mirror of https://github.com/coqui-ai/TTS.git
840 lines
34 KiB
Python
840 lines
34 KiB
Python
import math
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import random
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from dataclasses import dataclass, field
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from itertools import chain
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from typing import Dict, List, Tuple
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import torch
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from coqpit import Coqpit
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from torch import nn
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from torch.cuda.amp.autocast_mode import autocast
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from torch.nn import functional as F
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from TTS.tts.layers.glow_tts.duration_predictor import DurationPredictor
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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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from TTS.tts.models.base_tts import BaseTTS
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from TTS.tts.utils.helpers import generate_path, maximum_path, rand_segments, segment, sequence_mask
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from TTS.tts.utils.speakers import SpeakerManager
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from TTS.tts.utils.languages import LanguageManager
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from TTS.tts.utils.synthesis import synthesis
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from TTS.tts.utils.visual import plot_alignment
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from TTS.utils.trainer_utils import get_optimizer, get_scheduler
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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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@dataclass
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class VitsArgs(Coqpit):
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"""VITS model arguments.
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Args:
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num_chars (int):
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Number of characters in the vocabulary. Defaults to 100.
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out_channels (int):
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Number of output channels. Defaults to 513.
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spec_segment_size (int):
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Decoder input segment size. Defaults to 32 `(32 * hoplength = waveform length)`.
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hidden_channels (int):
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Number of hidden channels of the model. Defaults to 192.
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hidden_channels_ffn_text_encoder (int):
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Number of hidden channels of the feed-forward layers of the text encoder transformer. Defaults to 256.
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num_heads_text_encoder (int):
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Number of attention heads of the text encoder transformer. Defaults to 2.
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num_layers_text_encoder (int):
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Number of transformer layers in the text encoder. Defaults to 6.
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kernel_size_text_encoder (int):
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Kernel size of the text encoder transformer FFN layers. Defaults to 3.
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dropout_p_text_encoder (float):
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Dropout rate of the text encoder. Defaults to 0.1.
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dropout_p_duration_predictor (float):
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Dropout rate of the duration predictor. Defaults to 0.1.
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kernel_size_posterior_encoder (int):
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Kernel size of the posterior encoder's WaveNet layers. Defaults to 5.
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dilatation_posterior_encoder (int):
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Dilation rate of the posterior encoder's WaveNet layers. Defaults to 1.
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num_layers_posterior_encoder (int):
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Number of posterior encoder's WaveNet layers. Defaults to 16.
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kernel_size_flow (int):
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Kernel size of the Residual Coupling layers of the flow network. Defaults to 5.
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dilatation_flow (int):
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Dilation rate of the Residual Coupling WaveNet layers of the flow network. Defaults to 1.
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num_layers_flow (int):
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Number of Residual Coupling WaveNet layers of the flow network. Defaults to 6.
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resblock_type_decoder (str):
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Type of the residual block in the decoder network. Defaults to "1".
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resblock_kernel_sizes_decoder (List[int]):
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Kernel sizes of the residual blocks in the decoder network. Defaults to `[3, 7, 11]`.
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resblock_dilation_sizes_decoder (List[List[int]]):
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Dilation sizes of the residual blocks in the decoder network. Defaults to `[[1, 3, 5], [1, 3, 5], [1, 3, 5]]`.
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upsample_rates_decoder (List[int]):
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Upsampling rates for each concecutive upsampling layer in the decoder network. The multiply of these
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values must be equal to the kop length used for computing spectrograms. Defaults to `[8, 8, 2, 2]`.
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upsample_initial_channel_decoder (int):
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Number of hidden channels of the first upsampling convolution layer of the decoder network. Defaults to 512.
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upsample_kernel_sizes_decoder (List[int]):
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Kernel sizes for each upsampling layer of the decoder network. Defaults to `[16, 16, 4, 4]`.
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use_sdp (bool):
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Use Stochastic Duration Predictor. Defaults to True.
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noise_scale (float):
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Noise scale used for the sample noise tensor in training. Defaults to 1.0.
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inference_noise_scale (float):
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Noise scale used for the sample noise tensor in inference. Defaults to 0.667.
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length_scale (float):
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Scale factor for the predicted duration values. Smaller values result faster speech. Defaults to 1.
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noise_scale_dp (float):
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Noise scale used by the Stochastic Duration Predictor sample noise in training. Defaults to 1.0.
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inference_noise_scale_dp (float):
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Noise scale for the Stochastic Duration Predictor in inference. Defaults to 0.8.
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max_inference_len (int):
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Maximum inference length to limit the memory use. Defaults to None.
