linter and test updates for speaker_encoder, gmm_Attention

layers/common_layers.py

@@ -136,6 +136,8 @@ class GravesAttention(nn.Module):
         self.attention_weights = torch.zeros(inputs.shape[0], inputs.shape[1]).to(inputs.device)
         self.mu_prev = torch.zeros(inputs.shape[0], self.K).to(inputs.device)
 
+    # pylint: disable=R0201
+    # pylint: disable=unused-argument
     def preprocess_inputs(self, inputs):
         return None
 
@@ -376,8 +378,7 @@ def init_attn(attn_type, query_dim, embedding_dim, attention_dim,
                                  attention_location_kernel_size, windowing,
                                  norm, forward_attn, trans_agent,
                                  forward_attn_mask)
-    elif attn_type == "graves":
+    if attn_type == "graves":
         return GravesAttention(query_dim, attn_K)
-    else:
     raise RuntimeError(
         " [!] Given Attention Type '{attn_type}' is not exist.")

models/tacotron2.py

@@ -27,6 +27,7 @@ class Tacotron2(nn.Module):
                  separate_stopnet=True,
                  bidirectional_decoder=False):
         super(Tacotron2, self).__init__()
+        self.postnet_output_dim = postnet_output_dim
         self.decoder_output_dim = decoder_output_dim
         self.n_frames_per_step = r
         self.bidirectional_decoder = bidirectional_decoder
@@ -50,7 +51,7 @@ class Tacotron2(nn.Module):
                                location_attn, attn_K, separate_stopnet, proj_speaker_dim)
         if self.bidirectional_decoder:
             self.decoder_backward = copy.deepcopy(self.decoder)
-        self.postnet = Postnet(self.decoder_output_dim)
+        self.postnet = Postnet(self.postnet_output_dim)
 
     def _init_states(self):
         self.speaker_embeddings = None

speaker_encoder/compute_embeddings.py

@@ -6,51 +6,41 @@ import numpy as np
 from tqdm import tqdm
 
 import torch
-from torch.utils.data import DataLoader
-from TTS.datasets.preprocess import get_preprocessor_by_name
-from TTS.speaker_encoder.dataset import MyDataset
 from TTS.speaker_encoder.model import SpeakerEncoder
-from TTS.speaker_encoder.visual import plot_embeddings
 from TTS.utils.audio import AudioProcessor
 from TTS.utils.generic_utils import load_config
 
 parser = argparse.ArgumentParser(
-    description='Compute embedding vectors for each wav file in a dataset. ')
+    description="Compute embedding vectors for each wav file in a dataset. "
-parser.add_argument(
-    'model_path',
-    type=str,
-    help='Path to model outputs (checkpoint, tensorboard etc.).')
-parser.add_argument(
-    'config_path',
-    type=str,
-    help='Path to config file for training.',
 )
 parser.add_argument(
-    'data_path',
+    "model_path", type=str, help="Path to model outputs (checkpoint, tensorboard etc.)."
-    type=str,
-    help='Defines the data path. It overwrites config.json.')
-parser.add_argument(
-    'output_path',
-    type=str,
-    help='path for training outputs.')
-parser.add_argument(
-    '--use_cuda', type=bool, help='flag to set cuda.', default=False
 )
+parser.add_argument(
+    "config_path", type=str, help="Path to config file for training.",
+)
+parser.add_argument(
+    "data_path", type=str, help="Defines the data path. It overwrites config.json."
+)
+parser.add_argument("output_path", type=str, help="path for training outputs.")
+parser.add_argument("--use_cuda", type=bool, help="flag to set cuda.", default=False)
 args = parser.parse_args()
 
 
 c = load_config(args.config_path)
-ap = AudioProcessor(**c['audio'])
+ap = AudioProcessor(**c["audio"])
 
