mirror of https://github.com/coqui-ai/TTS.git
Clean up old code
This commit is contained in:
Edresson Casanova
parent
3a524b0597
commit
dcd0d1f6a1
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@ -117,7 +117,9 @@ def load_tts_samples(
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if eval_split:
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if meta_file_val:
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meta_data_eval = formatter(root_path, meta_file_val, ignored_speakers=ignored_speakers)
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meta_data_eval = [{**item, **{"language": language, "speech_style": speech_style}} for item in meta_data_eval]
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meta_data_eval = [
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{**item, **{"language": language, "speech_style": speech_style}} for item in meta_data_eval
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]
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else:
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meta_data_eval, meta_data_train = split_dataset(meta_data_train, eval_split_max_size, eval_split_size)
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meta_data_eval_all += meta_data_eval
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@ -1,6 +1,8 @@
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import torch
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from torch import nn
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# pylint: disable=W0223
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class GradientReversalFunction(torch.autograd.Function):
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"""Revert gradient without any further input modification.
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Adapted from: https://github.com/Tomiinek/Multilingual_Text_to_Speech/"""
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@ -30,16 +32,15 @@ class ReversalClassifier(nn.Module):
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"""
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def __init__(self, in_channels, out_channels, hidden_channels, gradient_clipping_bounds=0.25, scale_factor=1.0):
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super(ReversalClassifier, self).__init__()
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super().__init__()
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self._lambda = scale_factor
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self._clipping = gradient_clipping_bounds
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self._out_channels = out_channels
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self._classifier = nn.Sequential(
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nn.Linear(in_channels, hidden_channels),
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nn.ReLU(),
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nn.Linear(hidden_channels, out_channels)
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nn.Linear(in_channels, hidden_channels), nn.ReLU(), nn.Linear(hidden_channels, out_channels)
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)
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self.test = nn.Linear(in_channels, hidden_channels)
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def forward(self, x, labels, x_mask=None):
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x = GradientReversalFunction.apply(x, self._lambda, self._clipping)
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x = self._classifier(x)
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@ -55,7 +56,7 @@ class ReversalClassifier(nn.Module):
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ml = torch.max(x_mask)
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input_mask = torch.arange(ml, device=predictions.device)[None, :] < x_mask[:, None]
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target = labels.repeat(ml.int().item(), 1).transpose(0,1)
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target = labels.repeat(ml.int().item(), 1).transpose(0, 1)
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target[~input_mask] = ignore_index
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return nn.functional.cross_entropy(predictions.transpose(1,2), target, ignore_index=ignore_index)
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return nn.functional.cross_entropy(predictions.transpose(1, 2), target, ignore_index=ignore_index)
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@ -12,9 +12,9 @@ from trainer.torch import DistributedSampler, DistributedSamplerWrapper
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from TTS.model import BaseTrainerModel
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from TTS.tts.datasets.dataset import TTSDataset
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from TTS.tts.utils.emotions import get_speech_style_balancer_weights
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from TTS.tts.utils.languages import LanguageManager, get_language_balancer_weights
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from TTS.tts.utils.speakers import SpeakerManager, get_speaker_balancer_weights, get_speaker_manager
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from TTS.tts.utils.emotions import get_speech_style_balancer_weights
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from TTS.tts.utils.synthesis import synthesis
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from TTS.tts.utils.visual import plot_alignment, plot_spectrogram
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@ -17,7 +17,9 @@ from trainer.trainer_utils import get_optimizer, get_scheduler
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from TTS.tts.configs.shared_configs import CharactersConfig
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from TTS.tts.datasets.dataset import TTSDataset, _parse_sample
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from TTS.tts.layers.generic.classifier import ReversalClassifier
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from TTS.tts.layers.glow_tts.duration_predictor import DurationPredictor
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from TTS.tts.layers.tacotron.gst_layers import GST
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from TTS.tts.layers.vits.discriminator import VitsDiscriminator
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from TTS.tts.layers.vits.networks import PosteriorEncoder, ResidualCouplingBlocks, TextEncoder
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from TTS.tts.layers.vits.stochastic_duration_predictor import StochasticDurationPredictor
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@ -33,9 +35,6 @@ from TTS.tts.utils.visual import plot_alignment
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from TTS.vocoder.models.hifigan_generator import HifiganGenerator
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from TTS.vocoder.utils.generic_utils import plot_results
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from TTS.tts.layers.tacotron.gst_layers import GST
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from TTS.tts.layers.generic.classifier import ReversalClassifier
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##############################
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# IO / Feature extraction
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##############################
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@ -503,7 +502,6 @@ class VitsArgs(Coqpit):
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external_emotions_embs_file: str = None
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emotion_embedding_dim: int = 0
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num_emotions: int = 0
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emotion_just_encoder: bool = False
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# prosody encoder
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use_prosody_encoder: bool = False
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@ -615,7 +613,7 @@ class Vits(BaseTTS):
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dp_cond_embedding_dim = self.cond_embedding_dim if self.args.condition_dp_on_speaker else 0
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if self.args.emotion_just_encoder and (self.args.use_emotion_embedding or self.args.use_external_emotions_embeddings):
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if self.args.use_emotion_embedding or self.args.use_external_emotions_embeddings:
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dp_cond_embedding_dim += self.args.emotion_embedding_dim
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if self.args.use_prosody_encoder:
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@ -796,12 +794,6 @@ class Vits(BaseTTS):
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if self.num_emotions > 0:
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print(" > initialization of emotion-embedding layers.")