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init_discriminator (bool):
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Initialize the disciminator network if set True. Set False for inference. Defaults to True.
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use_spectral_norm_disriminator (bool):
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Use spectral normalization over weight norm in the discriminator. Defaults to False.
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use_speaker_embedding (bool):
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Enable/Disable speaker embedding for multi-speaker models. Defaults to False.
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num_speakers (int):
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Number of speakers for the speaker embedding layer. Defaults to 0.
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speakers_file (str):
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Path to the speaker mapping file for the Speaker Manager. Defaults to None.
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speaker_embedding_channels (int):
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Number of speaker embedding channels. Defaults to 256.
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use_d_vector_file (bool):
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Enable/Disable the use of d-vectors for multi-speaker training. Defaults to False.
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d_vector_file (str):
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Path to the file including pre-computed speaker embeddings. Defaults to None.
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d_vector_dim (int):
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Number of d-vector channels. Defaults to 0.
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detach_dp_input (bool):
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Detach duration predictor's input from the network for stopping the gradients. Defaults to True.
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"""
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num_chars: int = 100
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out_channels: int = 513
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spec_segment_size: int = 32
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hidden_channels: int = 192
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hidden_channels_ffn_text_encoder: int = 768
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num_heads_text_encoder: int = 2
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num_layers_text_encoder: int = 6
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kernel_size_text_encoder: int = 3
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dropout_p_text_encoder: float = 0.1
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dropout_p_duration_predictor: float = 0.5
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kernel_size_posterior_encoder: int = 5
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dilation_rate_posterior_encoder: int = 1
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num_layers_posterior_encoder: int = 16
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kernel_size_flow: int = 5
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dilation_rate_flow: int = 1
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num_layers_flow: int = 4
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resblock_type_decoder: str = "1"
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resblock_kernel_sizes_decoder: List[int] = field(default_factory=lambda: [3, 7, 11])
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resblock_dilation_sizes_decoder: List[List[int]] = field(default_factory=lambda: [[1, 3, 5], [1, 3, 5], [1, 3, 5]])
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upsample_rates_decoder: List[int] = field(default_factory=lambda: [8, 8, 2, 2])
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upsample_initial_channel_decoder: int = 512
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upsample_kernel_sizes_decoder: List[int] = field(default_factory=lambda: [16, 16, 4, 4])
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use_sdp: bool = True
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noise_scale: float = 1.0
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inference_noise_scale: float = 0.667
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length_scale: float = 1
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noise_scale_dp: float = 1.0
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inference_noise_scale_dp: float = 1.0
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max_inference_len: int = None
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init_discriminator: bool = True
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use_spectral_norm_disriminator: bool = False
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use_speaker_embedding: bool = False
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num_speakers: int = 0
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speakers_file: str = None
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d_vector_file: str = None
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speaker_embedding_channels: int = 256
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use_d_vector_file: bool = False
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d_vector_file: str = None
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d_vector_dim: int = 0
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detach_dp_input: bool = True
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use_language_embedding: bool = False
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embedded_language_dim: int = 4
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num_languages: int = 0
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class Vits(BaseTTS):
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"""VITS TTS model
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Paper::
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https://arxiv.org/pdf/2106.06103.pdf
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Paper Abstract::
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Several recent end-to-end text-to-speech (TTS) models enabling single-stage training and parallel
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sampling have been proposed, but their sample quality does not match that of two-stage TTS systems.
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In this work, we present a parallel endto-end TTS method that generates more natural sounding audio than
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current two-stage models. Our method adopts variational inference augmented with normalizing flows and
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an adversarial training process, which improves the expressive power of generative modeling. We also propose a
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stochastic duration predictor to synthesize speech with diverse rhythms from input text. With the
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uncertainty modeling over latent variables and the stochastic duration predictor, our method expresses the
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natural one-to-many relationship in which a text input can be spoken in multiple ways
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with different pitches and rhythms. A subjective human evaluation (mean opinion score, or MOS)
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on the LJ Speech, a single speaker dataset, shows that our method outperforms the best publicly
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available TTS systems and achieves a MOS comparable to ground truth.
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Check :class:`TTS.tts.configs.vits_config.VitsConfig` for class arguments.