-wav_files = glob.glob(args.data_path + '/**/*.wav', recursive=True)
+wav_files = glob.glob(args.data_path + "/**/*.wav", recursive=True)
-output_files = [wav_file.replace(args.data_path, args.output_path).replace(
+output_files = [
-    '.wav', '.npy') for wav_file in wav_files]
+    wav_file.replace(args.data_path, args.output_path).replace(".wav", ".npy")
+    for wav_file in wav_files
+]
 
 for output_file in output_files:
     os.makedirs(os.path.dirname(output_file), exist_ok=True)
 
 model = SpeakerEncoder(**c.model)
-model.load_state_dict(torch.load(args.model_path)['model'])
+model.load_state_dict(torch.load(args.model_path)["model"])
 model.eval()
 if args.use_cuda:
     model.cuda()

speaker_encoder/dataset.py

@@ -1,23 +1,12 @@
-import os
 import numpy as np
-import collections
 import torch
 import random
 from torch.utils.data import Dataset
 
-from TTS.utils.text import text_to_sequence, phoneme_to_sequence, pad_with_eos_bos
-from TTS.utils.data import prepare_data, prepare_tensor, prepare_stop_target
-
 
 class MyDataset(Dataset):
-    def __init__(self,
+    def __init__(self, ap, meta_data, voice_len=1.6, num_speakers_in_batch=64,
-                 ap,
+                 num_utter_per_speaker=10, skip_speakers=False, verbose=False):
-                 meta_data,
-                 voice_len=1.6,
-                 num_speakers_in_batch=64,
-                 num_utter_per_speaker=10,  
-                 skip_speakers=False,
-                 verbose=False):
         """
         Args:
             ap (TTS.utils.AudioProcessor): audio processor object.
@@ -29,6 +18,7 @@ class MyDataset(Dataset):
         self.sample_rate = ap.sample_rate
         self.voice_len = voice_len
         self.seq_len = int(voice_len * self.sample_rate)
+        self.num_speakers_in_batch = num_speakers_in_batch
         self.num_utter_per_speaker = num_utter_per_speaker
         self.skip_speakers = skip_speakers
         self.ap = ap
@@ -47,16 +37,16 @@ class MyDataset(Dataset):
     def load_data(self, idx):
         text, wav_file, speaker_name = self.items[idx]
         wav = np.asarray(self.load_wav(wav_file), dtype=np.float32)
-        mel = self.ap.melspectrogram(wav).astype('float32')
+        mel = self.ap.melspectrogram(wav).astype("float32")
         # sample seq_len
 
         assert text.size > 0, self.items[idx][1]
         assert wav.size > 0, self.items[idx][1]
 
         sample = {
-            'mel': mel,
+            "mel": mel,
-            'item_idx': self.items[idx][1],
+            "item_idx": self.items[idx][1],
-            'speaker_name': speaker_name
+            "speaker_name": speaker_name,
         }
         return sample
 
@@ -64,26 +54,32 @@ class MyDataset(Dataset):
         """
         Find unique speaker ids and create a dict mapping utterances from speaker id
         """
-        speakers = list(set([item[-1] for item in self.items]))
+        speakers = list({item[-1] for item in self.items})
         self.speaker_to_utters = {}
         self.speakers = []
         for speaker in speakers:
             speaker_utters = [item[1] for item in self.items if item[2] == speaker]
             if len(speaker_utters) < self.num_utter_per_speaker and self.skip_speakers:
-                print(f" [!] Skipped speaker {speaker}. Not enough utterances {self.num_utter_per_speaker} vs {len(speaker_utters)}.")
+                print(
+                    f" [!] Skipped speaker {speaker}. Not enough utterances {self.num_utter_per_speaker} vs {len(speaker_utters)}."
+                )
             else:
                 self.speakers.append(speaker)
                 self.speaker_to_utters[speaker] = speaker_utters
 
     def __len__(self):
-        return int(1e+10)
+        return int(1e10)
 
