Instructions to use nullonesix/training with libraries, inference providers, notebooks, and local apps. Follow these links to get started.
- Libraries
- Transformers
How to use nullonesix/training with Transformers:
# Use a pipeline as a high-level helper from transformers import pipeline pipe = pipeline("automatic-speech-recognition", model="nullonesix/training")# Load model directly from transformers import AutoProcessor, AutoModelForSpeechSeq2Seq processor = AutoProcessor.from_pretrained("nullonesix/training") model = AutoModelForSpeechSeq2Seq.from_pretrained("nullonesix/training", device_map="auto") - Notebooks
- Google Colab
- Kaggle
| #!/usr/bin/env python | |
| # coding=utf-8 | |
| # Copyright 2023 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. | |
| """ | |
| Evaluating a Whisper model on one or more evaluation datasets. | |
| """ | |
| # You can also adapt this script for your own speech recognition validation. Pointers for this are left as comments. | |
| import logging | |
| import os | |
| import string | |
| import sys | |
| import time | |
| from dataclasses import field | |
| from functools import partial | |
| from pathlib import Path | |
| from typing import Any, Dict, List, Optional, Union | |
| import datasets | |
| import evaluate | |
| import flax | |
| import jax | |
| import jax.numpy as jnp | |
| import numpy as np | |
| import optax | |
| import torch | |
| import transformers | |
| from datasets import Dataset, DatasetDict, IterableDatasetDict, load_dataset | |
| from flax import jax_utils | |
| from flax.jax_utils import pad_shard_unpad | |
| from flax.training.common_utils import get_metrics, onehot | |
| from torch.utils.data import DataLoader | |
| from tqdm import tqdm | |
| from transformers import ( | |
| HfArgumentParser, | |
| Seq2SeqTrainingArguments, | |
| WhisperConfig, | |
| WhisperFeatureExtractor, | |
| WhisperProcessor, | |
| WhisperTokenizerFast, | |
| is_tensorboard_available, | |
| is_wandb_available, | |
| ) | |
| from transformers.models.whisper.english_normalizer import EnglishTextNormalizer | |
| from transformers.utils import check_min_version, send_example_telemetry | |
| from transformers.utils.versions import require_version | |
| from distil_whisper import FlaxWhisperForConditionalGeneration | |
| # Will error if the minimal version of Transformers is not installed. Remove at your own risks. | |
| check_min_version("4.27.0.dev0") | |
| require_version( | |
| "datasets>=1.18.0", | |
| "To fix: pip install -r examples/flax/speech-recogintion/requirements.txt", | |
| ) | |
| logger = logging.getLogger(__name__) | |
| class ModelArguments: | |
| """ | |
| Arguments pertaining to which model/config/tokenizer we are going to fine-tune from. | |
| """ | |
| model_name_or_path: str = field( | |
| metadata={"help": ("Path to pretrained model or model identifier from huggingface.co/models")} | |
| ) | |
| config_name: Optional[str] = field( | |
| default=None, | |
| metadata={"help": "Pretrained config name or path if not the same as model_name"}, | |
| ) | |
| tokenizer_name: Optional[str] = field( | |
| default=None, | |
| metadata={"help": "Pretrained tokenizer name or path if not the same as model_name"}, | |
| ) | |
| feature_extractor_name: Optional[str] = field( | |
| default=None, | |
| metadata={"help": "feature extractor name or path if not the same as model_name"}, | |
| ) | |
| processor_name: Optional[str] = field( | |
| default=None, | |
| metadata={"help": "processor name or path if not the same as model_name"}, | |
| ) | |
| cache_dir: Optional[str] = field( | |
| default=None, | |
| metadata={"help": ("Where to store the pretrained models downloaded from huggingface.co")}, | |
| ) | |
| use_fast_tokenizer: bool = field( | |
| default=True, | |
| metadata={"help": ("Whether to use one of the fast tokenizer (backed by the tokenizers library) or not.")}, | |
| ) | |
| model_revision: str = field( | |
| default="main", | |
| metadata={"help": ("The specific model version to use (can be a branch name, tag name or commit id).")}, | |
| ) | |
| subfolder: str = field( | |
| default="", | |
| metadata={ | |
| "help": "In case the relevant files are located inside a subfolder of the model repo on huggingface.co, you can" | |
| "specify the folder name here." | |
| }, | |
| ) | |
| use_auth_token: bool = field( | |
| default=False, | |
| metadata={ | |
| "help": ( | |
| "Will use the token generated when running `transformers-cli login`" | |
| " (necessary to use this script with private models)." | |
| ) | |
| }, | |
| ) | |
| dtype: Optional[str] = field( | |
| default="float32", | |