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Examples:
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>>> from TTS.tts.configs.vits_config import VitsConfig
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>>> from TTS.tts.models.vits import Vits
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>>> config = VitsConfig()
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>>> model = Vits(config)
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"""
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# pylint: disable=dangerous-default-value
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def __init__(self, config: Coqpit, speaker_manager: SpeakerManager = None):
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super().__init__(config)
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self.END2END = True
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self.speaker_manager = speaker_manager
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if config.__class__.__name__ == "VitsConfig":
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# loading from VitsConfig
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if "num_chars" not in config:
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_, self.config, num_chars = self.get_characters(config)
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config.model_args.num_chars = num_chars
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else:
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self.config = config
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config.model_args.num_chars = config.num_chars
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args = self.config.model_args
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elif isinstance(config, VitsArgs):
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# loading from VitsArgs
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self.config = config
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args = config
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else:
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raise ValueError("config must be either a VitsConfig or VitsArgs")
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self.args = args
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self.init_multispeaker(config)
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self.init_multilingual(config)
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self.length_scale = args.length_scale
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self.noise_scale = args.noise_scale
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self.inference_noise_scale = args.inference_noise_scale
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self.inference_noise_scale_dp = args.inference_noise_scale_dp
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self.noise_scale_dp = args.noise_scale_dp
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self.max_inference_len = args.max_inference_len
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self.spec_segment_size = args.spec_segment_size
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self.text_encoder = TextEncoder(
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args.num_chars,
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args.hidden_channels,
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args.hidden_channels,
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args.hidden_channels_ffn_text_encoder,
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args.num_heads_text_encoder,
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args.num_layers_text_encoder,
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args.kernel_size_text_encoder,
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args.dropout_p_text_encoder,
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language_emb_dim=self.embedded_language_dim
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)
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self.posterior_encoder = PosteriorEncoder(
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args.out_channels,
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args.hidden_channels,
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args.hidden_channels,
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kernel_size=args.kernel_size_posterior_encoder,
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dilation_rate=args.dilation_rate_posterior_encoder,
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num_layers=args.num_layers_posterior_encoder,
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cond_channels=self.embedded_speaker_dim,
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)
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self.flow = ResidualCouplingBlocks(
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args.hidden_channels,
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args.hidden_channels,
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kernel_size=args.kernel_size_flow,
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dilation_rate=args.dilation_rate_flow,
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num_layers=args.num_layers_flow,
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cond_channels=self.embedded_speaker_dim,
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)
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if args.use_sdp:
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self.duration_predictor = StochasticDurationPredictor(
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args.hidden_channels + self.embedded_language_dim,
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192,
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3,
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args.dropout_p_duration_predictor,
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4,
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cond_channels=self.embedded_speaker_dim,
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language_emb_dim=self.embedded_language_dim,
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)
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else:
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self.duration_predictor = DurationPredictor(
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args.hidden_channels + self.embedded_language_dim,
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256,
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3,
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args.dropout_p_duration_predictor,
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cond_channels=self.embedded_speaker_dim,
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language_emb_dim=self.embedded_language_dim,
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)
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self.waveform_decoder = HifiganGenerator(
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args.hidden_channels,
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1,
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args.resblock_type_decoder,
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args.resblock_dilation_sizes_decoder,
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args.resblock_kernel_sizes_decoder,
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args.upsample_kernel_sizes_decoder,
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args.upsample_initial_channel_decoder,
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args.upsample_rates_decoder,
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inference_padding=0,
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cond_channels=self.embedded_speaker_dim,
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conv_pre_weight_norm=False,
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conv_post_weight_norm=False,
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conv_post_bias=False,
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)
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if args.init_discriminator:
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self.disc = VitsDiscriminator(use_spectral_norm=args.use_spectral_norm_disriminator)
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def init_multispeaker(self, config: Coqpit):
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"""Initialize multi-speaker modules of a model. A model can be trained either with a speaker embedding layer
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or with external `d_vectors` computed from a speaker encoder model.
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Args:
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config (Coqpit): Model configuration.
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data (List, optional): Dataset items to infer number of speakers. Defaults to None.
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"""
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self.embedded_speaker_dim = 0
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if hasattr(config, "model_args"):
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config = config.model_args
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self.num_speakers = config.num_speakers
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if config.use_speaker_embedding:
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self._init_speaker_embedding(config)
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if config.use_d_vector_file:
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self._init_d_vector(config)
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def _init_speaker_embedding(self, config):
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# pylint: disable=attribute-defined-outside-init
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if config.speakers_file is not None:
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self.speaker_manager = SpeakerManager(speaker_id_file_path=config.speakers_file)
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if self.num_speakers > 0:
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print(" > initialization of speaker-embedding layers.")