     def __sample_speaker(self):
         speaker = random.sample(self.speakers, 1)[0]
         if self.num_utter_per_speaker > len(self.speaker_to_utters[speaker]):
-            utters = random.choices(self.speaker_to_utters[speaker], k=self.num_utter_per_speaker)
+            utters = random.choices(
+                self.speaker_to_utters[speaker], k=self.num_utter_per_speaker
+            )
         else:
-            utters = random.sample(self.speaker_to_utters[speaker], self.num_utter_per_speaker)
+            utters = random.sample(
+                self.speaker_to_utters[speaker], self.num_utter_per_speaker
+            )
         return speaker, utters
 
     def __sample_speaker_utterances(self, speaker):
@@ -92,7 +88,7 @@ class MyDataset(Dataset):
         """
         feats = []
         labels = []
-        for idx in range(self.num_utter_per_speaker):
+        for _ in range(self.num_utter_per_speaker):
             # TODO:dummy but works
             while True:
                 if len(self.speaker_to_utters[speaker]) > 0:
@@ -104,11 +100,10 @@ class MyDataset(Dataset):
                 wav = self.load_wav(utter)
                 if wav.shape[0] - self.seq_len > 0:
                     break
-                else:
                 self.speaker_to_utters[speaker].remove(utter)
 
             offset = random.randint(0, wav.shape[0] - self.seq_len)
-            mel = self.ap.melspectrogram(wav[offset:offset+self.seq_len])
+            mel = self.ap.melspectrogram(wav[offset : offset + self.seq_len])
             feats.append(torch.FloatTensor(mel))
             labels.append(speaker)
         return feats, labels

speaker_encoder/loss.py

@@ -5,9 +5,8 @@ import torch.nn.functional as F
 
 # adapted from https://github.com/cvqluu/GE2E-Loss
 class GE2ELoss(nn.Module):
-
+    def __init__(self, init_w=10.0, init_b=-5.0, loss_method="softmax"):
-    def __init__(self, init_w=10.0, init_b=-5.0, loss_method='softmax'):
+        """
-        '''
         Implementation of the Generalized End-to-End loss defined in https://arxiv.org/abs/1710.10467 [1]
         Accepts an input of size (N, M, D)
             where N is the number of speakers in the batch,
@@ -16,24 +15,27 @@ class GE2ELoss(nn.Module):
         Args:
             - init_w (float): defines the initial value of w in Equation (5) of [1]
             - init_b (float): definies the initial value of b in Equation (5) of [1]
-        '''
+        """
         super(GE2ELoss, self).__init__()
+        # pylint: disable=E1102
         self.w = nn.Parameter(torch.tensor(init_w))
+        # pylint: disable=E1102
         self.b = nn.Parameter(torch.tensor(init_b))
         self.loss_method = loss_method
 
-        assert self.loss_method in ['softmax', 'contrast']
+        assert self.loss_method in ["softmax", "contrast"]
 
-        if self.loss_method == 'softmax':
+        if self.loss_method == "softmax":
             self.embed_loss = self.embed_loss_softmax
-        if self.loss_method == 'contrast':
+        if self.loss_method == "contrast":
             self.embed_loss = self.embed_loss_contrast
 
+    # pylint: disable=R0201
     def calc_new_centroids(self, dvecs, centroids, spkr, utt):
-        '''
+        """
         Calculates the new centroids excluding the reference utterance
-        '''
+        """
-        excl = torch.cat((dvecs[spkr, :utt], dvecs[spkr, utt+1:]))
+        excl = torch.cat((dvecs[spkr, :utt], dvecs[spkr, utt + 1 :]))
         excl = torch.mean(excl, 0)
         new_centroids = []
         for i, centroid in enumerate(centroids):
@@ -44,26 +46,36 @@ class GE2ELoss(nn.Module):
         return torch.stack(new_centroids)
 