| metadata={ | |
| "help": ( | |
| "Floating-point format in which the model weights should be initialized" | |
| " and trained. Choose one of `[float32, float16, bfloat16]`." | |
| ) | |
| }, | |
| ) | |
| load_with_scan: Optional[bool] = field( | |
| default=False, | |
| metadata={ | |
| "help": ( | |
| "Whether to load the model with scan enabled. Required when the model was saved with scan enabled" | |
| ) | |
| }, | |
| ) | |
| return_timestamps: bool = field( | |
| default=False, metadata={"help": "Whether or not to predict timestamps in the generation step."} | |
| ) | |
| class DataTrainingArguments: | |
| """ | |
| Arguments pertaining to what data we are going to input our model for training and eval. | |
| """ | |
| dataset_name: str = field( | |
| default=None, | |
| metadata={ | |
| "help": "The name of the dataset to use (via the datasets library). Load and combine " | |
| "multiple datasets by separating dataset hours by a '+' symbol." | |
| }, | |
| ) | |
| dataset_config_name: Optional[str] = field( | |
| default=None, | |
| metadata={"help": "The configuration name of the dataset to use (via the datasets library)."}, | |
| ) | |
| dataset_split_name: Optional[str] = field( | |
| default=None, | |
| metadata={"help": "The split name of the dataset to use (via the datasets library)."}, | |
| ) | |
| dataset_cache_dir: Optional[str] = field( | |
| default=None, | |
| metadata={"help": "Path to cache directory for saving and loading datasets"}, | |
| ) | |
| overwrite_cache: bool = field( | |
| default=False, | |
| metadata={"help": "Overwrite the cached training and evaluation sets"}, | |
| ) | |
| preprocessing_num_workers: Optional[int] = field( | |
| default=None, | |
| metadata={"help": "The number of processes to use for the preprocessing."}, | |
| ) | |
| audio_column_name: str = field( | |
| default="audio", | |
| metadata={"help": "The name of the dataset column containing the audio data. Defaults to 'audio'"}, | |
| ) | |
| text_column_name: str = field( | |
| default=None, | |
| metadata={"help": "The name of the dataset column containing the text data. Defaults to `text`."}, | |
| ) | |
| max_duration_in_seconds: float = field( | |
| default=30.0, | |
| metadata={"help": "Filter audio files that are longer than `max_duration_in_seconds` seconds"}, | |
| ) | |
| min_duration_in_seconds: float = field( | |
| default=0.0, | |
| metadata={"help": "Filter audio files that are shorter than `min_duration_in_seconds` seconds"}, | |
| ) | |
| max_label_length: int = field( | |
| default=128, | |
| metadata={"help": "Truncate transcriptions that are longer `max_label_length` tokens."}, | |
| ) | |
| pad_target_to_multiple_of: Optional[int] = field( | |
| default=None, | |
| metadata={ | |
| "help": ( | |
| "If set will pad the target sequence to a multiple of the provided" | |
| " value. This is important to avoid triggering recompilations on TPU." | |
| " If unspecified, will default to padding the targets to max length." | |
| ) | |
| }, | |
| ) | |
| preprocessing_only: bool = field( | |
| default=False, | |
| metadata={ | |
| "help": ( | |
| "Whether to only do data preprocessing and skip training. This is" | |
| " especially useful when data preprocessing errors out in distributed" | |
| " training due to timeout. In this case, one should run the" | |
| " preprocessing in a non-distributed setup with" | |
| " `preprocessing_only=True` so that the cached datasets can" | |
| " consequently be loaded in distributed training" | |
| ) | |
| }, | |
| ) | |
| wandb_project: str = field( | |
| default="distil-whisper", | |
| metadata={"help": "The name of the wandb project."}, | |
| ) | |
| wandb_name: str = field( | |
| default=None, | |
| metadata={"help": "The name of the wandb run."}, | |
| ) | |
| wandb_job_type: str = field( | |
| default="distil-whisper", | |
| metadata={"help": "The name of the wandb job type."}, | |
| ) | |
| wandb_dir: str = field( | |
| default=None, | |
| metadata={"help": "The absolute path to save the wandb logs."}, | |
| ) | |
| save_code_to_wandb: bool = field( | |
| default=False, | |
| metadata={ | |
| "help": ( | |
| "Whether to save main script to wandb. This is valuable for improving" | |
| " experiment reproducibility and to diff code across experiments in" | |
| " the UI." | |
| ) | |
| }, | |
| ) | |
| streaming: bool = field( | |
| default=True, | |
| metadata={"help": "Whether to use Datasets' streaming mode to load and the data."}, | |
| ) | |
| max_eval_samples: Optional[int] = field( | |
| default=None, | |
| metadata={"help": "For debugging purposes, truncate the number of eval examples to this value if set."}, | |
