jax-projects/wav2vec2/run_wav2vec2_pretrain_flax.py [283:363]:
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def main():
    # See all possible arguments in src/transformers/training_args.py
    # or by passing the --help flag to this script.
    # We now keep distinct sets of args, for a cleaner separation of concerns.

    parser = HfArgumentParser((ModelArguments, DataTrainingArguments, TrainingArguments))

    model_args, data_args, training_args = parser.parse_args_into_dataclasses()
    configure_logger(model_args, training_args)

    # Downloading and loading a dataset from the hub.
    datasets = load_dataset(data_args.dataset_name, data_args.dataset_config_name, cache_dir=model_args.cache_dir)

    if "validation" not in datasets.keys():
        # make sure only "validation" and "train" keys remain"
        datasets = DatasetDict()
        datasets["validation"] = load_dataset(
            data_args.dataset_name,
            data_args.dataset_config_name,
            split=f"{data_args.train_split_name}[:{data_args.validation_split_percentage}%]",
            cache_dir=model_args.cache_dir,
        )
        datasets["train"] = load_dataset(
            data_args.dataset_name,
            data_args.dataset_config_name,
            split=f"{data_args.train_split_name}[{data_args.validation_split_percentage}%:]",
            cache_dir=model_args.cache_dir,
        )
    else:
        # make sure only "validation" and "train" keys remain"
        datasets = DatasetDict()
        datasets["validation"] = load_dataset(
            data_args.dataset_name,
            data_args.dataset_config_name,
            split="validation",
            cache_dir=model_args.cache_dir,
        )
        datasets["train"] = load_dataset(
            data_args.dataset_name,
            data_args.dataset_config_name,
            split=f"{data_args.train_split_name}",
            cache_dir=model_args.cache_dir,
        )

    # only normalized-inputs-training is supported
    feature_extractor = Wav2Vec2FeatureExtractor.from_pretrained(
        model_args.model_name_or_path, cache_dir=model_args.cache_dir, do_normalize=True
    )

    def prepare_dataset(batch):
        # check that all files have the correct sampling rate
        batch["speech"], _ = librosa.load(batch[data_args.speech_file_column], sr=feature_extractor.sampling_rate)
        return batch

    # load audio files into numpy arrays
    vectorized_datasets = datasets.map(
        prepare_dataset, num_proc=data_args.preprocessing_num_workers, remove_columns=datasets["train"].column_names
    )

    # filter audio files that are too long
    vectorized_datasets = vectorized_datasets.filter(
        lambda data: len(data["speech"]) < int(data_args.max_duration_in_seconds * feature_extractor.sampling_rate)
    )

    def normalize(batch):
        return feature_extractor(batch["speech"], sampling_rate=feature_extractor.sampling_rate)

    # normalize and transform to `BatchFeatures`
    vectorized_datasets = vectorized_datasets.map(
        normalize,
        batched=True,
        num_proc=data_args.preprocessing_num_workers,
        load_from_cache_file=not data_args.overwrite_cache,
        remove_columns=vectorized_datasets["train"].column_names,
    )

    # pretraining is only supported for "newer" stable layer norm architecture
    # apply_spec_augment has to be True, mask_feature_prob has to be 0.0
    config = Wav2Vec2Config.from_pretrained(
        model_args.model_name_or_path,
        cache_dir=model_args.cache_dir,
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wav2vec2/run_pretrain.py [287:367]:
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def main():
    # See all possible arguments in src/transformers/training_args.py
    # or by passing the --help flag to this script.
    # We now keep distinct sets of args, for a cleaner separation of concerns.

    parser = HfArgumentParser((ModelArguments, DataTrainingArguments, TrainingArguments))

    model_args, data_args, training_args = parser.parse_args_into_dataclasses()
    configure_logger(model_args, training_args)

    # Downloading and loading a dataset from the hub.
    datasets = load_dataset(data_args.dataset_name, data_args.dataset_config_name, cache_dir=model_args.cache_dir)

    if "validation" not in datasets.keys():
        # make sure only "validation" and "train" keys remain"
        datasets = DatasetDict()
        datasets["validation"] = load_dataset(
            data_args.dataset_name,
            data_args.dataset_config_name,
            split=f"{data_args.train_split_name}[:{data_args.validation_split_percentage}%]",
            cache_dir=model_args.cache_dir,
        )
        datasets["train"] = load_dataset(
            data_args.dataset_name,
            data_args.dataset_config_name,
            split=f"{data_args.train_split_name}[{data_args.validation_split_percentage}%:]",
            cache_dir=model_args.cache_dir,
        )
    else:
        # make sure only "validation" and "train" keys remain"
        datasets = DatasetDict()
        datasets["validation"] = load_dataset(
            data_args.dataset_name,
            data_args.dataset_config_name,
            split="validation",
            cache_dir=model_args.cache_dir,
        )
        datasets["train"] = load_dataset(
            data_args.dataset_name,
            data_args.dataset_config_name,
            split=f"{data_args.train_split_name}",
            cache_dir=model_args.cache_dir,
        )

    # only normalized-inputs-training is supported
    feature_extractor = Wav2Vec2FeatureExtractor.from_pretrained(
        model_args.model_name_or_path, cache_dir=model_args.cache_dir, do_normalize=True
    )

    def prepare_dataset(batch):
        # check that all files have the correct sampling rate
        batch["speech"], _ = librosa.load(batch[data_args.speech_file_column], sr=feature_extractor.sampling_rate)
        return batch

    # load audio files into numpy arrays
    vectorized_datasets = datasets.map(
        prepare_dataset, num_proc=data_args.preprocessing_num_workers, remove_columns=datasets["train"].column_names
    )

    # filter audio files that are too long
    vectorized_datasets = vectorized_datasets.filter(
        lambda data: len(data["speech"]) < int(data_args.max_duration_in_seconds * feature_extractor.sampling_rate)
    )

    def normalize(batch):
        return feature_extractor(batch["speech"], sampling_rate=feature_extractor.sampling_rate)

    # normalize and transform to `BatchFeatures`
    vectorized_datasets = vectorized_datasets.map(
        normalize,
        batched=True,
        num_proc=data_args.preprocessing_num_workers,
        load_from_cache_file=not data_args.overwrite_cache,
        remove_columns=vectorized_datasets["train"].column_names,
    )

    # pretraining is only supported for "newer" stable layer norm architecture
    # apply_spec_augment has to be True, mask_feature_prob has to be 0.0
    config = Wav2Vec2Config.from_pretrained(
        model_args.model_name_or_path,
        cache_dir=model_args.cache_dir,
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