     def calc_cosine_sim(self, dvecs, centroids):
-        '''
+        """
         Make the cosine similarity matrix with dims (N,M,N)
-        '''
+        """
         cos_sim_matrix = []
         for spkr_idx, speaker in enumerate(dvecs):
             cs_row = []
             for utt_idx, utterance in enumerate(speaker):
                 new_centroids = self.calc_new_centroids(
-                    dvecs, centroids, spkr_idx, utt_idx)
+                    dvecs, centroids, spkr_idx, utt_idx
+                )
                 # vector based cosine similarity for speed
-                cs_row.append(torch.clamp(torch.mm(utterance.unsqueeze(1).transpose(0, 1), new_centroids.transpose(
+                cs_row.append(
-                    0, 1)) / (torch.norm(utterance) * torch.norm(new_centroids, dim=1)), 1e-6))
+                    torch.clamp(
+                        torch.mm(
+                            utterance.unsqueeze(1).transpose(0, 1),
+                            new_centroids.transpose(0, 1),
+                        )
+                        / (torch.norm(utterance) * torch.norm(new_centroids, dim=1)),
+                        1e-6,
+                    )
+                )
             cs_row = torch.cat(cs_row, dim=0)
             cos_sim_matrix.append(cs_row)
         return torch.stack(cos_sim_matrix)
 
+    # pylint: disable=R0201
     def embed_loss_softmax(self, dvecs, cos_sim_matrix):
-        '''
+        """
         Calculates the loss on each embedding $L(e_{ji})$ by taking softmax
-        '''
+        """
         N, M, _ = dvecs.shape
         L = []
         for j in range(N):
@@ -74,10 +86,11 @@ class GE2ELoss(nn.Module):
             L.append(L_row)
         return torch.stack(L)
 
+    # pylint: disable=R0201
     def embed_loss_contrast(self, dvecs, cos_sim_matrix):
-        ''' 
+        """
         Calculates the loss on each embedding $L(e_{ji})$ by contrast loss with closest centroid
-        '''
+        """
         N, M, _ = dvecs.shape
         L = []
         for j in range(N):
@@ -85,17 +98,21 @@ class GE2ELoss(nn.Module):
             for i in range(M):
                 centroids_sigmoids = torch.sigmoid(cos_sim_matrix[j, i])
                 excl_centroids_sigmoids = torch.cat(
-                    (centroids_sigmoids[:j], centroids_sigmoids[j+1:]))
+                    (centroids_sigmoids[:j], centroids_sigmoids[j + 1 :])
+                )
                 L_row.append(
-                    1. - torch.sigmoid(cos_sim_matrix[j, i, j]) + torch.max(excl_centroids_sigmoids))
+                    1.0
+                    - torch.sigmoid(cos_sim_matrix[j, i, j])
+                    + torch.max(excl_centroids_sigmoids)
+                )
             L_row = torch.stack(L_row)
             L.append(L_row)
         return torch.stack(L)
 
     def forward(self, dvecs):
-        '''
+        """
         Calculates the GE2E loss for an input of dimensions (num_speakers, num_utts_per_speaker, dvec_feats)
-        '''
+        """
         centroids = torch.mean(dvecs, 1)
         cos_sim_matrix = self.calc_cosine_sim(dvecs, centroids)
         torch.clamp(self.w, 1e-6)

speaker_encoder/model.py

@@ -13,7 +13,7 @@ class LSTMWithProjection(nn.Module):
 
     def forward(self, x):
         self.lstm.flatten_parameters()
-        o, (h, c) = self.lstm(x)
+        o, (_, _) = self.lstm(x)
         return self.linear(o)
 
