', num_spks=2, num_layers=3, hidden_size=896, dropout=0.0, non_linear='relu', bidirectional=True): super(SepRNN, self).__init__() if (non_linear not in ['relu', 'sigmoid', 'tanh', 'none']): raise ValueError('Unsupported non-linear type:{}'.format(non_linear)) self.num_spks = num_spks rnn = rnn.upper() if (rnn not in ['RNN', 'LSTM', 'GRU']): raise ValueError('Unsupported rnn type: {}'.format(rnn)) self.rnn = getattr(torch.nn, rnn)(input_dim, hidden_size, num_layers, batch_first=True, dropout=dropout, bidirectional=bidirectional) self.drops = torch.nn.Dropout(p=dropout) self.linear = torch.nn.ModuleList([torch.nn.Linear(((hidden_size * 2) if bidirectional else hidden_size), num_bins) for _ in range(self.num_spks)]) self.non_linear = {'relu': torch.nn.functional.relu, 'sigmoid': torch.sigmoid, 'tanh': torch.nn.functional.tanh, 'none': torch.nn.Identity()}[non_linear] self.num_bins = num_bins def forward(self, x, train=True): assert isinstance(x, PackedSequence) (x, _) = self.rnn(x) (x, len_x) = pad_packed_sequence(x, batch_first=True) x = self.drops(x) m = [] for linear in self.linear: y = linear(x) y = self.non_linear(y) if (not train): y = y.view((- 1), self.num_bins) m.append(y) return m
def main(): output_dir = '{}/wav{}/{}/{}'.format(args.tgt_dir, args.sample_rate, args.mode, args.part) if os.path.exists(output_dir): raise ValueError('Warning: {} already exists, please check!') else: os.makedirs(output_dir) wav_dir = '{}/wav{}/{}/{}'.format(args.src_dir, args.sample_rate, args.mode, args.part) assert os.path.exists(wav_dir) for cond in ['s1', 's2', 'mix_clean', 'mix_both', 'mix_single', 'noise']: filelist = [f for f in os.listdir('{}/{}'.format(wav_dir, cond)) if f.endswith('.wav')] filelist.sort() cond_dir = '{}/{}'.format(output_dir, cond) if (not os.path.exists(cond_dir)): os.makedirs(cond_dir) wav_scp_file = open('{}/wav.scp'.format(cond_dir), 'w') utt2spk_file = open('{}/utt2spk'.format(cond_dir), 'w') for f in filelist: uttname = f.strip('.wav') wav_scp_file.write('{} {}/{}/{}\n'.format(uttname, wav_dir, cond, f)) utt2spk_file.write('{} {}\n'.format(uttname, uttname)) wav_scp_file.close() utt2spk_file.close() shutil.copyfile('{}/utt2spk'.format(cond_dir), '{}/spk2utt'.format(cond_dir)) return 0
def main(): output_dir = '{}/wav{}/{}'.format(args.tgt_dir, args.sample_rate, args.part) if os.path.exists(output_dir): raise ValueError('Warning: {} already exists, please check!') else: os.makedirs(output_dir) if ((args.part == 'train') or (args.part == 'dev')): dset = 'trainset_28spk_wav_16k' elif (args.part == 'test'): dset = 'testset_wav_16k' for cond in ['clean', 'noisy']: wav_dir = '{}/{}_{}'.format(args.src_dir, cond, dset) filelist = [f for f in os.listdir(wav_dir) if f.endswith('.wav')] filelist.sort() cond_dir = '{}/{}'.format(output_dir, cond) if (not os.path.exists(cond_dir)): os.makedirs(cond_dir) wav_scp_file = open('{}/wav.scp'.format(cond_dir), 'w') utt2spk_file = open('{}/utt2spk'.format(cond_dir), 'w') for f in filelist: uttname = f.strip('.wav') if (uttname.startswith('p226') or uttname.startswith('p287')): if (args.part == 'train'): continue elif (args.part == 'dev'): continue wav_scp_file.write('{} {}/{}\n'.format(uttname, wav_dir, f)) utt2spk_file.write('{} {}\n'.format(uttname, uttname)) wav_scp_file.close() utt2spk_file.close() shutil.copyfile('{}/utt2spk'.format(cond_dir), '{}/spk2utt'.format(cond_dir)) return 0
class RandomDataset(Dataset): def __init__(self, **kwargs): self.class_num = 48 def __getitem__(self, idx): samples = random.randint((EXAMPLE_WAV_MIN_SEC * SAMPLE_RATE), (EXAMPLE_WAV_MAX_SEC * SAMPLE_RATE)) wav = torch.randn(samples) label = random.randint(0, (self.class_num - 1)) return (wav, label) def __len__(self): return EXAMPLE_DATASET_SIZE def collate_fn(self, samples): (wavs, labels) = ([], []) for (wav, label) in samples: wavs.append(wav) labels.append(label) return (wavs, labels)
class DownstreamExpert(nn.Module): '\n Used to handle downstream-specific operations\n eg. downstream forward, metric computation, contents to log\n ' def __init__(self, upstream_dim, upstream_rate, downstream_expert, expdir, **kwargs): "\n Args:\n upstream_dim: int\n Different upstream will give different representation dimension\n You might want to first project them to the same dimension\n\n upstream_rate: int\n 160: for upstream with 10 ms per frame\n 320: for upstream with 20 ms per frame\n \n downstream_expert: dict\n The 'downstream_expert' field specified in your downstream config file\n eg. downstream/example/config.yaml\n\n expdir: string\n The expdir from command-line argument, you should save all results into\n this directory, like some logging files.\n\n **kwargs: dict\n All the arguments specified by the argparser in run_downstream.py\n and all the other fields in config.yaml, in case you need it.\n \n Note1. Feel free to add new argument for __init__ as long as it is\n a command-line argument or a config field. You can check the constructor\n code in downstream/runner.py\n " super(DownstreamExpert, self).__init__() self.upstream_dim = upstream_dim self.datarc = downstream_expert['datarc'] self.modelrc = downstream_expert['modelrc'] self.train_dataset = RandomDataset(**self.datarc) self.dev_dataset = RandomDataset(**self.datarc) self.test_dataset = RandomDataset(**self.datarc) self.connector = nn.Linear(upstream_dim, self.modelrc['input_dim']) self.model = Model(output_class_num=self.train_dataset.class_num, **self.modelrc) self.objective = nn.CrossEntropyLoss() self.register_buffer('best_score', torch.zeros(1)) def get_dataloader(self, split, epoch: int=0): "\n Args:\n split: string\n 'train'\n will always be called before the training loop\n\n 'dev', 'test', or more\n defined by the 'eval_dataloaders' field in your downstream config\n these will be called before the evaluation loops during the training loop\n\n Return:\n a torch.utils.data.DataLoader returning each batch in the format of:\n\n [wav1, wav2, ...], your_other_contents1, your_other_contents2, ...\n\n where wav1, wav2 ... are in variable length\n each wav is torch.FloatTensor in cpu with:\n 1. dim() == 1\n 2. sample_rate == 16000\n 3. directly loaded by torchaudio\n " if (split == 'train'): return self._get_train_dataloader(self.train_dataset, epoch) elif (split == 'dev'): return self._get_eval_dataloader(self.dev_dataset) elif (split == 'test'): return self._get_eval_dataloader(self.test_dataset) def _get_train_dataloader(self, dataset, epoch: int): from s3prl.utility.data import get_ddp_sampler sampler = get_ddp_sampler(dataset, epoch) return DataLoader(dataset, batch_size=self.datarc['train_batch_size'], shuffle=(sampler is None), sampler=sampler, num_workers=self.datarc['num_workers'], collate_fn=dataset.collate_fn) def _get_eval_dataloader(self, dataset): return DataLoader(dataset, batch_size=self.datarc['eval_batch_size'], shuffle=False, num_workers=self.datarc['num_workers'], collate_fn=dataset.collate_fn) def forward(self, split, features, your_other_contents1, records, **kwargs): "\n Args:\n split: string\n 'train'\n when the forward is inside the training loop\n\n 'dev', 'test' or more\n when the forward is inside the evaluation loop\n\n features:\n list of unpadded features [feat1, feat2, ...]\n each feat is in torch.FloatTensor and already\n put in the device assigned by command-line args\n\n your_other_contents1, ... :\n in the order defined by your dataloader (dataset + collate_fn)\n these are all in cpu, and you can move them to the same device\n as features\n\n records:\n defaultdict(list), by appending contents into records,\n these contents can be averaged and logged on Tensorboard\n later by self.log_records (also customized by you)\n\n Note1. downstream/runner.py will call self.log_records\n 1. every `log_step` during training\n 2. once after evalute the whole dev/test dataloader\n\n Note2. `log_step` is defined in your downstream config\n eg. downstream/example/config.yaml\n\n Return:\n loss:\n the loss to be optimized, should not be detached\n a single scalar in torch.FloatTensor\n " features = pad_sequence(features, batch_first=True) features = self.connector(features) predicted = self.model(features) utterance_labels = your_other_contents1 labels = torch.LongTensor(utterance_labels).to(features.device) loss = self.objective(predicted, labels) predicted_classid = predicted.max(dim=(- 1)).indices records['loss'].append(loss.item()) records['acc'] += (predicted_classid == labels).view((- 1)).cpu().float().tolist() return loss def log_records(self, split, records, logger, global_step, batch_ids, total_batch_num, **kwargs): "\n Args:\n split: string\n 'train':\n records and batchids contain contents for `log_step` batches\n `log_step` is defined in your downstream config\n eg. downstream/example/config.yaml\n\n 'dev', 'test' or more:\n records and batchids contain contents for the entire evaluation dataset\n\n records:\n defaultdict(list), contents already prepared by self.forward\n\n logger:\n Tensorboard SummaryWriter\n please use f'{your_task_name}/{split}-{key}' as key name to log your contents,\n preventing conflict with the logging of other tasks\n\n global_step:\n The global_step when training, which is helpful for Tensorboard logging\n\n batch_ids:\n The batches contained in records when enumerating over the dataloader\n\n total_batch_num:\n The total amount of batches in the dataloader\n \n Return:\n a list of string\n Each string is a filename we wish to use to save the current model\n according to the evaluation result, like the best.ckpt on the dev set\n You can return nothing or an empty list when no need to save the checkpoint\n " save_names = [] for (key, values) in records.items(): average = torch.FloatTensor(values).mean().item() logger.add_scalar(f'example/{split}-{key}', average, global_step=global_step) if ((split == 'dev') and (key == 'acc') and (average > self.best_score)): self.best_score = (torch.ones(1) * average) save_names.append(f'{split}-best.ckpt') return save_names
class Model(nn.Module): def __init__(self, input_dim, output_class_num, **kwargs): super(Model, self).__init__() self.linear = nn.Linear(input_dim, output_class_num) def forward(self, features): pooled = features.mean(dim=1) predicted = self.linear(pooled) return predicted
class FluentCommandsDataset(Dataset): def __init__(self, df, base_path, Sy_intent): self.df = df self.base_path = base_path self.max_length = (SAMPLE_RATE * EXAMPLE_WAV_MAX_SEC) self.Sy_intent = Sy_intent def __len__(self): return len(self.df) def __getitem__(self, idx): wav_path = os.path.join(self.base_path, self.df.loc[idx].path) (wav, sr) = torchaudio.load(wav_path) wav = wav.squeeze(0) label = [] for slot in ['action', 'object', 'location']: value = self.df.loc[idx][slot] label.append(self.Sy_intent[slot][value]) return (wav.numpy(), np.array(label), Path(wav_path).stem) def collate_fn(self, samples): return zip(*samples)
def get_downstream_model(input_dim, output_dim, config): model_cls = eval(config['select']) model_conf = config.get(config['select'], {}) model = model_cls(input_dim, output_dim, **model_conf) return model
class FrameLevel(nn.Module): def __init__(self, input_dim, output_dim, hiddens=None, activation='ReLU', **kwargs): super().__init__() latest_dim = input_dim self.hiddens = [] if (hiddens is not None): for dim in hiddens: self.hiddens += [nn.Linear(latest_dim, dim), getattr(nn, activation)()] latest_dim = dim self.hiddens = nn.Sequential(*self.hiddens) self.linear = nn.Linear(latest_dim, output_dim) def forward(self, hidden_state, features_len=None): hidden_state = self.hiddens(hidden_state) logit = self.linear(hidden_state) return (logit, features_len)
class UtteranceLevel(nn.Module): def __init__(self, input_dim, output_dim, pooling='MeanPooling', activation='ReLU', pre_net=None, post_net={'select': 'FrameLevel'}, **kwargs): super().__init__() latest_dim = input_dim self.pre_net = (get_downstream_model(latest_dim, latest_dim, pre_net) if isinstance(pre_net, dict) else None) self.pooling = eval(pooling)(input_dim=latest_dim, activation=activation) self.post_net = get_downstream_model(latest_dim, output_dim, post_net) def forward(self, hidden_state, features_len=None): if (self.pre_net is not None): (hidden_state, features_len) = self.pre_net(hidden_state, features_len) (pooled, features_len) = self.pooling(hidden_state, features_len) (logit, features_len) = self.post_net(pooled, features_len) return (logit, features_len)
class MeanPooling(nn.Module): def __init__(self, **kwargs): super(MeanPooling, self).__init__() def forward(self, feature_BxTxH, features_len, **kwargs): ' \n Arguments\n feature_BxTxH - [BxTxH] Acoustic feature with shape \n features_len - [B] of feature length\n ' agg_vec_list = [] for i in range(len(feature_BxTxH)): agg_vec = torch.mean(feature_BxTxH[i][:features_len[i]], dim=0) agg_vec_list.append(agg_vec) return (torch.stack(agg_vec_list), torch.ones(len(feature_BxTxH)).long())
class AttentivePooling(nn.Module): ' Attentive Pooling module incoporate attention mask' def __init__(self, input_dim, activation, **kwargs): super(AttentivePooling, self).__init__() self.sap_layer = AttentivePoolingModule(input_dim, activation) def forward(self, feature_BxTxH, features_len): ' \n Arguments\n feature_BxTxH - [BxTxH] Acoustic feature with shape \n features_len - [B] of feature length\n ' device = feature_BxTxH.device len_masks = torch.lt(torch.arange(features_len.max()).unsqueeze(0).to(device), features_len.unsqueeze(1)) (sap_vec, _) = self.sap_layer(feature_BxTxH, len_masks) return (sap_vec, torch.ones(len(feature_BxTxH)).long())
class AttentivePoolingModule(nn.Module): '\n Implementation of Attentive Pooling \n ' def __init__(self, input_dim, activation='ReLU', **kwargs): super(AttentivePoolingModule, self).__init__() self.W_a = nn.Linear(input_dim, input_dim) self.W = nn.Linear(input_dim, 1) self.act_fn = getattr(nn, activation)() self.softmax = nn.functional.softmax def forward(self, batch_rep, att_mask): '\n input:\n batch_rep : size (B, T, H), B: batch size, T: sequence length, H: Hidden dimension\n \n attention_weight:\n att_w : size (B, T, 1)\n \n return:\n utter_rep: size (B, H)\n ' att_logits = self.W(self.act_fn(self.W_a(batch_rep))).squeeze((- 1)) att_logits = (att_mask + att_logits) att_w = self.softmax(att_logits, dim=(- 1)).unsqueeze((- 1)) utter_rep = torch.sum((batch_rep * att_w), dim=1) return (utter_rep, att_w)
class VCC18SegmentalDataset(Dataset): def __init__(self, dataframe, base_path, idtable='', valid=False): self.base_path = Path(base_path) self.dataframe = dataframe self.segments_durations = 1 if Path.is_file(idtable): self.idtable = torch.load(idtable) for (i, judge_i) in enumerate(self.dataframe['JUDGE']): self.dataframe['JUDGE'][i] = self.idtable[judge_i] elif (not valid): self.gen_idtable(idtable) def __len__(self): return len(self.dataframe) def __getitem__(self, idx): (wav_name, mean, mos, judge_id) = self.dataframe.loc[idx] wav_path = ((self.base_path / 'Converted_speech_of_submitted_systems') / wav_name) (wav, _) = apply_effects_file(str(wav_path), [['channels', '1'], ['rate', '16000'], ['norm']]) wav = wav.view((- 1)) wav_segments = unfold_segments(wav, self.segments_durations) system_name = (wav_name[:3] + wav_name[(- 8):(- 4)]) return (wav_segments, mean, system_name, mos, judge_id) def collate_fn(self, samples): (wavs_segments, means, system_names, moss, judge_ids) = zip(*samples) flattened_wavs_segments = [wav_segment for wav_segments in wavs_segments for wav_segment in wav_segments] wav_segments_lengths = [len(wav_segments) for wav_segments in wavs_segments] prefix_sums = list(accumulate(wav_segments_lengths, initial=0)) segment_judge_ids = [] for i in range((len(prefix_sums) - 1)): segment_judge_ids.extend(([judge_ids[i]] * (prefix_sums[(i + 1)] - prefix_sums[i]))) return (torch.stack(flattened_wavs_segments), prefix_sums, torch.FloatTensor(means), system_names, torch.FloatTensor(moss), torch.LongTensor(segment_judge_ids)) def gen_idtable(self, idtable_path): if (idtable_path == ''): idtable_path = './idtable.pkl' self.idtable = {} count = 0 for (i, judge_i) in enumerate(self.dataframe['JUDGE']): if (judge_i not in self.idtable.keys()): self.idtable[judge_i] = count count += 1 self.dataframe['JUDGE'][i] = self.idtable[judge_i] else: self.dataframe['JUDGE'][i] = self.idtable[judge_i] torch.save(self.idtable, idtable_path)
class VCC16SegmentalDataset(Dataset): def __init__(self, wav_list, base_path): self.wav_dir = Path(base_path) self.wav_list = wav_list self.segments_durations = 1 def __len__(self): return len(self.wav_list) def __getitem__(self, idx): wav_name = self.wav_list[idx] wav_path = (self.wav_dir / wav_name) (wav, _) = apply_effects_file(str(wav_path), [['channels', '1'], ['rate', '16000'], ['norm']]) wav = wav.view((- 1)) wav_segments = unfold_segments(wav, self.segments_durations) system_name = wav_name.name.split('_')[0] return (wav_segments, system_name) def collate_fn(self, samples): (wavs_segments, system_names) = zip(*samples) flattened_wavs_segments = [wav_segment for wav_segments in wavs_segments for wav_segment in wav_segments] wav_segments_lengths = [len(wav_segments) for wav_segments in wavs_segments] prefix_sums = list(accumulate(wav_segments_lengths, initial=0)) return (torch.stack(flattened_wavs_segments), prefix_sums, None, system_names, None, None)
def unfold_segments(tensor, tgt_duration, sample_rate=16000): seg_lengths = int((tgt_duration * sample_rate)) src_lengths = len(tensor) step = (seg_lengths // 2) tgt_lengths = (seg_lengths if (src_lengths <= seg_lengths) else (((src_lengths // step) + 1) * step)) pad_lengths = (tgt_lengths - src_lengths) padded_tensor = torch.cat([tensor, torch.zeros(pad_lengths)]) segments = padded_tensor.unfold(0, seg_lengths, step).unbind(0) return segments
