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| from dataclasses import dataclass, field | |
| from typing import List, Optional | |
| from coqpit import Coqpit | |
| from TTS.vc.configs.shared_configs import BaseVCConfig | |
| class FreeVCAudioConfig(Coqpit): | |
| """Audio configuration | |
| Args: | |
| max_wav_value (float): | |
| The maximum value of the waveform. | |
| input_sample_rate (int): | |
| The sampling rate of the input waveform. | |
| output_sample_rate (int): | |
| The sampling rate of the output waveform. | |
| filter_length (int): | |
| The length of the filter. | |
| hop_length (int): | |
| The hop length. | |
| win_length (int): | |
| The window length. | |
| n_mel_channels (int): | |
| The number of mel channels. | |
| mel_fmin (float): | |
| The minimum frequency of the mel filterbank. | |
| mel_fmax (Optional[float]): | |
| The maximum frequency of the mel filterbank. | |
| """ | |
| max_wav_value: float = field(default=32768.0) | |
| input_sample_rate: int = field(default=16000) | |
| output_sample_rate: int = field(default=24000) | |
| filter_length: int = field(default=1280) | |
| hop_length: int = field(default=320) | |
| win_length: int = field(default=1280) | |
| n_mel_channels: int = field(default=80) | |
| mel_fmin: float = field(default=0.0) | |
| mel_fmax: Optional[float] = field(default=None) | |
| class FreeVCArgs(Coqpit): | |
| """FreeVC model arguments | |
| Args: | |
| spec_channels (int): | |
| The number of channels in the spectrogram. | |
| inter_channels (int): | |
| The number of channels in the intermediate layers. | |
| hidden_channels (int): | |
| The number of channels in the hidden layers. | |
| filter_channels (int): | |
| The number of channels in the filter layers. | |
| n_heads (int): | |
| The number of attention heads. | |
| n_layers (int): | |
| The number of layers. | |
| kernel_size (int): | |
| The size of the kernel. | |
| p_dropout (float): | |
| The dropout probability. | |
| resblock (str): | |
| The type of residual block. | |
| resblock_kernel_sizes (List[int]): | |
| The kernel sizes for the residual blocks. | |
| resblock_dilation_sizes (List[List[int]]): | |
| The dilation sizes for the residual blocks. | |
| upsample_rates (List[int]): | |
| The upsample rates. | |
| upsample_initial_channel (int): | |
| The number of channels in the initial upsample layer. | |
| upsample_kernel_sizes (List[int]): | |
| The kernel sizes for the upsample layers. | |
| n_layers_q (int): | |
| The number of layers in the quantization network. | |
| use_spectral_norm (bool): | |
| Whether to use spectral normalization. | |
| gin_channels (int): | |
| The number of channels in the global conditioning vector. | |
| ssl_dim (int): | |
| The dimension of the self-supervised learning embedding. | |
| use_spk (bool): | |
| Whether to use external speaker encoder. | |
| """ | |
| spec_channels: int = field(default=641) | |
| inter_channels: int = field(default=192) | |
| hidden_channels: int = field(default=192) | |
| filter_channels: int = field(default=768) | |
| n_heads: int = field(default=2) | |
| n_layers: int = field(default=6) | |
| kernel_size: int = field(default=3) | |
| p_dropout: float = field(default=0.1) | |
| resblock: str = field(default="1") | |
| resblock_kernel_sizes: List[int] = field(default_factory=lambda: [3, 7, 11]) | |
| resblock_dilation_sizes: List[List[int]] = field(default_factory=lambda: [[1, 3, 5], [1, 3, 5], [1, 3, 5]]) | |
| upsample_rates: List[int] = field(default_factory=lambda: [10, 8, 2, 2]) | |
| upsample_initial_channel: int = field(default=512) | |
| upsample_kernel_sizes: List[int] = field(default_factory=lambda: [16, 16, 4, 4]) | |
| n_layers_q: int = field(default=3) | |
| use_spectral_norm: bool = field(default=False) | |
| gin_channels: int = field(default=256) | |
| ssl_dim: int = field(default=1024) | |
| use_spk: bool = field(default=False) | |
| num_spks: int = field(default=0) | |
| segment_size: int = field(default=8960) | |
| class FreeVCConfig(BaseVCConfig): | |
| """Defines parameters for FreeVC End2End TTS model. | |
| Args: | |
| model (str): | |
| Model name. Do not change unless you know what you are doing. | |
| model_args (FreeVCArgs): | |
| Model architecture arguments. Defaults to `FreeVCArgs()`. | |
| audio (FreeVCAudioConfig): | |
| Audio processing configuration. Defaults to `FreeVCAudioConfig()`. | |
| grad_clip (List): | |
| Gradient clipping thresholds for each optimizer. Defaults to `[1000.0, 1000.0]`. | |
| lr_gen (float): | |
| Initial learning rate for the generator. Defaults to 0.0002. | |