 
@@ -22,16 +22,16 @@ class SpeakerEncoder(nn.Module):
         super().__init__()
         layers = []
         layers.append(LSTMWithProjection(input_dim, lstm_dim, proj_dim))
-        for _ in range(num_lstm_layers-1):
+        for _ in range(num_lstm_layers - 1):
             layers.append(LSTMWithProjection(proj_dim, lstm_dim, proj_dim))
         self.layers = nn.Sequential(*layers)
         self._init_layers()
 
     def _init_layers(self):
         for name, param in self.layers.named_parameters():
-            if 'bias' in name:
+            if "bias" in name:
                 nn.init.constant_(param, 0.0)
-            elif 'weight' in name:
+            elif "weight" in name:
                 nn.init.xavier_normal_(param)
 
     def forward(self, x):
@@ -81,7 +81,8 @@ class SpeakerEncoder(nn.Module):
             if embed is None:
                 embed = self.inference(frames)
             else:
-                embed[cur_iter <= num_iters, :] += self.inference(frames[cur_iter <= num_iters, :, :])
+                embed[cur_iter <= num_iters, :] += self.inference(
+                    frames[cur_iter <= num_iters, :, :]
+                )
         return embed / num_iters
 
-

speaker_encoder/tests.py

@@ -4,22 +4,21 @@ import torch as T
 
 from TTS.speaker_encoder.model import SpeakerEncoder
 from TTS.speaker_encoder.loss import GE2ELoss
-from TTS.speaker_encoder.dataset import MyDataset
-from TTS.utils.audio import AudioProcessor
-from torch.utils.data import DataLoader
-from TTS.datasets.preprocess import libri_tts
 from TTS.utils.generic_utils import load_config
 
 
 file_path = os.path.dirname(os.path.realpath(__file__)) + "/../tests/"
-c = load_config(os.path.join(file_path, 'test_config.json'))
+c = load_config(os.path.join(file_path, "test_config.json"))
 
 
 class SpeakerEncoderTests(unittest.TestCase):
+    # pylint: disable=R0201
     def test_in_out(self):
         dummy_input = T.rand(4, 20, 80)  # B x T x D
         dummy_hidden = [T.rand(2, 4, 128), T.rand(2, 4, 128)]
-        model = SpeakerEncoder(input_dim=80, proj_dim=256, lstm_dim=768, num_lstm_layers=3)
+        model = SpeakerEncoder(
+            input_dim=80, proj_dim=256, lstm_dim=768, num_lstm_layers=3
+        )
         # computing d vectors
         output = model.forward(dummy_input)
         assert output.shape[0] == 4
@@ -35,8 +34,10 @@ class SpeakerEncoderTests(unittest.TestCase):
         # check normalization
         output_norm = T.nn.functional.normalize(output, dim=1, p=2)
         assert_diff = (output_norm - output).sum().item()
-        assert output.type() == 'torch.FloatTensor'
+        assert output.type() == "torch.FloatTensor"
-        assert abs(assert_diff) < 1e-4, f" [!] output_norm has wrong values - {assert_diff}"
+        assert (
+            abs(assert_diff) < 1e-4
+        ), f" [!] output_norm has wrong values - {assert_diff}"
         # compute d for a given batch
         dummy_input = T.rand(1, 240, 80)  # B x T x D
         output = model.compute_embedding(dummy_input, num_frames=160, overlap=0.5)
@@ -45,23 +46,29 @@ class SpeakerEncoderTests(unittest.TestCase):
         assert len(output.shape) == 2
 
 
-
 class GE2ELossTests(unittest.TestCase):
+    # pylint: disable=R0201
     def test_in_out(self):
         # check random input
         dummy_input = T.rand(4, 5, 64)  # num_speaker x num_utterance x dim
-        loss = GE2ELoss(loss_method='softmax')
+        loss = GE2ELoss(loss_method="softmax")
         output = loss.forward(dummy_input)
-        assert output.item() >= 0. 
+        assert output.item() >= 0.0
         # check all zeros
         dummy_input = T.ones(4, 5, 64)  # num_speaker x num_utterance x dim
-        loss = GE2ELoss(loss_method='softmax')
+        loss = GE2ELoss(loss_method="softmax")
         output = loss.forward(dummy_input)
         # check speaker loss with orthogonal d-vectors
         dummy_input = T.empty(3, 64)
         dummy_input = T.nn.init.orthogonal(dummy_input)
-        dummy_input = T.cat([dummy_input[0].repeat(5, 1, 1).transpose(0, 1), dummy_input[1].repeat(5, 1, 1).transpose(0, 1), dummy_input[2].repeat(5, 1, 1).transpose(0, 1)])  # num_speaker x num_utterance x dim
+        dummy_input = T.cat(
-        loss = GE2ELoss(loss_method='softmax')
+            [
+                dummy_input[0].repeat(5, 1, 1).transpose(0, 1),
+                dummy_input[1].repeat(5, 1, 1).transpose(0, 1),
+                dummy_input[2].repeat(5, 1, 1).transpose(0, 1),
+            ]
+        )  # num_speaker x num_utterance x dim
+        loss = GE2ELoss(loss_method="softmax")
         output = loss.forward(dummy_input)
         assert output.item() < 0.005
 