class DownstreamExpert(nn.Module): def __init__(self, upstream_dim, downstream_expert, **kwargs): super(DownstreamExpert, self).__init__() self.upstream_dim = upstream_dim self.datarc = downstream_expert['datarc'] self.modelrc = downstream_expert['modelrc'] idtable = (Path(kwargs['expdir']) / 'idtable.pkl') self.train_dataset = VCC18SegmentalDataset(preprocess(self.datarc['vcc2018_file_path'], 'train_judge.csv'), self.datarc['vcc2018_file_path'], idtable=idtable) self.dev_dataset = VCC18SegmentalDataset(preprocess(self.datarc['vcc2018_file_path'], 'valid_judge.csv'), self.datarc['vcc2018_file_path'], idtable=idtable, valid=False) self.vcc2018_test_dataset = VCC18SegmentalDataset(preprocess(self.datarc['vcc2018_file_path'], 'test_judge.csv'), self.datarc['vcc2018_file_path'], idtable=idtable, valid=False) self.vcc2018_system_mos = pd.read_csv(Path(self.datarc['vcc2018_file_path'], 'VCC2018_Results/system_mos_all_trackwise.csv')) self.vcc2016_test_dataset = VCC16SegmentalDataset(list(Path.iterdir(Path(self.datarc['vcc2016_file_path'], 'unified_speech'))), Path(self.datarc['vcc2016_file_path'], 'unified_speech')) self.vcc2016_system_mos = pd.read_csv(Path(self.datarc['vcc2016_file_path'], 'system_mos.csv'), index_col=False) self.connector = nn.Linear(upstream_dim, self.modelrc['projector_dim']) self.model = Model(input_dim=self.modelrc['projector_dim'], clipping=(self.modelrc['clipping'] if ('clipping' in self.modelrc) else False), attention_pooling=(self.modelrc['attention_pooling'] if ('attention_pooling' in self.modelrc) else False), num_judges=5000) self.objective = nn.MSELoss() self.segment_weight = self.modelrc['segment_weight'] self.bias_weight = self.modelrc['bias_weight'] self.best_scores = {'dev_loss': np.inf, 'dev_LCC': (- np.inf), 'dev_SRCC': (- np.inf), 'vcc2016_test_LCC': (- np.inf), 'vcc2016_test_SRCC': (- np.inf)} def get_dataloader(self, mode): if (mode == 'train'): return self._get_train_dataloader(self.train_dataset) elif (mode == 'dev'): return self._get_eval_dataloader(self.dev_dataset) elif (mode == 'vcc2018_test'): return self._get_eval_dataloader(self.vcc2018_test_dataset) elif (mode == 'vcc2016_test'): return self._get_eval_dataloader(self.vcc2016_test_dataset) def _get_train_dataloader(self, dataset): sampler = (DistributedSampler(dataset) if is_initialized() else None) return DataLoader(dataset, batch_size=self.datarc['train_batch_size'], shuffle=(sampler is None), sampler=sampler, num_workers=self.datarc['num_workers'], collate_fn=dataset.collate_fn) def _get_eval_dataloader(self, dataset): return DataLoader(dataset, batch_size=self.datarc['eval_batch_size'], shuffle=False, num_workers=self.datarc['num_workers'], collate_fn=dataset.collate_fn) def forward(self, mode, features, prefix_sums, means, system_names, moses, judge_ids, records, **kwargs): features = torch.stack(features) features = self.connector(features) uttr_scores = [] bias_scores = [] if (mode == 'train'): means = means.to(features.device) judge_ids = judge_ids.to(features.device) moses = moses.to(features.device) (segments_scores, segments_bias_scores) = self.model(features, judge_ids=judge_ids) segments_loss = 0 uttr_loss = 0 bias_loss = 0 for i in range((len(prefix_sums) - 1)): current_segment_scores = segments_scores[prefix_sums[i]:prefix_sums[(i + 1)]] current_bias_scores = segments_bias_scores[prefix_sums[i]:prefix_sums[(i + 1)]] uttr_score = current_segment_scores.mean(dim=(- 1)) uttr_scores.append(uttr_score.detach().cpu()) bias_score = current_bias_scores.mean(dim=(- 1)) bias_scores.append(bias_score.detach().cpu()) segments_loss += self.objective(current_segment_scores, means[i]) uttr_loss += self.objective(uttr_score, means[i]) bias_loss += self.objective(bias_score, moses[i]) segments_loss /= (len(prefix_sums) - 1) uttr_loss /= (len(prefix_sums) - 1) bias_loss /= (len(prefix_sums) - 1) loss = (((self.segment_weight * segments_loss) + (self.bias_weight * bias_loss)) + uttr_loss) records['segment loss'].append(segments_loss.item()) records['utterance loss'].append(uttr_loss.item()) records['bias loss'].append(bias_loss.item()) records['total loss'].append(loss.item()) records['pred_scores'] += uttr_scores records['true_scores'] += means.detach().cpu().tolist() if ((mode == 'dev') or (mode == 'vcc2018_test')): means = means.to(features.device) segments_scores = self.model(features) segments_loss = 0 uttr_loss = 0 for i in range((len(prefix_sums) - 1)): current_segment_scores = segments_scores[prefix_sums[i]:prefix_sums[(i + 1)]] uttr_score = current_segment_scores.mean(dim=(- 1)) uttr_scores.append(uttr_score.detach().cpu()) segments_loss += self.objective(current_segment_scores, means[i]) uttr_loss += self.objective(uttr_score, means[i]) segments_loss /= (len(prefix_sums) - 1) uttr_loss /= (len(prefix_sums) - 1) loss = (segments_loss + uttr_loss) records['total loss'].append(loss.item()) records['pred_scores'] += uttr_scores records['true_scores'] += means.detach().cpu().tolist() if (mode == 'vcc2016_test'): segments_scores = self.model(features) for i in range((len(prefix_sums) - 1)): current_segment_scores = segments_scores[prefix_sums[i]:prefix_sums[(i + 1)]] uttr_score = current_segment_scores.mean(dim=(- 1)) uttr_scores.append(uttr_score.detach().cpu()) if (len(records['system']) == 0): records['system'].append(defaultdict(list)) for i in range(len(system_names)): records['system'][0][system_names[i]].append(uttr_scores[i].tolist()) if (mode == 'train'): return loss return 0 def log_records(self, mode, records, logger, global_step, batch_ids, total_batch_num, **kwargs): save_names = [] if (mode == 'train'): avg_uttr_loss = torch.FloatTensor(records['utterance loss']).mean().item() avg_frame_loss = torch.FloatTensor(records['segment loss']).mean().item() avg_bias_loss = torch.FloatTensor(records['bias loss']).mean().item() logger.add_scalar(f'wav2MOS_segment_MBNet/{mode}-utterance loss', avg_uttr_loss, global_step=global_step) logger.add_scalar(f'wav2MOS_segment_MBNet/{mode}-segment loss', avg_frame_loss, global_step=global_step) logger.add_scalar(f'wav2MOS_segment_MBNet/{mode}-bias loss', avg_bias_loss, global_step=global_step) if ((mode == 'train') or (mode == 'dev')): avg_total_loss = torch.FloatTensor(records['total loss']).mean().item() logger.add_scalar(f'wav2MOS_segment_MBNet/{mode}-total loss', avg_total_loss, global_step=global_step) if ((mode == 'dev') or (mode == 'vcc2018_test')): all_pred_scores = records['pred_scores'] all_true_scores = records['true_scores'] all_pred_scores = np.array(all_pred_scores) all_true_scores = np.array(all_true_scores) MSE = np.mean(((all_true_scores - all_pred_scores) ** 2)) logger.add_scalar(f'wav2MOS_segment_MBNet/{mode}-Utterance level MSE', MSE, global_step=global_step) (pearson_rho, _) = pearsonr(all_true_scores, all_pred_scores) logger.add_scalar(f'wav2MOS_segment_MBNet/{mode}-Utterance level LCC', pearson_rho, global_step=global_step) (spearman_rho, _) = spearmanr(all_true_scores.T, all_pred_scores.T) logger.add_scalar(f'wav2MOS_segment_MBNet/{mode}-Utterance level SRCC', spearman_rho, global_step=global_step) tqdm.write(f'[{mode}] Utterance-level MSE = {MSE:.4f}') tqdm.write(f'[{mode}] Utterance-level LCC = {pearson_rho:.4f}') tqdm.write(f'[{mode}] Utterance-level SRCC = {spearman_rho:.4f}') if ((mode == 'dev') or (mode == 'vcc2018_test')): system_level_mos = self.vcc2018_system_mos if (mode == 'vcc2016_test'): system_level_mos = self.vcc2016_system_mos if ((mode == 'dev') or (mode == 'vcc2018_test') or (mode == 'vcc2016_test')): all_system_pred_scores = [] all_system_true_scores = [] for (key, values) in records['system'][0].items(): all_system_pred_scores.append(np.mean(values)) all_system_true_scores.append(system_level_mos[key].iloc[0]) all_system_pred_scores = np.array(all_system_pred_scores) all_system_true_scores = np.array(all_system_true_scores) MSE = np.mean(((all_system_true_scores - all_system_pred_scores) ** 2)) (pearson_rho, _) = pearsonr(all_system_true_scores, all_system_pred_scores) (spearman_rho, _) = spearmanr(all_system_true_scores, all_system_pred_scores) tqdm.write(f'[{mode}] System-level MSE = {MSE:.4f}') tqdm.write(f'[{mode}] System-level LCC = {pearson_rho:.4f}') tqdm.write(f'[{mode}] System-level SRCC = {spearman_rho:.4f}') logger.add_scalar(f'wav2MOS_segment_MBNet/{mode}-System level MSE', MSE, global_step=global_step) logger.add_scalar(f'wav2MOS_segment_MBNet/{mode}-System level LCC', pearson_rho, global_step=global_step) logger.add_scalar(f'wav2MOS_segment_MBNet/{mode}-System level SRCC', spearman_rho, global_step=global_step) if (mode == 'dev'): if (avg_total_loss < self.best_scores['dev_loss']): self.best_scores[mode] = avg_total_loss save_names.append(f'{mode}-best.ckpt') if (pearson_rho > self.best_scores['dev_LCC']): self.best_scores['dev_LCC'] = pearson_rho save_names.append(f'{mode}-LCC-best.ckpt') if (spearman_rho > self.best_scores['dev_SRCC']): self.best_scores['dev_SRCC'] = spearman_rho save_names.append(f'{mode}-SRCC-best.ckpt') if (mode == 'vcc2016_test'): if (pearson_rho > self.best_scores['vcc2016_test_LCC']): self.best_scores['vcc2016_test_LCC'] = pearson_rho save_names.append(f'{mode}-LCC-best.ckpt') if (spearman_rho > self.best_scores['vcc2016_test_SRCC']): self.best_scores['vcc2016_test_SRCC'] = spearman_rho save_names.append(f'{mode}-SRCC-best.ckpt') return save_names
def preprocess(base_path, txt_file): dataframe = pd.read_csv(Path(base_path, txt_file), index_col=False) return dataframe
class SelfAttentionPooling(nn.Module): '\n Implementation of SelfAttentionPooling\n Original Paper: Self-Attention Encoding and Pooling for Speaker Recognition\n https://arxiv.org/pdf/2008.01077v1.pdf\n ' def __init__(self, input_dim): super(SelfAttentionPooling, self).__init__() self.W = nn.Linear(input_dim, 1) def forward(self, batch_rep): '\n input:\n batch_rep : size (N, T, H), N: batch size, T: sequence length, H: Hidden dimension\n\n attention_weight:\n att_w : size (N, T, 1)\n\n return:\n utter_rep: size (N, H)\n ' softmax = nn.functional.softmax att_w = softmax(self.W(batch_rep).squeeze((- 1))).unsqueeze((- 1)) utter_rep = torch.sum((batch_rep * att_w), dim=1) return utter_rep
class Model(nn.Module): def __init__(self, input_dim, clipping=False, attention_pooling=False, num_judges=5000, **kwargs): super(Model, self).__init__() self.mean_net_linear = nn.Linear(input_dim, 1) self.mean_net_clipping = clipping self.mean_net_pooling = (SelfAttentionPooling(input_dim) if attention_pooling else None) self.bias_net_linear = nn.Linear(input_dim, 1) self.bias_net_pooling = (SelfAttentionPooling(input_dim) if attention_pooling else None) self.judge_embbeding = nn.Embedding(num_embeddings=num_judges, embedding_dim=input_dim) def forward(self, features, judge_ids=None): if (self.mean_net_pooling is not None): x = self.mean_net_pooling(features) segment_score = self.mean_net_linear(x) else: x = self.mean_net_linear(features) segment_score = x.squeeze((- 1)).mean(dim=(- 1)) if self.mean_net_clipping: segment_score = ((torch.tanh(segment_score) * 2) + 3) if (judge_ids is None): return segment_score.squeeze((- 1)) else: time = features.shape[1] judge_features = self.judge_embbeding(judge_ids) judge_features = torch.stack([judge_features for i in range(time)], dim=1) bias_features = (features + judge_features) if (self.bias_net_pooling is not None): y = self.bias_net_pooling(bias_features) bias_score = self.bias_net_linear(y) else: y = self.bias_net_linear(bias_features) bias_score = y.squeeze((- 1)).mean(dim=(- 1)) bias_score = (bias_score + segment_score) return (segment_score.squeeze((- 1)), bias_score.squeeze((- 1)))
class MOSEIDataset(Dataset): def __init__(self, split, data, path): self.split = split self.data = data self.path = path def __getitem__(self, idx): wav_path = os.path.join(self.path, 'Segmented_Audio', self.split, self.data[idx][0]) (wav, sr) = torchaudio.load(wav_path) label = self.data[idx][1] return (wav.view((- 1)), torch.tensor(label).long()) def __len__(self): return len(self.data) def collate_fn(self, samples): (wavs, labels) = ([], []) for (wav, label) in samples: wavs.append(wav) labels.append(label) return (wavs, labels)
class DownstreamExpert(nn.Module): '\n Used to handle downstream-specific operations\n eg. downstream forward, metric computation, contents to log\n ' def __init__(self, upstream_dim, downstream_expert, expdir, **kwargs): "\n Args:\n upstream_dim: int\n Different upstream will give different representation dimension\n You might want to first project them to the same dimension\n\n downstream_expert: dict\n The 'downstream_expert' field specified in your downstream config file\n eg. downstream/downstream/example/config.yaml\n\n **kwargs: dict\n The arguments specified by the argparser in run_downstream.py\n in case you need it.\n " super(DownstreamExpert, self).__init__() self.upstream_dim = upstream_dim self.datarc = downstream_expert['datarc'] self.modelrc = downstream_expert['modelrc'] (self.train_data, self.dev_data, self.test_data) = ([], [], []) df = pd.read_csv(self.datarc['label_path'], encoding='latin-1') for row in df.itertuples(): filename = (((row.file + '_') + str(row.index)) + '.wav') if (self.datarc['num_class'] == 2): label = row.label2a elif (self.datarc['num_class'] == 3): label = (row.label2b + 1) elif (self.datarc['num_class'] == 6): label = row.label6 elif (self.datarc['num_class'] == 7): label = (row.label7 + 3) else: raise ValueError('Unsupported num_class') if (row.split == 0): self.train_data.append((filename, label)) elif (row.split == 1): self.dev_data.append((filename, label)) elif (row.split == 2): self.test_data.append((filename, label)) self.train_dataset = MOSEIDataset('train', self.train_data, self.datarc['data_dir']) self.dev_dataset = MOSEIDataset('dev', self.dev_data, self.datarc['data_dir']) self.test_dataset = MOSEIDataset('test', self.test_data, self.datarc['data_dir']) self.connector = nn.Linear(upstream_dim, self.modelrc['input_dim']) self.model = Model(output_class_num=self.datarc['num_class'], **self.modelrc) self.objective = nn.CrossEntropyLoss() self.expdir = expdir self.logging = os.path.join(self.expdir, 'log.log') self.best = defaultdict((lambda : 0)) self.answer = [] def _get_train_dataloader(self, dataset, epoch: int): from s3prl.utility.data import get_ddp_sampler sampler = get_ddp_sampler(dataset, epoch) return DataLoader(dataset, batch_size=self.datarc['train_batch_size'], shuffle=(sampler is None), sampler=sampler, num_workers=self.datarc['num_workers'], collate_fn=dataset.collate_fn) def _get_eval_dataloader(self, dataset): return DataLoader(dataset, batch_size=self.datarc['eval_batch_size'], shuffle=False, num_workers=self.datarc['num_workers'], collate_fn=dataset.collate_fn) '\n Datalaoder Specs:\n Each dataloader should output a list in the following format:\n\n [[wav1, wav2, ...], your_other_contents1, your_other_contents2, ...]\n\n where wav1, wav2 ... are in variable length\n each wav is torch.FloatTensor in cpu with:\n 1. dim() == 1\n 2. sample_rate == 16000\n 3. directly loaded by torchaudio without any preprocessing\n ' def get_train_dataloader(self, epoch: int): return self._get_train_dataloader(self.train_dataset, epoch) def get_dev_dataloader(self): return self._get_eval_dataloader(self.dev_dataset) def get_test_dataloader(self): return self._get_eval_dataloader(self.test_dataset) def get_dataloader(self, mode, epoch: int=0): if (mode == 'train'): return eval(f'self.get_{mode}_dataloader')(epoch) return eval(f'self.get_{mode}_dataloader')() def forward(self, mode, features, labels, records, **kwargs): "\n This function will be used in both train/dev/test, you can use\n self.training (bool) to control the different behavior for\n training or evaluation (dev/test)\n\n Args:\n mode: str\n 'train' or 'dev' or 'test'\n\n features:\n list of unpadded features [feat1, feat2, ...]\n each feat is in torch.FloatTensor and already\n put in the device assigned by command-line args\n\n records:\n defaultdict(list), by dumping contents into records,\n these contents can be averaged and logged on Tensorboard\n later by self.log_records\n\n Note1. downstream/runner.py will call self.log_records\n 1. every log_step during training\n 2. once after evalute the whole dev/test dataloader\n\n Note2. log_step is defined in your downstream config\n\n Return:\n loss:\n the loss to be optimized, should not be detached\n a single scalar in torch.FloatTensor\n " features = pad_sequence(features, batch_first=True) features = self.connector(features) predicted = self.model(features) utterance_labels = labels labels = torch.LongTensor(utterance_labels).to(features.device) loss = self.objective(predicted, labels) predicted_classid = predicted.max(dim=(- 1)).indices records['acc'] += (predicted_classid == labels).view((- 1)).cpu().float().tolist() records['predicted'] += predicted_classid.cpu().float().tolist() records['original'] += labels.cpu().float().tolist() if (not self.training): pass return loss def log_records(self, mode, records, logger, global_step, **kwargs): "\n This function will be used in both train/dev/test, you can use\n self.training (bool) to control the different behavior for\n training or evaluation (dev/test)\n\n Args:\n mode: str\n 'train' or 'dev' or 'test'\n\n records:\n defaultdict(list), contents already prepared by self.forward\n\n logger:\n Tensorboard SummaryWriter\n please use f'{prefix}your_content_name' as key name\n to log your customized contents\n\n global_step:\n global_step in runner, which is helpful for Tensorboard logging\n " prefix = f'mosei/{mode}-' average = torch.FloatTensor(records['acc']).mean().item() f1 = sklearn.metrics.f1_score(records['original'], records['predicted'], average='macro') logger.add_scalar(f'{prefix}acc', average, global_step=global_step) if (mode in ['dev', 'test']): print(f'{prefix}acc: {average}') message = f'''{mode} at step {global_step}: {average} (acc), {f1} (f1) ''' save_ckpt = [] if (average > self.best[prefix]): self.best[prefix] = average message = f'New best on {message}' name = prefix.split('/')[(- 1)].split('-')[0] save_ckpt.append(f'{name}-best.ckpt') if (mode in ['dev', 'test']): with open((Path(self.expdir) / f'{mode}_predict.txt'), 'w') as file: line = [f'''{f} ''' for f in records['predicted']] file.writelines(line) if (mode in ['dev', 'test']): with open((Path(self.expdir) / f'{mode}_truth.txt'), 'w') as file: line = [f'''{f} ''' for f in records['original']] file.writelines(line) with open(self.logging, 'a') as f: f.write(message) if (not self.training): pass return save_ckpt
class Model(nn.Module): def __init__(self, input_dim, output_class_num, **kwargs): super(Model, self).__init__() self.linear = nn.Linear(input_dim, output_class_num) def forward(self, features): pooled = features.mean(dim=1) predicted = self.linear(pooled) return predicted
def label2a(a): if (a < 0): return 0 return 1
def label2b(a): if (a < 0): return (- 1) if (a > 0): return 1 return 0
def label7(a): if (a < (- 2)): return (- 3) if (a < (- 1)): return (- 2) if (a < 0): return (- 1) if (a == 0): return 0 if (a <= 1): return 1 if (a <= 2): return 2 return 3
class DownstreamExpert(PhoneExpert): '\n Basically the same as the phone linear expert\n ' def __init__(self, upstream_dim, downstream_expert, **kwargs): super(DownstreamExpert, self).__init__(upstream_dim, downstream_expert, **kwargs) delattr(self, 'model') self.model = Model(input_dim=self.upstream_dim, output_class_num=self.train_dataset.class_num, **self.modelrc)
