add Coqpit configs for the TTS models

This commit is contained in:
Eren Gölge 2021-05-06 16:03:09 +02:00
parent 7227e8f1d2
commit 7663bc63c1
9 changed files with 233 additions and 415 deletions

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import importlib
import os
from inspect import isclass
# import all files under configs/
configs_dir = os.path.dirname(__file__)
for file in os.listdir(configs_dir):
path = os.path.join(configs_dir, file)
if not file.startswith("_") and not file.startswith(".") and (file.endswith(".py") or os.path.isdir(path)):
config_name = file[: file.find(".py")] if file.endswith(".py") else file
module = importlib.import_module("TTS.tts.configs." + config_name)
for attribute_name in dir(module):
attribute = getattr(module, attribute_name)
if isclass(attribute):
# Add the class to this package's variables
globals()[attribute_name] = attribute

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{
"attention_heads": 4,
"attention_norm": "sigmoid",
"attention_type": "original",
"audio_config": {
"clip_norm": true,
"do_trim_silence": true,
"fft_size": 1024,
"frame_length_ms": null,
"frame_shift_ms": null,
"griffin_lim_iters": 60,
"hop_length": 256,
"max_norm": 4,
"mel_fmax": 7600,
"mel_fmin": 50,
"min_level_db": -100,
"num_mels": 80,
"power": 1.5,
"preemphasis": 0,
"ref_level_db": 20,
"sample_rate": 22050,
"signal_norm": true,
"spec_gain": 1,
"stats_path": "/home/erogol/Data/LJSpeech-1.1/scale_stats.npy",
"symmetric_norm": true,
"trim_db": 60,
"win_length": 1024
},
"bidirectional_decoder": false,
"compute_input_seq_cache": false,
"ddc_r": 7,
"decoder_diff_spec_alpha": 0.25,
"decoder_loss_alpha": 0.5,
"decoder_ssim_alpha": 0.5,
"double_decoder_consistency": true,
"enable_eos_bos_chars": false,
"forward_attn_mask": false,
"ga_alpha": 5,
"grad_clip": 1,
"gradual_training": [
[
0,
7,
64
],
[
1,
5,
64
],
[
50000,
3,
32
],
[
130000,
2,
32
],
[
290000,
1,
32
]
],
"location_attn": true,
"lr": 0.0001,
"memory_size": -1,
"noam_schedule": false,
"phoneme_cache_path": "/home/erogol/Models/phoneme_cache/",
"phoneme_language": "en-us",
"postnet_diff_spec_alpha": 0.25,
"postnet_loss_alpha": 0.25,
"postnet_ssim_alpha": 0.25,
"prenet_dropout": false,
"prenet_type": "original",
"r": 7,
"separate_stopnet": true,
"seq_len_norm": false,
"stopnet": true,
"stopnet_pos_weight": 15,
"test_sentences_file": null,
"text_cleaner": "phoneme_cleaners",
"training_config": {
"batch_group_size": 4,
"batch_size": 32,
"checkpoint": true,
"datasets": [
{
"meta_file_train": "metadata.csv",
"meta_file_val": null,
"name": "ljspeech",
"path": "/home/erogol/Data/LJSpeech-1.1/"
}
],
"epochs": 1000,
"eval_batch_size": 16,
"keep_after": 10000,
"keep_all_best": false,
"loss_masking": true,
"max_seq_len": 153,
"min_seq_len": 6,
"mixed_precision": true,
"model": "Tacotron2",
"num_loader_workers": 4,
"num_val_loader_workers": 4,
"output_path": "/home/erogol/Models/LJSpeech/",
"print_eval": false,
"print_step": 25,
"run_description": "tacotron2 with DDC and differential spectral loss.",
"run_eval": true,
"run_name": "ljspeech-ddc",
"save_step": 10000,
"tb_model_param_stats": false,
"tb_plot_step": 100,
"test_delay_epochs": 10,
"use_noise_augment": true
},
"transition_agent": false,
"use_forward_attn": false,
"use_phonemes": true,
"warmup_steps": 4000,
"wd": 0.000001,
"windowing": false
}

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{
"model": "glow_tts",
"run_name": "glow-tts-gatedconv",
"run_description": "glow-tts model training with gated conv.",
// AUDIO PARAMETERS
"audio":{
"fft_size": 1024, // number of stft frequency levels. Size of the linear spectogram frame.