speaker_encoder/train.py

@@ -5,24 +5,21 @@ import time
 import traceback
 
 import torch
-from torch import optim
 from torch.utils.data import DataLoader
 from TTS.datasets.preprocess import load_meta_data
 from TTS.speaker_encoder.dataset import MyDataset
-from TTS.speaker_encoder.generic_utils import save_best_model, save_checkpoint
 from TTS.speaker_encoder.loss import GE2ELoss
 from TTS.speaker_encoder.model import SpeakerEncoder
 from TTS.speaker_encoder.visual import plot_embeddings
+from TTS.speaker_encoder.generic_utils import save_best_model, save_checkpoint
 from TTS.utils.audio import AudioProcessor
 from TTS.utils.generic_utils import (NoamLR, check_update, copy_config_file,
                                      count_parameters,
                                      create_experiment_folder, get_git_branch,
-                                     gradual_training_scheduler, load_config,
+                                     load_config,
-                                     remove_experiment_folder, set_init_dict,
+                                     remove_experiment_folder, set_init_dict)
-                                     setup_model, split_dataset)
 from TTS.utils.logger import Logger
 from TTS.utils.radam import RAdam
-from TTS.utils.visual import plot_alignment, plot_spectrogram
 
 torch.backends.cudnn.enabled = True
 torch.backends.cudnn.benchmark = True
@@ -34,10 +31,6 @@ print(" > Number of GPUs: ", num_gpus)
 
 
 def setup_loader(ap, is_val=False, verbose=False):
-    global meta_data_train
-    global meta_data_eval
-    if "meta_data_train" not in globals():
-        meta_data_train, meta_data_eval = load_meta_data(c.datasets)
     if is_val:
         loader = None
     else:
@@ -63,12 +56,11 @@ def train(model, criterion, optimizer, scheduler, ap, global_step):
     best_loss = float('inf')
     avg_loss = 0
     end_time = time.time()
-    for num_iter, data in enumerate(data_loader):
+    for _, data in enumerate(data_loader):
         start_time = time.time()
 
         # setup input data
         inputs = data[0]
-        labels = data[1]
         loader_time = time.time() - end_time
         global_step += 1
 
@@ -132,68 +124,11 @@ def train(model, criterion, optimizer, scheduler, ap, global_step):
     return avg_loss, global_step
 