class Model(nn.Module): def __init__(self, input_dim, output_class_num, hidden_size, dropout, **kwargs): super(Model, self).__init__() self.in_linear = nn.Linear(input_dim, hidden_size) self.out_linear = nn.Linear(hidden_size, output_class_num) self.drop = nn.Dropout(dropout) self.act_fn = nn.functional.relu def forward(self, features): hidden = self.in_linear(features) hidden = self.drop(hidden) hidden = self.act_fn(hidden) predicted = self.out_linear(hidden) return predicted
class PhoneDataset(Dataset): def __init__(self, split, bucket_size, libri_root, phone_path, bucket_file, sample_rate=16000, train_dev_seed=1337, **kwargs): super(PhoneDataset, self).__init__() self.libri_root = libri_root self.phone_path = phone_path self.sample_rate = sample_rate self.class_num = 41 self.Y = {} phone_file = open(os.path.join(phone_path, 'converted_aligned_phones.txt')).readlines() for line in phone_file: line = line.strip('\n').split(' ') self.Y[line[0]] = [int(p) for p in line[1:]] if ((split == 'train') or (split == 'dev')): usage_list = open(os.path.join(phone_path, 'train_split.txt')).readlines() random.seed(train_dev_seed) random.shuffle(usage_list) percent = int((len(usage_list) * 0.9)) usage_list = (usage_list[:percent] if (split == 'train') else usage_list[percent:]) elif (split == 'test'): usage_list = open(os.path.join(phone_path, 'test_split.txt')).readlines() else: raise ValueError("Invalid 'split' argument for dataset: PhoneDataset!") usage_list = {line.strip('\n'): None for line in usage_list} print(((((('[Dataset] - # phone classes: ' + str(self.class_num)) + ', number of data for ') + split) + ': ') + str(len(usage_list)))) assert os.path.isdir(bucket_file), 'Please first run `preprocess/generate_len_for_bucket.py to get bucket file.' table = pd.read_csv(os.path.join(bucket_file, 'train-clean-100.csv')).sort_values(by=['length'], ascending=False) X = table['file_path'].tolist() X_lens = table['length'].tolist() self.X = [] (batch_x, batch_len) = ([], []) for (x, x_len) in zip(X, X_lens): if (self._parse_x_name(x) in usage_list): batch_x.append(x) batch_len.append(x_len) if (len(batch_x) == bucket_size): if ((bucket_size >= 2) and (max(batch_len) > HALF_BATCHSIZE_TIME)): self.X.append(batch_x[:(bucket_size // 2)]) self.X.append(batch_x[(bucket_size // 2):]) else: self.X.append(batch_x) (batch_x, batch_len) = ([], []) if (len(batch_x) > 1): if (self._parse_x_name(x) in usage_list): self.X.append(batch_x) def _parse_x_name(self, x): return x.split('/')[(- 1)].split('.')[0] def _load_wav(self, wav_path): (wav, sr) = torchaudio.load(os.path.join(self.libri_root, wav_path)) return wav.view((- 1)) def __len__(self): return len(self.X) def __getitem__(self, index): wav_batch = [self._load_wav(x_file) for x_file in self.X[index]] label_batch = [torch.LongTensor(self.Y[self._parse_x_name(x_file)]) for x_file in self.X[index]] return (wav_batch, label_batch) def collate_fn(self, items): return (items[0][0], items[0][1])
class Model(nn.Module): def __init__(self, input_dim, output_class_num, **kwargs): super(Model, self).__init__() self.linear = nn.Linear(input_dim, output_class_num) def forward(self, features): predicted = self.linear(features) return predicted
class DownstreamExpert(PhoneExpert): '\n Basically the same as the phone linear expert\n ' def __init__(self, upstream_dim, downstream_expert, **kwargs): super(DownstreamExpert, self).__init__(upstream_dim, downstream_expert, **kwargs) delattr(self, 'model') self.model = Model(input_dim=self.upstream_dim, output_class_num=self.train_dataset.class_num, **self.modelrc)
class QUESST14Dataset(Dataset): 'QUESST 2014 dataset (English-only).' def __init__(self, split, **kwargs): dataset_root = Path(kwargs['dataset_root']) doc_paths = english_audio_paths(dataset_root, 'language_key_utterances.lst') query_paths = english_audio_paths(dataset_root, f'language_key_{split}.lst') self.dataset_root = dataset_root self.n_queries = len(query_paths) self.n_docs = len(doc_paths) self.data = (query_paths + doc_paths) def __len__(self): return len(self.data) def __getitem__(self, idx): audio_path = self.data[idx] (wav, _) = apply_effects_file(str(audio_path), [['channels', '1'], ['rate', '16000'], ['gain', '-3.0']]) wav = wav.squeeze(0) return (wav.numpy(), audio_path.with_suffix('').name) def collate_fn(self, samples): 'Collate a mini-batch of data.' (wavs, audio_names) = zip(*samples) return (wavs, audio_names)
def english_audio_paths(dataset_root_path, lst_name): 'Extract English audio paths.' audio_paths = [] with open(((dataset_root_path / 'scoring') / lst_name)) as f: for line in f: (audio_path, lang) = tuple(line.strip().split()) if (lang != 'nnenglish'): continue audio_path = re.sub('^.*?\\/', '', audio_path) audio_paths.append((dataset_root_path / audio_path)) return audio_paths
class DownstreamExpert(nn.Module): '\n Used to handle downstream-specific operations\n eg. downstream forward, metric computation, contents to log\n ' def __init__(self, upstream_dim: int, downstream_expert: dict, expdir: str, **kwargs): super(DownstreamExpert, self).__init__() self.upstream_dim = upstream_dim self.datarc = downstream_expert['datarc'] self.modelrc = downstream_expert['modelrc'] self.expdir = Path(expdir) self.train_dataset = QUESST14Trainset('dev', **self.datarc) self.valid_dataset = QUESST14Trainset('eval', **self.datarc) self.test_dataset = None self.model = Model(input_dim=upstream_dim, **self.modelrc) def get_dataloader(self, mode): if (mode == 'train'): return DataLoader(self.train_dataset, sampler=WeightedRandomSampler(weights=self.train_dataset.sample_weights, num_samples=len(self.train_dataset.sample_weights), replacement=True), batch_size=self.datarc['batch_size'], drop_last=True, num_workers=self.datarc['num_workers'], collate_fn=self.train_dataset.collate_fn) if (mode == 'valid'): return DataLoader(self.valid_dataset, sampler=WeightedRandomSampler(weights=self.valid_dataset.sample_weights, num_samples=self.datarc['valid_size'], replacement=True), batch_size=self.datarc['batch_size'], drop_last=True, num_workers=self.datarc['num_workers'], collate_fn=self.valid_dataset.collate_fn) if (mode in ['dev', 'eval']): self.test_dataset = QUESST14Testset(mode, **self.datarc) return DataLoader(self.test_dataset, shuffle=False, batch_size=self.datarc['batch_size'], drop_last=False, num_workers=self.datarc['num_workers'], collate_fn=self.test_dataset.collate_fn) raise NotImplementedError def forward(self, mode, features, infos, records, **kwargs): if (mode in ['train', 'valid']): features = torch.stack(features) (prefix_sums, labels) = infos labels = torch.cat(labels).to(features.device) embs = self.model(features) query_embs = embs[:self.datarc['batch_size']] audio_embs = embs[self.datarc['batch_size']:] max_similarities = torch.empty(len(labels)).to(labels.device) for i in range(self.datarc['batch_size']): similarities = F.cosine_similarity(query_embs[i:(i + 1)], audio_embs[prefix_sums[i]:prefix_sums[(i + 1)]]) max_similarities[i] = similarities.max() pos_similarities = max_similarities[(labels > 0)] neg_similarities = max_similarities[(labels < 0)] pos_loss = (1 - pos_similarities).sum() neg_loss = neg_similarities.clamp(0).sum() loss = ((pos_loss + neg_loss) / self.datarc['batch_size']) records['loss'].append(loss.item()) records['similarity-positive'] += pos_similarities.tolist() records['similarity-negative'] += neg_similarities.tolist() return loss elif (mode in ['dev', 'eval']): audio_tensors = torch.stack(features) (prefix_sums, audio_names) = infos embs = self.model(audio_tensors) embs = embs.detach().cpu() for i in range(len(audio_names)): records['embs'].append(embs[prefix_sums[i]:prefix_sums[(i + 1)]]) records['audio_names'].append(audio_names[i]) else: raise NotImplementedError def log_records(self, mode, records, logger, global_step, **kwargs): 'Log training, validation information or test on a dataset.' if (mode in ['train', 'valid']): prefix = f'quesst14_embedding/{mode}' for (key, val) in records.items(): average = (sum(val) / len(val)) logger.add_scalar(f'{prefix}-{key}', average, global_step=global_step) elif (mode in ['dev', 'eval']): query_embs = records['embs'][:self.test_dataset.n_queries] doc_embs = records['embs'][self.test_dataset.n_queries:] query_names = records['audio_names'][:self.test_dataset.n_queries] doc_names = records['audio_names'][self.test_dataset.n_queries:] results = {} for (query_emb, query_name) in zip(tqdm(query_embs, desc='Query', ncols=0), query_names): query_emb = query_emb[0:1].cuda() scores = [] for (doc_emb, doc_name) in zip(tqdm(doc_embs, desc='Doc', ncols=0, leave=False), doc_names): with torch.no_grad(): doc_emb = doc_emb.cuda() similarities = F.cosine_similarity(query_emb, doc_emb) score = similarities.max().detach().cpu() scores.append(score) scores = torch.stack(scores) if (scores.std() < 0.1): scores = torch.zeros_like(scores) else: scores = (((scores - scores.mean()) / (scores.std() + 1e-06)) + 0.5) results[query_name] = list(zip(doc_names, scores.tolist())) score_thresh = 0.5 root = etree.Element('stdlist', termlist_filename='benchmark.stdlist.xml', indexing_time='1.00', language='english', index_size='1', system_id='benchmark') for (query_name, doc_scores) in results.items(): term_list = etree.SubElement(root, 'detected_termlist', termid=query_name, term_search_time='1.0', oov_term_count='1') for (doc_name, score) in doc_scores: etree.SubElement(term_list, 'term', file=doc_name, channel='1', tbeg='0.000', dur='0.00', score=f'{score:.4f}', decision=('YES' if (score > score_thresh) else 'NO')) etree.ElementTree(root).write(str((self.expdir / 'benchmark.stdlist.xml')), encoding='UTF-8', pretty_print=True) else: raise NotImplementedError
class Model(nn.Module): def __init__(self, input_dim, bottleneck_dim, hidden_dim, num_layers, **kwargs): super(Model, self).__init__() self.connector = nn.Linear(input_dim, bottleneck_dim) self.rnn = nn.LSTM(input_size=bottleneck_dim, hidden_size=hidden_dim, num_layers=num_layers, batch_first=True) self.attention_linear = nn.Linear(hidden_dim, 1) def forward(self, features): hiddens = F.relu(self.connector(features)) (lstm_outputs, _) = self.rnn(hiddens) hiddens = torch.tanh(lstm_outputs) attention_weights = F.softmax(self.attention_linear(hiddens), dim=1) embeds = torch.sum((hiddens * attention_weights), dim=1) return embeds
class QUESST14Testset(Dataset): 'QUESST 2014 testing dataset (English-only).' def __init__(self, split, **kwargs): assert (split in ['dev', 'eval']) dataset_root = Path(kwargs['quesst2014_root']) doc_paths = get_audio_paths(dataset_root, 'language_key_utterances.lst') query_paths = get_audio_paths(dataset_root, f'language_key_{split}.lst') self.dataset_root = dataset_root self.n_queries = len(query_paths) self.n_docs = len(doc_paths) self.data = (query_paths + doc_paths) def __len__(self): return len(self.data) def __getitem__(self, idx): audio_path = self.data[idx] (wav, _) = apply_effects_file(str(audio_path), [['channels', '1'], ['rate', '16000'], ['norm']]) wav = wav.squeeze(0) src_len = len(wav) tgt_len = (48000 if (src_len <= 48000) else (((src_len // 12000) + 1) * 12000)) wav = torch.cat([wav, torch.zeros((tgt_len - src_len))]) segments = wav.unfold(0, 48000, 12000).unbind(0) return (segments, audio_path.with_suffix('').name) def collate_fn(self, samples): 'Collate a mini-batch of data.' (segments, audio_names) = zip(*samples) lengths = [len(segs) for segs in segments] prefix_sums = list(accumulate(lengths, initial=0)) flattened = [seg for segs in segments for seg in segs] return (flattened, (prefix_sums, audio_names))
def get_audio_paths(dataset_root_path, lst_name): 'Extract audio paths.' audio_paths = [] with open(((dataset_root_path / 'scoring') / lst_name)) as f: for line in f: (audio_path, lang) = tuple(line.strip().split()) if (lang != 'nnenglish'): continue audio_path = re.sub('^.*?\\/', '', audio_path) audio_paths.append((dataset_root_path / audio_path)) return audio_paths
class QUESST14Trainset(Dataset): 'QUESST 2014 training dataset.' def __init__(self, split, **kwargs): dataset_root = Path(kwargs['quesst2014_root']) scoring_root = (dataset_root / 'scoring') split_root = (scoring_root / f'groundtruth_quesst14_{split}') query2positives = parse_rttm((split_root / f'quesst14_{split}.rttm')) audio_names = parse_lst((scoring_root / 'language_key_utterances.lst')) query_names = parse_lst((scoring_root / f'language_key_{split}.lst')) print(f'[QUESST2014] # of audios: {len(audio_names)}') print(f'[QUESST2014] # of queries: {len(query_names)}') audio_set = set(audio_names) query2negatives = {query_name: list((audio_set - set((query2positives[query_name] if (query_name in query2positives) else [])))) for query_name in query_names} positive_pairs = [(query_name, audio_name) for query_name in query_names for audio_name in (set(query2positives[query_name]) & audio_set)] negative_pairs = [(query_name, list(negative_audio_set)) for (query_name, negative_audio_set) in query2negatives.items()] print(f'[QUESST2014] # of positive pairs: {len(positive_pairs)}') print(f'[QUESST2014] # of negative pairs: {len(negative_pairs)}') self.audio_root = (dataset_root / 'Audio') self.query_root = (dataset_root / f'{split}_queries') self.max_dur = 3.0 self.positive_pairs = positive_pairs self.negative_pairs = negative_pairs def __len__(self): return (len(self.positive_pairs) + len(self.negative_pairs)) def __getitem__(self, idx): if (idx < len(self.positive_pairs)): (query_name, audio_name) = self.positive_pairs[idx] else: (query_name, audio_names) = self.negative_pairs[(idx - len(self.positive_pairs))] audio_name = random.sample(audio_names, 1)[0] query_path = (self.query_root / query_name).with_suffix('.wav') audio_path = (self.audio_root / audio_name).with_suffix('.wav') query_tensor = path2tensor(query_path) audio_tensor = path2tensor(audio_path) query_segment = crop_segment(query_tensor, self.max_dur) audio_segments = unfold_segments(audio_tensor, self.max_dur) label = torch.LongTensor([(1 if (idx < len(self.positive_pairs)) else (- 1))]) return (query_segment, audio_segments, label) def collate_fn(self, samples): 'Collate a mini-batch of data.' (query_segments, segments_list, labels) = zip(*samples) flattened = [segment for segments in segments_list for segment in segments] lengths = [len(segments) for segments in segments_list] prefix_sums = list(accumulate(lengths, initial=0)) return ((list(query_segments) + flattened), (prefix_sums, labels)) @property def sample_weights(self): 'Sample weights to balance positive and negative data.' n_pos = len(self.positive_pairs) n_neg = len(self.negative_pairs) return (([(1 / n_pos)] * n_pos) + ([(1 / n_neg)] * n_neg))
def parse_rttm(rttm_path): 'Parse audio and query pairs from *.rttm.' pattern = re.compile('LEXEME\\s+(quesst14_[0-9]+).*?(quesst14_(dev|eval)_[0-9]+)') query2audios = defaultdict(list) with open(rttm_path) as fd: for line in fd: match = pattern.match(line) if (match is None): continue query2audios[match.group(2)].append(match.group(1)) return query2audios
def parse_lst(lst_path): 'Extract audio names of nnenglish.' audio_names = [] with open(lst_path) as fd: for line in fd: (audio_path, lang) = tuple(line.strip().split()) if (lang != 'nnenglish'): continue audio_name = Path(audio_path).with_suffix('').name audio_names.append(audio_name) return audio_names
def path2tensor(filepath): (tensor, _) = apply_effects_file(str(filepath), [['channels', '1'], ['rate', '16000'], ['norm']]) return tensor.squeeze(0)
def crop_segment(tensor, tgt_dur, sample_rate=16000): src_dur = (len(tensor) / sample_rate) random_shift = random.uniform(0, (src_dur - tgt_dur)) (audio_tensor, _) = apply_effects_tensor(tensor.unsqueeze(0), sample_rate, [['pad', f'{tgt_dur}', f'{tgt_dur}'], ['trim', f'{(tgt_dur + random_shift)}', f'{tgt_dur}']]) return audio_tensor.squeeze(0)
def unfold_segments(tensor, tgt_dur, sample_rate=16000): seg_len = int((tgt_dur * sample_rate)) src_len = len(tensor) hop_len = (seg_len // 4) tgt_len = (seg_len if (src_len <= seg_len) else (((src_len // hop_len) + 1) * hop_len)) pad_len = (tgt_len - src_len) front_pad_len = random.randint(0, pad_len) tail_pad_len = (pad_len - front_pad_len) padded_tensor = torch.cat([torch.zeros(front_pad_len), tensor, torch.zeros(tail_pad_len)]) segments = padded_tensor.unfold(0, seg_len, hop_len).unbind(0) return segments
class ModelEntry(): def __init__(self, model, name, trainable, interfaces): self.model = model self.name = name self.trainable = trainable self.interfaces = interfaces