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self.embedded_speaker_dim = config.speaker_embedding_channels
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self.emb_g = nn.Embedding(self.num_speakers, self.embedded_speaker_dim)
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def _init_d_vector(self, config):
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# pylint: disable=attribute-defined-outside-init
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if hasattr(self, "emb_g"):
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raise ValueError("[!] Speaker embedding layer already initialized before d_vector settings.")
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self.speaker_manager = SpeakerManager(d_vectors_file_path=config.d_vector_file)
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self.embedded_speaker_dim = config.d_vector_dim
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def init_multilingual(self, config: Coqpit, data: List = None):
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"""Initialize multilingual modules of a model.
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Args:
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config (Coqpit): Model configuration.
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data (List, optional): Dataset items to infer number of speakers. Defaults to None.
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"""
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if hasattr(config, "model_args"):
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config = config.model_args
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# init language manager
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self.language_manager = LanguageManager(config, data=data)
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# init language embedding layer
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if config.use_language_embedding:
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self.embedded_language_dim = config.embedded_language_dim
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self.emb_l = nn.Embedding(self.language_manager.num_languages, self.embedded_language_dim)
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torch.nn.init.xavier_uniform_(self.emb_l.weight)
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else:
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self.embedded_language_dim = 0
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self.emb_l = None
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@staticmethod
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def _set_cond_input(aux_input: Dict):
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"""Set the speaker conditioning input based on the multi-speaker mode."""
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sid, g, lid = None, None, None
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if "speaker_ids" in aux_input and aux_input["speaker_ids"] is not None:
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sid = aux_input["speaker_ids"]
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if sid.ndim == 0:
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sid = sid.unsqueeze_(0)
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if "d_vectors" in aux_input and aux_input["d_vectors"] is not None:
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g = F.normalize(aux_input["d_vectors"]).unsqueeze(-1)
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if g.ndim == 2:
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g = g.unsqueeze_(0)
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if "language_ids" in aux_input and aux_input["language_ids"] is not None:
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lid = aux_input["language_ids"]
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if lid.ndim == 0:
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lid = lid.unsqueeze_(0)
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return sid, g, lid
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def get_aux_input(self, aux_input: Dict):
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sid, g = self._set_cond_input(aux_input)
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return {"speaker_id": sid, "style_wav": None, "d_vector": g}
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def forward(
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self,
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x: torch.tensor,
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x_lengths: torch.tensor,
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y: torch.tensor,
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y_lengths: torch.tensor,
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aux_input={"d_vectors": None, "speaker_ids": None, "language_ids": None},
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) -> Dict:
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"""Forward pass of the model.
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Args:
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x (torch.tensor): Batch of input character sequence IDs.
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x_lengths (torch.tensor): Batch of input character sequence lengths.
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y (torch.tensor): Batch of input spectrograms.
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y_lengths (torch.tensor): Batch of input spectrogram lengths.
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aux_input (dict, optional): Auxiliary inputs for multi-speaker training. Defaults to {"d_vectors": None, "speaker_ids": None}.
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Returns:
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Dict: model outputs keyed by the output name.
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Shapes:
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- x: :math:`[B, T_seq]`