| ) | |
| log_audio: Optional[bool] = field( | |
| default=False, | |
| metadata={"help": "For debugging purposes, record the audio samples as well as the ground truths / preds."}, | |
| ) | |
| def shift_tokens_right(label_ids: np.array, decoder_start_token_id: int) -> np.ndarray: | |
| """ | |
| Shift label ids one token to the right. | |
| """ | |
| shifted_label_ids = np.zeros_like(label_ids) | |
| shifted_label_ids[:, 1:] = label_ids[:, :-1] | |
| shifted_label_ids[:, 0] = decoder_start_token_id | |
| return shifted_label_ids | |
| class FlaxDataCollatorSpeechSeq2SeqWithPadding: | |
| """ | |
| Data collator that will dynamically pad the inputs received. | |
| Args: | |
| processor ([`Wav2Vec2Processor`]) | |
| The processor used for proccessing the data. | |
| decoder_start_token_id (:obj: `int`) | |
| The begin-of-sentence of the decoder. | |
| input_padding (:obj:`bool`, :obj:`str` or :class:`~transformers.tokenization_utils_base.PaddingStrategy`, `optional`, defaults to :obj:`True`): | |
| Select a strategy to pad the returned input sequences (according to the model's padding side and padding index) | |
| among: | |
| * :obj:`True` or :obj:`'longest'`: Pad to the longest sequence in the batch (or no padding if only a single | |
| sequence if provided). | |
| * :obj:`'max_length'`: Pad to a maximum length specified with the argument :obj:`max_length` or to the | |
| maximum acceptable input length for the model if that argument is not provided. | |
| * :obj:`False` or :obj:`'do_not_pad'` (default): No padding (i.e., can output a batch with sequences of | |
| different lengths). | |
| target_padding (:obj:`bool`, :obj:`str` or :class:`~transformers.tokenization_utils_base.PaddingStrategy`, `optional`, defaults to :obj:`True`): | |
| Select a strategy to pad the returned target sequences (according to the model's padding side and padding index). | |
| See above for details. | |
| max_target_length (:obj:`int`, `optional`): | |
| Maximum length of the ``labels`` of the returned list and optionally padding length (see above). | |
| log_audio (:obj:`bool`): | |
| Whether we're logging audio samples as part of our eval. If so, will forward on the audio samples to the batch. | |
| audio_column_name (:obj:`str`): | |
| Name of the audio column in the dataset. Only relevant if logging audio samples. | |
| """ | |
| processor: Any | |
| decoder_start_token_id: int | |
| input_padding: Union[bool, str] = "max_length" | |
| target_padding: Union[bool, str] = "max_length" | |
| max_target_length: Optional[int] = None | |
| log_audio: Optional[bool] = False | |
| audio_column_name: Optional[str] = "audio" | |
| def __call__(self, features: List[Dict[str, Union[List[int], np.ndarray]]]) -> Dict[str, np.ndarray]: | |
| # split inputs and labels since they have to be of different lengths and need | |
| # different padding methods | |
| model_input_name = self.processor.model_input_names[0] | |
| # dataloader returns a list of features which we convert to a dict | |
| input_features = {model_input_name: [feature[model_input_name] for feature in features]} | |
| label_features = {"input_ids": [feature["labels"] for feature in features]} | |
| # reformat list to dict and set to pytorch format | |
| batch = self.processor.feature_extractor.pad( | |
| input_features, | |
| padding=self.input_padding, | |
| return_tensors="np", | |
| ) | |
| labels_batch = self.processor.tokenizer.pad( | |
| label_features, | |
| max_length=self.max_target_length, | |
| padding=self.target_padding, | |
| return_tensors="np", | |
| ) | |
| # if bos token is appended in previous tokenization step, | |
| # cut bos token here as it's append later anyways | |
| labels = labels_batch["input_ids"] | |
| if (labels[:, 0] == self.decoder_start_token_id).all().item(): | |
| labels = labels[:, 1:] | |
| labels_batch.attention_mask = labels_batch.attention_mask[:, 1:] | |
| decoder_input_ids = shift_tokens_right(labels, self.decoder_start_token_id) | |
| # replace padding with -100 to ignore correctly when computing the loss | |
| labels = np.ma.array(labels, mask=np.not_equal(labels_batch.attention_mask, 1)) | |
| labels = labels.filled(fill_value=-100) | |
| batch["labels"] = labels | |
| batch["decoder_input_ids"] = decoder_input_ids | |
| if self.log_audio: | |
| audio_samples = [feature[self.audio_column_name] for feature in features] | |
| batch["audio"] = audio_samples | |
| return batch | |
| def get_data_loader( | |
| dataset: Dataset, | |
| batch_size: int, | |
| data_collator: FlaxDataCollatorSpeechSeq2SeqWithPadding, | |
| dataloader_num_workers: int = 0, | |