class Runner(): '\n Used to handle high-level concepts of a ML experiment\n eg. training loop, evaluation loop, upstream propagation, optimization, logging, checkpoint saving\n ' def __init__(self, args, config): self.args = args self.config = config self.init_ckpt = (torch.load(self.args.init_ckpt, map_location='cpu') if self.args.init_ckpt else {}) self.upstream = self._get_upstream() self.featurizer = self._get_featurizer() self.downstream = self._get_downstream() self.all_entries = [self.upstream, self.featurizer, self.downstream] def _load_weight(self, model, name): init_weight = self.init_ckpt.get(name) if init_weight: show(f'[Runner] - Loading {name} weights from the previous experiment') model.load_state_dict(init_weight) def _init_model(self, model, name, trainable, interfaces=None): for interface in (interfaces or []): assert hasattr(model, interface), interface self._load_weight(model, name) if (is_initialized() and trainable and any((p.requires_grad for p in model.parameters()))): model = DDP(model, device_ids=[self.args.local_rank], find_unused_parameters=True) for interface in (interfaces or []): setattr(model, interface, getattr(model.module, interface)) return ModelEntry(model, name, trainable, interfaces) def _get_upstream(self): if (('from_hf_hub' in self.args) and (self.args.from_hf_hub == True)): from huggingface_hub import snapshot_download print(f'[Runner] - Downloading upstream model {self.args.upstream} from the Hugging Face Hub') filepath = snapshot_download(self.args.upstream, self.args.upstream_revision, use_auth_token=True) sys.path.append(filepath) dependencies = (Path(filepath) / 'requirements.txt').resolve() print('[Dependency] - The downloaded upstream model requires the following dependencies. Please make sure they are installed:') for (idx, line) in enumerate((Path(filepath) / 'requirements.txt').open().readlines()): print(f'{idx}. {line.strip()}') print(f'You can install them by:') print() print(f'pip install -r {dependencies}') print() from expert import UpstreamExpert Upstream = UpstreamExpert ckpt_path = os.path.join(filepath, self.args.upstream_model_name) else: Upstream = getattr(hub, self.args.upstream) ckpt_path = self.args.upstream_ckpt upstream_refresh = self.args.upstream_refresh if (is_initialized() and (get_rank() > 0)): torch.distributed.barrier() upstream_refresh = False model = Upstream(ckpt=ckpt_path, model_config=self.args.upstream_model_config, refresh=upstream_refresh).to(self.args.device) if (is_initialized() and (get_rank() == 0)): torch.distributed.barrier() return self._init_model(model=model, name='Upstream', trainable=self.args.upstream_trainable, interfaces=['get_downsample_rates']) def _get_featurizer(self): model = Featurizer(upstream=self.upstream.model, feature_selection=self.args.upstream_feature_selection, layer_selection=self.args.upstream_layer_selection, upstream_device=self.args.device, normalize=self.args.upstream_feature_normalize).to(self.args.device) return self._init_model(model=model, name='Featurizer', trainable=True, interfaces=['output_dim', 'downsample_rate']) def _get_downstream(self): expert = importlib.import_module(f's3prl.downstream.{self.args.downstream}.expert') Downstream = getattr(expert, 'DownstreamExpert') model = Downstream(upstream_dim=self.featurizer.model.output_dim, upstream_rate=self.featurizer.model.downsample_rate, **self.config, **vars(self.args)).to(self.args.device) return self._init_model(model=model, name='Downstream', trainable=True, interfaces=['get_dataloader', 'log_records']) def _get_optimizer(self, model_params): optimizer = get_optimizer(model_params, self.config['runner']['total_steps'], self.config['optimizer']) self._load_weight(optimizer, 'Optimizer') return optimizer def _get_scheduler(self, optimizer): scheduler = get_scheduler(optimizer, self.config['runner']['total_steps'], self.config['scheduler']) self._load_weight(scheduler, 'Scheduler') return scheduler def _create_model_card(self, path): model_card = MODEL_CARD_MARKDOWN.format(upstream_model=self.args.upstream) with open(os.path.join(path, 'README.md'), 'w') as f: f.write(model_card) def train(self): trainable_models = [] trainable_paras = [] for entry in self.all_entries: if entry.trainable: entry.model.train() trainable_models.append(entry.model) trainable_paras += list(entry.model.parameters()) else: entry.model.eval() amp = self.config['runner'].get('fp16', False) if amp: print('[Runner] - Enabled fp16 training') scaler = torch.cuda.amp.GradScaler() optimizer = self._get_optimizer(trainable_models) scheduler = None if self.config.get('scheduler'): scheduler = self._get_scheduler(optimizer) specaug = None if self.config.get('specaug'): from .specaug import SpecAug specaug = SpecAug(**self.config['specaug']) tqdm_file = (sys.stderr if is_leader_process() else open(os.devnull, 'w')) pbar = tqdm(total=self.config['runner']['total_steps'], dynamic_ncols=True, desc='overall', file=tqdm_file) init_step = self.init_ckpt.get('Step') if init_step: pbar.n = init_step if is_leader_process(): logger = SummaryWriter(self.args.expdir) batch_ids = [] backward_steps = 0 records = defaultdict(list) epoch = self.init_ckpt.get('Epoch', 0) train_split = self.config['runner'].get('train_dataloader', 'train') while (pbar.n < pbar.total): try: dataloader = self.downstream.model.get_dataloader(train_split, epoch=epoch) except TypeError as e: if ("unexpected keyword argument 'epoch'" in str(e)): dataloader = self.downstream.model.get_dataloader(train_split) if (hasattr(dataloader, 'sampler') and isinstance(dataloader.sampler, DistributedSampler)): dataloader.sampler.set_epoch(epoch) else: raise for (batch_id, (wavs, *others)) in enumerate(tqdm(dataloader, dynamic_ncols=True, desc='train', file=tqdm_file)): try: if (pbar.n >= pbar.total): break global_step = (pbar.n + 1) wavs = [torch.FloatTensor(wav).to(self.args.device) for wav in wavs] with torch.cuda.amp.autocast(enabled=amp): if self.upstream.trainable: features = self.upstream.model(wavs) else: with torch.no_grad(): features = self.upstream.model(wavs) features = self.featurizer.model(wavs, features) if specaug: (features, _) = specaug(features) loss = self.downstream.model(train_split, features, *others, records=records) batch_ids.append(batch_id) gradient_accumulate_steps = self.config['runner'].get('gradient_accumulate_steps') loss = (loss / gradient_accumulate_steps) if amp: scaler.scale(loss).backward() else: loss.backward() del loss except RuntimeError as e: if ('CUDA out of memory' in str(e)): print(f'[Runner] - CUDA out of memory at step {global_step}') if is_initialized(): raise with torch.cuda.device(self.args.device): torch.cuda.empty_cache() optimizer.zero_grad() continue else: raise backward_steps += 1 if ((backward_steps % gradient_accumulate_steps) > 0): continue if amp: scaler.unscale_(optimizer) grad_norm = torch.nn.utils.clip_grad_norm_(trainable_paras, self.config['runner']['gradient_clipping']) if amp: scaler.step(optimizer) scaler.update() elif math.isnan(grad_norm): print(f'[Runner] - grad norm is NaN at step {global_step}') else: optimizer.step() optimizer.zero_grad() if scheduler: scheduler.step() if (not is_leader_process()): batch_ids = [] records = defaultdict(list) continue if ((global_step % self.config['runner']['log_step']) == 0): self.downstream.model.log_records(train_split, records=records, logger=logger, global_step=global_step, batch_ids=batch_ids, total_batch_num=len(dataloader)) batch_ids = [] records = defaultdict(list) save_names = [] if ((global_step % self.config['runner']['eval_step']) == 0): for split in self.config['runner']['eval_dataloaders']: save_names += self.evaluate(split, logger, global_step) if ((global_step % self.config['runner']['save_step']) == 0): def check_ckpt_num(directory): max_keep = self.config['runner']['max_keep'] ckpt_pths = glob.glob(f'{directory}/states-*.ckpt') if (len(ckpt_pths) >= max_keep): ckpt_pths = sorted(ckpt_pths, key=(lambda pth: int(pth.split('-')[(- 1)].split('.')[0]))) for ckpt_pth in ckpt_pths[:((len(ckpt_pths) - max_keep) + 1)]: os.remove(ckpt_pth) check_ckpt_num(self.args.expdir) save_names.append(f'states-{global_step}.ckpt') if (len(save_names) > 0): all_states = {'Optimizer': optimizer.state_dict(), 'Step': global_step, 'Epoch': epoch, 'Args': self.args, 'Config': self.config} for entry in self.all_entries: if entry.trainable: all_states[entry.name] = get_model_state(entry.model) if scheduler: all_states['Scheduler'] = scheduler.state_dict() if is_initialized(): all_states['WorldSize'] = get_world_size() save_paths = [os.path.join(self.args.expdir, name) for name in save_names] tqdm.write(f'[Runner] - Save the checkpoint to:') for (i, path) in enumerate(save_paths): tqdm.write(f'{(i + 1)}. {path}') torch.save(all_states, path) pbar.update(1) epoch += 1 pbar.close() if self.args.push_to_hf_hub: self.push_to_huggingface_hub() if is_leader_process(): logger.close() def evaluate(self, split=None, logger=None, global_step=0): 'evaluate function will always be called on a single process even during distributed training' not_during_training = ((split is None) and (logger is None) and (global_step == 0)) if not_during_training: split = self.args.evaluate_split tempdir = tempfile.mkdtemp() logger = SummaryWriter(tempdir) random.seed(self.args.seed) np.random.seed(self.args.seed) torch.manual_seed(self.args.seed) if torch.cuda.is_available(): torch.cuda.manual_seed_all(self.args.seed) with torch.cuda.device(self.args.device): torch.cuda.empty_cache() trainings = [] for entry in self.all_entries: trainings.append(entry.model.training) entry.model.eval() dataloader = self.downstream.model.get_dataloader(split) evaluate_ratio = float(self.config['runner'].get('evaluate_ratio', 1)) evaluate_steps = round((len(dataloader) * evaluate_ratio)) batch_ids = [] records = defaultdict(list) for (batch_id, (wavs, *others)) in enumerate(tqdm(dataloader, dynamic_ncols=True, desc=split, total=evaluate_steps)): if (batch_id > evaluate_steps): break wavs = [torch.FloatTensor(wav).to(self.args.device) for wav in wavs] with torch.no_grad(): features = self.upstream.model(wavs) features = self.featurizer.model(wavs, features) self.downstream.model(split, features, *others, records=records, batch_id=batch_id) batch_ids.append(batch_id) save_names = self.downstream.model.log_records(split, records=records, logger=logger, global_step=global_step, batch_ids=batch_ids, total_batch_num=len(dataloader)) batch_ids = [] records = defaultdict(list) if torch.cuda.is_available(): with torch.cuda.device(self.args.device): torch.cuda.empty_cache() for (entry, training) in zip(self.all_entries, trainings): if training: entry.model.train() if not_during_training: logger.close() shutil.rmtree(tempdir) return ([] if (type(save_names) is not list) else save_names) def inference(self): filepath = Path(self.args.evaluate_split) assert filepath.is_file(), filepath filename = filepath.stem if hasattr(self.downstream.model, 'load_audio'): wav = self.downstream.model.load_audio(filepath) else: (wav, sr) = torchaudio.load(str(filepath)) assert (sr == SAMPLE_RATE), sr wavs = [wav.view((- 1)).to(self.args.device)] for entry in self.all_entries: entry.model.eval() with torch.no_grad(): features = self.upstream.model(wavs) features = self.featurizer.model(wavs, features) self.downstream.model.inference(features, [filename]) def push_to_huggingface_hub(self): 'Creates a downstream repository on the Hub and pushes training artifacts to it.' if (self.args.hf_hub_org.lower() != 'none'): organization = self.args.hf_hub_org else: organization = os.environ.get('HF_USERNAME') huggingface_token = HfFolder.get_token() print(f'[Runner] - Organisation to push fine-tuned model to: {organization}') if (self.args.hub == 'huggingface'): model_info = HfApi().model_info(self.args.upstream, token=huggingface_token) downstream_model_id = model_info.sha upstream_model_id = model_info.modelId.replace('/', '__') else: upstream_model_id = self.args.upstream.replace('/', '__') downstream_model_id = str(uuid.uuid4())[:8] repo_name = f'{upstream_model_id}__{downstream_model_id}' repo_url = HfApi().create_repo(token=huggingface_token, name=repo_name, organization=organization, exist_ok=True, private=False) print(f'[Runner] - Created Hub repo: {repo_url}') HF_HUB_DIR = 'hf_hub' REPO_ROOT_DIR = os.path.join(self.args.expdir, HF_HUB_DIR, repo_name) REPO_TASK_DIR = os.path.join(REPO_ROOT_DIR, self.args.downstream, self.args.expname) print(f'[Runner] - Cloning Hub repo to {REPO_ROOT_DIR}') model_repo = Repository(local_dir=REPO_ROOT_DIR, clone_from=repo_url, use_auth_token=huggingface_token) model_repo.git_pull() shutil.copytree(self.args.expdir, REPO_TASK_DIR, dirs_exist_ok=True, ignore=shutil.ignore_patterns(HF_HUB_DIR)) checkpoints = list(Path(REPO_TASK_DIR).glob('*best*.ckpt')) if (len(checkpoints) == 0): print('[Runner] - Did not find a best checkpoint! Using the final checkpoint instead ...') CKPT_PATH = os.path.join(REPO_TASK_DIR, f"states-{self.config['runner']['total_steps']}.ckpt") elif (len(checkpoints) > 1): print(f'[Runner] - More than one best checkpoint found! Using {checkpoints[0]} as default ...') CKPT_PATH = checkpoints[0] else: print(f'[Runner] - Found best checkpoint {checkpoints[0]}!') CKPT_PATH = checkpoints[0] shutil.move(CKPT_PATH, os.path.join(REPO_TASK_DIR, 'model.ckpt')) model_repo.lfs_track('*.ckpt') self._create_model_card(REPO_ROOT_DIR) print('[Runner] - Pushing model files to the Hub ...') model_repo.push_to_hub() print('[Runner] - Training run complete!')
class SeparationDataset(Dataset): def __init__(self, data_dir, rate=16000, src=['mix_clean'], tgt=['s1', 's2'], n_fft=512, hop_length=320, win_length=512, window='hann', center=True): super(SeparationDataset, self).__init__() "\n Args:\n data_dir (str):\n prepared data directory\n\n rate (int):\n audio sample rate\n\n src and tgt (list(str)):\n the input and desired output.\n LibriMix offeres different options for the users. For\n clean source separation, src=['mix_clean'] and tgt=['s1', 's2'].\n Please see https://github.com/JorisCos/LibriMix for details\n\n n_fft (int):\n length of the windowed signal after padding with zeros.\n\n hop_length (int):\n number of audio samples between adjacent STFT columns.\n\n win_length (int):\n length of window for each frame\n\n window (str):\n type of window function, only support Hann window now\n\n center (bool):\n whether to pad input on both sides so that the\n t-th frame is centered at time t * hop_length\n\n The STFT related parameters are the same as librosa.\n " self.data_dir = data_dir self.rate = rate self.src = src self.tgt = tgt self.n_fft = n_fft self.hop_length = hop_length self.win_length = win_length self.window = window self.center = center self.n_srcs = len(self.tgt) assert ((len(self.src) == 1) and (len(self.tgt) == 2)) cond_list = ['s1', 's2', 'noise', 'mix_clean', 'mix_both', 'mix_single'] reco2path = {} for cond in (src + tgt): assert (cond in cond_list) assert os.path.exists('{}/{}/wav.scp'.format(self.data_dir, cond)) with open('{}/{}/wav.scp'.format(self.data_dir, cond), 'r') as fh: content = fh.readlines() for line in content: line = line.strip('\n') (uttname, path) = line.split() if (uttname not in reco2path): reco2path[uttname] = {} reco2path[uttname][cond] = path self.reco2path = reco2path self.recolist = list(self.reco2path.keys()) self.recolist.sort() def __len__(self): return len(self.recolist) def __getitem__(self, i): reco = self.recolist[i] src_path = self.reco2path[reco][self.src[0]] (src_samp, rate) = librosa.load(src_path, sr=None) assert (rate == self.rate) src_feat = np.transpose(librosa.stft(src_samp, n_fft=self.n_fft, hop_length=self.hop_length, win_length=self.win_length, window=self.window, center=self.center)) (tgt_samp_list, tgt_feat_list) = ([], []) for j in range(self.n_srcs): tgt_path = self.reco2path[reco][self.tgt[j]] (tgt_samp, rate) = librosa.load(tgt_path, sr=None) assert (rate == self.rate) tgt_feat = np.transpose(librosa.stft(tgt_samp, n_fft=self.n_fft, hop_length=self.hop_length, win_length=self.win_length, window=self.window, center=self.center)) tgt_samp_list.append(tgt_samp) tgt_feat_list.append(tgt_feat) '\n reco (str):\n name of the utterance\n\n src_sample (ndarray):\n audio samples for the source [T, ]\n\n src_feat (ndarray):\n the STFT feature map for the source with shape [T1, D]\n\n tgt_samp_list (list(ndarray)):\n list of audio samples for the targets\n\n tgt_feat_list (list(ndarray)):\n list of STFT feature map for the targets\n ' return (reco, src_samp, src_feat, tgt_samp_list, tgt_feat_list) def collate_fn(self, batch): sorted_batch = sorted(batch, key=(lambda x: (- x[1].shape[0]))) bs = len(sorted_batch) uttname_list = [sorted_batch[i][0] for i in range(bs)] source_attr = {} mix_magnitude_list = [torch.from_numpy(np.abs(sorted_batch[i][2])) for i in range(bs)] mix_phase_list = [torch.from_numpy(np.angle(sorted_batch[i][2])) for i in range(bs)] mix_stft_list = [torch.from_numpy(sorted_batch[i][2]) for i in range(bs)] mix_magnitude = pad_sequence(mix_magnitude_list, batch_first=True) mix_phase = pad_sequence(mix_phase_list, batch_first=True) mix_stft = pad_sequence(mix_stft_list, batch_first=True) source_attr['magnitude'] = mix_magnitude source_attr['phase'] = mix_phase source_attr['stft'] = mix_stft target_attr = {} target_attr['magnitude'] = [] target_attr['phase'] = [] for j in range(self.n_srcs): tgt_magnitude_list = [torch.from_numpy(np.abs(sorted_batch[i][4][j])) for i in range(bs)] tgt_phase_list = [torch.from_numpy(np.angle(sorted_batch[i][4][j])) for i in range(bs)] tgt_magnitude = pad_sequence(tgt_magnitude_list, batch_first=True) tgt_phase = pad_sequence(tgt_phase_list, batch_first=True) target_attr['magnitude'].append(tgt_magnitude) target_attr['phase'].append(tgt_phase) wav_length = torch.from_numpy(np.array([len(sorted_batch[i][1]) for i in range(bs)])) source_wav_list = [torch.from_numpy(sorted_batch[i][1]) for i in range(bs)] source_wav = pad_sequence(source_wav_list, batch_first=True) target_wav_list = [] for j in range(self.n_srcs): target_wav_list.append(pad_sequence([torch.from_numpy(sorted_batch[i][3][j]) for i in range(bs)], batch_first=True)) feat_length = torch.from_numpy(np.array([stft.size(0) for stft in mix_stft_list])) '\n source_wav_list (list(tensor)):\n list of audio samples for the source\n\n uttname_list (list(str)):\n list of utterance names\n\n source_attr (dict):\n dictionary containing magnitude and phase information for the sources\n\n source_wav (tensor):\n padded version of source_wav_list, with size [bs, max_T]\n\n target_attr (dict):\n dictionary containing magnitude and phase information for the targets\n\n feat_length (tensor):\n length of the STFT feature for each utterance\n\n wav_length (tensor):\n number of samples in each utterance\n ' return (source_wav_list, uttname_list, source_attr, source_wav, target_attr, target_wav_list, feat_length, wav_length)
class SepRNN(torch.nn.Module): def __init__(self, input_dim, num_bins, rnn='lstm', num_spks=2, num_layers=3, hidden_size=896, dropout=0.0, non_linear='relu', bidirectional=True): super(SepRNN, self).__init__() if (non_linear not in ['relu', 'sigmoid', 'tanh']): raise ValueError('Unsupported non-linear type:{}'.format(non_linear)) self.num_spks = num_spks rnn = rnn.upper() if (rnn not in ['RNN', 'LSTM', 'GRU']): raise ValueError('Unsupported rnn type: {}'.format(rnn)) self.rnn = getattr(torch.nn, rnn)(input_dim, hidden_size, num_layers, batch_first=True, dropout=dropout, bidirectional=bidirectional) self.drops = torch.nn.Dropout(p=dropout) self.linear = torch.nn.ModuleList([torch.nn.Linear(((hidden_size * 2) if bidirectional else hidden_size), num_bins) for _ in range(self.num_spks)]) self.non_linear = {'relu': torch.nn.functional.relu, 'sigmoid': torch.nn.functional.sigmoid, 'tanh': torch.nn.functional.tanh}[non_linear] self.num_bins = num_bins def forward(self, x, train=True): assert isinstance(x, PackedSequence) (x, _) = self.rnn(x) (x, len_x) = pad_packed_sequence(x, batch_first=True) x = self.drops(x) m = [] for linear in self.linear: y = linear(x) y = self.non_linear(y) if (not train): y = y.view((- 1), self.num_bins) m.append(y) return m
