mirror of https://github.com/coqui-ai/TTS.git
update train scripts for coqpit
This commit is contained in:
Eren Gölge
parent
045f1c3e76
commit
78b3825d0b
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@ -15,7 +15,7 @@ from torch.optim import Adam
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from torch.utils.data import DataLoader
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from torch.utils.data.distributed import DistributedSampler
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from TTS.utils.arguments import parse_arguments, process_args
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from TTS.utils.arguments import init_training
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from TTS.utils.audio import AudioProcessor
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from TTS.utils.distribute import init_distributed
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from TTS.utils.generic_utils import KeepAverage, count_parameters, remove_experiment_folder, set_init_dict
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@ -131,12 +131,12 @@ def train(model, criterion, optimizer, scheduler, scaler, ap, global_step, epoch
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if c.mixed_precision:
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scaler.scale(loss).backward()
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scaler.unscale_(optimizer)
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grad_norm = torch.nn.utils.clip_grad_norm_(model.parameters(), c.clip_grad)
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grad_norm = torch.nn.utils.clip_grad_norm_(model.parameters(), c.grad_clip)
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scaler.step(optimizer)
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scaler.update()
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else:
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loss.backward()
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grad_norm = torch.nn.utils.clip_grad_norm_(model.parameters(), c.clip_grad)
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grad_norm = torch.nn.utils.grad_clip_norm_(model.parameters(), c.clip_grad)
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optimizer.step()
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# schedule update
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@ -311,7 +311,7 @@ def main(args): # pylint: disable=redefined-outer-name
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eval_data, train_data = load_wav_data(c.data_path, c.eval_split_size)
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# setup audio processor
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ap = AudioProcessor(**c.audio)
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ap = AudioProcessor(**c.audio.to_dict())
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# DISTRUBUTED
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if num_gpus > 1:
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@ -416,9 +416,7 @@ def main(args): # pylint: disable=redefined-outer-name
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if __name__ == "__main__":
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args = parse_arguments(sys.argv)
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c, OUT_PATH, AUDIO_PATH, c_logger, tb_logger = process_args(args, model_class="vocoder")
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args, c, OUT_PATH, AUDIO_PATH, c_logger, tb_logger = init_training(sys.argv)
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try:
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main(args)
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except KeyboardInterrupt:
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@ -11,7 +11,7 @@ import torch
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from torch.utils.data import DataLoader
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from TTS.tts.utils.visual import plot_spectrogram
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from TTS.utils.arguments import parse_arguments, process_args
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from TTS.utils.arguments import init_training
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from TTS.utils.audio import AudioProcessor
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from TTS.utils.generic_utils import KeepAverage, count_parameters, remove_experiment_folder, set_init_dict
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from TTS.utils.radam import RAdam
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@ -307,29 +307,7 @@ def main(args): # pylint: disable=redefined-outer-name
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global train_data, eval_data
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# setup audio processor
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ap = AudioProcessor(**c.audio)
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# print(f" > Loading wavs from: {c.data_path}")
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# if c.feature_path is not None:
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# print(f" > Loading features from: {c.feature_path}")
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# eval_data, train_data = load_wav_feat_data(
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# c.data_path, c.feature_path, c.eval_split_size
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# )
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# else:
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# mel_feat_path = os.path.join(OUT_PATH, "mel")
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# feat_data = find_feat_files(mel_feat_path)
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# if feat_data:
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# print(f" > Loading features from: {mel_feat_path}")
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# eval_data, train_data = load_wav_feat_data(
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# c.data_path, mel_feat_path, c.eval_split_size
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# )
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# else:
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# print(" > No feature data found. Preprocessing...")