"win_length": 1024, // stft window length in ms.
"hop_length": 256, // stft window hop-lengh in ms.
"frame_length_ms": null, // stft window length in ms.If null, 'win_length' is used.
"frame_shift_ms": null, // stft window hop-lengh in ms. If null, 'hop_length' is used.
// Audio processing parameters
"sample_rate": 22050, // DATASET-RELATED: wav sample-rate. If different than the original data, it is resampled.
"preemphasis": 0.0, // pre-emphasis to reduce spec noise and make it more structured. If 0.0, no -pre-emphasis.
"ref_level_db": 0, // reference level db, theoretically 20db is the sound of air.
// Griffin-Lim
"power": 1.1, // value to sharpen wav signals after GL algorithm.
"griffin_lim_iters": 60,// #griffin-lim iterations. 30-60 is a good range. Larger the value, slower the generation.
// Silence trimming
"do_trim_silence": true,// enable trimming of slience of audio as you load it. LJspeech (false), TWEB (false), Nancy (true)
"trim_db": 60, // threshold for timming silence. Set this according to your dataset.
// MelSpectrogram parameters
"num_mels": 80, // size of the mel spec frame.
"mel_fmin": 50.0, // minimum freq level for mel-spec. ~50 for male and ~95 for female voices. Tune for dataset!!
"mel_fmax": 7600.0, // maximum freq level for mel-spec. Tune for dataset!!
"spec_gain": 1.0, // scaler value appplied after log transform of spectrogram.
// Normalization parameters
"signal_norm": true, // normalize spec values. Mean-Var normalization if 'stats_path' is defined otherwise range normalization defined by the other params.
"min_level_db": -100, // lower bound for normalization
"symmetric_norm": true, // move normalization to range [-1, 1]
"max_norm": 1.0, // scale normalization to range [-max_norm, max_norm] or [0, max_norm]
"clip_norm": true, // clip normalized values into the range.
"stats_path": "/home/erogol/Data/LJSpeech-1.1/scale_stats.npy" // DO NOT USE WITH MULTI_SPEAKER MODEL. scaler stats file computed by 'compute_statistics.py'. If it is defined, mean-std based notmalization is used and other normalization params are ignored
},
// VOCABULARY PARAMETERS
// if custom character set is not defined,
// default set in symbols.py is used
// "characters":{
// "pad": "_",
// "eos": "~",
// "bos": "^",
// "characters": "ABCDEFGHIJKLMNOPQRSTUVWXYZabcdefghijklmnopqrstuvwxyz!'(),-.:;? ",
// "punctuations":"!'(),-.:;? ",
// "phonemes":"iyɨʉɯuɪʏʊeøɘəɵɤoɛœɜɞʌɔæɐaɶɑɒᵻʘɓǀɗǃʄǂɠǁʛpbtdʈɖcɟkɡʔɴŋɲɳnɱmʙrʀⱱɾɽɸβfvθðszʃʒʂʐçʝxɣχʁħʕhɦɬɮʋɹɻjɰlɭʎʟˈˌːˑʍwɥʜʢʡɕʑɺɧɚ˞ɫ"
// },
"add_blank": false, // if true add a new token after each token of the sentence. This increases the size of the input sequence, but has considerably improved the prosody of the GlowTTS model.
// DISTRIBUTED TRAINING
"apex_amp_level": null, // APEX amp optimization level. "O1" is currently supported.
"distributed":{
"backend": "nccl",
"url": "tcp:\/\/localhost:54323"
},
"reinit_layers": [], // give a list of layer names to restore from the given checkpoint. If not defined, it reloads all heuristically matching layers.
// MODEL PARAMETERS
"use_mas": false, // use Monotonic Alignment Search if true. Otherwise use pre-computed attention alignments.
// TRAINING
"batch_size": 32, // Batch size for training. Lower values than 32 might cause hard to learn attention. It is overwritten by 'gradual_training'.
"eval_batch_size":16,
"r": 1, // Number of decoder frames to predict per iteration. Set the initial values if gradual training is enabled.
"loss_masking": true, // enable / disable loss masking against the sequence padding.
// VALIDATION
"run_eval": true,
"test_delay_epochs": 0, //Until attention is aligned, testing only wastes computation time.
"test_sentences_file": null, // set a file to load sentences to be used for testing. If it is null then we use default english sentences.
// OPTIMIZER
"noam_schedule": true, // use noam warmup and lr schedule.