def main(): output_dir = '{}/wav{}/{}/{}'.format(args.tgt_dir, args.sample_rate, args.mode, args.part) if os.path.exists(output_dir): raise ValueError('Warning: {} already exists, please check!'.format(output_dir)) else: os.makedirs(output_dir) wav_dir = '{}/wav{}/{}/{}'.format(args.src_dir, args.sample_rate, args.mode, args.part) assert os.path.exists(wav_dir) for cond in ['s1', 's2', 'mix_clean', 'mix_both', 'mix_single', 'noise']: if (not os.path.exists('{}/{}'.format(wav_dir, cond))): continue filelist = [f for f in os.listdir('{}/{}'.format(wav_dir, cond)) if f.endswith('.wav')] filelist.sort() cond_dir = '{}/{}'.format(output_dir, cond) if (not os.path.exists(cond_dir)): os.makedirs(cond_dir) wav_scp_file = open('{}/wav.scp'.format(cond_dir), 'w') utt2spk_file = open('{}/utt2spk'.format(cond_dir), 'w') for f in filelist: uttname = f.strip('.wav') wav_scp_file.write('{} {}/{}/{}\n'.format(uttname, wav_dir, cond, f)) utt2spk_file.write('{} {}\n'.format(uttname, uttname)) wav_scp_file.close() utt2spk_file.close() shutil.copyfile('{}/utt2spk'.format(cond_dir), '{}/spk2utt'.format(cond_dir)) return 0
def get_utt2path(wav_scp_file): utt2path = {} with open(wav_scp_file, 'r') as fh: content = fh.readlines() for line in content: line = line.strip('\n') (utt, path) = line.split() utt2path[utt] = path return utt2path
def main(): random.seed(args.seed) with open('{}/s1/utt2spk'.format(args.src_dir), 'r') as fh: content = fh.readlines() uttlist = [] for line in content: line = line.strip('\n') utt = line.split()[0] uttlist.append(utt) uttlist.sort() num_utt_ori = len(uttlist) random.shuffle(uttlist) uttlist = uttlist[:args.sample] uttlist.sort() print('Selecting {} utts from {} utts'.format(len(uttlist), num_utt_ori)) for dset in ['mix_both', 'mix_clean', 'mix_single', 'noise', 's1', 's2']: (src_dset, tgt_dset) = ('{}/{}'.format(args.src_dir, dset), '{}/{}'.format(args.tgt_dir, dset)) os.makedirs(tgt_dset) with open('{}/utt2spk'.format(tgt_dset), 'w') as fh: for utt in uttlist: fh.write('{} {}\n'.format(utt, utt)) with open('{}/spk2utt'.format(tgt_dset), 'w') as fh: for utt in uttlist: fh.write('{} {}\n'.format(utt, utt)) utt2path = get_utt2path('{}/wav.scp'.format(src_dset)) with open('{}/wav.scp'.format(tgt_dset), 'w') as fh: for utt in uttlist: fh.write('{} {}\n'.format(utt, utt2path[utt])) return 0
class SeparationDataset(Dataset): def __init__(self, data_dir, rate=16000, src=['mix_clean'], tgt=['s1', 's2'], n_fft=512, hop_length=320, win_length=512, window='hann', center=True): super(SeparationDataset, self).__init__() "\n Args:\n data_dir (str):\n prepared data directory\n\n rate (int):\n audio sample rate\n\n src and tgt (list(str)):\n the input and desired output.\n LibriMix offeres different options for the users. For\n clean source separation, src=['mix_clean'] and tgt=['s1', 's2'].\n Please see https://github.com/JorisCos/LibriMix for details\n\n n_fft (int):\n length of the windowed signal after padding with zeros.\n\n hop_length (int):\n number of audio samples between adjacent STFT columns.\n\n win_length (int):\n length of window for each frame\n\n window (str):\n type of window function, only support Hann window now\n\n center (bool):\n whether to pad input on both sides so that the\n t-th frame is centered at time t * hop_length\n\n The STFT related parameters are the same as librosa.\n " self.data_dir = data_dir self.rate = rate self.src = src self.tgt = tgt self.n_fft = n_fft self.hop_length = hop_length self.win_length = win_length self.window = window self.center = center self.n_srcs = len(self.tgt) assert ((len(self.src) == 1) and (len(self.tgt) == 2)) cond_list = ['s1', 's2', 'noise', 'mix_clean', 'mix_both', 'mix_single'] reco2path = {} for cond in (src + tgt): assert (cond in cond_list) assert os.path.exists('{}/{}/wav.scp'.format(self.data_dir, cond)) with open('{}/{}/wav.scp'.format(self.data_dir, cond), 'r') as fh: content = fh.readlines() for line in content: line = line.strip('\n') (uttname, path) = line.split() if (uttname not in reco2path): reco2path[uttname] = {} reco2path[uttname][cond] = path self.reco2path = reco2path self.recolist = list(self.reco2path.keys()) self.recolist.sort() def __len__(self): return len(self.recolist) def __getitem__(self, i): reco = self.recolist[i] src_path = self.reco2path[reco][self.src[0]] (src_samp, rate) = librosa.load(src_path, sr=None) assert (rate == self.rate) src_feat = np.transpose(librosa.stft(src_samp, n_fft=self.n_fft, hop_length=self.hop_length, win_length=self.win_length, window=self.window, center=self.center)) (tgt_samp_list, tgt_feat_list) = ([], []) for j in range(self.n_srcs): tgt_path = self.reco2path[reco][self.tgt[j]] (tgt_samp, rate) = librosa.load(tgt_path, sr=None) assert (rate == self.rate) tgt_feat = np.transpose(librosa.stft(tgt_samp, n_fft=self.n_fft, hop_length=self.hop_length, win_length=self.win_length, window=self.window, center=self.center)) tgt_samp_list.append(tgt_samp) tgt_feat_list.append(tgt_feat) '\n reco (str):\n name of the utterance\n\n src_sample (ndarray):\n audio samples for the source [T, ]\n\n src_feat (ndarray):\n the STFT feature map for the source with shape [T1, D]\n\n tgt_samp_list (list(ndarray)):\n list of audio samples for the targets\n\n tgt_feat_list (list(ndarray)):\n list of STFT feature map for the targets\n ' return (reco, src_samp, src_feat, tgt_samp_list, tgt_feat_list) def collate_fn(self, batch): sorted_batch = sorted(batch, key=(lambda x: (- x[1].shape[0]))) bs = len(sorted_batch) uttname_list = [sorted_batch[i][0] for i in range(bs)] source_attr = {} mix_magnitude_list = [torch.from_numpy(np.abs(sorted_batch[i][2])) for i in range(bs)] mix_phase_list = [torch.from_numpy(np.angle(sorted_batch[i][2])) for i in range(bs)] mix_stft_list = [torch.from_numpy(sorted_batch[i][2]) for i in range(bs)] mix_magnitude = pad_sequence(mix_magnitude_list, batch_first=True) mix_phase = pad_sequence(mix_phase_list, batch_first=True) mix_stft = pad_sequence(mix_stft_list, batch_first=True) source_attr['magnitude'] = mix_magnitude source_attr['phase'] = mix_phase source_attr['stft'] = mix_stft target_attr = {} target_attr['magnitude'] = [] target_attr['phase'] = [] for j in range(self.n_srcs): tgt_magnitude_list = [torch.from_numpy(np.abs(sorted_batch[i][4][j])) for i in range(bs)] tgt_phase_list = [torch.from_numpy(np.angle(sorted_batch[i][4][j])) for i in range(bs)] tgt_magnitude = pad_sequence(tgt_magnitude_list, batch_first=True) tgt_phase = pad_sequence(tgt_phase_list, batch_first=True) target_attr['magnitude'].append(tgt_magnitude) target_attr['phase'].append(tgt_phase) wav_length = torch.from_numpy(np.array([len(sorted_batch[i][1]) for i in range(bs)])) source_wav_list = [torch.from_numpy(sorted_batch[i][1]) for i in range(bs)] source_wav = pad_sequence(source_wav_list, batch_first=True) target_wav_list = [] for j in range(self.n_srcs): target_wav_list.append(pad_sequence([torch.from_numpy(sorted_batch[i][3][j]) for i in range(bs)], batch_first=True)) feat_length = torch.from_numpy(np.array([stft.size(0) for stft in mix_stft_list])) '\n source_wav_list (list(tensor)):\n list of audio samples for the source\n\n uttname_list (list(str)):\n list of utterance names\n\n source_attr (dict):\n dictionary containing magnitude and phase information for the sources\n\n source_wav (tensor):\n padded version of source_wav_list, with size [bs, max_T]\n\n target_attr (dict):\n dictionary containing magnitude and phase information for the targets\n\n feat_length (tensor):\n length of the STFT feature for each utterance\n\n wav_length (tensor):\n number of samples in each utterance\n ' return (source_wav_list, uttname_list, source_attr, source_wav, target_attr, target_wav_list, feat_length, wav_length)
class SepRNN(torch.nn.Module): def __init__(self, input_dim, num_bins, rnn='lstm', num_spks=2, num_layers=3, hidden_size=896, dropout=0.0, non_linear='relu', bidirectional=True): super(SepRNN, self).__init__() if (non_linear not in ['relu', 'sigmoid', 'tanh', 'none']): raise ValueError('Unsupported non-linear type:{}'.format(non_linear)) self.num_spks = num_spks rnn = rnn.upper() if (rnn not in ['RNN', 'LSTM', 'GRU']): raise ValueError('Unsupported rnn type: {}'.format(rnn)) self.rnn = getattr(torch.nn, rnn)(input_dim, hidden_size, num_layers, batch_first=True, dropout=dropout, bidirectional=bidirectional) self.drops = torch.nn.Dropout(p=dropout) self.linear = torch.nn.ModuleList([torch.nn.Linear(((hidden_size * 2) if bidirectional else hidden_size), num_bins) for _ in range(self.num_spks)]) self.non_linear = {'relu': torch.nn.functional.relu, 'sigmoid': torch.sigmoid, 'tanh': torch.nn.functional.tanh, 'none': torch.nn.Identity()}[non_linear] self.num_bins = num_bins def forward(self, x, train=True): assert isinstance(x, PackedSequence) (x, _) = self.rnn(x) (x, len_x) = pad_packed_sequence(x, batch_first=True) x = self.drops(x) m = [] for linear in self.linear: y = linear(x) y = self.non_linear(y) if (not train): y = y.view((- 1), self.num_bins) m.append(y) return m
def main(): output_dir = '{}/wav{}/{}/{}'.format(args.tgt_dir, args.sample_rate, args.mode, args.part) if os.path.exists(output_dir): raise ValueError('Warning: {} already exists, please check!'.format(output_dir)) else: os.makedirs(output_dir) wav_dir = '{}/wav{}/{}/{}'.format(args.src_dir, args.sample_rate, args.mode, args.part) assert os.path.exists(wav_dir) for cond in ['s1', 's2', 'mix_clean', 'mix_both', 'mix_single', 'noise']: if (not os.path.exists('{}/{}'.format(wav_dir, cond))): continue filelist = [f for f in os.listdir('{}/{}'.format(wav_dir, cond)) if f.endswith('.wav')] filelist.sort() cond_dir = '{}/{}'.format(output_dir, cond) if (not os.path.exists(cond_dir)): os.makedirs(cond_dir) wav_scp_file = open('{}/wav.scp'.format(cond_dir), 'w') utt2spk_file = open('{}/utt2spk'.format(cond_dir), 'w') for f in filelist: uttname = f.strip('.wav') wav_scp_file.write('{} {}/{}/{}\n'.format(uttname, wav_dir, cond, f)) utt2spk_file.write('{} {}\n'.format(uttname, uttname)) wav_scp_file.close() utt2spk_file.close() shutil.copyfile('{}/utt2spk'.format(cond_dir), '{}/spk2utt'.format(cond_dir)) return 0
def get_utt2path(wav_scp_file): utt2path = {} with open(wav_scp_file, 'r') as fh: content = fh.readlines() for line in content: line = line.strip('\n') (utt, path) = line.split() utt2path[utt] = path return utt2path
def main(): random.seed(args.seed) with open('{}/s1/utt2spk'.format(args.src_dir), 'r') as fh: content = fh.readlines() uttlist = [] for line in content: line = line.strip('\n') utt = line.split()[0] uttlist.append(utt) uttlist.sort() num_utt_ori = len(uttlist) random.shuffle(uttlist) uttlist = uttlist[:args.sample] uttlist.sort() print('Selecting {} utts from {} utts'.format(len(uttlist), num_utt_ori)) for dset in ['mix_both', 'mix_clean', 'mix_single', 'noise', 's1', 's2']: (src_dset, tgt_dset) = ('{}/{}'.format(args.src_dir, dset), '{}/{}'.format(args.tgt_dir, dset)) os.makedirs(tgt_dset) with open('{}/utt2spk'.format(tgt_dset), 'w') as fh: for utt in uttlist: fh.write('{} {}\n'.format(utt, utt)) with open('{}/spk2utt'.format(tgt_dset), 'w') as fh: for utt in uttlist: fh.write('{} {}\n'.format(utt, utt)) utt2path = get_utt2path('{}/wav.scp'.format(src_dset)) with open('{}/wav.scp'.format(tgt_dset), 'w') as fh: for utt in uttlist: fh.write('{} {}\n'.format(utt, utt2path[utt])) return 0
class DownstreamExpert(SpeakerExpert): '\n Basically the same as the speaker utterance expert, except handles the speaker frame-wise label\n ' def __init__(self, upstream_dim, downstream_expert, expdir, **kwargs): super(DownstreamExpert, self).__init__(upstream_dim, downstream_expert, expdir, **kwargs) def forward(self, mode, features, labels, records, **kwargs): '\n Args:\n features:\n list of unpadded features [feat1, feat2, ...]\n each feat is in torch.FloatTensor and already\n put in the device assigned by command-line args\n\n labels:\n the frame-wise spekaer labels\n\n records:\n defaultdict(list), by appending contents into records,\n these contents can be averaged and logged on Tensorboard\n later by self.log_records every log_step\n\n Return:\n loss:\n the loss to be optimized, should not be detached\n ' lengths = torch.LongTensor([len(l) for l in features]) features = pad_sequence(features, batch_first=True) labels = labels.unsqueeze((- 1)).expand(features.size(0), features.size(1)).to(features.device) predicted = self.model(features) class_num = predicted.size((- 1)) loss = self.objective(predicted.reshape((- 1), class_num), labels.reshape((- 1))) predicted_classid = predicted.max(dim=(- 1)).indices sames = (predicted_classid == labels) for (s, l) in zip(sames, lengths): records['acc'] += s[:l].tolist() return loss
class SpeakerDataset(Dataset): def __init__(self, split, bucket_size, libri_root, split_file, bucket_file, sample_rate=16000, train_dev_seed=1337, **kwargs): self.libri_root = libri_root self.split_file = split_file self.sample_rate = sample_rate assert os.path.isdir(bucket_file), 'Please first run `preprocess/generate_len_for_bucket.py to get bucket file.' self.table = pd.read_csv(os.path.join(bucket_file, 'train-clean-100.csv')).sort_values(by=['length'], ascending=False) X = self.table['file_path'].tolist() X_lens = self.table['length'].tolist() if (((split == 'train') or (split == 'dev')) and os.path.isfile(os.path.join(split_file, 'train_split.txt'))): usage_list = open(os.path.join(split_file, 'train_split.txt')).readlines() random.seed(train_dev_seed) random.shuffle(usage_list) percent = int((len(usage_list) * 0.9)) usage_list = (usage_list[:percent] if (split == 'train') else usage_list[percent:]) elif ((split == 'test') and os.path.isfile(os.path.join(split_file, 'test_split.txt'))): usage_list = open(os.path.join(split_file, 'test_split.txt')).readlines() else: raise NotImplementedError('Invalid `split` argument!') usage_list = {line.strip('\n'): None for line in usage_list} self.X = [] (batch_x, batch_len) = ([], []) for (x, x_len) in zip(X, X_lens): if (self._parse_x_name(x) in usage_list): batch_x.append(x) batch_len.append(x_len) if (len(batch_x) == bucket_size): if ((bucket_size >= 2) and (max(batch_len) > HALF_BATCHSIZE_TIME)): self.X.append(batch_x[:(bucket_size // 2)]) self.X.append(batch_x[(bucket_size // 2):]) else: self.X.append(batch_x) (batch_x, batch_len) = ([], []) if (len(batch_x) > 1): if (self._parse_x_name(x) in usage_list): self.X.append(batch_x) print('[Dataset] - Computing speaker class...') speakers = self._get_all_speakers(X) self.speaker2idx = self._compute_speaker2idx(speakers) self.class_num = len(self.speaker2idx) print(((((('[Dataset] - # possible speaker classes: ' + str(self.class_num)) + ', number of data for ') + split) + ': ') + str(len(usage_list)))) def _parse_x_name(self, x): return x.split('/')[(- 1)].split('.')[0] def _load_wav(self, wav_path): (wav, sr) = torchaudio.load(os.path.join(self.libri_root, wav_path)) return wav.view((- 1)) def _get_speaker_from_path(self, x): return x.split('/')[(- 1)].split('.')[0].split('-')[0] def _get_all_speakers(self, X): speaker_set = {} for x in X: speaker = self._get_speaker_from_path(x) if (speaker not in speaker_set): speaker_set[speaker] = 0 else: speaker_set[speaker] += 1 return speaker_set def _compute_speaker2idx(self, speakers): idx = 0 speaker2idx = {} for speaker in sorted(speakers): if ((speaker not in speaker2idx) and (speakers[speaker] > SPEAKER_THRESHOLD)): speaker2idx[speaker] = idx idx += 1 return speaker2idx def __len__(self): return len(self.X) def __getitem__(self, index): wav_batch = [self._load_wav(x_file) for x_file in self.X[index]] label_batch = torch.LongTensor([self.speaker2idx[self._get_speaker_from_path(x_file)] for x_file in self.X[index]]) return (wav_batch, label_batch) def collate_fn(self, items): return (items[0][0], items[0][1])
class DownstreamExpert(nn.Module): '\n Used to handle downstream-specific operations\n eg. downstream forward, metric computation, contents to log\n ' def __init__(self, upstream_dim, downstream_expert, expdir, **kwargs): super(DownstreamExpert, self).__init__() self.upstream_dim = upstream_dim self.datarc = downstream_expert['datarc'] self.modelrc = downstream_expert['modelrc'] self.train_dataset = SpeakerDataset('train', self.datarc['train_batch_size'], **self.datarc) self.dev_dataset = SpeakerDataset('dev', self.datarc['eval_batch_size'], **self.datarc) self.test_dataset = SpeakerDataset('test', self.datarc['eval_batch_size'], **self.datarc) self.model = Model(input_dim=self.upstream_dim, output_class_num=self.train_dataset.class_num, **self.modelrc) self.objective = nn.CrossEntropyLoss() self.logging = os.path.join(expdir, 'log.log') self.best = defaultdict((lambda : 0)) def _get_train_dataloader(self, dataset): return DataLoader(dataset, batch_size=1, shuffle=True, num_workers=self.datarc['num_workers'], drop_last=False, pin_memory=True, collate_fn=dataset.collate_fn) def _get_eval_dataloader(self, dataset): return DataLoader(dataset, batch_size=1, shuffle=False, num_workers=self.datarc['num_workers'], drop_last=False, pin_memory=True, collate_fn=dataset.collate_fn) '\n Datalaoder Specs:\n Each dataloader should output in the following format:\n\n [[wav1, wav2, ...], your_other_contents1, your_other_contents2, ...]\n\n where wav1, wav2 ... are in variable length\n each wav is torch.FloatTensor in cpu with dim()==1 and sample_rate==16000\n ' def get_train_dataloader(self): return self._get_train_dataloader(self.train_dataset) def get_dev_dataloader(self): return self._get_eval_dataloader(self.dev_dataset) def get_test_dataloader(self): return self._get_eval_dataloader(self.test_dataset) def get_dataloader(self, mode): return eval(f'self.get_{mode}_dataloader')() def forward(self, mode, features, labels, records, **kwargs): '\n Args:\n features:\n list of unpadded features [feat1, feat2, ...]\n each feat is in torch.FloatTensor and already\n put in the device assigned by command-line args\n\n labels:\n the utterance-wise spekaer labels\n\n records:\n defaultdict(list), by appending contents into records,\n these contents can be averaged and logged on Tensorboard\n later by self.log_records every log_step\n\n Return:\n loss:\n the loss to be optimized, should not be detached\n ' features = torch.stack([f.mean(dim=0) for f in features], dim=0) labels = labels.to(features.device) predicted = self.model(features) loss = self.objective(predicted, labels) predicted_classid = predicted.max(dim=(- 1)).indices records['acc'] += (predicted_classid == labels).view((- 1)).cpu().float().tolist() return loss def log_records(self, mode, records, logger, global_step, **kwargs): "\n Args:\n records:\n defaultdict(list), contents already appended\n\n logger:\n Tensorboard SummaryWriter\n please use f'{prefix}your_content_name' as key name\n to log your customized contents\n\n global_step:\n global_step in runner, which is helpful for Tensorboard logging\n " prefix = f'libri_speaker/{mode}-' average = torch.FloatTensor(records['acc']).mean().item() logger.add_scalar(f'{prefix}acc', average, global_step=global_step) message = f'''{prefix}|step:{global_step}|acc:{average} ''' save_ckpt = [] if (average > self.best[prefix]): self.best[prefix] = average message = f'best|{message}' name = prefix.split('/')[(- 1)].split('-')[0] save_ckpt.append(f'best-states-{name}.ckpt') with open(self.logging, 'a') as f: f.write(message) print(message) return save_ckpt
