#!/usr/bin/env python # Copyright 2021 The HuggingFace Inc. team. All rights reserved. # # Licensed under the Apache License, Version 2.0 (the "License"); # you may not use this file except in compliance with the License. # You may obtain a copy of the License at # # http://www.apache.org/licenses/LICENSE-2.0 # # Unless required by applicable law or agreed to in writing, software # distributed under the License is distributed on an "AS IS" BASIS, # WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. # See the License for the specific language governing permissions and # limitations under the License. # /// script # dependencies = [ # "transformers @ git+https://github.com/huggingface/transformers.git", # "datasets[audio]>=1.14.0", # "evaluate", # "librosa", # "torchaudio", # "torch>=1.6", # ] # /// import logging import os import sys from dataclasses import dataclass, field from random import randint import datasets import evaluate import numpy as np from datasets import DatasetDict, load_dataset import transformers from transformers import ( AutoConfig, AutoFeatureExtractor, AutoModelForAudioClassification, HfArgumentParser, Trainer, TrainingArguments, set_seed, ) from transformers.utils import check_min_version from transformers.utils.versions import require_version logger = logging.getLogger(__name__) # Will error if the minimal version of Transformers is not installed. Remove at your own risks. check_min_version("4.57.0.dev0") require_version("datasets>=1.14.0", "To fix: pip install -r examples/pytorch/audio-classification/requirements.txt") def random_subsample(wav: np.ndarray, max_length: float, sample_rate: int = 16000): """Randomly sample chunks of `max_length` seconds from the input audio""" sample_length = int(round(sample_rate * max_length)) if len(wav) <= sample_length: return wav random_offset = randint(0, len(wav) - sample_length - 1) return wav[random_offset : random_offset + sample_length] @dataclass class DataTrainingArguments: """ Arguments pertaining to what data we are going to input our model for training and eval. Using `HfArgumentParser` we can turn this class into argparse arguments to be able to specify them on the command line. """ dataset_name: str | None = field(default=None, metadata={"help": "Name of a dataset from the datasets package"}) dataset_config_name: str | None = field( default=None, metadata={"help": "The configuration name of the dataset to use (via the datasets library)."} ) train_file: str | None = field( default=None, metadata={"help": "A file containing the training audio paths and labels."} ) eval_file: str | None = field( default=None, metadata={"help": "A file containing the validation audio paths and labels."} ) train_split_name: str = field( default="train", metadata={ "help": "The name of the training data set split to use (via the datasets library). Defaults to 'train'" }, ) eval_split_name: str = field( default="validation", metadata={ "help": ( "The name of the training data set split to use (via the datasets library). Defaults to 'validation'" ) }, ) audio_column_name: str = field( default="audio", metadata={"help": "The name of the dataset column containing the audio data. Defaults to 'audio'"}, ) label_column_name: str = field( default="label", metadata={"help": "The name of the dataset column containing the labels. Defaults to 'label'"} ) max_train_samples: int | None = field( default=None, metadata={ "help": ( "For debugging purposes or quicker training, truncate the number of training examples to this " "value if set." ) }, ) max_eval_samples: int | None = field( default=None, metadata={ "help": ( "For debugging purposes or quicker training, truncate the number of evaluation examples to this " "value if set." ) }, ) max_length_seconds: float = field( default=20, metadata={"help": "Audio clips will be randomly cut to this length during training if the value is set."}, ) @dataclass class ModelArguments: """ Arguments pertaining to which model/config/tokenizer we are going to fine-tune from. """ model_name_or_path: str = field( default="facebook/wav2vec2-base", metadata={"help": "Path to pretrained model or model identifier from huggingface.co/models"}, ) config_name: str | None = field( default=None, metadata={"help": "Pretrained config name or path if not the same as model_name"} ) cache_dir: str | None = field( default=None, metadata={"help": "Where do you want to store the pretrained models downloaded from the Hub"} ) model_revision: str = field( default="main", metadata={"help": "The specific model version to use (can be a branch name, tag name or commit id)."