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self.emb_emotion = nn.Embedding(self.num_emotions, self.args.emotion_embedding_dim)
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if not self.args.emotion_just_encoder:
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self.cond_embedding_dim += self.args.emotion_embedding_dim
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if self.args.use_external_emotions_embeddings:
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if not self.args.emotion_just_encoder:
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self.cond_embedding_dim += self.args.emotion_embedding_dim
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def get_aux_input(self, aux_input: Dict):
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sid, g, lid, eid, eg = self._set_cond_input(aux_input)
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@ -983,13 +975,6 @@ class Vits(BaseTTS):
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if self.args.use_emotion_embedding and eid is not None and eg is None:
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eg = self.emb_emotion(eid).unsqueeze(-1) # [b, h, 1]
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# concat the emotion embedding and speaker embedding
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if eg is not None and (self.args.use_emotion_embedding or self.args.use_external_emotions_embeddings) and not self.args.emotion_just_encoder:
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if g is None:
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g = eg
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else:
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g = torch.cat([g, eg], dim=1) # [b, h1+h2, 1]
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# language embedding
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lang_emb = None
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if self.args.use_language_embedding and lid is not None:
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@ -1004,15 +989,15 @@ class Vits(BaseTTS):
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if self.args.use_prosody_encoder:
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pros_emb = self.prosody_encoder(z).transpose(1, 2)
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_, l_pros_speaker = self.speaker_reversal_classifier(pros_emb.transpose(1, 2), sid, x_mask=None)
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# print("Encoder input", x.shape)
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x, m_p, logs_p, x_mask = self.text_encoder(x, x_lengths, lang_emb=lang_emb, emo_emb=eg, pros_emb=pros_emb)
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# print("X shape:", x.shape, "m_p shape:", m_p.shape, "x_mask:", x_mask.shape, "x_lengths:", x_lengths.shape)
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# flow layers
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z_p = self.flow(z, y_mask, g=g)
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# print("Y mask:", y_mask.shape)
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# duration predictor
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g_dp = g if self.args.condition_dp_on_speaker else None
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if eg is not None and (self.args.use_emotion_embedding or self.args.use_external_emotions_embeddings) and self.args.emotion_just_encoder:
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if eg is not None and (self.args.use_emotion_embedding or self.args.use_external_emotions_embeddings):
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if g_dp is None:
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g_dp = eg
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else:
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@ -1130,13 +1115,6 @@ class Vits(BaseTTS):
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if self.args.use_emotion_embedding and eid is not None and eg is None:
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eg = self.emb_emotion(eid).unsqueeze(-1) # [b, h, 1]
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# concat the emotion embedding and speaker embedding
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if eg is not None and (self.args.use_emotion_embedding or self.args.use_external_emotions_embeddings) and not self.args.emotion_just_encoder:
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if g is None:
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g = eg
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else:
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g = torch.cat([g, eg], dim=1) # [b, h1+h2, 1]
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# language embedding
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lang_emb = None
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if self.args.use_language_embedding and lid is not None:
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@ -1154,7 +1132,7 @@ class Vits(BaseTTS):
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# duration predictor
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g_dp = g if self.args.condition_dp_on_speaker else None
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if eg is not None and (self.args.use_emotion_embedding or self.args.use_external_emotions_embeddings) and self.args.emotion_just_encoder:
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if eg is not None and (self.args.use_emotion_embedding or self.args.use_external_emotions_embeddings):
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if g_dp is None:
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g_dp = eg
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else:
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@ -1176,9 +1154,7 @@ class Vits(BaseTTS):
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lang_emb=lang_emb,
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)
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else:
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logw = self.duration_predictor(
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x, x_mask, g=g_dp, lang_emb=lang_emb
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)
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logw = self.duration_predictor(x, x_mask, g=g_dp, lang_emb=lang_emb)
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w = torch.exp(logw) * x_mask * self.length_scale
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w_ceil = torch.ceil(w)
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@ -1195,13 +1171,23 @@ class Vits(BaseTTS):
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z = self.flow(z_p, y_mask, g=g, reverse=True)