"grad_clip": 5.0, // upper limit for gradients for clipping.
"epochs": 10000, // total number of epochs to train.
"lr": 1e-3, // Initial learning rate. If Noam decay is active, maximum learning rate.
"wd": 0.000001, // Weight decay weight.
"warmup_steps": 4000, // Noam decay steps to increase the learning rate from 0 to "lr"
"seq_len_norm": false, // Normalize eash sample loss with its length to alleviate imbalanced datasets. Use it if your dataset is small or has skewed distribution of sequence lengths.
"encoder_type": "gatedconv",
// TENSORBOARD and LOGGING
"print_step": 25, // Number of steps to log training on console.
"tb_plot_step": 100, // Number of steps to plot TB training figures.
"print_eval": false, // If True, it prints intermediate loss values in evalulation.
"save_step": 5000, // Number of training steps expected to save traninpg stats and checkpoints.
"checkpoint": true, // If true, it saves checkpoints per "save_step"
"keep_all_best": false, // If true, keeps all best_models after keep_after steps
"keep_after": 10000, // Global step after which to keep best models if keep_all_best is true
"tb_model_param_stats": false, // true, plots param stats per layer on tensorboard. Might be memory consuming, but good for debugging.
"apex_amp_level": null,
// DATA LOADING
"text_cleaner": "phoneme_cleaners",
"enable_eos_bos_chars": false, // enable/disable beginning of sentence and end of sentence chars.
"num_loader_workers": 4, // number of training data loader processes. Don't set it too big. 4-8 are good values.
"num_val_loader_workers": 4, // number of evaluation data loader processes.
"batch_group_size": 0, //Number of batches to shuffle after bucketing.
"min_seq_len": 3, // DATASET-RELATED: minimum text length to use in training
"max_seq_len": 500, // DATASET-RELATED: maximum text length
"compute_f0": false, // compute f0 values in data-loader
"compute_input_seq_cache": false, // if true, text sequences are computed before starting training. If phonemes are enabled, they are also computed at this stage.
// PATHS
"output_path": "/home/erogol/Models/LJSpeech/",
// PHONEMES
"phoneme_cache_path": "/home/erogol/Models/phoneme_cache/", // phoneme computation is slow, therefore, it caches results in the given folder.
"use_phonemes": true, // use phonemes instead of raw characters. It is suggested for better pronounciation.
"phoneme_language": "en-us", // depending on your target language, pick one from https://github.com/bootphon/phonemizer#languages
// MULTI-SPEAKER and GST
"use_speaker_embedding": false, // use speaker embedding to enable multi-speaker learning.
"style_wav_for_test": null, // path to style wav file to be used in TacotronGST inference.
"use_gst": false, // TACOTRON ONLY: use global style tokens
// DATASETS
"datasets": // List of datasets. They all merged and they get different speaker_ids.
[
{
"name": "ljspeech",
"path": "/home/erogol/Data/LJSpeech-1.1/",
"meta_file_train": "metadata.csv",
"meta_file_val": null
// "path_for_attn": "/home/erogol/Data/LJSpeech-1.1/alignments/"
}
]
}

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{
"model": "glow_tts",
"run_name": "glow-tts-residual_bn_conv",
"run_description": "glow-tts model training with residual BN conv.",
// AUDIO PARAMETERS
"audio":{
"fft_size": 1024, // number of stft frequency levels. Size of the linear spectogram frame.
"win_length": 1024, // stft window length in ms.
"hop_length": 256, // stft window hop-lengh in ms.
"frame_length_ms": null, // stft window length in ms.If null, 'win_length' is used.
"frame_shift_ms": null, // stft window hop-lengh in ms. If null, 'hop_length' is used.
// Audio processing parameters
"sample_rate": 22050, // DATASET-RELATED: wav sample-rate. If different than the original data, it is resampled.
"preemphasis": 0.0, // pre-emphasis to reduce spec noise and make it more structured. If 0.0, no -pre-emphasis.
"ref_level_db": 0, // reference level db, theoretically 20db is the sound of air.
// Griffin-Lim
"power": 1.1, // value to sharpen wav signals after GL algorithm.
"griffin_lim_iters": 60,// #griffin-lim iterations. 30-60 is a good range. Larger the value, slower the generation.