 
-# def evaluate(model, criterion, ap, global_step, epoch):
-#     data_loader = setup_loader(ap, is_val=True)
-#     model.eval()
-#     epoch_time = 0
-#     avg_loss = 0
-#     print("\n > Validation")
-#     with torch.no_grad():
-#         if data_loader is not None:
-#             for num_iter, data in enumerate(data_loader):
-#                 start_time = time.time()
-
-#                 # setup input data
-#                 inputs = data[0]
-#                 labels = data[1]
-
-#                 # dispatch data to GPU
-#                 if use_cuda:
-#                     inputs = inputs.cuda()
-#                     # labels = labels.cuda()
-
-#                 # forward pass
-#                 outputs = model.forward(inputs)
-
-#                 # loss computation
-#                 loss = criterion(outputs.reshape(
-#                     c.num_speakers_in_batch, outputs.shape[0] // c.num_speakers_in_batch, -1))
-#                 step_time = time.time() - start_time
-#                 epoch_time += step_time
-
-#                 if num_iter % c.print_step == 0:
-#                     print(
-#                         "   | > Loss: {:.5f}  ".format(loss.item()),
-#                         flush=True)
-
-#                 avg_loss += float(loss.item())
-
-#             eval_figures = {
-#                 "prediction": plot_spectrogram(const_spec, ap),
-#                 "ground_truth": plot_spectrogram(gt_spec, ap),
-#                 "alignment": plot_alignment(align_img)
-#             }
-#             tb_logger.tb_eval_figures(global_step, eval_figures)
-
-#             # Sample audio
-#             if c.model in ["Tacotron", "TacotronGST"]:
-#                 eval_audio = ap.inv_spectrogram(const_spec.T)
-#             else:
-#                 eval_audio = ap.inv_mel_spectrogram(const_spec.T)
-#             tb_logger.tb_eval_audios(
-#                 global_step, {"ValAudio": eval_audio}, c.audio["sample_rate"])
-
-#             # compute average losses
-#             avg_loss /= (num_iter + 1)
-
-#             # Plot Validation Stats
-#             epoch_stats = {"GE2Eloss": avg_loss}
-#             tb_logger.tb_eval_stats(global_step, epoch_stats)
-#     return avg_loss
-
-
-# FIXME: move args definition/parsing inside of main?
 def main(args):  # pylint: disable=redefined-outer-name
+    # pylint: disable=global-variable-undefined
+    global meta_data_train
+    global meta_data_eval
+
     ap = AudioProcessor(**c.audio)
     model = SpeakerEncoder(input_dim=40,
                            proj_dim=128,
@@ -211,7 +146,7 @@ def main(args):  # pylint: disable=redefined-outer-name
             if c.reinit_layers:
                 raise RuntimeError
             model.load_state_dict(checkpoint['model'])
-        except:
+        except KeyError:
             print(" > Partial model initialization.")
             model_dict = model.state_dict()
             model_dict = set_init_dict(model_dict, checkpoint, c)
@@ -239,6 +174,9 @@ def main(args):  # pylint: disable=redefined-outer-name
     num_params = count_parameters(model)
     print("\n > Model has {} parameters".format(num_params), flush=True)
 
+    # pylint: disable=redefined-outer-name
+    meta_data_train, meta_data_eval = load_meta_data(c.datasets)
+
     global_step = args.restore_step
     train_loss, global_step = train(model, criterion, optimizer, scheduler, ap,
                                     global_step)

speaker_encoder/visual.py

@@ -3,10 +3,12 @@ import numpy as np
 import matplotlib
 import matplotlib.pyplot as plt
 
-matplotlib.use('Agg')
+matplotlib.use("Agg")
 
 
-colormap = np.array([
+colormap = (
+    np.array(
+        [
             [76, 255, 0],
             [0, 127, 70],
             [255, 0, 0],
@@ -20,11 +22,15 @@ colormap = np.array([
             [33, 0, 127],
             [0, 0, 0],
             [183, 183, 183],
-], dtype=np.float) / 255 
+        ],
+        dtype=np.float,
+    )
+    / 255
+)
 
 
 def plot_embeddings(embeddings, num_utter_per_speaker):
-    embeddings = embeddings[:10*num_utter_per_speaker]
+    embeddings = embeddings[: 10 * num_utter_per_speaker]
     model = umap.UMAP()
     projection = model.fit_transform(embeddings)
     num_speakers = embeddings.shape[0] // num_utter_per_speaker
@@ -32,7 +38,7 @@ def plot_embeddings(embeddings, num_utter_per_speaker):
     colors = [colormap[i] for i in ground_truth]
 