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# # preprocessing feature data from given wav files
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# preprocess_wav_files(OUT_PATH, CONFIG, ap)
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# eval_data, train_data = load_wav_feat_data(
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# c.data_path, mel_feat_path, c.eval_split_size
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# )
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ap = AudioProcessor(**c.audio.to_dict())
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print(f" > Loading wavs from: {c.data_path}")
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if c.feature_path is not None:
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@ -438,9 +416,7 @@ def main(args): # pylint: disable=redefined-outer-name
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if __name__ == "__main__":
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args = parse_arguments(sys.argv)
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c, OUT_PATH, AUDIO_PATH, c_logger, tb_logger = process_args(args, model_class="vocoder")
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args, c, OUT_PATH, AUDIO_PATH, c_logger, tb_logger = init_training(sys.argv)
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try:
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main(args)
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except KeyboardInterrupt:
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@ -1,164 +0,0 @@
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{
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"run_name": "hifigan",
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"run_description": "hifigan mean-var scaling",
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// AUDIO PARAMETERS
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"audio":{
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"fft_size": 1024, // number of stft frequency levels. Size of the linear spectogram frame.
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"win_length": 1024, // stft window length in ms.
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"hop_length": 256, // stft window hop-lengh in ms.
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"frame_length_ms": null, // stft window length in ms.If null, 'win_length' is used.
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"frame_shift_ms": null, // stft window hop-lengh in ms. If null, 'hop_length' is used.
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// Audio processing parameters
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"sample_rate": 22050, // DATASET-RELATED: wav sample-rate. If different than the original data, it is resampled.
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"preemphasis": 0.0, // pre-emphasis to reduce spec noise and make it more structured. If 0.0, no -pre-emphasis.
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"ref_level_db": 20, // reference level db, theoretically 20db is the sound of air.
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"log_func": "np.log10",
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"do_sound_norm": true,
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// Silence trimming
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"do_trim_silence": false,// enable trimming of slience of audio as you load it. LJspeech (false), TWEB (false), Nancy (true)
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"trim_db": 60, // threshold for timming silence. Set this according to your dataset.
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// MelSpectrogram parameters
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"num_mels": 80, // size of the mel spec frame.
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"mel_fmin": 50.0, // minimum freq level for mel-spec. ~50 for male and ~95 for female voices. Tune for dataset!!
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"mel_fmax": 7600.0, // maximum freq level for mel-spec. Tune for dataset!!
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"spec_gain": 1.0, // scaler value appplied after log transform of spectrogram.
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// Normalization parameters
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"signal_norm": true, // normalize spec values. Mean-Var normalization if 'stats_path' is defined otherwise range normalization defined by the other params.
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"min_level_db": -100, // lower bound for normalization
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"symmetric_norm": true, // move normalization to range [-1, 1]
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"max_norm": 4.0, // scale normalization to range [-max_norm, max_norm] or [0, max_norm]
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"clip_norm": true, // clip normalized values into the range.
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"stats_path": "/home/erogol/.local/share/tts/tts_models--en--ljspeech--speedy-speech-wn/scale_stats.npy"
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},
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// DISTRIBUTED TRAINING
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"distributed":{
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"backend": "nccl",
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"url": "tcp:\/\/localhost:54324"
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},
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// MODEL PARAMETERS
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"use_pqmf": false,
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// LOSS PARAMETERS
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"use_stft_loss": false,
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"use_subband_stft_loss": false,
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"use_mse_gan_loss": true,
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"use_hinge_gan_loss": false,
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"use_feat_match_loss": true, // use only with melgan discriminators
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"use_l1_spec_loss": true,
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// loss weights
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"stft_loss_weight": 0,
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"subband_stft_loss_weight": 0,
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"mse_G_loss_weight": 1,
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"hinge_G_loss_weight": 0,
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"feat_match_loss_weight": 10,
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"l1_spec_loss_weight": 45,
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// multiscale stft loss parameters
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// "stft_loss_params": {
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// "n_ffts": [1024, 2048, 512],
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// "hop_lengths": [120, 240, 50],
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// "win_lengths": [600, 1200, 240]
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// },
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"l1_spec_loss_params": {
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"use_mel": true,
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"sample_rate": 22050,
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"n_fft": 1024,
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"hop_length": 256,
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"win_length": 1024,
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"n_mels": 80,
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"mel_fmin": 0.0,
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"mel_fmax": null
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},
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"target_loss": "avg_G_loss", // loss value to pick the best model to save after each epoch
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// DISCRIMINATOR
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"discriminator_model": "hifigan_discriminator",
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//"discriminator_model_params":{
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// "peroids": [2, 3, 5, 7, 11],
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// "base_channels": 16,
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// "max_channels":512,
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// "downsample_factors":[4, 4, 4]
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//},
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"steps_to_start_discriminator": 0, // steps required to start GAN trainining.1
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"diff_samples_for_G_and_D": false, // draw a new sample from the dataset for the D pass.