| pin_memory: bool = True, | |
| ) -> DataLoader: | |
| """ | |
| Returns batches of size `batch_size` from `dataset`. If `drop_last` is set to `False`, the final batch may be incomplete, | |
| and range in size from 1 to `batch_size`. Shuffle batches if `shuffle` is `True`. | |
| Args: | |
| dataset (Dataset): dataset from which to load the data. | |
| batch_size (int): how many samples per batch to load. | |
| data_collator (FlaxDataCollatorSpeechSeq2SeqWithPadding, optional): merges a list of samples to form a | |
| mini-batch of Tensor(s). Used when using batched loading from a map-style dataset. | |
| dataloader_num_workers (int, optional): how many subprocesses to use for data | |
| loading. ``0`` means that the data will be loaded in the main process. | |
| (default: ``0``) | |
| pin_memory (bool, optional): If ``True``, the data loader will copy Tensors | |
| into device/CUDA pinned memory before returning them. If your data elements | |
| are a custom type, or your :attr:`collate_fn` returns a batch that is a custom type, | |
| see the example below. | |
| """ | |
| data_loader = DataLoader( | |
| dataset, | |
| batch_size=batch_size, | |
| drop_last=False, | |
| pin_memory=pin_memory, | |
| collate_fn=data_collator, | |
| num_workers=dataloader_num_workers, | |
| ) | |
| return data_loader | |
| def write_metric(summary_writer, eval_metrics, step, prefix="eval"): | |
| for metric_name, value in eval_metrics.items(): | |
| summary_writer.scalar(f"{prefix}/{metric_name}", value, step) | |
| def write_wandb_metric(wandb_logger, metrics, train_time, prefix): | |
| log_metrics = {} | |
| for k, v in metrics.items(): | |
| log_metrics[f"{prefix}/{k}"] = v | |
| log_metrics[f"{prefix}/time"] = train_time | |
| wandb_logger.log(log_metrics) # TODO(SG): bug with wandb means we can't log the step count | |
| def convert_audio_to_wandb(wandb_logger, audio): | |
| return wandb_logger.Audio(audio["array"][:, np.newaxis], sample_rate=audio["sampling_rate"]) | |
| def write_wandb_pred( | |
| wandb_logger, | |
| eval_audios, | |
| pred_str, | |
| label_str, | |
| norm_pred_str, | |
| norm_label_str, | |
| prefix="eval", | |
| num_lines=200000, | |
| ): | |
| columns = ["Target", "Pred", "Norm Target", "Norm Pred"] | |
| # convert str data to a wandb compatible format | |
| str_data = [[label_str[i], pred_str[i], norm_label_str[i], norm_pred_str[i]] for i in range(len(pred_str))] | |
| if len(eval_audios) > 0: | |
| columns.insert(0, "Audio") | |
| str_data = [ | |
| [ | |
| convert_audio_to_wandb(wandb_logger, eval_audios[i]), | |
| *str_data[i], | |
| ] | |
| for i in range(len(pred_str)) | |
| ] | |
| # log as a table with the appropriate headers | |
| wandb_logger.log( | |
| {f"{prefix}/all_predictions": wandb_logger.Table(columns=columns, data=str_data[:num_lines])}, | |
| ) | |
| # log incorrect normalised predictions | |
| str_data = np.asarray(str_data) | |
| str_data_incorrect = str_data[str_data[:, -2] != str_data[:, -1]] | |
| # log as a table with the appropriate headers | |
| wandb_logger.log( | |
| {f"{prefix}/incorrect_predictions": wandb_logger.Table(columns=columns, data=str_data_incorrect[:num_lines])}, | |
| ) | |
| def convert_dataset_str_to_list( | |
| dataset_names, dataset_config_names, splits=None, text_column_names=None, dataset_hours=None, default_split="train" | |
| ): | |
| if isinstance(dataset_names, str): | |
| dataset_names = dataset_names.split("+") | |
| # we assume that all the datasets we're using derive from the distil-whisper org on the Hub - prepend the org name if necessary | |
| for i in range(len(dataset_names)): | |
| ds_name = dataset_names[i] | |
| dataset_names[i] = f"distil-whisper/{ds_name}" if "/" not in ds_name else ds_name | |
| dataset_config_names = dataset_config_names.split("+") | |
| splits = splits.split("+") if splits is not None else None | |
| text_column_names = text_column_names.split("+") if text_column_names is not None else None | |
| dataset_hours = dataset_hours.split("+") if dataset_hours is not None else None | |
| # basic checks to ensure we've got the right number of datasets/configs/splits/columns/probs | |
| if len(dataset_names) != len(dataset_config_names): | |
| raise ValueError( | |
| f"Ensure one config is passed for each dataset, got {len(dataset_names)} datasets and" | |
| f" {len(dataset_config_names)} configs." | |
| ) | |
| if splits is not None and len(splits) != len(dataset_names): | |
| raise ValueError( | |
| f"Ensure one split is passed for each dataset, got {len(dataset_names)} datasets and {len(splits)} splits." | |
| ) | |
