예제 #1
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def main():
    """Run `tts` model training directly by a `config.json` file."""
    # init trainer args
    train_args = TrainingArgs()
    parser = train_args.init_argparse(arg_prefix="")

    # override trainer args from comman-line args
    args, config_overrides = parser.parse_known_args()
    train_args.parse_args(args)

    # load config.json and register
    if args.config_path or args.continue_path:
        if args.config_path:
            # init from a file
            config = load_config(args.config_path)
            if len(config_overrides) > 0:
                config.parse_known_args(config_overrides, relaxed_parser=True)
        elif args.continue_path:
            # continue from a prev experiment
            config = load_config(
                os.path.join(args.continue_path, "config.json"))
            if len(config_overrides) > 0:
                config.parse_known_args(config_overrides, relaxed_parser=True)
        else:
            # init from console args
            from TTS.config.shared_configs import BaseTrainingConfig  # pylint: disable=import-outside-toplevel

            config_base = BaseTrainingConfig()
            config_base.parse_known_args(config_overrides)
            config = register_config(config_base.model)()

    # load training samples
    if "feature_path" in config and config.feature_path:
        # load pre-computed features
        print(f" > Loading features from: {config.feature_path}")
        eval_samples, train_samples = load_wav_feat_data(
            config.data_path, config.feature_path, config.eval_split_size)
    else:
        # load data raw wav files
        eval_samples, train_samples = load_wav_data(config.data_path,
                                                    config.eval_split_size)

    # setup audio processor
    ap = AudioProcessor(**config.audio)

    # init the model from config
    model = setup_model(config)

    # init the trainer and 🚀
    trainer = Trainer(
        train_args,
        config,
        config.output_path,
        model=model,
        train_samples=train_samples,
        eval_samples=eval_samples,
        training_assets={"audio_processor": ap},
        parse_command_line_args=False,
    )
    trainer.fit()
예제 #2
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def main():
    """
    Call train.py as a new process and pass command arguments
    """
    parser = TrainingArgs().init_argparse(arg_prefix="")
    parser.add_argument("--script",
                        type=str,
                        help="Target training script to distibute.")
    args, unargs = parser.parse_known_args()

    num_gpus = torch.cuda.device_count()
    group_id = time.strftime("%Y_%m_%d-%H%M%S")

    # set arguments for train.py
    folder_path = pathlib.Path(__file__).parent.absolute()
    if os.path.exists(os.path.join(folder_path, args.script)):
        command = [os.path.join(folder_path, args.script)]
    else:
        command = [args.script]
    command.append("--continue_path={}".format(args.continue_path))
    command.append("--restore_path={}".format(args.restore_path))
    command.append("--config_path={}".format(args.config_path))
    command.append("--group_id=group_{}".format(group_id))
    command.append("--use_ddp=true")
    command += unargs
    command.append("")

    # run processes
    processes = []
    for i in range(num_gpus):
        my_env = os.environ.copy()
        my_env["PYTHON_EGG_CACHE"] = "/tmp/tmp{}".format(i)
        command[-1] = "--rank={}".format(i)
        # prevent stdout for processes with rank != 0
        stdout = None
        p = subprocess.Popen(["python3"] + command, stdout=stdout, env=my_env)  # pylint: disable=consider-using-with
        processes.append(p)
        print(command)

    for p in processes:
        p.wait()
예제 #3
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    optimizer="Adam",
    lr_scheduler=None,
    lr=3e-5,
)

# init audio processor
ap = AudioProcessor(**config.audio.to_dict())

# load training samples
train_samples, eval_samples = load_tts_samples(dataset_config, eval_split=True)

# init speaker manager for multi-speaker training
# it mainly handles speaker-id to speaker-name for the model and the data-loader
speaker_manager = SpeakerManager()
speaker_manager.set_speaker_ids_from_data(train_samples + eval_samples)

# init model
model = Tacotron2(config, speaker_manager)

# init the trainer and 🚀
trainer = Trainer(
    TrainingArgs(),
    config,
    output_path,
    model=model,
    train_samples=train_samples,
    eval_samples=eval_samples,
    training_assets={"audio_processor": ap},
)
trainer.fit()
예제 #4
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import os

from TTS.trainer import Trainer, TrainingArgs, init_training
from TTS.tts.configs import AlignTTSConfig, BaseDatasetConfig

output_path = os.path.dirname(os.path.abspath(__file__))
dataset_config = BaseDatasetConfig(
    name="ljspeech", meta_file_train="metadata.csv", path=os.path.join(output_path, "../LJSpeech-1.1/")
)
config = AlignTTSConfig(
    batch_size=32,
    eval_batch_size=16,
    num_loader_workers=4,
    num_eval_loader_workers=4,
    run_eval=True,
    test_delay_epochs=-1,
    epochs=1000,
    text_cleaner="english_cleaners",
    use_phonemes=False,
    phoneme_language="en-us",
    phoneme_cache_path=os.path.join(output_path, "phoneme_cache"),
    print_step=25,
    print_eval=True,
    mixed_precision=False,
    output_path=output_path,
    datasets=[dataset_config],
)
args, config, output_path, _, c_logger, dashboard_logger = init_training(TrainingArgs(), config)
trainer = Trainer(args, config, output_path, c_logger, dashboard_logger)
trainer.fit()
예제 #5
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import os