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- x_lengths: :math:`[B]`
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- y: :math:`[B, C, T_spec]`
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- y_lengths: :math:`[B]`
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- d_vectors: :math:`[B, C, 1]`
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- speaker_ids: :math:`[B]`
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"""
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outputs = {}
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sid, g, lid = self._set_cond_input(aux_input)
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# speaker embedding
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if self.args.use_speaker_embedding and sid is not None and not self.use_d_vector:
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g = self.emb_g(sid).unsqueeze(-1) # [b, h, 1]
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# language embedding
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if self.args.use_language_embedding:
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lang_emb = self.emb_l(lid).unsqueeze(-1)
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x, m_p, logs_p, x_mask = self.text_encoder(x, x_lengths, lang_emb=lang_emb)
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# posterior encoder
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z, m_q, logs_q, y_mask = self.posterior_encoder(y, y_lengths, g=g)
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# flow layers
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z_p = self.flow(z, y_mask, g=g)
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# find the alignment path
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attn_mask = torch.unsqueeze(x_mask, -1) * torch.unsqueeze(y_mask, 2)
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with torch.no_grad():
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|
o_scale = torch.exp(-2 * logs_p)
|
|
logp1 = torch.sum(-0.5 * math.log(2 * math.pi) - logs_p, [1]).unsqueeze(-1) # [b, t, 1]
|
|
logp2 = torch.einsum("klm, kln -> kmn", [o_scale, -0.5 * (z_p ** 2)])
|
|
logp3 = torch.einsum("klm, kln -> kmn", [m_p * o_scale, z_p])
|
|
logp4 = torch.sum(-0.5 * (m_p ** 2) * o_scale, [1]).unsqueeze(-1) # [b, t, 1]
|
|
logp = logp2 + logp3 + logp1 + logp4
|
|
attn = maximum_path(logp, attn_mask.squeeze(1)).unsqueeze(1).detach()
|
|
|
|
# duration predictor
|
|
attn_durations = attn.sum(3)
|
|
if self.args.use_sdp:
|
|
loss_duration = self.duration_predictor(
|
|
x.detach() if self.args.detach_dp_input else x,
|
|
x_mask,
|
|
attn_durations,
|
|
g=g.detach() if self.args.detach_dp_input and g is not None else g,
|
|
lang_emb=lang_emb.detach() if self.args.detach_dp_input and lang_emb is not None else lang_emb,
|
|
)
|
|
loss_duration = loss_duration / torch.sum(x_mask)
|
|
else:
|
|
attn_log_durations = torch.log(attn_durations + 1e-6) * x_mask
|
|
log_durations = self.duration_predictor(
|
|
x.detach() if self.args.detach_dp_input else x,
|
|
x_mask,
|
|
g=g.detach() if self.args.detach_dp_input and g is not None else g,
|
|
lang_emb=lang_emb.detach() if self.args.detach_dp_input and lang_emb is not None else lang_emb,
|
|
)
|
|
loss_duration = torch.sum((log_durations - attn_log_durations) ** 2, [1, 2]) / torch.sum(x_mask)
|
|
outputs["loss_duration"] = loss_duration
|
|
|
|
# expand prior
|
|
m_p = torch.einsum("klmn, kjm -> kjn", [attn, m_p])
|
|
logs_p = torch.einsum("klmn, kjm -> kjn", [attn, logs_p])
|
|
|
|
# select a random feature segment for the waveform decoder
|
|
z_slice, slice_ids = rand_segments(z, y_lengths, self.spec_segment_size)
|
|
o = self.waveform_decoder(z_slice, g=g)
|
|
outputs.update(
|
|
{
|
|
"model_outputs": o,
|
|
"alignments": attn.squeeze(1),
|
|
"slice_ids": slice_ids,
|
|
"z": z,
|
|
"z_p": z_p,
|
|
"m_p": m_p,
|
|
"logs_p": logs_p,
|
|
"m_q": m_q,
|
|
"logs_q": logs_q,
|
|
}
|
|
)
|
|
return outputs
|
|
|
|
def inference(self, x, aux_input={"d_vectors": None, "speaker_ids": None, "language_ids": None}):
|
|
"""
|
|
Shapes:
|
|
- x: :math:`[B, T_seq]`
|
|
- d_vectors: :math:`[B, C, 1]`
|
|
- speaker_ids: :math:`[B]`
|
|
"""
|
|
sid, g, lid = self._set_cond_input(aux_input)
|
|
x_lengths = torch.tensor(x.shape[1:2]).to(x.device)
|
|
|
|
# speaker embedding
|
|
if self.args.use_speaker_embedding and sid is not None and not self.use_d_vector:
|
|
g = self.emb_g(sid).unsqueeze(-1)
|
|
|
|
# language embedding
|
|
if self.args.use_language_embedding and lid is not None:
|
|
lang_emb = self.emb_l(lid).unsqueeze(-1)
|
|
|
|
x, m_p, logs_p, x_mask = self.text_encoder(x, x_lengths, lang_emb=lang_emb)
|
|
|
|
if self.args.use_sdp:
|
|
logw = self.duration_predictor(x, x_mask, g=g, reverse=True, noise_scale=self.inference_noise_scale_dp, lang_emb=lang_emb)
|
|
else:
|
|
logw = self.duration_predictor(x, x_mask, g=g, lang_emb=lang_emb)
|
|
|
|
w = torch.exp(logw) * x_mask * self.length_scale
|
|
w_ceil = torch.ceil(w)
|
|
y_lengths = torch.clamp_min(torch.sum(w_ceil, [1, 2]), 1).long()
|
|
y_mask = sequence_mask(y_lengths, None).to(x_mask.dtype)
|
|
attn_mask = torch.unsqueeze(x_mask, 2) * torch.unsqueeze(y_mask, -1)
|
|
attn = generate_path(w_ceil.squeeze(1), attn_mask.squeeze(1).transpose(1, 2))
|
|
|
|
m_p = torch.matmul(attn.transpose(1, 2), m_p.transpose(1, 2)).transpose(1, 2)
|
|
logs_p = torch.matmul(attn.transpose(1, 2), logs_p.transpose(1, 2)).transpose(1, 2)
|
|
|
|
z_p = m_p + torch.randn_like(m_p) * torch.exp(logs_p) * self.inference_noise_scale
|
|
z = self.flow(z_p, y_mask, g=g, reverse=True)
|
|
o = self.waveform_decoder((z * y_mask)[:, :, : self.max_inference_len], g=g)
|
|
|
|
outputs = {"model_outputs": o, "alignments": attn.squeeze(1), "z": z, "z_p": z_p, "m_p": m_p, "logs_p": logs_p}
|
|
return outputs
|
|
|
|
def voice_conversion(self, y, y_lengths, sid_src, sid_tgt):
|
|
"""TODO: create an end-point for voice conversion"""
|
|
assert self.num_speakers > 0, "num_speakers have to be larger than 0."