| if text_column_names is not None and len(text_column_names) != len(dataset_names): | |
| raise ValueError( | |
| f"Ensure one text column name is passed for each dataset, got {len(dataset_names)} datasets and" | |
| f" {len(text_column_names)} text column names." | |
| ) | |
| if dataset_hours is not None: | |
| if len(dataset_hours) != len(dataset_names): | |
| raise ValueError( | |
| f"Ensure one probability is passed for each dataset, got {len(dataset_names)} datasets and " | |
| f"{len(dataset_hours)} hours." | |
| ) | |
| dataset_hours = [float(ds_hours) for ds_hours in dataset_hours] | |
| else: | |
| dataset_hours = [None] * len(dataset_names) | |
| text_column_names = ( | |
| text_column_names if text_column_names is not None else ["text" for _ in range(len(dataset_names))] | |
| ) | |
| splits = splits if splits is not None else [default_split for _ in range(len(dataset_names))] | |
| dataset_names_dict = [] | |
| for i, ds_name in enumerate(dataset_names): | |
| dataset_names_dict.append( | |
| { | |
| "name": ds_name, | |
| "config": dataset_config_names[i], | |
| "split": splits[i], | |
| "text_column_name": text_column_names[i], | |
| "hours": dataset_hours[i], | |
| } | |
| ) | |
| return dataset_names_dict | |
| class FlaxWhisperFeatureExtractor(WhisperFeatureExtractor): | |
| def _np_extract_fbank_features(self, waveform: np.array) -> np.ndarray: | |
| """ | |
| Compute the log-mel spectrogram of the provided audio using torch filters. Using the torch implementation | |
| computes stft filter banks approx 5x faster than its numpy counterpart, which is the native implementation | |
| in transformers, and matches to within 1e-5 abs tolerance. | |
| """ | |
| waveform = torch.from_numpy(waveform).type(torch.float32) | |
| window = torch.hann_window(self.n_fft) | |
| stft = torch.stft(waveform, self.n_fft, self.hop_length, window=window, return_complex=True) | |
| magnitudes = stft[..., :-1].abs() ** 2 | |
| mel_filters = torch.from_numpy(self.mel_filters).type(torch.float32) | |
| mel_spec = mel_filters.T @ magnitudes | |
| log_spec = torch.clamp(mel_spec, min=1e-10).log10() | |
| log_spec = torch.maximum(log_spec, log_spec.max() - 8.0) | |
| log_spec = (log_spec + 4.0) / 4.0 | |
| return log_spec.numpy() | |
| def main(): | |
| # 1. Parse input arguments | |
| # 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, Seq2SeqTrainingArguments)) | |
| 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() | |
| # Sending telemetry. Tracking the example usage helps us better allocate resources to maintain them. The | |
| # information sent is the one passed as arguments along with your JAX/Flax versions. | |
| send_example_telemetry("run_flax_speech_recognition_seq2seq", model_args, data_args, framework="flax") | |
| # 2. Setup logging | |
| # Make one log on every process with the configuration for debugging. | |
| logging.basicConfig( | |
| format="%(asctime)s - %(levelname)s - %(name)s - %(message)s", | |
| datefmt="%m/%d/%Y %H:%M:%S", | |
| handlers=[logging.StreamHandler(sys.stdout)], | |
| ) | |
| # Set the verbosity to info of the Transformers logger. | |
| # We only want one process per machine to log things on the screen. | |
| logger.setLevel(logging.INFO if jax.process_index() == 0 else logging.ERROR) | |
| if jax.process_index() == 0: | |
| datasets.utils.logging.set_verbosity_warning() | |
| transformers.utils.logging.set_verbosity_info() | |
| else: | |
| datasets.utils.logging.set_verbosity_error() | |
| transformers.utils.logging.set_verbosity_error() | |
| logger.info("Evaluation parameters %s", training_args) | |
| # Enable tensorboard only on the master node | |
| has_tensorboard = is_tensorboard_available() | |
| if "tensorboard" in training_args.report_to: | |
| if has_tensorboard and jax.process_index() == 0: | |
| try: | |
| from flax.metrics.tensorboard import SummaryWriter | |
| summary_writer = SummaryWriter(log_dir=Path(training_args.output_dir)) | |
| except ImportError as ie: | |
| has_tensorboard = False | |
| logger.warning( | |
| "Unable to display metrics through TensorBoard because some" f" package are not installed: {ie}" | |
| ) | |
| else: | |
| logger.warning( | |
| "Unable to display metrics through TensorBoard because the package is" | |
| " not installed: Please run `pip install tensorboard` to enable." | |
| ) | |
| # Enable wandb only on the master node | |
| has_wandb = is_wandb_available() | |
| if "wandb" in training_args.report_to: | |
| if has_wandb and jax.process_index() == 0: | |
| import wandb as wandb_logger | |
| # Set up wandb run | |
| wandb_logger.init( | |
| project=data_args.wandb_project, | |