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// GENERATOR
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"generator_model": "hifigan_generator",
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"generator_model_params": {
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"upsample_factors":[8,8,2,2],
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"upsample_kernel_sizes": [16,16,4,4],
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"upsample_initial_channel": 512,
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"resblock_kernel_sizes": [3,7,11],
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"resblock_dilation_sizes": [[1,3,5], [1,3,5], [1,3,5]],
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"resblock_type": "1"
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},
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// DATASET
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"data_path": "/home/erogol/gdrive/Datasets/LJSpeech-1.1/wavs/",
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"feature_path": null,
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// "feature_path": "/home/erogol/gdrive/Datasets/non-binary-voice-files/tacotron-DCA/",
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"seq_len": 8192,
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"pad_short": 2000,
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"conv_pad": 0,
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"use_noise_augment": false,
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"use_cache": true,
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"reinit_layers": [], // give a list of layer names to restore from the given checkpoint. If not defined, it reloads all heuristically matching layers.
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// TRAINING
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"batch_size": 16, // Batch size for training. Lower values than 32 might cause hard to learn attention. It is overwritten by 'gradual_training'.
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// VALIDATION
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"run_eval": true,
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"test_delay_epochs": 10, //Until attention is aligned, testing only wastes computation time.
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"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.
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// OPTIMIZER
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"epochs": 10000, // total number of epochs to train.
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"wd": 0.0, // Weight decay weight.
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"gen_clip_grad": -1, // Generator gradient clipping threshold. Apply gradient clipping if > 0
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"disc_clip_grad": -1, // Discriminator gradient clipping threshold.
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"lr_gen": 0.0002, // Initial learning rate. If Noam decay is active, maximum learning rate.
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"lr_disc": 0.0002,
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"optimizer": "AdamW",
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"optimizer_params":{
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"betas": [0.8, 0.99],
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"weight_decay": 0.0
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},
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"lr_scheduler_gen": "ExponentialLR", // one of the schedulers from https://pytorch.org/docs/stable/optim.html#how-to-adjust-learning-rate
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"lr_scheduler_gen_params": {
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"gamma": 0.999,
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"last_epoch": -1
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},
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"lr_scheduler_disc": "ExponentialLR", // one of the schedulers from https://pytorch.org/docs/stable/optim.html#how-to-adjust-learning-rate
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"lr_scheduler_disc_params": {
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"gamma": 0.999,
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"last_epoch": -1
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},
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// TENSORBOARD and LOGGING
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"print_step": 25, // Number of steps to log traning on console.
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"print_eval": false, // If True, it prints loss values for each step in eval run.
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"save_step": 25000, // Number of training steps expected to plot training stats on TB and save model checkpoints.
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"checkpoint": true, // If true, it saves checkpoints per "save_step"
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"tb_model_param_stats": true, // true, plots param stats per layer on tensorboard. Might be memory consuming, but good for debugging.
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// DATA LOADING
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"num_loader_workers": 8, // number of training data loader processes. Don't set it too big. 4-8 are good values.
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"num_val_loader_workers": 4, // number of evaluation data loader processes.
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"eval_split_size": 10,
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// PATHS
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"output_path": "/home/erogol/gdrive/Trainings/LJSpeech/"
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}
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