| lr_disc (float): | |
| Initial learning rate for the discriminator. Defaults to 0.0002. | |
| lr_scheduler_gen (str): | |
| Name of the learning rate scheduler for the generator. One of the `torch.optim.lr_scheduler.*`. Defaults to | |
| `ExponentialLR`. | |
| lr_scheduler_gen_params (dict): | |
| Parameters for the learning rate scheduler of the generator. Defaults to `{'gamma': 0.999875, "last_epoch":-1}`. | |
| lr_scheduler_disc (str): | |
| Name of the learning rate scheduler for the discriminator. One of the `torch.optim.lr_scheduler.*`. Defaults to | |
| `ExponentialLR`. | |
| lr_scheduler_disc_params (dict): | |
| Parameters for the learning rate scheduler of the discriminator. Defaults to `{'gamma': 0.999875, "last_epoch":-1}`. | |
| scheduler_after_epoch (bool): | |
| If true, step the schedulers after each epoch else after each step. Defaults to `False`. | |
| optimizer (str): | |
| Name of the optimizer to use with both the generator and the discriminator networks. One of the | |
| `torch.optim.*`. Defaults to `AdamW`. | |
| kl_loss_alpha (float): | |
| Loss weight for KL loss. Defaults to 1.0. | |
| disc_loss_alpha (float): | |
| Loss weight for the discriminator loss. Defaults to 1.0. | |
| gen_loss_alpha (float): | |
| Loss weight for the generator loss. Defaults to 1.0. | |
| feat_loss_alpha (float): | |
| Loss weight for the feature matching loss. Defaults to 1.0. | |
| mel_loss_alpha (float): | |
| Loss weight for the mel loss. Defaults to 45.0. | |
| return_wav (bool): | |
| If true, data loader returns the waveform as well as the other outputs. Do not change. Defaults to `True`. | |
| compute_linear_spec (bool): | |
| If true, the linear spectrogram is computed and returned alongside the mel output. Do not change. Defaults to `True`. | |
| use_weighted_sampler (bool): | |
| If true, use weighted sampler with bucketing for balancing samples between datasets used in training. Defaults to `False`. | |
| weighted_sampler_attrs (dict): | |
| Key retuned by the formatter to be used for weighted sampler. For example `{"root_path": 2.0, "speaker_name": 1.0}` sets sample probabilities | |
| by overweighting `root_path` by 2.0. Defaults to `{}`. | |
| weighted_sampler_multipliers (dict): | |
| Weight each unique value of a key returned by the formatter for weighted sampling. | |
| For example `{"root_path":{"/raid/datasets/libritts-clean-16khz-bwe-coqui_44khz/LibriTTS/train-clean-100/":1.0, "/raid/datasets/libritts-clean-16khz-bwe-coqui_44khz/LibriTTS/train-clean-360/": 0.5}`. | |
| It will sample instances from `train-clean-100` 2 times more than `train-clean-360`. Defaults to `{}`. | |
| r (int): | |
| Number of spectrogram frames to be generated at a time. Do not change. Defaults to `1`. | |
| add_blank (bool): | |
| If true, a blank token is added in between every character. Defaults to `True`. | |
| test_sentences (List[List]): | |
| List of sentences with speaker and language information to be used for testing. | |
| language_ids_file (str): | |
| Path to the language ids file. | |
| use_language_embedding (bool): | |
| If true, language embedding is used. Defaults to `False`. | |
| Note: | |
| Check :class:`TTS.tts.configs.shared_configs.BaseTTSConfig` for the inherited parameters. | |
| Example: | |
| >>> from TTS.vc.configs.freevc_config import FreeVCConfig | |
| >>> config = FreeVCConfig() | |
| """ | |
| model: str = "freevc" | |
| # model specific params | |
| model_args: FreeVCArgs = field(default_factory=FreeVCArgs) | |
| audio: FreeVCAudioConfig = field(default_factory=FreeVCAudioConfig) | |
| # optimizer | |
| # TODO with training support | |
| # loss params | |
| # TODO with training support | |
| # data loader params | |
| return_wav: bool = True | |
| compute_linear_spec: bool = True | |
| # sampler params | |
| use_weighted_sampler: bool = False # TODO: move it to the base config | |
| weighted_sampler_attrs: dict = field(default_factory=lambda: {}) | |
| weighted_sampler_multipliers: dict = field(default_factory=lambda: {}) | |
| # overrides | |
| r: int = 1 # DO NOT CHANGE | |
| add_blank: bool = True | |
| # multi-speaker settings | |
| # use speaker embedding layer | |
| num_speakers: int = 0 | |
| speakers_file: str = None | |
| speaker_embedding_channels: int = 256 | |
| # use d-vectors | |
| use_d_vector_file: bool = False | |
| d_vector_file: List[str] = None | |
| d_vector_dim: int = None | |
| def __post_init__(self): | |
| for key, val in self.model_args.items(): | |
| if hasattr(self, key): | |
| self[key] = val | |