class Model(nn.Module): def __init__(self, input_dim, output_class_num, **kwargs): super(Model, self).__init__() self.linear = nn.Linear(input_dim, output_class_num) def forward(self, features): predicted = self.linear(features) return predicted
class SpeechCommandsBaseDataset(Dataset): '12-class Speech Commands base dataset.' def __init__(self): self.class2index = {CLASSES[i]: i for i in range(len(CLASSES))} self.class_num = 12 self.data = [] def __getitem__(self, idx): (class_name, audio_path) = self.data[idx] (wav, _) = apply_effects_file(str(audio_path), EFFECTS) wav = wav.squeeze(0).numpy() fileid = '-'.join(Path(audio_path).parts[(- 2):]) return (wav, self.class2index[class_name], fileid) def __len__(self): return len(self.data) def collate_fn(self, samples): 'Collate a mini-batch of data.' return zip(*samples)
class SpeechCommandsDataset(SpeechCommandsBaseDataset): 'Training and validation dataset.' def __init__(self, data_list, **kwargs): super().__init__() data = [((class_name, audio_path) if (class_name in self.class2index.keys()) else ('_unknown_', audio_path)) for (class_name, audio_path) in data_list] data += [('_silence_', audio_path) for audio_path in Path(kwargs['speech_commands_root'], '_background_noise_').glob('*.wav')] class_counts = {class_name: 0 for class_name in CLASSES} for (class_name, _) in data: class_counts[class_name] += 1 sample_weights = [(len(data) / class_counts[class_name]) for (class_name, _) in data] self.data = data self.sample_weights = sample_weights def __getitem__(self, idx): (wav, label, stem) = super().__getitem__(idx) if (label == self.class2index['_silence_']): random_offset = randint(0, (len(wav) - 16000)) wav = wav[random_offset:(random_offset + 16000)] return (wav, label, stem)
class SpeechCommandsTestingDataset(SpeechCommandsBaseDataset): 'Testing dataset.' def __init__(self, **kwargs): super().__init__() self.data = [(class_dir.name, audio_path) for class_dir in Path(kwargs['speech_commands_test_root']).iterdir() if class_dir.is_dir() for audio_path in class_dir.glob('*.wav')]
class DownstreamExpert(nn.Module): '\n Used to handle downstream-specific operations\n eg. downstream forward, metric computation, contents to log\n ' def __init__(self, upstream_dim: int, downstream_expert: dict, expdir: str, **kwargs): super(DownstreamExpert, self).__init__() self.upstream_dim = upstream_dim self.datarc = downstream_expert['datarc'] self.modelrc = downstream_expert['modelrc'] (train_list, valid_list) = split_dataset(self.datarc['speech_commands_root']) self.train_dataset = SpeechCommandsDataset(train_list, **self.datarc) self.dev_dataset = SpeechCommandsDataset(valid_list, **self.datarc) self.test_dataset = SpeechCommandsTestingDataset(**self.datarc) model_cls = eval(self.modelrc['select']) model_conf = self.modelrc.get(self.modelrc['select'], {}) self.projector = nn.Linear(upstream_dim, self.modelrc['projector_dim']) self.model = model_cls(input_dim=self.modelrc['projector_dim'], output_dim=self.train_dataset.class_num, **model_conf) self.objective = nn.CrossEntropyLoss() self.expdir = expdir self.register_buffer('best_score', torch.zeros(1)) def _get_balanced_train_dataloader(self, dataset, drop_last=False): sampler = WeightedRandomSampler(dataset.sample_weights, len(dataset.sample_weights)) if is_initialized(): sampler = DistributedSamplerWrapper(sampler) return DataLoader(dataset, sampler=sampler, batch_size=self.datarc['batch_size'], drop_last=drop_last, num_workers=self.datarc['num_workers'], collate_fn=dataset.collate_fn) def _get_balanced_dev_dataloader(self, dataset, drop_last=False): return DataLoader(dataset, sampler=WeightedRandomSampler(dataset.sample_weights, len(dataset.sample_weights)), batch_size=self.datarc['batch_size'], drop_last=drop_last, num_workers=self.datarc['num_workers'], collate_fn=dataset.collate_fn) def _get_dataloader(self, dataset): return DataLoader(dataset, shuffle=False, batch_size=self.datarc['batch_size'], drop_last=False, num_workers=self.datarc['num_workers'], collate_fn=dataset.collate_fn) def get_dataloader(self, mode): if (mode == 'train'): return self._get_balanced_train_dataloader(self.train_dataset, drop_last=True) elif (mode == 'dev'): return self._get_balanced_dev_dataloader(self.dev_dataset, drop_last=False) elif (mode == 'test'): return self._get_dataloader(self.test_dataset) else: raise NotImplementedError def forward(self, mode, features, labels, filenames, records, **kwargs): device = features[0].device features_len = torch.IntTensor([len(feat) for feat in features]).to(device=device) features = pad_sequence(features, batch_first=True) features = self.projector(features) (predicted, _) = self.model(features, features_len) labels = torch.LongTensor(labels).to(features.device) loss = self.objective(predicted, labels) predicted_classid = predicted.max(dim=(- 1)).indices records['loss'].append(loss.item()) records['acc'] += (predicted_classid == labels).view((- 1)).cpu().float().tolist() records['filename'] += filenames records['predict'] += [CLASSES[idx] for idx in predicted_classid.cpu().tolist()] records['truth'] += [CLASSES[idx] for idx in labels.cpu().tolist()] return loss def log_records(self, mode, records, logger, global_step, **kwargs): save_names = [] for key in ['loss', 'acc']: values = records[key] average = (sum(values) / len(values)) logger.add_scalar(f'speech_commands/{mode}-{key}', average, global_step=global_step) with open(Path(self.expdir, 'log.log'), 'a') as f: if (key == 'acc'): print(f'{mode} {key}: {average}') f.write(f'''{mode} at step {global_step}: {average} ''') if ((mode == 'dev') and (average > self.best_score)): self.best_score = (torch.ones(1) * average) f.write(f'''New best on {mode} at step {global_step}: {average} ''') save_names.append(f'{mode}-best.ckpt') with open((Path(self.expdir) / f'{mode}_predict.txt'), 'w') as file: lines = [f'''{f} {i} ''' for (f, i) in zip(records['filename'], records['predict'])] file.writelines(lines) with open((Path(self.expdir) / f'{mode}_truth.txt'), 'w') as file: lines = [f'''{f} {i} ''' for (f, i) in zip(records['filename'], records['truth'])] file.writelines(lines) return save_names
def split_dataset(root_dir: Union[(str, Path)], max_uttr_per_class=((2 ** 27) - 1)) -> Tuple[(List[Tuple[(str, str)]], List[Tuple[(str, str)]])]: 'Split Speech Commands into 3 set.\n \n Args:\n root_dir: speech commands dataset root dir\n max_uttr_per_class: predefined value in the original paper\n \n Return:\n train_list: [(class_name, audio_path), ...]\n valid_list: as above\n ' (train_list, valid_list) = ([], []) for entry in Path(root_dir).iterdir(): if ((not entry.is_dir()) or (entry.name == '_background_noise_')): continue for audio_path in entry.glob('*.wav'): speaker_hashed = re.sub('_nohash_.*$', '', audio_path.name) hashed_again = hashlib.sha1(speaker_hashed.encode('utf-8')).hexdigest() percentage_hash = ((int(hashed_again, 16) % (max_uttr_per_class + 1)) * (100.0 / max_uttr_per_class)) if (percentage_hash < 10): valid_list.append((entry.name, audio_path)) elif (percentage_hash < 20): pass else: train_list.append((entry.name, audio_path)) return (train_list, valid_list)
class Model(nn.Module): '\n Not used in SUPERB Benchmark\n ' def __init__(self, input_dim, output_class_num, **kwargs): super(Model, self).__init__() hidden_dim = kwargs['hidden_dim'] self.connector = nn.Linear(input_dim, hidden_dim) self.fc1 = nn.Linear(hidden_dim, hidden_dim) self.fc2 = nn.Linear(hidden_dim, output_class_num) def forward(self, features): features = F.relu(self.connector(features)) features = self.fc1(features) pooled = features.mean(dim=1) predicted = self.fc2(pooled) return predicted
class AdditionalDataset(): @classmethod def from_tsv(cls, file, key, bpe_tokenizer=None, pre_tokenizer=None): data = [] with open(file, 'r') as file: reader = csv.DictReader(file, delimiter='\t', quotechar=None, doublequote=False, lineterminator='\n', quoting=csv.QUOTE_NONE) for line in reader: data.append(line[key]) return cls(data, bpe_tokenizer, pre_tokenizer) def __init__(self, data, dictionary, bpe_tokenizer=None, pre_tokenizer=None): self.data = data self.bpe_tokenizer = bpe_tokenizer self.pre_tokenizer = pre_tokenizer self.dictionary = dictionary def _create_target(self, index): tokenized = self._tokenize_text(self.data[index]) target = self.dictionary.encode_line(tokenized, add_if_not_exist=False, append_eos=True).long() return target def get_addtional_input(self, id_list): target = [self._create_target(id) for id in id_list] batched_target = fairseq.data.data_utils.collate_tokens(target, self.dictionary.pad(), self.dictionary.eos(), left_pad=False, move_eos_to_beginning=False) target_lengths = torch.tensor([t.size(0) for t in target], dtype=torch.long) prev_output_tokens = fairseq.data.data_utils.collate_tokens(target, self.dictionary.pad(), self.dictionary.eos(), left_pad=False, move_eos_to_beginning=True) ntokens = sum((t.size(0) for t in target)) return {'target': batched_target, 'prev_output_tokens': prev_output_tokens, 'target_lengths': target_lengths, 'ntokens': ntokens} def _tokenize_text(self, text): if (self.pre_tokenizer is not None): text = self.pre_tokenizer.encode(text) if (self.bpe_tokenizer is not None): text = self.bpe_tokenizer.encode(text) return text
class S3prl_SpeechToTextTask(SpeechToTextTask): def __init__(self, *args, **kwargs): super().__init__(*args, **kwargs) def load_dataset(self, split, max_feature_len=(- 1), epoch=1, combine=False, **kwargs): is_train_split = split.startswith('train') pre_tokenizer = self.build_tokenizer(self.args) bpe_tokenizer = self.build_bpe(self.args) self.datasets[split] = S3prl_SpeechToTextDatasetCreator.from_tsv(self.args.data, self.data_cfg, split, self.tgt_dict, pre_tokenizer, bpe_tokenizer, is_train_split=is_train_split, epoch=epoch, seed=self.args.seed, upstream_rate=self.upstream_rate, max_feature_len=max_feature_len) def build_model(self, args, input_dim): args.input_feat_per_channel = input_dim args.input_channels = self.data_cfg.input_channels return super(SpeechToTextTask, self).build_model(args)
class S3prl_SpeechToTextDatasetCreator(SpeechToTextDatasetCreator): def __init__(self, *args, **kwargs): super().__init__(*args, **kwargs) KEY_SAMPLE_RATE = 'sr' DEFAULT_SAMPLE_RATE = 16000 @classmethod def _from_list(cls, split_name: str, is_train_split, samples: List[List[Dict]], data_cfg: S2TDataConfig, tgt_dict, pre_tokenizer, bpe_tokenizer, upstream_rate, max_feature_len) -> SpeechToTextDataset: (audio_paths, n_frames, src_texts, tgt_texts, ids) = ([], [], [], [], []) (speakers, src_langs, tgt_langs) = ([], [], []) srs = [] for s in samples: ids.extend([ss[cls.KEY_ID] for ss in s]) audio_paths.extend([op.join(data_cfg.audio_root, ss[cls.KEY_AUDIO]) for ss in s]) n_frames.extend([int(ss[cls.KEY_N_FRAMES]) for ss in s]) tgt_texts.extend([ss[cls.KEY_TGT_TEXT] for ss in s]) src_texts.extend([ss.get(cls.KEY_SRC_TEXT, cls.DEFAULT_SRC_TEXT) for ss in s]) speakers.extend([ss.get(cls.KEY_SPEAKER, cls.DEFAULT_SPEAKER) for ss in s]) src_langs.extend([ss.get(cls.KEY_SRC_LANG, cls.DEFAULT_LANG) for ss in s]) tgt_langs.extend([ss.get(cls.KEY_TGT_LANG, cls.DEFAULT_LANG) for ss in s]) srs.extend([int(ss.get(cls.KEY_SAMPLE_RATE, cls.DEFAULT_SAMPLE_RATE)) for ss in s]) return S3prl_SpeechToTextDataset(split_name, is_train_split, data_cfg, audio_paths, n_frames, src_texts=src_texts, tgt_texts=tgt_texts, speakers=speakers, src_langs=src_langs, tgt_langs=tgt_langs, srs=srs, ids=ids, tgt_dict=tgt_dict, pre_tokenizer=pre_tokenizer, bpe_tokenizer=bpe_tokenizer, upstream_rate=upstream_rate, max_feature_len=max_feature_len) @classmethod def from_tsv(cls, root: str, data_cfg: S2TDataConfig, splits: str, tgt_dict, pre_tokenizer, bpe_tokenizer, is_train_split: bool, epoch: int, seed: int, upstream_rate: int, max_feature_len: int) -> SpeechToTextDataset: _splits = splits.split(',') assert (len(_splits) == 1), 'do not support multiple files training' split = _splits[0] tsv_path = op.join(root, f'{split}.tsv') if (not op.isfile(tsv_path)): raise FileNotFoundError(f'Dataset not found: {tsv_path}') with open(tsv_path) as f: reader = csv.DictReader(f, delimiter='\t', quotechar=None, doublequote=False, lineterminator='\n', quoting=csv.QUOTE_NONE) samples = [dict(e) for e in reader] assert (len(samples) > 0) return cls._from_list(split, is_train_split, [samples], data_cfg, tgt_dict, pre_tokenizer, bpe_tokenizer, upstream_rate, max_feature_len)
class S3prl_SpeechToTextDataset(SpeechToTextDataset): TARGET_RATE = 16000 def __init__(self, *args, srs=Optional[List[int]], upstream_rate=160, max_feature_len=(- 1), **kwargs): super().__init__(*args, **kwargs) self.srs = srs self.max_feature_len = max_feature_len self.max_wav_len = (max_feature_len * upstream_rate) self.resamplers = {} for sr in set(srs): self.resamplers[sr] = torchaudio.transforms.Resample(orig_freq=sr, new_freq=self.TARGET_RATE) for i in range(len(self.n_frames)): new_n_frames = (((self.n_frames[i] * self.TARGET_RATE) / self.srs[i]) / upstream_rate) if ((self.max_feature_len > 0) and (new_n_frames > max_feature_len)): new_n_frames = max_feature_len self.n_frames[i] = int(new_n_frames) def __getitem__(self, index: int) -> Tuple[(str, int, torch.Tensor, Optional[torch.Tensor])]: (source, sr) = torchaudio.load(self.audio_paths[index]) assert (self.srs[index] == sr) source = self.resamplers[sr](source) source = torch.mean(source, dim=0) source = source.view((- 1)) if (self.feature_transforms is not None): assert (not self.data_cfg.use_audio_input) source = self.feature_transforms(source) source = source.float() if (self.max_feature_len > 0): if (source.size(0) > self.max_wav_len): print(f'wav too long({source.size(0)}), truncate to {self.max_wav_len} (id={index})') source = source[:self.max_wav_len] target = None if (self.tgt_texts is not None): tokenized = self.tokenize_text(self.tgt_texts[index]) target = self.tgt_dict.encode_line(tokenized, add_if_not_exist=False, append_eos=True).long() if self.data_cfg.prepend_tgt_lang_tag: lang_tag = self.LANG_TAG_TEMPLATE.format(self.tgt_langs[index]) lang_tag_idx = self.tgt_dict.index(lang_tag) target = torch.cat((torch.LongTensor([lang_tag_idx]), target), 0) return (self.ids[index], index, source, target) def collater(self, samples: List[Tuple[(str, int, torch.Tensor, torch.Tensor)]]): ids = [sample[0] for sample in samples] samples = [sample[1:] for sample in samples] output_dict = super().collater(samples) output_dict['utt_id'] = ids wavs = [] for i in range(output_dict['nsentences']): wav = output_dict['net_input']['src_tokens'][i] length = output_dict['net_input']['src_lengths'][i].item() wavs.append(wav[:length].numpy()) return (wavs, output_dict)
class DownstreamExpert(nn.Module): '\n Used to handle downstream-specific operations\n eg. downstream forward, metric computation, contents to log\n ' def __init__(self, upstream_dim, upstream_rate, downstream_expert, expdir, **kwargs): "\n Args:\n upstream_dim: int\n Different upstream will give different representation dimension\n You might want to first project them to the same dimension\n\n upstream_rate: int\n 160: for upstream with 10 ms per frame\n 320: for upstream with 20 ms per frame\n \n downstream_expert: dict\n The 'downstream_expert' field specified in your downstream config file\n eg. downstream/example/config.yaml\n\n expdir: string\n The expdir from command-line argument, you should save all results into\n this directory, like some logging files.\n\n **kwargs: dict\n All the arguments specified by the argparser in run_downstream.py\n and all the other fields in config.yaml, in case you need it.\n \n Note1. Feel free to add new argument for __init__ as long as it is\n a command-line argument or a config field. You can check the constructor\n code in downstream/runner.py\n " super(DownstreamExpert, self).__init__() print(downstream_expert) self.expdir = expdir self.src_lang = downstream_expert['src_lang'] self.tgt_lang = downstream_expert['tgt_lang'] self.post_process = downstream_expert['post_process'] self.output_prefix = downstream_expert['output_prefix'] self.upstream_rate = upstream_rate self.datarc = downstream_expert['datarc'] self.max_positions = downstream_expert['modelrc']['max_source_positions'] self.upstream_rate = downstream_expert.get('upstream_rate', upstream_rate) if (self.upstream_rate < 0): self.upstream_rate = upstream_rate assert ((self.upstream_rate % upstream_rate) == 0) self.downsample_ratio = int((self.upstream_rate / upstream_rate)) self.downsample_method = downstream_expert.get('downsample_method', 'drop') if (self.downsample_method == 'concat'): upstream_dim *= self.downsample_ratio self.task = S3prl_SpeechToTextTask.setup_task(Namespace(**downstream_expert['taskrc'])) self.task.upstream_rate = self.upstream_rate self.data_dir = downstream_expert['taskrc']['data'] self.criterion = self.task.build_criterion(Namespace(**downstream_expert['criterionrc'])) modelrc = Namespace(**downstream_expert['modelrc']) assert (modelrc.arch in fairseq.models.ARCH_CONFIG_REGISTRY) fairseq.models.ARCH_CONFIG_REGISTRY[modelrc.arch](modelrc) self.model = self.task.build_model(modelrc, upstream_dim) self.generator = self.task.build_generator([self.model], Namespace(**downstream_expert['generatorrc'])) self.batch_itr = {} self.use_asr = downstream_expert['taskrc']['use_asr'] if self.use_asr: rc = downstream_expert['asrrc'] self.asr_datarc = rc['datarc'] self.asr_weight = rc['weight'] self.asr_dict = Dictionary.load(f"{self.data_dir}/{rc['vocab_file']}") asr_bperc = rc['bpe_tokenizer'].copy() asr_bperc['sentencepiece_model'] = f"{self.data_dir}/{asr_bperc['sentencepiece_model']}" self.asr_bpe = encoders.build_bpe(Namespace(**asr_bperc)) self.asr_task = S3prl_SpeechToTextTask.setup_task(Namespace(**downstream_expert['taskrc'])) self.asr_dict.add_symbol('<blank>') self.asr_task.tgt_dict = self.asr_dict self.asr_head = nn.Linear(modelrc.encoder_embed_dim, len(self.asr_dict)) self.additional_dataset = {} self.register_buffer('best_score', torch.zeros(1)) def get_dataloader(self, split, epoch: int=0): "\n Args:\n mode: string\n 'train', 'dev' or 'test'\n\n Return:\n a torch.utils.data.DataLoader returning each batch in the format of:\n\n [wav1, wav2, ...], your_other_contents1, your_other_contents2, ...