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o = self.waveform_decoder((z * y_mask)[:, :, : self.max_inference_len], g=g)
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outputs = {"model_outputs": o, "alignments": attn.squeeze(1), "z": z, "z_p": z_p, "m_p": m_p, "logs_p": logs_p, "durations": w_ceil}
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outputs = {
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"model_outputs": o,
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"alignments": attn.squeeze(1),
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"z": z,
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"z_p": z_p,
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"m_p": m_p,
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"logs_p": logs_p,
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"durations": w_ceil,
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}
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return outputs
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def compute_style_feature(self, style_wav_path):
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style_wav, sr = torchaudio.load(style_wav_path)
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if sr != self.config.audio.sample_rate:
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raise RuntimeError(" [!] Style reference need to have sampling rate equal to {self.config.audio.sample_rate} !!")
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raise RuntimeError(
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" [!] Style reference need to have sampling rate equal to {self.config.audio.sample_rate} !!"
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)
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y = wav_to_spec(
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style_wav,
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self.config.audio.fft_size,
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@ -1371,7 +1357,7 @@ class Vits(BaseTTS):
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or self.args.use_emotion_encoder_as_loss,
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gt_cons_emb=self.model_outputs_cache["gt_cons_emb"],
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syn_cons_emb=self.model_outputs_cache["syn_cons_emb"],
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loss_spk_reversal_classifier=self.model_outputs_cache["loss_spk_reversal_classifier"]
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loss_spk_reversal_classifier=self.model_outputs_cache["loss_spk_reversal_classifier"],
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)
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return self.model_outputs_cache, loss_dict
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@ -1539,7 +1525,11 @@ class Vits(BaseTTS):
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emotion_ids = None
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# get numerical speaker ids from speaker names
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if self.speaker_manager is not None and self.speaker_manager.ids and (self.args.use_speaker_embedding or self.args.use_prosody_encoder):
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if (
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self.speaker_manager is not None
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and self.speaker_manager.ids
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and (self.args.use_speaker_embedding or self.args.use_prosody_encoder)
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):
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speaker_ids = [self.speaker_manager.ids[sn] for sn in batch["speaker_names"]]
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if speaker_ids is not None:
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@ -1,10 +1,10 @@
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import json
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import os
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import torch
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import numpy as np
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from typing import Any, List
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import fsspec
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import numpy as np
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import torch
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from coqpit import Coqpit
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from TTS.config import get_from_config_or_model_args_with_default
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@ -95,7 +95,9 @@ class SpeakerManager(EmbeddingManager):
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SpeakerEncoder: Speaker encoder object.
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"""
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speaker_manager = None
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if get_from_config_or_model_args_with_default(config, "use_speaker_embedding", False) or get_from_config_or_model_args_with_default(config, "use_prosody_encoder", False):
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if get_from_config_or_model_args_with_default(
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config, "use_speaker_embedding", False
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) or get_from_config_or_model_args_with_default(config, "use_prosody_encoder", False):
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if samples:
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speaker_manager = SpeakerManager(data_items=samples)
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if get_from_config_or_model_args_with_default(config, "speaker_file", None):
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@ -168,10 +168,9 @@ def synthesis(
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style_feature = style_wav
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else:
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style_feature = compute_style_feature(style_wav, model.ap, cuda=use_cuda)
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if hasattr(model, 'compute_style_feature'):
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if hasattr(model, "compute_style_feature"):
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style_feature = model.compute_style_feature(style_wav)
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# convert text to sequence of token IDs
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text_inputs = np.asarray(
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model.tokenizer.text_to_ids(text, language=language_id),
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