|
|
g_src = self.emb_g(sid_src).unsqueeze(-1)
|
|
g_tgt = self.emb_g(sid_tgt).unsqueeze(-1)
|
|
z, _, _, y_mask = self.enc_q(y, y_lengths, g=g_src)
|
|
z_p = self.flow(z, y_mask, g=g_src)
|
|
z_hat = self.flow(z_p, y_mask, g=g_tgt, reverse=True)
|
|
o_hat = self.waveform_decoder(z_hat * y_mask, g=g_tgt)
|
|
return o_hat, y_mask, (z, z_p, z_hat)
|
|
|
|
def train_step(self, batch: dict, criterion: nn.Module, optimizer_idx: int) -> Tuple[Dict, Dict]:
|
|
"""Perform a single training step. Run the model forward pass and compute losses.
|
|
|
|
Args:
|
|
batch (Dict): Input tensors.
|
|
criterion (nn.Module): Loss layer designed for the model.
|
|
optimizer_idx (int): Index of optimizer to use. 0 for the generator and 1 for the discriminator networks.
|
|
|
|
Returns:
|
|
Tuple[Dict, Dict]: Model ouputs and computed losses.
|
|
"""
|
|
# pylint: disable=attribute-defined-outside-init
|
|
if optimizer_idx not in [0, 1]:
|
|
raise ValueError(" [!] Unexpected `optimizer_idx`.")
|
|
|
|
if optimizer_idx == 0:
|
|
text_input = batch["text_input"]
|
|
text_lengths = batch["text_lengths"]
|
|
mel_lengths = batch["mel_lengths"]
|
|
linear_input = batch["linear_input"]
|
|
d_vectors = batch["d_vectors"]
|
|
speaker_ids = batch["speaker_ids"]
|
|
language_ids = batch["language_ids"]
|
|
waveform = batch["waveform"]
|
|
|
|
# generator pass
|
|
outputs = self.forward(
|
|
text_input,
|
|
text_lengths,
|
|
linear_input.transpose(1, 2),
|
|
mel_lengths,
|
|
aux_input={"d_vectors": d_vectors, "speaker_ids": speaker_ids, "language_ids": language_ids},
|
|
)
|
|
|
|
# cache tensors for the discriminator
|
|
self.y_disc_cache = None
|
|
self.wav_seg_disc_cache = None
|
|
self.y_disc_cache = outputs["model_outputs"]
|
|
wav_seg = segment(
|
|
waveform.transpose(1, 2),
|
|
outputs["slice_ids"] * self.config.audio.hop_length,
|
|
self.args.spec_segment_size * self.config.audio.hop_length,
|
|
)
|
|
self.wav_seg_disc_cache = wav_seg
|
|
outputs["waveform_seg"] = wav_seg
|
|
|
|
# compute discriminator scores and features
|
|
outputs["scores_disc_fake"], outputs["feats_disc_fake"], _, outputs["feats_disc_real"] = self.disc(
|
|
outputs["model_outputs"], wav_seg
|
|
)
|
|
|
|
# compute losses
|
|
with autocast(enabled=False): # use float32 for the criterion
|
|
loss_dict = criterion[optimizer_idx](
|
|
waveform_hat=outputs["model_outputs"].float(),
|
|
waveform=wav_seg.float(),
|
|
z_p=outputs["z_p"].float(),
|
|
logs_q=outputs["logs_q"].float(),
|
|
m_p=outputs["m_p"].float(),
|
|
logs_p=outputs["logs_p"].float(),
|
|
z_len=mel_lengths,
|
|
scores_disc_fake=outputs["scores_disc_fake"],
|
|
feats_disc_fake=outputs["feats_disc_fake"],
|
|
feats_disc_real=outputs["feats_disc_real"],
|
|
loss_duration=outputs["loss_duration"],
|
|
)
|
|
|
|
# handle the duration loss
|
|
if self.args.use_sdp:
|
|
loss_dict["nll_duration"] = outputs["nll_duration"]
|
|
loss_dict["loss"] += outputs["nll_duration"]
|
|
else:
|
|
loss_dict["loss_duration"] = outputs["loss_duration"]
|
|
loss_dict["loss"] += outputs["loss_duration"]
|
|
|
|
elif optimizer_idx == 1:
|
|
# discriminator pass
|
|
outputs = {}
|
|
|
|
# compute scores and features
|
|
outputs["scores_disc_fake"], _, outputs["scores_disc_real"], _ = self.disc(
|
|
self.y_disc_cache.detach(), self.wav_seg_disc_cache
|
|
)