| name=data_args.wandb_name, | |
| job_type=data_args.wandb_job_type, | |
| dir=data_args.wandb_dir, | |
| save_code=data_args.save_code_to_wandb, | |
| ) | |
| else: | |
| logger.warning("Wandb logging requires wandb to be installed. Run `pip install wandb` to enable.") | |
| # 3. Load dataset | |
| raw_datasets = IterableDatasetDict() if data_args.streaming else DatasetDict() | |
| # Convert lists of dataset names/configs/splits to a dict | |
| # names: "librispeech_asr+gigaspeech", configs: "all+l", splits: "validation.clean+validation" | |
| # -> [{"name: "librispeech_asr": "config": "all", "split": "validation.clean"}, {"name: "gigaspeech": "config": "l", "split": "validation"} | |
| dataset_names_dict = convert_dataset_str_to_list( | |
| data_args.dataset_name, | |
| data_args.dataset_config_name, | |
| splits=data_args.dataset_split_name, | |
| text_column_names=data_args.text_column_name, | |
| ) | |
| if len(dataset_names_dict) == 1: | |
| # load a single eval set | |
| dataset_dict = dataset_names_dict[0] | |
| raw_datasets["eval"] = load_dataset( | |
| dataset_dict["name"], | |
| dataset_dict["config"], | |
| split=dataset_dict["split"], | |
| cache_dir=data_args.dataset_cache_dir, | |
| use_auth_token=True if model_args.use_auth_token else None, | |
| streaming=data_args.streaming, | |
| ) | |
| if dataset_dict["text_column_name"] not in list(raw_datasets["eval"].features.keys()): | |
| raise ValueError( | |
| f"--text column name {dataset_dict['text_column_name']} not found in the evaluation " | |
| f"dataset {dataset_dict['name']}. Ensure `text_column_name` is set to the correct column " | |
| f"for the target text. Should be one of {' '.join(list(raw_datasets['eval'].features.keys()))}" | |
| ) | |
| if dataset_dict["text_column_name"] != "text": | |
| raw_datasets["eval"] = raw_datasets["eval"].rename_column(dataset_dict["text_column_name"], "text") | |
| else: | |
| # load multiple eval sets | |
| for dataset_dict in tqdm(dataset_names_dict, desc="Loading datasets..."): | |
| # Clean-up the dataset name for pretty logging | |
| # ("distil-whisper/librispeech_asr", "validation.clean") -> "librispeech_asr/validation-clean" | |
| pretty_name = f"{dataset_dict['name'].split('/')[-1]}/{dataset_dict['split'].replace('.', '-')}" | |
| raw_datasets[pretty_name] = load_dataset( | |
| dataset_dict["name"], | |
| dataset_dict["config"], | |
| split=dataset_dict["split"], | |
| cache_dir=data_args.dataset_cache_dir, | |
| use_auth_token=True if model_args.use_auth_token else None, | |
| streaming=data_args.streaming, | |
| ) | |
| if dataset_dict["text_column_name"] not in list(raw_datasets[pretty_name].features.keys()): | |
| raise ValueError( | |
| f"`--text_column_name` {dataset_dict['text_column_name']} not found in the evaluation " | |
| f"dataset {dataset_dict['name']}. Ensure `text_column_name` is set to the correct column " | |
| f"for the target text. Should be one of {' '.join(list(raw_datasets[pretty_name].features.keys()))}" | |
| ) | |
| if dataset_dict["text_column_name"] != "text": | |
| raw_datasets[pretty_name] = raw_datasets[pretty_name].rename_column( | |
| dataset_dict["text_column_name"], "text" | |
| ) | |
| # 5. Load pretrained model, tokenizer, and feature extractor | |
| config = WhisperConfig.from_pretrained( | |
| (model_args.config_name if model_args.config_name else model_args.model_name_or_path), | |
| cache_dir=model_args.cache_dir, | |
| revision=model_args.model_revision, | |
| use_auth_token=True if model_args.use_auth_token else None, | |
| ) | |
| feature_extractor = FlaxWhisperFeatureExtractor.from_pretrained( | |
| (model_args.feature_extractor_name if model_args.feature_extractor_name else model_args.model_name_or_path), | |
| cache_dir=model_args.cache_dir, | |
| revision=model_args.model_revision, | |
| use_auth_token=True if model_args.use_auth_token else None, | |
| ) | |
| tokenizer = WhisperTokenizerFast.from_pretrained( | |
| (model_args.tokenizer_name if model_args.tokenizer_name else model_args.model_name_or_path), | |
| cache_dir=model_args.cache_dir, | |
| use_fast=model_args.use_fast_tokenizer, | |
| revision=model_args.model_revision, | |
| use_auth_token=True if model_args.use_auth_token else None, | |
| ) | |
| processor = WhisperProcessor.from_pretrained( | |
| (model_args.processor_name if model_args.processor_name else model_args.model_name_or_path), | |
| cache_dir=model_args.cache_dir, | |
| revision=model_args.model_revision, | |
| use_auth_token=True if model_args.use_auth_token else None, | |
| ) | |
| model, params = FlaxWhisperForConditionalGeneration.from_pretrained( | |