     fig, ax = plt.subplots(figsize=(16, 10))
-    im = ax.scatter(projection[:, 0], projection[:, 1], c=colors)
+    _ = ax.scatter(projection[:, 0], projection[:, 1], c=colors)
     plt.gca().set_aspect("equal", "datalim")
     plt.title("UMAP projection")
     plt.tight_layout()

tests/outputs/dummy_model_config.json

@@ -44,6 +44,8 @@
     "prenet_dropout": true,        // ONLY TACOTRON2 - enable/disable dropout at prenet. 
     "use_forward_attn": true,      // ONLY TACOTRON2 - if it uses forward attention. In general, it aligns faster.
     "forward_attn_mask": false, 
+    "attention_type": "original",
+    "attention_heads": 5,
     "bidirectional_decoder": false,
     "transition_agent": false,     // ONLY TACOTRON2 - enable/disable transition agent of forward attention.
     "location_attn": false,        // ONLY TACOTRON2 - enable_disable location sensitive attention. It is enabled for TACOTRON by default.

tests/test_layers.py

@@ -5,7 +5,8 @@ from TTS.layers.tacotron import Prenet, CBHG, Decoder, Encoder
 from TTS.layers.losses import L1LossMasked
 from TTS.utils.generic_utils import sequence_mask
 
-#pylint: disable=unused-variable
+# pylint: disable=unused-variable
+
 
 class PrenetTests(unittest.TestCase):
     def test_in_out(self):
@@ -49,6 +50,8 @@ class DecoderTests(unittest.TestCase):
             memory_size=4,
             attn_windowing=False,
             attn_norm="sigmoid",
+            attn_K=5,
+            attn_type="original",
             prenet_type='original',
             prenet_dropout=True,
             forward_attn=True,
@@ -77,6 +80,8 @@ class DecoderTests(unittest.TestCase):
             memory_size=4,
             attn_windowing=False,
             attn_norm="sigmoid",
+            attn_K=5,
+            attn_type="graves",
             prenet_type='original',
             prenet_dropout=True,
             forward_attn=True,

train.py

@@ -117,7 +117,8 @@ def format_data(data):
 
 def train(model, criterion, criterion_st, optimizer, optimizer_st, scheduler,
           ap, global_step, epoch):
-    data_loader = setup_loader(ap, model.decoder.r, is_val=False, verbose=(epoch == 0))
+    data_loader = setup_loader(ap, model.decoder.r, is_val=False,
+                               verbose=(epoch == 0))
     model.train()
     epoch_time = 0
     train_values = {

utils/audio.py

@@ -15,7 +15,7 @@ class AudioProcessor(object):
                  ref_level_db=None,
                  num_freq=None,
                  power=None,
-                 preemphasis=None,
+                 preemphasis=0.0,
                  signal_norm=None,
                  symmetric_norm=None,
                  max_norm=None,
@@ -48,7 +48,7 @@ class AudioProcessor(object):
         self.do_trim_silence = do_trim_silence
         self.sound_norm = sound_norm
         self.n_fft, self.hop_length, self.win_length = self._stft_parameters()
-        assert min_level_db ~= 0.0, " [!] min_level_db is 0"
+        assert min_level_db != 0.0, " [!] min_level_db is 0"
         members = vars(self)
         for key, value in members.items():
             print(" | > {}:{}".format(key, value))
@@ -132,12 +132,12 @@ class AudioProcessor(object):
 
     def apply_preemphasis(self, x):
         if self.preemphasis == 0:
-            raise RuntimeError(" !! Preemphasis is applied with factor 0.0. ")
+            raise RuntimeError(" [!] Preemphasis is set 0.0.")
         return scipy.signal.lfilter([1, -self.preemphasis], [1], x)
 
     def apply_inv_preemphasis(self, x):
         if self.preemphasis == 0:
-            raise RuntimeError(" !! Preemphasis is applied with factor 0.0. ")
+            raise RuntimeError(" [!] Preemphasis is set 0.0.")
         return scipy.signal.lfilter([1], [1, -self.preemphasis], x)
 
     def spectrogram(self, y):