|
|
|
|
# compute loss
|
|
with autocast(enabled=False): # use float32 for the criterion
|
|
loss_dict = criterion[optimizer_idx](
|
|
outputs["scores_disc_real"],
|
|
outputs["scores_disc_fake"],
|
|
)
|
|
return outputs, loss_dict
|
|
|
|
def _log(self, ap, batch, outputs, name_prefix="train"): # pylint: disable=unused-argument,no-self-use
|
|
y_hat = outputs[0]["model_outputs"]
|
|
y = outputs[0]["waveform_seg"]
|
|
figures = plot_results(y_hat, y, ap, name_prefix)
|
|
sample_voice = y_hat[0].squeeze(0).detach().cpu().numpy()
|
|
audios = {f"{name_prefix}/audio": sample_voice}
|
|
|
|
alignments = outputs[0]["alignments"]
|
|
align_img = alignments[0].data.cpu().numpy().T
|
|
|
|
figures.update(
|
|
{
|
|
"alignment": plot_alignment(align_img, output_fig=False),
|
|
}
|
|
)
|
|
|
|
return figures, audios
|
|
|
|
def train_log(
|
|
self, batch: dict, outputs: dict, logger: "Logger", assets: dict, steps: int
|
|
): # pylint: disable=no-self-use
|
|
"""Create visualizations and waveform examples.
|
|
|
|
For example, here you can plot spectrograms and generate sample sample waveforms from these spectrograms to
|
|
be projected onto Tensorboard.
|
|
|
|
Args:
|
|
ap (AudioProcessor): audio processor used at training.
|
|
batch (Dict): Model inputs used at the previous training step.
|
|
outputs (Dict): Model outputs generated at the previoud training step.
|
|
|
|
Returns:
|
|
Tuple[Dict, np.ndarray]: training plots and output waveform.
|
|
"""
|
|
ap = assets["audio_processor"]
|
|
self._log(ap, batch, outputs, "train")
|
|
|
|
@torch.no_grad()
|
|
def eval_step(self, batch: dict, criterion: nn.Module, optimizer_idx: int):
|
|
return self.train_step(batch, criterion, optimizer_idx)
|
|
|
|
def eval_log(self, batch: dict, outputs: dict, logger: "Logger", assets: dict, steps: int) -> None:
|
|
ap = assets["audio_processor"]
|
|
return self._log(ap, batch, outputs, "eval")
|
|
|
|
@torch.no_grad()
|
|
def test_run(self, ap) -> Tuple[Dict, Dict]:
|
|
"""Generic test run for `tts` models used by `Trainer`.
|
|
|
|
You can override this for a different behaviour.
|
|
|
|
Returns:
|
|
Tuple[Dict, Dict]: Test figures and audios to be projected to Tensorboard.
|
|
"""
|
|
print(" | > Synthesizing test sentences.")
|
|
test_audios = {}
|
|
test_figures = {}
|
|
test_sentences = self.config.test_sentences
|
|
|
|
for idx, s_info in enumerate(test_sentences):
|
|
try:
|
|
aux_inputs = self.get_aux_input_from_test_setences(s_info)
|
|
wav, alignment, _, _ = synthesis(
|
|
self,
|
|
aux_inputs["text"],
|
|
self.config,
|
|
"cuda" in str(next(self.parameters()).device),
|
|
ap,
|
|
speaker_id=aux_inputs["speaker_id"],
|
|
d_vector=aux_inputs["d_vector"],
|
|
style_wav=aux_inputs["style_wav"],
|
|
language_id=aux_inputs["language_id"],
|
|
enable_eos_bos_chars=self.config.enable_eos_bos_chars,
|
|
use_griffin_lim=True,
|
|
do_trim_silence=False,
|
|
).values()
|
|
|
|
test_audios["{}-audio".format(idx)] = wav
|
|
test_figures["{}-alignment".format(idx)] = plot_alignment(alignment.T, output_fig=False)
|
|
except: # pylint: disable=bare-except
|
|
print(" !! Error creating Test Sentence -", idx)
|
|
return test_figures, test_audios
|
|
|
|
def get_optimizer(self) -> List:
|
|
"""Initiate and return the GAN optimizers based on the config parameters.