\n\n where wav1, wav2 ... are in variable length\n each wav is torch.FloatTensor in cpu with:\n 1. dim() == 1\n 2. sample_rate == 16000\n 3. directly loaded by torchaudio\n " data_split = self.datarc[split] if (data_split not in self.batch_itr): self.task.load_dataset(split=data_split, max_feature_len=self.max_positions) self.batch_itr[data_split] = self.task.get_batch_iterator(self.task.dataset(data_split), max_tokens=self.datarc['max_tokens'], max_positions=self.max_positions, num_workers=self.datarc['num_workers'], ignore_invalid_inputs=False, epoch=(epoch + 1)) return self.batch_itr[data_split].next_epoch_itr() def forward(self, mode, features, input_dict, records, **kwargs): "\n Args:\n mode: string\n 'train', 'dev' or 'test' for this forward step\n\n features:\n list of unpadded features [feat1, feat2, ...]\n each feat is in torch.FloatTensor and already\n put in the device assigned by command-line args\n\n your_other_contents1, ... :\n in the order defined by your dataloader (dataset + collate_fn)\n these are all in cpu, and you can move them to the same device\n as features\n\n records:\n defaultdict(list), by appending contents into records,\n these contents can be averaged and logged on Tensorboard\n later by self.log_records (also customized by you)\n\n Note1. downstream/runner.py will call self.log_records\n 1. every `log_step` during training\n 2. once after evalute the whole dev/test dataloader\n\n Note2. `log_step` is defined in your downstream config\n eg. downstream/example/config.yaml\n\n Return:\n loss:\n the loss to be optimized, should not be detached\n a single scalar in torch.FloatTensor\n " device = features[0].device features = self.downsample(features) features_length = torch.LongTensor([len(feature) for feature in features]) features = pad_sequence(features, batch_first=True, padding_value=0.0) input_dict['net_input']['src_tokens'] = features input_dict['net_input']['src_lengths'] = features_length input_dict = fairseq.utils.move_to_cuda(input_dict, device=device) if self.use_asr: asr_input_dict = self._create_asr_input_dict(input_dict, mode) asr_input_dict = fairseq.utils.move_to_cuda(asr_input_dict, device=device) loss = torch.FloatTensor(0) if (mode in ['train', 'dev']): encoder_out = self.model.encoder(src_tokens=input_dict['net_input']['src_tokens'], src_lengths=input_dict['net_input']['src_lengths']) st_decoder_out = self.model.decoder(prev_output_tokens=input_dict['net_input']['prev_output_tokens'], encoder_out=encoder_out) (st_loss, _) = self.criterion.compute_loss(self.model, st_decoder_out, input_dict) loss = st_loss if self.use_asr: asr_loss = self.count_asr_loss(encoder_out, asr_input_dict) loss = (((1 - self.asr_weight) * st_loss) + (self.asr_weight * asr_loss)) loss /= input_dict['nsentences'] records['loss'].append(loss.item()) if self.use_asr: records['st_loss'].append(st_loss.item()) records['asr_loss'].append(asr_loss.item()) if (mode in ['dev', 'test']): records['ids'] += input_dict['id'].cpu().tolist() records['utt_ids'] += input_dict['utt_id'] (hyps, refs) = self._inference_step(input_dict) records['hyps'] += hyps records['refs'] += refs if self.use_asr: (asr_hyps, asr_refs) = self._inference_step_asr(asr_input_dict) records['asr_hyps'] += asr_hyps records['asr_refs'] += asr_refs return loss def downsample(self, features): if (self.downsample_ratio == 1): return features new_features = [] for feature in features: if (self.downsample_method == 'drop'): feature = feature[::self.downsample_ratio] elif (self.downsample_method == 'concat'): N = (feature.size(0) % self.downsample_ratio) if (N != 0): feature = F.pad(feature, (0, 0, 0, (self.downsample_ratio - N))) feature = feature.view((feature.size(0) // self.downsample_ratio), (feature.size(1) * self.downsample_ratio)) elif (self.downsample_method == 'average'): N = (feature.size(0) % self.downsample_ratio) if (N != 0): feature = F.pad(feature, (0, 0, 0, (self.downsample_ratio - N))) feature = feature.view((feature.size(0) // self.downsample_ratio), self.downsample_ratio, feature.size(1)).mean(dim=1) else: raise NotImplementedError new_features.append(feature) return new_features def _create_asr_input_dict(self, input_dict, mode): if (mode not in self.additional_dataset): dataset = AdditionalDataset.from_tsv(f'{self.data_dir}/{self.datarc[mode]}.tsv', self.asr_datarc['key'], self.asr_dict, self.asr_bpe) self.additional_dataset[mode] = dataset additional_data = self.additional_dataset[mode].get_addtional_input(input_dict['id']) asr_input_dict = input_dict.copy() asr_input_dict['net_input'] = input_dict['net_input'].copy() asr_input_dict['net_input']['prev_output_tokens'] = additional_data['prev_output_tokens'] asr_input_dict['target'] = additional_data['target'] asr_input_dict['target_lengths'] = additional_data['target_lengths'] asr_input_dict['ntokens'] = additional_data['ntokens'] return asr_input_dict def count_asr_loss(self, encoder_out, input_dict): hidden = encoder_out['encoder_out'][0] log_prob = self.asr_head(hidden).log_softmax(2) hidden_length = self.model.encoder.subsample.get_out_seq_lens_tensor(input_dict['net_input']['src_lengths']) targets = input_dict['target'] target_lengths = input_dict['target_lengths'] loss = nn.functional.ctc_loss(log_prob, targets, hidden_length, target_lengths, blank=self.asr_dict.index('<blank>'), reduction='sum', zero_infinity=True) return loss def _decode(self, toks, dictionary): toks = toks[(toks != dictionary.pad())] s = dictionary.string(toks.int().cpu(), self.post_process) return (s if s else '<unk>') def _inference_step(self, input_dict): output = self.generator.generate([self.model], input_dict) hyps = [] refs = [] for i in range(len(output)): hyps.append(self._decode(output[i][0]['tokens'], self.task.target_dictionary)) refs.append(self._decode(input_dict['target'][i], self.task.target_dictionary)) return (hyps, refs) def _inference_step_asr(self, input_dict): encoder_out = self.model.encoder(src_tokens=input_dict['net_input']['src_tokens'], src_lengths=input_dict['net_input']['src_lengths']) hidden = encoder_out['encoder_out'][0] logit = self.asr_head(hidden) predict = logit.argmax(dim=(- 1)).transpose(0, 1) hyps = [] refs = [] for i in range(len(predict)): predict_ids = predict[i].unique_consecutive() predict_ids = predict_ids[(predict_ids != self.asr_dict.index('<blank>'))] hyps.append(self._decode(predict_ids, self.asr_dict)) refs.append(self._decode(input_dict['target'][i], self.asr_dict)) return (hyps, refs) def _metric(self, hyps, refs): tok = ('zh' if (self.tgt_lang == 'zh') else '13a') bleu = sacrebleu.corpus_bleu(hyps, [refs], tokenize=tok) return bleu def _asr_metric(self, hyps, refs): ce = 0 we = 0 c_total = 0 w_total = 0 for (hyp, ref) in zip(hyps, refs): normalized_hyp = hyp.translate(str.maketrans('', '', ''.join(list((set(string.punctuation) - set("'-")))))).lower() normalized_ref = ref.translate(str.maketrans('', '', ''.join(list((set(string.punctuation) - set("'-")))))).lower() ce += editdistance.eval(normalized_hyp, normalized_ref) c_total += len(normalized_ref) hyp_w = normalized_hyp.split() ref_w = normalized_ref.split() we += editdistance.eval(hyp_w, ref_w) w_total += len(ref_w) cer = (ce / c_total) wer = (we / w_total) return (cer, wer) def log_records(self, mode, records, logger, global_step, batch_ids, total_batch_num, **kwargs): "\n Args:\n mode: string\n 'train':\n records and batchids contain contents for `log_step` batches\n `log_step` is defined in your downstream config\n eg. downstream/example/config.yaml\n\n 'dev' or 'test' :\n records and batchids contain contents for the entire evaluation dataset\n\n records:\n defaultdict(list), contents already prepared by self.forward\n\n logger:\n Tensorboard SummaryWriter\n please use f'{your_task_name}/{mode}-{key}' as key name to log your contents,\n preventing conflict with the logging of other tasks\n\n global_step:\n The global_step when training, which is helpful for Tensorboard logging\n\n batch_ids:\n The batches contained in records when enumerating over the dataloader\n\n total_batch_num:\n The total amount of batches in the dataloader\n \n Return:\n a list of string\n Each string is a filename we wish to use to save the current model\n according to the evaluation result, like the best.ckpt on the dev set\n You can return nothing or an empty list when no need to save the checkpoint\n " save_names = [] if (mode in ['train', 'dev']): ave_loss = (sum(records['loss']) / len(records['loss'])) logger.add_scalar(f'st/{mode}-loss', ave_loss, global_step=global_step) if self.use_asr: ave_st_loss = (sum(records['st_loss']) / len(records['st_loss'])) logger.add_scalar(f'st/{mode}-st_loss', ave_st_loss, global_step=global_step) ave_asr_loss = (sum(records['asr_loss']) / len(records['asr_loss'])) logger.add_scalar(f'st/{mode}-asr_loss', ave_asr_loss, global_step=global_step) if (mode in ['dev', 'test']): bleu = self._metric(records['hyps'], records['refs']) logger.add_scalar(f'st/{mode}-bleu', bleu.score, global_step=global_step) for i in range(4): logger.add_scalar(f'st/{mode}-bleu{(i + 1)}', bleu.precisions[i], global_step=global_step) if ((bleu.score > self.best_score) and (mode == 'dev')): self.best_score = (torch.ones(1) * bleu.score) save_names.append(f'{mode}-best.ckpt') with open(f'{self.expdir}/{self.output_prefix}-st-{mode}.tsv', 'w') as f: print('utt_id', 'hyp', 'ref', sep='\t', file=f) results = list(zip(records['ids'], records['hyps'], records['refs'], records['utt_ids'])) results.sort(key=(lambda x: x[0])) for (idx, hyp, ref, utt_id) in results: print(utt_id, hyp, ref, sep='\t', file=f) print(bleu) if self.use_asr: (cer, wer) = self._asr_metric(records['asr_hyps'], records['asr_refs']) logger.add_scalar(f'st/{mode}-asr-cer', cer, global_step=global_step) logger.add_scalar(f'st/{mode}-asr-wer', wer, global_step=global_step) with open(f'{self.expdir}/{self.output_prefix}-asr-{mode}.tsv', 'w') as f: print('utt_id', 'hyp', 'ref', sep='\t', file=f) results = list(zip(records['ids'], records['asr_hyps'], records['asr_refs'], records['utt_ids'])) results.sort(key=(lambda x: x[0])) for (idx, hyp, ref, utt_id) in results: print(utt_id, hyp, ref, sep='\t', file=f) tqdm.write(f'[cer]:{cer}, [wer]:{wer}') return save_names
def verbose(args, text): if args.verbose: print(text)
def length(s): return len(s.split())
def verbose(args, text): if args.verbose: print(text)
def create_sentencepiece(filenames, model_type, vocab_size, output_prefix): sp.SentencePieceTrainer.train(input=','.join(filenames), model_prefix=output_prefix, vocab_size=vocab_size, model_type=model_type, character_coverage=1.0, unk_id=UNK_TOKEN_ID, bos_id=BOS_TOKEN_ID, eos_id=EOS_TOKEN_ID, pad_id=PAD_TOKEN_ID) spm = sp.SentencePieceProcessor(model_file=f'{output_prefix}.model') vocab = {i: spm.IdToPiece(i) for i in range(spm.GetPieceSize())} assert (vocab.get(UNK_TOKEN_ID) == UNK_TOKEN) assert (vocab.get(BOS_TOKEN_ID) == BOS_TOKEN) assert (vocab.get(EOS_TOKEN_ID) == EOS_TOKEN) assert (vocab.get(PAD_TOKEN_ID) == PAD_TOKEN) vocab = {i: s for (i, s) in vocab.items() if (s not in {UNK_TOKEN, BOS_TOKEN, EOS_TOKEN, PAD_TOKEN})} with open(f'{output_prefix}.txt', 'w') as f: for (_, s) in sorted(vocab.items(), key=(lambda x: x[0])): print(f'{s} 1', file=f)
def verbose(args, text): if args.verbose: print(text)
class SpeakerVerifi_train(Dataset): def __init__(self, vad_config, key_list, file_path, meta_data, max_timestep=None, n_jobs=12): self.roots = file_path self.root_key = key_list self.max_timestep = max_timestep self.vad_c = vad_config self.dataset = [] self.all_speakers = [] for index in range(len(self.root_key)): cache_path = ((Path(os.path.dirname(__file__)) / '.wav_lengths') / f'{self.root_key[index]}_length.pt') cache_path.parent.mkdir(exist_ok=True) root = Path(self.roots[index]) if (not cache_path.is_file()): def trimmed_length(path): (wav_sample, _) = apply_effects_file(path, EFFECTS) wav_sample = wav_sample.squeeze(0) length = wav_sample.shape[0] return length wav_paths = find_files(root) wav_lengths = Parallel(n_jobs=n_jobs)((delayed(trimmed_length)(path) for path in tqdm.tqdm(wav_paths, desc='Preprocessing'))) wav_tags = [Path(path).parts[(- 3):] for path in wav_paths] torch.save([wav_tags, wav_lengths], str(cache_path)) else: (wav_tags, wav_lengths) = torch.load(str(cache_path)) wav_paths = [root.joinpath(*tag) for tag in wav_tags] speaker_dirs = [f.stem for f in root.iterdir() if f.is_dir()] self.all_speakers.extend(speaker_dirs) for (path, length) in zip(wav_paths, wav_lengths): if (length > self.vad_c['min_sec']): self.dataset.append(path) self.all_speakers.sort() self.speaker_num = len(self.all_speakers) def __len__(self): return len(self.dataset) def __getitem__(self, idx): path = self.dataset[idx] (wav, _) = apply_effects_file(str(path), EFFECTS) wav = wav.squeeze(0) length = wav.shape[0] if (self.max_timestep != None): if (length > self.max_timestep): start = random.randint(0, int((length - self.max_timestep))) wav = wav[start:(start + self.max_timestep)] tags = Path(path).parts[(- 3):] utterance_id = '-'.join(tags).replace('.wav', '') label = self.all_speakers.index(tags[0]) return (wav.numpy(), utterance_id, label) def collate_fn(self, samples): return zip(*samples)
class SpeakerVerifi_test(Dataset): def __init__(self, vad_config, file_path, meta_data): self.root = file_path self.meta_data = meta_data self.necessary_dict = self.processing() self.vad_c = vad_config self.dataset = self.necessary_dict['spk_paths'] self.pair_table = self.necessary_dict['pair_table'] def processing(self): pair_table = [] spk_paths = set() with open(self.meta_data, 'r') as f: usage_list = f.readlines() for pair in usage_list: list_pair = pair.split() pair_1 = os.path.join(self.root, list_pair[1]) pair_2 = os.path.join(self.root, list_pair[2]) spk_paths.add(pair_1) spk_paths.add(pair_2) one_pair = [list_pair[0], pair_1, pair_2] pair_table.append(one_pair) return {'spk_paths': list(spk_paths), 'total_spk_num': None, 'pair_table': pair_table} def __len__(self): return len(self.necessary_dict['spk_paths']) def __getitem__(self, idx): x_path = self.dataset[idx] x_name = x_path (wav, _) = apply_effects_file(x_path, EFFECTS) wav = wav.squeeze(0) return (wav.numpy(), x_name) def collate_fn(self, data_sample): (wavs, x_names) = zip(*data_sample) return (wavs, x_names)
class DownstreamExpert(nn.Module): '\n Used to handle downstream-specific operations\n eg. downstream forward, metric computation, contents to log\n\n Note 1.\n dataloaders should output in the following format:\n\n [[wav1, wav2, ...], your_other_contents, ...]\n\n where wav1, wav2 ... are in variable length\n and wav1 is in torch.FloatTensor\n ' def __init__(self, upstream_dim, downstream_expert, expdir, **kwargs): super(DownstreamExpert, self).__init__() self.upstream_dim = upstream_dim self.downstream = downstream_expert self.datarc = downstream_expert['datarc'] self.modelrc = downstream_expert['modelrc'] self.expdir = expdir train_file_path = ((Path(self.datarc['file_path']) / 'dev') / 'wav') test_file_path = ((Path(self.datarc['file_path']) / 'test') / 'wav') train_config = {'vad_config': self.datarc['vad_config'], 'file_path': [train_file_path], 'key_list': ['Voxceleb1'], 'meta_data': self.datarc['train_meta_data'], 'max_timestep': self.datarc['max_timestep']} self.train_dataset = SpeakerVerifi_train(**train_config) dev_config = {'vad_config': self.datarc['vad_config'], 'file_path': train_file_path, 'meta_data': self.datarc['dev_meta_data']} self.dev_dataset = SpeakerVerifi_test(**dev_config) test_config = {'vad_config': self.datarc['vad_config'], 'file_path': test_file_path, 'meta_data': self.datarc['test_meta_data']} self.test_dataset = SpeakerVerifi_test(**test_config) self.connector = nn.Linear(self.upstream_dim, self.modelrc['input_dim']) agg_dim = self.modelrc['module_config'][self.modelrc['module']].get('agg_dim', self.modelrc['input_dim']) ModelConfig = {'input_dim': self.modelrc['input_dim'], 'agg_dim': agg_dim, 'agg_module_name': self.modelrc['agg_module'], 'module_name': self.modelrc['module'], 'hparams': self.modelrc['module_config'][self.modelrc['module']], 'utterance_module_name': self.modelrc['utter_module']} self.model = Model(**ModelConfig) objective_config = {'speaker_num': self.train_dataset.speaker_num, 'hidden_dim': self.modelrc['input_dim'], **self.modelrc['LossConfig'][self.modelrc['ObjectiveLoss']]} self.objective = eval(self.modelrc['ObjectiveLoss'])(**objective_config) self.score_fn = nn.CosineSimilarity(dim=(- 1)) self.eval_metric = EER self.register_buffer('best_score', (torch.ones(1) * 100)) def get_dataloader(self, mode): "\n Args:\n mode: string\n 'train', 'dev' or 'test'\n\n Return:\n a torch.utils.data.DataLoader returning each batch in the format of:\n\n [wav1, wav2, ...], your_other_contents1, your_other_contents2, ...\n\n where wav1, wav2 ... are in variable length\n each wav is torch.FloatTensor in cpu with:\n 1. dim() == 1\n 2. sample_rate == 16000\n 3. directly loaded by torchaudio\n " if (mode == 'train'): return self._get_train_dataloader(self.train_dataset) elif (mode == 'dev'): return self._get_eval_dataloader(self.dev_dataset) elif (mode == 'test'): return self._get_eval_dataloader(self.test_dataset) def _get_train_dataloader(self, dataset): sampler = (DistributedSampler(dataset) if is_initialized() else None) return DataLoader(dataset, batch_size=self.datarc['train_batch_size'], shuffle=(sampler is None), sampler=sampler, num_workers=self.datarc['num_workers'], collate_fn=dataset.collate_fn) def _get_eval_dataloader(self, dataset): return DataLoader(dataset, batch_size=self.datarc['eval_batch_size'], shuffle=False, num_workers=self.datarc['num_workers'], collate_fn=dataset.collate_fn) def get_train_dataloader(self): return self._get_train_dataloader(self.train_dataset) def get_dev_dataloader(self): return self._get_eval_dataloader(self.dev_dataset) def get_test_dataloader(self): return self._get_eval_dataloader(self.test_dataset) def forward(self, mode, features, utter_idx, labels=None, records=None, **kwargs): '\n Args:\n features:\n the features extracted by upstream\n put in the device assigned by command-line args\n\n labels:\n the speaker labels\n\n records:\n defaultdict(list), by appending scalars into records,\n these scalars will be averaged and logged on Tensorboard\n\n logger:\n Tensorboard SummaryWriter, given here for logging/debugging\n convenience, please use "self.downstream/your_content_name" as key\n name to log your customized contents\n\n global_step:\n global_step in runner, which is helpful for Tensorboard logging\n\n Return:\n loss:\n the loss to be optimized, should not be detached\n ' features_pad = pad_sequence(features, batch_first=True) if (self.modelrc['module'] == 'XVector'): attention_mask = [torch.ones((feature.shape[0] - 14)) for feature in features] else: attention_mask = [torch.ones(feature.shape[0]) for feature in features] attention_mask_pad = pad_sequence(attention_mask, batch_first=True) attention_mask_pad = ((1.0 - attention_mask_pad) * (- 100000.0)) features_pad = self.connector(features_pad) if (mode == 'train'): agg_vec = self.model(features_pad, attention_mask_pad.cuda()) labels = torch.LongTensor(labels).to(features_pad.device) loss = self.objective(agg_vec, labels) records['loss'].append(loss.item()) return loss elif (mode in ['dev', 'test']): agg_vec = self.model.inference(features_pad, attention_mask_pad.cuda()) agg_vec = torch.nn.functional.normalize(agg_vec, dim=(- 1)) utt_name = utter_idx for idx in range(len(agg_vec)): records[utt_name[idx]] = agg_vec[idx].cpu().detach() return torch.tensor(0) def log_records(self, mode, records, logger, global_step, **kwargs): "\n Args:\n records:\n defaultdict(list), contents already appended\n\n logger:\n Tensorboard SummaryWriter\n please use f'{prefix}your_content_name' as key name\n to log your customized contents\n\n prefix:\n used to indicate downstream and train/test on Tensorboard\n eg. 