// Silence trimming
"do_trim_silence": true,// enable trimming of slience of audio as you load it. LJspeech (false), TWEB (false), Nancy (true)
"trim_db": 60, // threshold for timming silence. Set this according to your dataset.
// MelSpectrogram parameters
"num_mels": 80, // size of the mel spec frame.
"mel_fmin": 50.0, // minimum freq level for mel-spec. ~50 for male and ~95 for female voices. Tune for dataset!!
"mel_fmax": 7600.0, // maximum freq level for mel-spec. Tune for dataset!!
"spec_gain": 1.0, // scaler value appplied after log transform of spectrogram.00
// Normalization parameters
"signal_norm": false, // normalize spec values. Mean-Var normalization if 'stats_path' is defined otherwise range normalization defined by the other params.
"min_level_db": -100, // lower bound for normalization
"symmetric_norm": true, // move normalization to range [-1, 1]
"max_norm": 1.0, // scale normalization to range [-max_norm, max_norm] or [0, max_norm]
"clip_norm": true, // clip normalized values into the range.
"stats_path": null // DO NOT USE WITH MULTI_SPEAKER MODEL. scaler stats file computed by 'compute_statistics.py'. If it is defined, mean-std based notmalization is used and other normalization params are ignored
},
// VOCABULARY PARAMETERS
// if custom character set is not defined,
// default set in symbols.py is used
// "characters":{
// "pad": "_",
// "eos": "~",
// "bos": "^",
// "characters": "ABCDEFGHIJKLMNOPQRSTUVWXYZabcdefghijklmnopqrstuvwxyz!'(),-.:;? ",
// "punctuations":"!'(),-.:;? ",
// "phonemes":"iyɨʉɯuɪʏʊeøɘəɵɤoɛœɜɞʌɔæɐaɶɑɒᵻʘɓǀɗǃʄǂɠǁʛpbtdʈɖcɟkɡʔɴŋɲɳnɱmʙrʀⱱɾɽɸβfvθðszʃʒʂʐçʝxɣχʁħʕhɦɬɮʋɹɻjɰlɭʎʟˈˌːˑʍwɥʜʢʡɕʑɺɧɚ˞ɫ"
// },
"add_blank": false, // if true add a new token after each token of the sentence. This increases the size of the input sequence, but has considerably improved the prosody of the GlowTTS model.
// DISTRIBUTED TRAINING
"distributed":{
"backend": "nccl",
"url": "tcp:\/\/localhost:54321"
},
"reinit_layers": [], // give a list of layer names to restore from the given checkpoint. If not defined, it reloads all heuristically matching layers.
// MODEL PARAMETERS
// "use_mas": false, // use Monotonic Alignment Search if true. Otherwise use pre-computed attention alignments.
"hidden_channels_encoder": 192,
"hidden_channels_decoder": 192,
"hidden_channels_duration_predictor": 256,
"use_encoder_prenet": true,
"encoder_type": "rel_pos_transformer",
"encoder_params": {
"kernel_size":3,
"dropout_p": 0.1,
"num_layers": 6,
"num_heads": 2,
"hidden_channels_ffn": 768,
"input_length": null
},
// TRAINING
"batch_size": 32, // Batch size for training. Lower values than 32 might cause hard to learn attention. It is overwritten by 'gradual_training'.
"eval_batch_size":16,
"r": 1, // Number of decoder frames to predict per iteration. Set the initial values if gradual training is enabled.
"loss_masking": true, // enable / disable loss masking against the sequence padding.
"mixed_precision": true,
"data_dep_init_iter": 10,
// VALIDATION
"run_eval": true,
"test_delay_epochs": 0, //Until attention is aligned, testing only wastes computation time.
"test_sentences_file": null, // set a file to load sentences to be used for testing. If it is null then we use default english sentences.
// OPTIMIZER
"noam_schedule": true, // use noam warmup and lr schedule.
"grad_clip": 5.0, // upper limit for gradients for clipping.
"epochs": 10000, // total number of epochs to train.
"lr": 1e-3, // Initial learning rate. If Noam decay is active, maximum learning rate.
"wd": 0.000001, // Weight decay weight.
"warmup_steps": 4000, // Noam decay steps to increase the learning rate from 0 to "lr"
"seq_len_norm": false, // Normalize eash sample loss with its length to alleviate imbalanced datasets. Use it if your dataset is small or has skewed distribution of sequence lengths.
// TENSORBOARD and LOGGING
"print_step": 25, // Number of steps to log training on console.