|
|
|
|
It returnes 2 optimizers in a list. First one is for the generator and the second one is for the discriminator.
|
|
|
|
Returns:
|
|
List: optimizers.
|
|
"""
|
|
gen_parameters = chain(
|
|
self.text_encoder.parameters(),
|
|
self.posterior_encoder.parameters(),
|
|
self.flow.parameters(),
|
|
self.duration_predictor.parameters(),
|
|
self.waveform_decoder.parameters(),
|
|
)
|
|
# add the speaker embedding layer
|
|
if hasattr(self, "emb_g"):
|
|
gen_parameters = chain(gen_parameters, self.emb_g.parameters())
|
|
optimizer0 = get_optimizer(
|
|
self.config.optimizer, self.config.optimizer_params, self.config.lr_gen, parameters=gen_parameters
|
|
)
|
|
optimizer1 = get_optimizer(self.config.optimizer, self.config.optimizer_params, self.config.lr_disc, self.disc)
|
|
return [optimizer0, optimizer1]
|
|
|
|
def get_lr(self) -> List:
|
|
"""Set the initial learning rates for each optimizer.
|
|
|
|
Returns:
|
|
List: learning rates for each optimizer.
|
|
"""
|
|
return [self.config.lr_gen, self.config.lr_disc]
|
|
|
|
def get_scheduler(self, optimizer) -> List:
|
|
"""Set the schedulers for each optimizer.
|
|
|
|
Args:
|
|
optimizer (List[`torch.optim.Optimizer`]): List of optimizers.
|
|
|
|
Returns:
|
|
List: Schedulers, one for each optimizer.
|
|
"""
|
|
scheduler0 = get_scheduler(self.config.lr_scheduler_gen, self.config.lr_scheduler_gen_params, optimizer[0])
|
|
scheduler1 = get_scheduler(self.config.lr_scheduler_disc, self.config.lr_scheduler_disc_params, optimizer[1])
|
|
return [scheduler0, scheduler1]
|
|
|
|
def get_criterion(self):
|
|
"""Get criterions for each optimizer. The index in the output list matches the optimizer idx used in
|
|
`train_step()`"""
|
|
from TTS.tts.layers.losses import ( # pylint: disable=import-outside-toplevel
|
|
VitsDiscriminatorLoss,
|
|
VitsGeneratorLoss,
|
|
)
|
|
|
|
return [VitsGeneratorLoss(self.config), VitsDiscriminatorLoss(self.config)]
|
|
|
|
@staticmethod
|
|
def make_symbols(config):
|
|
"""Create a custom arrangement of symbols used by the model. The output list of symbols propagate along the
|
|
whole training and inference steps."""
|
|
_pad = config.characters["pad"]
|
|
_punctuations = config.characters["punctuations"]
|
|
_letters = config.characters["characters"]
|
|
_letters_ipa = config.characters["phonemes"]
|
|
symbols = [_pad] + list(_punctuations) + list(_letters)
|
|
if config.use_phonemes:
|
|
symbols += list(_letters_ipa)
|
|
return symbols
|
|
|
|
@staticmethod
|
|
def get_characters(config: Coqpit):
|
|
if config.characters is not None:
|
|
symbols = Vits.make_symbols(config)
|
|
else:
|
|
from TTS.tts.utils.text.symbols import ( # pylint: disable=import-outside-toplevel
|
|
parse_symbols,
|
|
phonemes,
|
|
symbols,
|
|
)
|
|
|
|
config.characters = parse_symbols()
|
|
if config.use_phonemes:
|
|
symbols = phonemes
|
|
num_chars = len(symbols) + getattr(config, "add_blank", False)
|
|
return symbols, config, num_chars
|
|
|
|
def load_checkpoint(
|
|
self, config, checkpoint_path, eval=False
|
|
): # pylint: disable=unused-argument, redefined-builtin
|
|
"""Load the model checkpoint and setup for training or inference"""
|
|
state = torch.load(checkpoint_path, map_location=torch.device("cpu"))
|
|
self.load_state_dict(state["model"])
|
|
if eval:
|
|
self.eval()
|
|
assert not self.training
|