| model_args.model_name_or_path, | |
| config=config, | |
| dtype=getattr(jnp, model_args.dtype), | |
| cache_dir=model_args.cache_dir, | |
| revision=model_args.model_revision, | |
| use_auth_token=True if model_args.use_auth_token else None, | |
| _do_init=False, | |
| subfolder=model_args.subfolder, | |
| # use_scan=model_args.load_with_scan, # Model might have (erroneously) been saved with scan still enabled | |
| ) | |
| if model.config.decoder_start_token_id is None: | |
| raise ValueError("Make sure that `config.decoder_start_token_id` is correctly defined") | |
| # disable scan if necessary (makes the inference step faster) | |
| if model_args.load_with_scan: | |
| model.disable_scan() # to disable scan in the nn.Module | |
| params = model.convert_scan_to_unroll(params) # to convert the scan params to unrolled | |
| # 6. Resample speech dataset: `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), | |
| ) | |
| # 7. Preprocessing the datasets. | |
| # We need to read the audio files as arrays and tokenize the targets. | |
| max_label_length = ( | |
| data_args.max_label_length if data_args.max_label_length is not None else model.config.max_length | |
| ) | |
| audio_column_name = data_args.audio_column_name | |
| num_workers = data_args.preprocessing_num_workers | |
| dataloader_num_workers = training_args.dataloader_num_workers | |
| model_input_name = feature_extractor.model_input_names[0] | |
| normalizer = EnglishTextNormalizer(tokenizer.english_spelling_normalizer) | |
| if data_args.max_eval_samples is not None: | |
| for split in raw_datasets: | |
| raw_datasets[split] = ( | |
| raw_datasets[split].take(data_args.max_eval_samples) | |
| if data_args.streaming | |
| else raw_datasets[split].select(range(data_args.max_eval_samples)) | |
| ) | |
| def prepare_dataset(batch): | |
| # process audio | |
| sample = batch[audio_column_name] | |
| inputs = feature_extractor(sample["array"], sampling_rate=sample["sampling_rate"]) | |
| # process audio length | |
| batch[model_input_name] = inputs.get(model_input_name)[0] | |
| # process targets | |
| input_str = batch["text"] | |
| batch["labels"] = tokenizer(input_str, max_length=max_label_length, truncation=True).input_ids | |
| return batch | |
| vectorized_datasets = IterableDatasetDict() if data_args.streaming else DatasetDict() | |
| for split in raw_datasets: | |
| raw_datasets_features = list(raw_datasets[split].features.keys()) | |
| if data_args.log_audio: | |
| # if logging audio samples preserve the audio column when mapping the dataset | |
| raw_datasets_features.remove(audio_column_name) | |
| map_fn = partial( | |
| raw_datasets[split].map, | |
| function=prepare_dataset, | |
| remove_columns=raw_datasets_features, | |
| ) | |
| vectorized_datasets[split] = ( | |
| map_fn(num_proc=num_workers, desc="preprocess eval dataset") | |
| if not data_args.streaming | |
| else map_fn() # In streaming, we can't run multiproc - errors out if we try to | |
| ) | |
| # for large datasets it is advised to run the preprocessing on a | |
| # single machine first with `args.preprocessing_only` since there will mostly likely | |
| # be a timeout when running the script in distributed mode. | |
| # In a second step `args.preprocessing_only` can then be set to `False` to load the | |
| # cached dataset | |
| if data_args.preprocessing_only: | |
| cache = {k: v.cache_files for k, v in vectorized_datasets.items()} | |
| logger.info(f"Data preprocessing finished. Files cached at {cache}.") | |
| return | |
| # 8. Load Metric | |
| metric = evaluate.load("wer") | |
| # convention is that we space all punctuation *except* apostrophes | |
| all_punctuation = list(string.punctuation.replace("'", "")) | |
| return_timestamps = model_args.return_timestamps | |
| def compute_metrics(preds, labels): | |
| # replace padded labels by the padding token | |
| for idx in range(len(labels)): | |
| labels[idx][labels[idx] == -100] = tokenizer.pad_token_id | |
| pred_str = tokenizer.batch_decode(preds, skip_special_tokens=True, decode_with_timestamps=return_timestamps) | |
| # we do not want to group tokens when computing the metrics | |
| label_str = tokenizer.batch_decode(labels, skip_special_tokens=True) | |
| # space punctuation for orthographic WER (c.f. ESB paper https://arxiv.org/abs/2210.13352) | |
| spaced_pred_str = [ | |
| pred_str[i].replace(punctuation, f" {punctuation} ") | |
| for punctuation in all_punctuation | |
| for i in range(len(pred_str)) | |
| ] | |
| spaced_label_str = [ | |
| label_str[i].replace(punctuation, f" {punctuation} ") | |
| for punctuation in all_punctuation | |