}, ) feature_extractor_name: str | None = field(default=None, metadata={"help": "Name or path of preprocessor config."}) freeze_feature_encoder: bool = field( default=True, metadata={"help": "Whether to freeze the feature encoder layers of the model."} ) attention_mask: bool = field( default=True, metadata={"help": "Whether to generate an attention mask in the feature extractor."} ) token: str = field( default=None, metadata={ "help": ( "The token to use as HTTP bearer authorization for remote files. If not specified, will use the token " "generated when running `hf auth login` (stored in `~/.huggingface`)." ) }, ) trust_remote_code: bool = field( default=False, metadata={ "help": ( "Whether to trust the execution of code from datasets/models defined on the Hub." " This option should only be set to `True` for repositories you trust and in which you have read the" " code, as it will execute code present on the Hub on your local machine." ) }, ) ignore_mismatched_sizes: bool = field( default=False, metadata={"help": "Will enable to load a pretrained model whose head dimensions are different."}, ) 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)) if len(sys.argv) == 2 and sys.argv[1].endswith(".json"): # If we pass only one argument to the script and it's the path to a json file, # let's parse it to get our arguments. model_args, data_args, training_args = parser.parse_json_file(json_file=os.path.abspath(sys.argv[1])) else: model_args, data_args, training_args = parser.parse_args_into_dataclasses() # Setup logging logging.basicConfig( format="%(asctime)s - %(levelname)s - %(name)s - %(message)s", datefmt="%m/%d/%Y %H:%M:%S", handlers=[logging.StreamHandler(sys.stdout)], ) if training_args.should_log: # The default of training_args.log_level is passive, so we set log level at info here to have that default. transformers.utils.logging.set_verbosity_info() log_level = training_args.get_process_log_level() logger.setLevel(log_level) transformers.utils.logging.set_verbosity(log_level) transformers.utils.logging.enable_default_handler() transformers.utils.logging.enable_explicit_format() # Log on each process the small summary: logger.warning( f"Process rank: {training_args.local_process_index}, device: {training_args.device}, n_gpu: {training_args.n_gpu}, " + f"distributed training: {training_args.parallel_mode.value == 'distributed'}, 16-bits training: {training_args.fp16}" ) logger.info(f"Training/evaluation parameters {training_args}") # Set seed before initializing model. set_seed(training_args.seed) # Initialize our dataset and prepare it for the audio classification task. raw_datasets = DatasetDict() raw_datasets["train"] = load_dataset( data_args.dataset_name, data_args.dataset_config_name, split=data_args.train_split_name, token=model_args.token, trust_remote_code=model_args.trust_remote_code, ) raw_datasets["eval"] = load_dataset( data_args.dataset_name, data_args.dataset_config_name, split=data_args.eval_split_name, token=model_args.token, trust_remote_code=model_args.trust_remote_code, ) if data_args.audio_column_name not in raw_datasets["train"].column_names: raise ValueError( f"--audio_column_name {data_args.audio_column_name} not found in dataset '{data_args.dataset_name}'. " "Make sure to set `--audio_column_name` to the correct audio column - one of " f"{', '.join(raw_datasets['train'].column_names)}." ) if data_args.label_column_name not in raw_datasets["train"].column_names: raise ValueError( f"--label_column_name {data_args.label_column_name} not found in dataset '{data_args.dataset_name}'. " "Make sure to set `--label_column_name` to the correct text column - one of " f"{', '.join(raw_datasets['train'].column_names)}." ) # Setting `return_attention_mask=True` is the way to get a correctly masked mean-pooling over # transformer outputs in the classifier, but it doesn't always lead to better accuracy feature_extractor = AutoFeatureExtractor.from_pretrained( model_args.feature_extractor_name or model_args.model_name_or_path, return_attention_mask=model_args.attention_mask, cache_dir=model_args.cache_dir, revision=model_args.model_revision, token=model_args.token, trust_remote_code=model_args.trust_remote_code, ) # `datasets` takes care of automatically loading and resampling the audio, # so we just need to set the correct target sampling rate. raw_datasets = raw_datasets.cast_column( data_args.audio_column_name, datasets.features.Audio(sampling_rate=feature_extractor.sampling_rate) ) model_input_name = feature_extractor.model_input_names[0] def train_transforms(batch): """Apply train_transforms across a batch.""" subsampled_wavs = [] for audio in batch[data_args.audio_column_name]: wav = random_subsample( audio["array"], max_length=data_args.max_length_seconds, sample_rate=feature_extractor.sampling_rate ) subsampled_wavs.append(wav) inputs = feature_extractor(subsampled_wavs, sampling_rate=feature_extractor.sampling_rate) output_batch = {model_input_name: inputs.get(model_input_name)} output_batch["labels"] = list(batch[data_args.label_column_name]) return output_batch def val_transforms(batch): """Apply val_transforms across a batch.""" wavs = [audio["array"] for audio in batch[data_args.audio_column_name]] inputs = feature_extractor(wavs, sampling_rate=feature_extractor.sampling_rate) output_batch = {model_input_name: inputs.get(model_input_name)} output_batch["labels"] = list(batch[data_args.label_column_name]) return output_batch # Prepare label mappings. # We'll include these in the model's config to get human readable labels in the Inference API. labels = raw_datasets["train"].features[data_args.label_column_name].names label2id, id2label = {}, {} for i, label in enumerate(labels): label2id[label] = str(i) id2label[str(i)] = label # Load the accuracy metric from the datasets package metric = evaluate.load("accuracy", cache_dir=model_args.cache_dir) # Define our compute_metrics function. It takes an `EvalPrediction` object (a namedtuple with # `predictions` and `label_ids` fields) and has to return a dictionary string to float. def compute_metrics(eval_pred): """Computes accuracy on a batch of predictions""" predictions = np.argmax(eval_pred.predictions, axis=1) return metric.compute(predictions=predictions, references=eval_pred.label_ids) config = AutoConfig.from_pretrained( model_args.config_name or model_args.model_name_or_path, num_labels=len(labels), label2id=label2id, id2label=id2label, finetuning_task="audio-classification", cache_dir=model_args.cache_dir, revision=model_args.model_revision, token=model_args.token, trust_remote_code=model_args.trust_remote_code, ) model = AutoModelForAudioClassification.from_pretrained( model_args.model_name_or_path, from_tf=bool(".ckpt" in model_args.model_name_or_path), config=config, cache_dir=model_args.cache_dir, revision=model_args.model_revision, token=model_args.token, trust_remote_code=model_args.trust_remote_code, ignore_mismatched_sizes=model_args.ignore_mismatched_sizes, ) # freeze the convolutional waveform encoder if model_args.freeze_feature_encoder: model.freeze_feature_encoder() if training_args.do_train: if data_args.max_train_samples is not None: raw_datasets["train"] = ( raw_datasets["train"].shuffle(seed=training_args.seed).select(range(data_args.max_train_samples)) ) # Set the training transforms raw_datasets["train"].set_transform(train_transforms, output_all_columns=False) if training_args.do_eval: if data_args.max_eval_samples is not None: raw_datasets["eval"] = ( raw_datasets["eval"].shuffle(seed=training_args.seed).select(range(data_args.max_eval_samples)) ) # Set the validation transforms raw_datasets["eval"].set_transform(val_transforms, output_all_columns=False) # Initialize our trainer trainer = Trainer( model=model, args=training_args, train_dataset=raw_datasets["train"] if training_args.do_train else None, eval_dataset=raw_datasets["eval"] if training_args.do_eval else None, compute_metrics=compute_metrics, processing_class=feature_extractor, ) # Training if training_args.do_train: checkpoint = None if training_args.resume_from_checkpoint is not None: checkpoint = training_args.resume_from_checkpoint train_result = trainer.train(resume_from_checkpoint=checkpoint) trainer.save_model() trainer.log_metrics("train", train_result.metrics) trainer.save_metrics("train", train_result.metrics) trainer.save_state() # Evaluation if training_args.do_eval: metrics = trainer.evaluate() trainer.log_metrics("eval", metrics) trainer.save_metrics("eval", metrics) # Write model card and (optionally) push to hub kwargs = { "finetuned_from": model_args.model_name_or_path, "tasks": "audio-classification", "dataset": data_args.dataset_name, "tags": ["audio-classification"], } if training_args.push_to_hub: trainer.push_to_hub(**kwargs) else: trainer.create_model_card(**kwargs) if __name__ == "__main__": main()