'phone/train-'\n\n global_step:\n global_step in runner, which is helpful for Tensorboard logging\n " save_names = [] if (mode == 'train'): loss = torch.FloatTensor(records['loss']).mean().item() logger.add_scalar(f'sv-voxceleb1/{mode}-loss', loss, global_step=global_step) print(f'sv-voxceleb1/{mode}-loss: {loss}') elif (mode in ['dev', 'test']): trials = self.test_dataset.pair_table labels = [] scores = [] def names(name): return '-'.join(name.split('/')[(- 3):]).split('.')[0] for (label, name1, name2) in trials: labels.append(label) score = self.score_fn(records[name1], records[name2]).numpy() records['scores'].extend([score]) records['pair_names'].extend([f'{names(name1)}_{names(name2)}']) scores.append(score) (eer, *others) = self.eval_metric(np.array(labels, dtype=int), np.array(scores)) logger.add_scalar(f'sv-voxceleb1/{mode}-EER', eer, global_step=global_step) print(f'sv-voxceleb1/{mode}-EER: {eer}') if ((eer < self.best_score) and (mode == 'dev')): self.best_score = (torch.ones(1) * eer) save_names.append(f'{mode}-best.ckpt') with open((Path(self.expdir) / f'{mode}_predict.txt'), 'w') as file: line = [f'''{name} {score} ''' for (name, score) in zip(records['pair_names'], records['scores'])] file.writelines(line) with open((Path(self.expdir) / f'{mode}_truth.txt'), 'w') as file: line = [f'''{name} {score} ''' for (name, score) in zip(records['pair_names'], records['labels'])] file.writelines(line) return save_names def separate_data(self, agg_vec): assert ((len(agg_vec) % 2) == 0) total_num = (len(agg_vec) // 2) feature1 = agg_vec[:total_num] feature2 = agg_vec[total_num:] return (feature1, feature2)
def collect_speaker_ids(roots, speaker_num): all_speaker = [] all_speaker.extend([f.path for f in os.scandir(roots) if f.is_dir()]) ids = [[speaker.split('/')[(- 3)], speaker.split('/')[(- 1)]] for speaker in all_speaker] vox1 = [] for id in ids: if (id[0] == roots.split('/')[(- 2)]): vox1.append(id[1]) dev_speaker = random.sample(vox1, k=speaker_num) vox1_train = [ids for ids in vox1 if (ids not in dev_speaker)] train_speaker = [] train_speaker.extend(vox1_train) return (train_speaker, dev_speaker)
def construct_dev_speaker_id_txt(dev_speakers, dev_txt_name): f = open(dev_txt_name, 'w') for dev in dev_speakers: f.write(dev) f.write('\n') f.close() return
def sample_wavs_and_dump_txt(root, dev_ids, numbers, meta_data_name): wav_list = [] count_positive = 0 print(f'generate {numbers} sample pairs') for _ in trange(numbers): prob = random.random() if (prob > 0.5): dev_id_pair = random.sample(dev_ids, 2) sample1 = '/'.join(random.choice(find_files(os.path.join(root, dev_id_pair[0]))).split('/')[(- 3):]) sample2 = '/'.join(random.choice(find_files(os.path.join(root, dev_id_pair[1]))).split('/')[(- 3):]) label = '0' wav_list.append(' '.join([label, sample1, sample2])) else: dev_id_pair = random.sample(dev_ids, 1) sample1 = '/'.join(random.choice(find_files(os.path.join(root, dev_id_pair[0]))).split('/')[(- 3):]) sample2 = '/'.join(random.choice(find_files(os.path.join(root, dev_id_pair[0]))).split('/')[(- 3):]) label = '1' count_positive += 1 wav_list.append(' '.join([label, sample1, sample2])) print('finish, then dump file ..') f = open(meta_data_name, 'w') for data in wav_list: f.write((data + '\n')) f.close() return wav_list
def EER(labels, scores): '\n labels: (N,1) value: 0,1\n\n scores: (N,1) value: -1 ~ 1\n\n ' (fpr, tpr, thresholds) = roc_curve(labels, scores) s = interp1d(fpr, tpr) a = (lambda x: ((1.0 - x) - interp1d(fpr, tpr)(x))) eer = brentq(a, 0.0, 1.0) thresh = interp1d(fpr, thresholds)(eer) return (eer, thresh)
def eer_yist_f(labels, scores): '\n Args:\n labels: (N,1) with value being 0 or 1\n scores: (N,1) within [-1, 1]\n\n Returns:\n equal_error_rates\n threshold\n ' joints = sorted(zip(scores, labels), key=(lambda x: x[0])) (sorted_scores, sorted_labels) = zip(*joints) total_ones = sum(sorted_labels) total_zeros = (len(sorted_labels) - total_ones) prefsum_ones = list(accumulate(sorted_labels, partial(_count_labels, label_to_count=1), initial=0)) prefsum_zeros = list(accumulate(sorted_labels, partial(_count_labels, label_to_count=0), initial=0)) ext_scores = [(- 1.0), *sorted_scores, 1.0] (thresh_left, thresh_right) = (0, len(ext_scores)) while True: if (thresh_left == thresh_right): break thresh_idx = ((thresh_left + thresh_right) // 2) nb_false_positives = (total_zeros - prefsum_zeros[thresh_idx]) nb_false_negatives = prefsum_ones[thresh_idx] if (nb_false_positives > nb_false_negatives): thresh_left = thresh_idx elif (nb_false_positives < nb_false_negatives): thresh_right = thresh_idx else: break thresh = ((ext_scores[thresh_idx] + ext_scores[(thresh_idx + 1)]) / 2) false_negative_ratio = (nb_false_negatives / len(labels)) false_positive_ratio = (nb_false_positives / len(labels)) equal_error_rate = ((false_positive_ratio + false_negative_ratio) / 2) return (equal_error_rate, thresh)
def _count_labels(counted_so_far, label, label_to_count=0): return ((counted_so_far + 1) if (label == label_to_count) else counted_so_far)
def compute_metrics(input_x_speaker, ylabel): wav1 = [] wav2 = [] for i in range(len(ylabel)): wav1.append(input_x_speaker[i].unsqueeze(0)) wav2.append(input_x_speaker[(len(ylabel) + i)].unsqueeze(0)) wav1 = torch.stack(wav1) wav2 = torch.stack(wav2) ylabel = torch.stack(ylabel).cpu().detach().long().tolist() scores = self.score_fn(wav1, wav2).squeeze().cpu().detach().tolist() return (scores, ylabel)
class DownstreamExpert(nn.Module): '\n Used to handle downstream-specific operations\n eg. downstream forward, metric computation, contents to log\n ' def __init__(self, upstream_dim: int, downstream_expert: dict, expdir: str, **kwargs): super(DownstreamExpert, self).__init__() self.upstream_dim = upstream_dim self.datarc = downstream_expert['datarc'] self.modelrc = downstream_expert['modelrc'] self.lossrc = downstream_expert['lossrc'] self.expdir = Path(expdir) self.train_dataset = None self.valid_dataset = None self.test_dataset = None self.model = Model(input_dim=upstream_dim, **self.modelrc) self.objective = nn.CosineEmbeddingLoss(**self.lossrc) def get_dataloader(self, mode): if (mode == 'train'): self.train_dataset = SWS2013Dataset('dev', **self.datarc) self.valid_dataset = SWS2013Dataset('eval', **self.datarc) return DataLoader(self.train_dataset, sampler=WeightedRandomSampler(weights=self.train_dataset.sample_weights, num_samples=len(self.train_dataset.sample_weights), replacement=True), batch_size=self.datarc['batch_size'], drop_last=True, num_workers=self.datarc['num_workers'], collate_fn=self.train_dataset.collate_fn) if (mode == 'valid'): return DataLoader(self.valid_dataset, sampler=WeightedRandomSampler(weights=self.valid_dataset.sample_weights, num_samples=self.datarc['valid_size'], replacement=True), batch_size=self.datarc['batch_size'], drop_last=True, num_workers=self.datarc['num_workers'], collate_fn=self.valid_dataset.collate_fn) if (mode in ['dev', 'eval']): self.test_dataset = QUESST14Dataset(mode, **self.datarc) return DataLoader(self.test_dataset, shuffle=False, batch_size=self.datarc['batch_size'], drop_last=False, num_workers=self.datarc['num_workers'], collate_fn=self.test_dataset.collate_fn) if (mode == 'sws2013_eval'): self.test_dataset = SWS2013Testset('eval', **self.datarc) return DataLoader(self.test_dataset, shuffle=False, batch_size=self.datarc['batch_size'], drop_last=False, num_workers=self.datarc['num_workers'], collate_fn=self.test_dataset.collate_fn) raise NotImplementedError def forward(self, mode, features, labels, records, **kwargs): if (mode in ['train', 'valid']): audio_tensors = torch.stack(features[:(len(features) // 2)]) query_tensors = torch.stack(features[(len(features) // 2):]) labels = torch.cat(labels).to(audio_tensors.device) audio_embs = self.model(audio_tensors) query_embs = self.model(query_tensors) loss = self.objective(audio_embs, query_embs, labels) records['loss'].append(loss.item()) with torch.no_grad(): similarities = F.cosine_similarity(audio_embs, query_embs) records['similarity-positive'] += similarities[(labels > 0)].tolist() records['similarity-negative'] += similarities[(labels < 0)].tolist() return loss elif (mode in ['dev', 'eval', 'sws2013_eval']): audio_tensors = torch.stack(features) (lengths, audio_names) = labels embs = self.model(audio_tensors) embs = embs.detach().cpu() offset = 0 for (length, audio_name) in zip(lengths, audio_names): records['embs'].append(embs[offset:(offset + length)]) records['audio_names'].append(audio_name) offset += length else: raise NotImplementedError def log_records(self, mode, records, logger, global_step, **kwargs): 'Log training, validation information or test on a dataset.' if (mode in ['train', 'valid']): prefix = f'sws2013/{mode}' for (key, val) in records.items(): average = (sum(val) / len(val)) logger.add_scalar(f'{prefix}-{key}', average, global_step=global_step) elif (mode in ['dev', 'eval', 'sws2013_eval']): query_embs = records['embs'][:self.test_dataset.n_queries] doc_embs = records['embs'][self.test_dataset.n_queries:] query_names = records['audio_names'][:self.test_dataset.n_queries] doc_names = records['audio_names'][self.test_dataset.n_queries:] results = {} for (query_emb, query_name) in zip(tqdm(query_embs, desc='Query', ncols=0), query_names): query_emb = query_emb[0:1].cuda() scores = [] for (doc_emb, doc_name) in zip(tqdm(doc_embs, desc='Doc', ncols=0, leave=False), doc_names): with torch.no_grad(): doc_emb = doc_emb.cuda() similarities = F.cosine_similarity(query_emb, doc_emb) score = similarities.max().detach().cpu() scores.append(score) scores = torch.stack(scores) scores = ((scores - scores.mean()) / (scores.std() + 1e-06)) results[query_name] = list(zip(doc_names, scores.tolist())) score_thresh = 0 root = etree.Element('stdlist', termlist_filename='benchmark.stdlist.xml', indexing_time='1.00', language='english', index_size='1', system_id='benchmark') for (query_name, doc_scores) in results.items(): term_list = etree.SubElement(root, 'detected_termlist', termid=query_name, term_search_time='1.0', oov_term_count='1') for (doc_name, score) in doc_scores: etree.SubElement(term_list, 'term', file=doc_name, channel='1', tbeg='0.000', dur='0.00', score=f'{score:.4f}', decision=('YES' if (score > score_thresh) else 'NO')) tree = etree.ElementTree(root) tree.write(str((self.expdir / 'benchmark.stdlist.xml')), encoding='UTF-8', pretty_print=True) else: raise NotImplementedError
class Model(nn.Module): def __init__(self, input_dim, bottleneck_dim, hidden_dim, **kwargs): super(Model, self).__init__() self.connector = nn.Linear(input_dim, bottleneck_dim) self.fc1 = nn.Linear(bottleneck_dim, hidden_dim) self.attention_linear = nn.Linear(hidden_dim, 1) def forward(self, features): hiddens = F.relu(self.connector(features)) hiddens = torch.tanh(self.fc1(hiddens)) attention_weights = F.softmax(self.attention_linear(hiddens), dim=1) embeds = torch.sum((hiddens * attention_weights), dim=1) return embeds
class QUESST14Dataset(Dataset): 'QUESST 2014 dataset (English-only).' def __init__(self, split, **kwargs): assert (split in ['dev', 'eval']) dataset_root = Path(kwargs['quesst2014_root']) doc_paths = get_audio_paths(dataset_root, 'language_key_utterances.lst') query_paths = get_audio_paths(dataset_root, f'language_key_{split}.lst') self.dataset_root = dataset_root self.n_queries = len(query_paths) self.n_docs = len(doc_paths) self.data = (query_paths + doc_paths) def __len__(self): return len(self.data) def __getitem__(self, idx): audio_path = self.data[idx] (wav, _) = apply_effects_file(str(audio_path), [['channels', '1'], ['rate', '16000'], ['norm'], ['vad', '-T', '0.25', '-p', '0.1'], ['reverse'], ['vad', '-T', '0.25', '-p', '0.1'], ['reverse'], ['pad', '0', '3']]) segments = wav.squeeze(0).unfold(0, 48000, 12000).unbind(0) return (segments, len(segments), audio_path.with_suffix('').name) def collate_fn(self, samples): 'Collate a mini-batch of data.' (segments, lengths, audio_names) = zip(*samples) segments = [seg for segs in segments for seg in segs] return (segments, (lengths, audio_names))
def get_audio_paths(dataset_root_path, lst_name): 'Extract audio paths.' audio_paths = [] with open(((dataset_root_path / 'scoring') / lst_name)) as f: for line in f: (audio_path, lang) = tuple(line.strip().split()) audio_path = re.sub('^.*?\\/', '', audio_path) audio_paths.append((dataset_root_path / audio_path)) return audio_paths
class SWS2013Dataset(Dataset): 'SWS 2013 dataset.' def __init__(self, split, **kwargs): assert (split in ['dev', 'eval']) dataset_root = Path(kwargs['sws2013_root']) split_root = (Path(kwargs['sws2013_scoring_root']) / f'sws2013_{split}') audio2dur = parse_ecf((split_root / 'sws2013.ecf.xml')) query2audios = parse_rttm((split_root / f'sws2013_{split}.rttm')) query2tensors = find_queries((dataset_root / f'{split}_queries')) print(f'[SWS2013] # of audios: {len(audio2dur)}') print(f'[SWS2013] # of queries: {len(query2tensors)}') all_audio_set = set(audio2dur.keys()) query2audio_set = {query: set((audio_info['audio'] for audio_info in audio_infos)) for (query, audio_infos) in query2audios.items()} query2audio_compl_set = {query: (all_audio_set - audio_set) for (query, audio_set) in query2audio_set.items()} (positive_pairs, negative_pairs) = ([], []) for (query, tensors) in query2tensors.items(): for query_tensor in tensors: negative_pairs.append({'query_tensor': query_tensor, 'audio_set': (query2audio_compl_set[query] if (query in query2audio_compl_set) else all_audio_set)}) if (query not in query2audios): continue for audio_info in query2audios[query]: positive_pairs.append({'query_tensor': query_tensor, 'audio': audio_info['audio'], 'offset': audio_info['offset'], 'duration': audio_info['duration']}) print(f'[SWS2013] # of positive pairs: {len(positive_pairs)}') self.audio_dir = (dataset_root / 'Audio') self.audio2dur = audio2dur self.max_dur = 3.0 self.positive_pairs = positive_pairs self.negative_pairs = negative_pairs def __len__(self): return (len(self.positive_pairs) + len(self.negative_pairs)) def __getitem__(self, idx): if (idx < len(self.positive_pairs)): pair = self.positive_pairs[idx] audio_path = (self.audio_dir / pair['audio']).with_suffix('.wav') audio_tensor = path2segment(audio_path, pair['duration'], self.max_dur, pair['offset']) else: pair = self.negative_pairs[(idx - len(self.positive_pairs))] sample_audio = random.sample(pair['audio_set'], 1)[0] audio_dur = self.audio2dur[sample_audio] audio_path = (self.audio_dir / sample_audio).with_suffix('.wav') audio_tensor = path2segment(audio_path, audio_dur, self.max_dur, 0.0) audio_tensor = audio_tensor.squeeze(0) query_tensor = tensor2segment(pair['query_tensor'], self.max_dur) label = torch.LongTensor([(1 if (idx < len(self.positive_pairs)) else (- 1))]) return (audio_tensor, query_tensor, label) def collate_fn(self, samples): 'Collate a mini-batch of data.' (audio_tensors, query_tensors, labels) = zip(*samples) return ((audio_tensors + query_tensors), labels) @property def sample_weights(self): 'Sample weights to balance positive and negative data.' n_pos = len(self.positive_pairs) n_neg = len(self.negative_pairs) return (([(1 / n_pos)] * n_pos) + ([(1 / n_neg)] * n_neg))
def parse_rttm(rttm_path): 'Parse audio and query pairs from *.rttm.' pattern = re.compile('LEXEME\\s+(sws2013_[0-9]+).*?([0-9]\\.[0-9]+)\\s+([0-9]\\.[0-9]+)\\s+(sws2013_(dev|eval)_[0-9]+)') query2audios = defaultdict(list) with open(rttm_path) as fd: for line in fd: match = pattern.match(line) if (match is None): continue query2audios[match.group(4)].append({'audio': match.group(1), 'offset': float(match.group(2)), 'duration': float(match.group(3))}) return query2audios
def parse_ecf(ecf_path): 'Find audios from sws2013.ecf.xml.' root = ET.parse(str(ecf_path)).getroot() audio2dur = {} for excerpt in root.findall('excerpt'): audio_name = excerpt.attrib['audio_filename'].replace('Audio/', '').replace('.wav', '') duration = float(excerpt.attrib['dur']) audio2dur[audio_name] = duration return audio2dur
def find_queries(query_dir_path): 'Find all queries under sws2013_dev & sws2013_eval.' pattern = re.compile('(_[0-9]{2})?\\.wav') query2tensors = defaultdict(list) for query_path in tqdm(list(query_dir_path.glob('*.wav')), ncols=0, desc='Load queries'): query_name = pattern.sub('', query_path.name) (wav_tensor, sample_rate) = apply_effects_file(str(query_path), [['channels', '1'], ['rate', '16000'], ['norm']]) (trimmed, _) = apply_effects_tensor(wav_tensor, sample_rate, [['vad', '-T', '0.25', '-p', '0.1'], ['reverse'], ['vad', '-T', '0.25', '-p', '0.1'], ['reverse']]) wav_tensor = (trimmed if (trimmed.size(1) >= (sample_rate * 0.5)) else wav_tensor) wav_tensor = wav_tensor.squeeze(0) query2tensors[query_name].append(wav_tensor) return query2tensors
def path2segment(filepath, src_dur, tgt_dur, offset): random_shift = random.uniform(0, (src_dur - tgt_dur)) (audio_tensor, _) = apply_effects_file(str(filepath), [['channels', '1'], ['rate', '16000'], ['norm'], ['pad', f'{tgt_dur}', f'{tgt_dur}'], ['trim', f'{((tgt_dur + offset) + random_shift)}', f'{tgt_dur}']]) return audio_tensor