from TTS.trainer import Trainer, TrainingArgs, init_training
from TTS.vocoder.configs import UnivnetConfig

output_path = os.path.dirname(os.path.abspath(__file__))
config = UnivnetConfig(
    batch_size=64,
    eval_batch_size=16,
    num_loader_workers=4,
    num_eval_loader_workers=4,
    run_eval=True,
    test_delay_epochs=-1,
    epochs=1000,
    seq_len=8192,
    pad_short=2000,
    use_noise_augment=True,
    eval_split_size=10,
    print_step=25,
    print_eval=False,
    mixed_precision=False,
    lr_gen=1e-4,
    lr_disc=1e-4,
    data_path=os.path.join(output_path, "../LJSpeech-1.1/wavs/"),
    output_path=output_path,
)
args, config, output_path, _, c_logger, dashboard_logger = init_training(
    TrainingArgs(), config)
trainer = Trainer(args, config, output_path, c_logger, dashboard_logger)
trainer.fit()
예제 #6
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def main():
    """Run `tts` model training directly by a `config.json` file."""
    # init trainer args
    train_args = TrainingArgs()
    parser = train_args.init_argparse(arg_prefix="")

    # override trainer args from comman-line args
    args, config_overrides = parser.parse_known_args()
    train_args.parse_args(args)

    # load config.json and register
    if args.config_path or args.continue_path:
        if args.config_path:
            # init from a file
            config = load_config(args.config_path)
            if len(config_overrides) > 0:
                config.parse_known_args(config_overrides, relaxed_parser=True)
        elif args.continue_path:
            # continue from a prev experiment
            config = load_config(
                os.path.join(args.continue_path, "config.json"))
            if len(config_overrides) > 0:
                config.parse_known_args(config_overrides, relaxed_parser=True)
        else:
            # init from console args
            from TTS.config.shared_configs import BaseTrainingConfig  # pylint: disable=import-outside-toplevel

            config_base = BaseTrainingConfig()
            config_base.parse_known_args(config_overrides)
            config = register_config(config_base.model)()

    # load training samples
    train_samples, eval_samples = load_tts_samples(config.datasets,
                                                   eval_split=True)

    # setup audio processor
    ap = AudioProcessor(**config.audio)

    # init speaker manager
    if check_config_and_model_args(config, "use_speaker_embedding", True):
        speaker_manager = SpeakerManager(data_items=train_samples +
                                         eval_samples)
        if hasattr(config, "model_args"):
            config.model_args.num_speakers = speaker_manager.num_speakers
        else:
            config.num_speakers = speaker_manager.num_speakers
    elif check_config_and_model_args(config, "use_d_vector_file", True):
        if check_config_and_model_args(config, "use_speaker_encoder_as_loss",
                                       True):
            speaker_manager = SpeakerManager(
                d_vectors_file_path=config.model_args.d_vector_file,
                encoder_model_path=config.model_args.
                speaker_encoder_model_path,
                encoder_config_path=config.model_args.
                speaker_encoder_config_path,
                use_cuda=torch.cuda.is_available(),
            )
        else:
            speaker_manager = SpeakerManager(
                d_vectors_file_path=get_from_config_or_model_args(
                    config, "d_vector_file"))
        config.num_speakers = speaker_manager.num_speakers
        if hasattr(config, "model_args"):
            config.model_args.num_speakers = speaker_manager.num_speakers
    else:
        speaker_manager = None

    if check_config_and_model_args(config, "use_language_embedding", True):
        language_manager = LanguageManager(config=config)
        if hasattr(config, "model_args"):
            config.model_args.num_languages = language_manager.num_languages
        else:
            config.num_languages = language_manager.num_languages
    else:
        language_manager = None

    # init the model from config
    model = setup_model(config, speaker_manager, language_manager)

    # init the trainer and 🚀
    trainer = Trainer(
        train_args,
        config,
        config.output_path,
        model=model,
        train_samples=train_samples,
        eval_samples=eval_samples,
        training_assets={"audio_processor": ap},
        parse_command_line_args=False,
    )
    trainer.fit()
예제 #7
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파일: train_vits.py 프로젝트: gerazov/TTS
    spec_gain=1.0,
    signal_norm=False,
    do_amp_to_db_linear=False,
)
config = VitsConfig(
    audio=audio_config,
    run_name="vits_ljspeech",
    batch_size=48,
    eval_batch_size=16,
    batch_group_size=5,
    num_loader_workers=4,
    num_eval_loader_workers=4,
    run_eval=True,
    test_delay_epochs=-1,
    epochs=1000,
    text_cleaner="english_cleaners",
    use_phonemes=True,
    phoneme_language="en-us",
    phoneme_cache_path=os.path.join(output_path, "phoneme_cache"),
    compute_input_seq_cache=True,
    print_step=25,
    print_eval=True,
    mixed_precision=True,
    max_seq_len=500000,
    output_path=output_path,
    datasets=[dataset_config],
)
args, config, output_path, _, c_logger, tb_logger = init_training(TrainingArgs(), config)
trainer = Trainer(args, config, output_path, c_logger, tb_logger, cudnn_benchmark=True)
trainer.fit()
예제 #8
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파일: training.py 프로젝트: stjordanis/TTS
def init_arguments():
    train_config = TrainingArgs()
    parser = train_config.init_argparse(arg_prefix="")
    return parser