"tb_plot_step": 100, // Number of steps to plot TB training figures.
"print_eval": false, // If True, it prints intermediate loss values in evalulation.
"save_step": 5000, // Number of training steps expected to save traninpg stats and checkpoints.
"checkpoint": true, // If true, it saves checkpoints per "save_step"
"keep_all_best": false, // If true, keeps all best_models after keep_after steps
"keep_after": 10000, // Global step after which to keep best models if keep_all_best is true
"tb_model_param_stats": false, // true, plots param stats per layer on tensorboard. Might be memory consuming, but good for debugging.
// DATA LOADING
"text_cleaner": "phoneme_cleaners",
"enable_eos_bos_chars": false, // enable/disable beginning of sentence and end of sentence chars.
"num_loader_workers": 4, // number of training data loader processes. Don't set it too big. 4-8 are good values.
"num_val_loader_workers": 4, // number of evaluation data loader processes.
"batch_group_size": 0, //Number of batches to shuffle after bucketing.
"min_seq_len": 3, // DATASET-RELATED: minimum text length to use in training
"max_seq_len": 500, // DATASET-RELATED: maximum text length
"compute_f0": false, // compute f0 values in data-loader
"use_noise_augment": true, //add a random noise to audio signal for augmentation at training .
"compute_input_seq_cache": true,
// PATHS
"output_path": "/home/erogol/Models/LJSpeech/",
// PHONEMES
"phoneme_cache_path": "/home/erogol/Models/phoneme_cache/", // phoneme computation is slow, therefore, it caches results in the given folder.
"use_phonemes": true, // use phonemes instead of raw characters. It is suggested for better pronounciation.
"phoneme_language": "en-us", // depending on your target language, pick one from https://github.com/bootphon/phonemizer#languages
// MULTI-SPEAKER and GST
"use_speaker_embedding": false, // use speaker embedding to enable multi-speaker learning.
"use_external_speaker_embedding_file": false,
"style_wav_for_test": null, // path to style wav file to be used in TacotronGST inference.
"use_gst": false, // TACOTRON ONLY: use global style tokens
// DATASETS
"datasets": // List of datasets. They all merged and they get different speaker_ids.
[
{
"name": "ljspeech",
"path": "/home/erogol/Data/LJSpeech-1.1/",
"meta_file_train": "metadata.csv",
"meta_file_val": null
// "path_for_attn": "/home/erogol/Data/LJSpeech-1.1/alignments/"
}
]
}

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from dataclasses import asdict, dataclass, field
from typing import List
from coqpit import MISSING, Coqpit, check_argument
from TTS.config import BaseAudioConfig, BaseDatasetConfig, BaseTrainingConfig
@dataclass
class GSTConfig(Coqpit):
"""Defines Global Style Toke module"""
gst_style_input_wav: str = None
gst_style_input_weights: dict = None
gst_embedding_dim: int = 256
gst_use_speaker_embedding: bool = False
gst_num_heads: int = 4
gst_num_style_tokens: int = 10
def check_values(
self,
):
"""Check config fields"""
c = asdict(self)
super().check_values()
check_argument("gst_style_input_weights", c, restricted=False)
check_argument("gst_style_input_wav", c, restricted=False)
check_argument("gst_embedding_dim", c, restricted=True, min_val=0, max_val=1000)
check_argument("gst_use_speaker_embedding", c, restricted=False)
check_argument("gst_num_heads", c, restricted=True, min_val=2, max_val=10)
check_argument("gst_num_style_tokens", c, restricted=True, min_val=1, max_val=1000)
@dataclass
class CharactersConfig:
"""Defines character or phoneme set used by the model"""
pad: str = None
eos: str = None
bos: str = None
characters: str = None
punctuations: str = None
phonemes: str = None
def check_values(
self,
):
"""Check config fields"""
c = asdict(self)
check_argument("pad", c, "characters", restricted=True)
check_argument("eos", c, "characters", restricted=True)
check_argument("bos", c, "characters", restricted=True)
check_argument("characters", c, "characters", restricted=True)
check_argument("phonemes", c, restricted=True)
check_argument("punctuations", c, "characters", restricted=True)
@dataclass
class BaseTTSConfig(BaseTrainingConfig):
"""Shared parameters among all the tts models."""
audio: BaseAudioConfig = field(default_factory=BaseAudioConfig)
# phoneme settings
use_phonemes: bool = False
phoneme_language: str = None
compute_input_seq_cache: bool = False
text_cleaner: str = MISSING
enable_eos_bos_chars: bool = False
test_sentences_file: str = ""
phoneme_cache_path: str = None
# vocabulary parameters
characters: CharactersConfig = None
# training params
batch_group_size: int = 0
loss_masking: bool = None
# dataloading
min_seq_len: int = 1
max_seq_len: int = float("inf")
compute_f0: bool = False
use_noise_augment: bool = False
add_blank: bool = False
# dataset
datasets: List[BaseDatasetConfig] = field(default_factory=lambda: [BaseDatasetConfig()])

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from dataclasses import dataclass, field
from .shared_configs import BaseTTSConfig
@dataclass
class SpeedySpeechConfig(BaseTTSConfig):
"""Defines parameters for Speedy Speech (feed-forward encoder-decoder) based models."""
model: str = "speedy_speech"
# model specific params
positional_encoding: bool = True
hidden_channels: int = 128
encoder_type: str = "residual_conv_bn"
encoder_params: dict = field(
default_factory=lambda: {
"kernel_size": 4,
"dilations": [1, 2, 4, 1, 2, 4, 1, 2, 4, 1, 2, 4, 1],
"num_conv_blocks": 2,
"num_res_blocks": 13,
}
)
decoder_type: str = "residual_conv_bn"
decoder_params: dict = field(
default_factory=lambda: {
"kernel_size": 4,
"dilations": [1, 2, 4, 8, 1, 2, 4, 8, 1, 2, 4, 8, 1, 2, 4, 8, 1],
"num_conv_blocks": 2,
"num_res_blocks": 17,
}
)
# multi-speaker settings
use_speaker_embedding: bool = False
use_external_speaker_embedding_file: bool = False
external_speaker_embedding_file: str = False
# optimizer parameters
noam_schedule: bool = False
warmup_steps: int = 4000
lr: float = 1e-4
wd: float = 1e-6
grad_clip: float = 5.0
# loss params
ssim_alpha: float = 1.0
huber_alpha: float = 1.0
l1_alpha: float = 1.0
# overrides
min_seq_len: int = 13
max_seq_len: int = 200
r: int = 1

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from dataclasses import dataclass
from TTS.tts.configs.tacotron_config import TacotronConfig
@dataclass
class Tacotron2Config(TacotronConfig):
"""Defines parameters for Tacotron2 based models."""
model: str = "tacotron2"

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from dataclasses import asdict, dataclass
from typing import List
from coqpit import check_argument
from .shared_configs import BaseTTSConfig, GSTConfig
@dataclass
class TacotronConfig(BaseTTSConfig):
"""Defines parameters for Tacotron based models."""
model: str = "tacotron"
gst: GSTConfig = None
gst_style_input: str = None
# model specific params
r: int = 2
gradual_training: List = None
memory_size: int = -1
prenet_type: str = "original"
prenet_dropout: bool = True
prenet_dropout_at_inference: bool = False
stopnet: bool = True
separate_stopnet: bool = True
stopnet_pos_weight: float = 10.0
# attention layers
attention_type: str = "original"
attention_heads: int = None
attention_norm: str = "sigmoid"
windowing: bool = False
use_forward_attn: bool = False
forward_attn_mask: bool = False
transition_agent: bool = False
location_attn: bool = True
# advance methods
bidirectional_decoder: bool = False
double_decoder_consistency: bool = False
ddc_r: int = 6
# multi-speaker settings
use_speaker_embedding: bool = False
use_external_speaker_embedding_file: bool = False
external_speaker_embedding_file: str = False
# optimizer parameters
noam_schedule: bool = False
warmup_steps: int = 4000
lr: float = 1e-4
wd: float = 1e-6
grad_clip: float = 5.0
seq_len_norm: bool = False
loss_masking: bool = True
# loss params
decoder_loss_alpha: float = 0.25
postnet_loss_alpha: float = 0.25
postnet_diff_spec_alpha: float = 0.25
decoder_diff_spec_alpha: float = 0.25
decoder_ssim_alpha: float = 0.25
postnet_ssim_alpha: float = 0.25
ga_alpha: float = 5.0
@dataclass
class Tacotron2Config(TacotronConfig):
"""Defines parameters for Tacotron2 based models."""
model: str = "tacotron2"