| for i in range(len(label_str)) | |
| ] | |
| wer_ortho = 100 * metric.compute(predictions=spaced_pred_str, references=spaced_label_str) | |
| # normalize everything and re-compute the WER | |
| norm_pred_str = [normalizer(pred) for pred in pred_str] | |
| norm_label_str = [normalizer(label) for label in label_str] | |
| # for logging, we need the pred/labels to match the norm_pred/norm_labels, so discard any filtered samples here | |
| pred_str = [pred_str[i] for i in range(len(norm_pred_str)) if len(norm_label_str[i]) > 0] | |
| label_str = [label_str[i] for i in range(len(norm_label_str)) if len(norm_label_str[i]) > 0] | |
| # filtering step to only evaluate the samples that correspond to non-zero normalized references: | |
| norm_pred_str = [norm_pred_str[i] for i in range(len(norm_pred_str)) if len(norm_label_str[i]) > 0] | |
| norm_label_str = [norm_label_str[i] for i in range(len(norm_label_str)) if len(norm_label_str[i]) > 0] | |
| wer = 100 * metric.compute(predictions=norm_pred_str, references=norm_label_str) | |
| return {"wer": wer, "wer_ortho": wer_ortho}, pred_str, label_str, norm_pred_str, norm_label_str | |
| data_collator = FlaxDataCollatorSpeechSeq2SeqWithPadding( | |
| processor=processor, | |
| decoder_start_token_id=model.config.decoder_start_token_id, | |
| input_padding="longest", | |
| target_padding="max_length", | |
| max_target_length=max_label_length, | |
| log_audio=data_args.log_audio, | |
| ) | |
| # Store some constants | |
| per_device_eval_batch_size = int(training_args.per_device_eval_batch_size) | |
| eval_batch_size = per_device_eval_batch_size * jax.device_count() | |
| # label smoothed cross entropy | |
| def loss_fn(logits, labels, label_smoothing_factor=0.0): | |
| """ | |
| The label smoothing implementation is adapted from Flax's official example: | |
| https://github.com/google/flax/blob/87a211135c6a377c8f29048a1cac3840e38b9da4/examples/wmt/train.py#L104 | |
| """ | |
| vocab_size = logits.shape[-1] | |
| confidence = 1.0 - label_smoothing_factor | |
| low_confidence = (1.0 - confidence) / (vocab_size - 1) | |
| normalizing_constant = -( | |
| confidence * jnp.log(confidence) + (vocab_size - 1) * low_confidence * jnp.log(low_confidence + 1e-20) | |
| ) | |
| soft_labels = onehot(labels, vocab_size, on_value=confidence, off_value=low_confidence) | |
| loss = optax.softmax_cross_entropy(logits, soft_labels) | |
| loss = loss - normalizing_constant | |
| # ignore padded tokens from loss, i.e. where labels are not set to -100 | |
| padding_mask = labels >= 0 | |
| loss = loss * padding_mask | |
| loss = loss.sum() | |
| num_labels = padding_mask.sum() | |
| return loss, num_labels | |
| # Define eval fn | |
| def eval_step(params, batch, label_smoothing_factor=0.0): | |
| labels = batch.pop("labels") | |
| logits = model(**batch, params=params, freeze_encoder=True, train=False)[0] | |
| loss, num_labels = loss_fn(logits, labels, label_smoothing_factor) | |
| num_labels = jax.lax.psum(num_labels, "batch") | |
| # true loss = total loss / total samples | |
| loss = jax.lax.psum(loss, "batch") | |
| loss = jax.tree_util.tree_map(lambda x: x / num_labels, loss) | |
| metrics = {"loss": loss} | |
| return metrics | |
| # Define generation function | |
| num_beams = ( | |
| training_args.generation_num_beams | |
| if training_args.generation_num_beams is not None | |
| else model.config.num_beams | |
| ) | |
| # forcing the language and task tokens helps the flax teacher model in its generations | |
| gen_kwargs = { | |
| "max_length": max_label_length, | |
| "num_beams": num_beams, | |
| "language": "<|en|>", | |
| "task": "transcribe", | |
| "return_timestamps": return_timestamps, | |
| } | |
| def generate_step(params, batch): | |
| output_ids = model.generate( | |
| batch[model_input_name], | |
| attention_mask=batch.get("attention_mask"), | |
| params=params, | |
| freeze_encoder=True, | |
| **gen_kwargs, | |
| ) | |
| return output_ids.sequences | |
| # Create parallel version of the eval and generate step | |
| p_eval_step = jax.pmap( | |
| partial(eval_step, label_smoothing_factor=training_args.label_smoothing_factor), | |
| "batch", | |
| ) | |
| p_generate_step = jax.pmap(generate_step, "batch") | |
| # Replicate params on each device | |
| params = jax_utils.replicate(params) | |
| def eval_step(split="eval"): | |
| # ======================== Evaluating ============================== | |
| eval_metrics = [] | |
| eval_preds = [] | |
| eval_labels = [] | |
| eval_audios = [] | |
| eval_start = time.time() | |
| eval_loader = get_data_loader( | |
| vectorized_datasets[split], | |
| batch_size=eval_batch_size, | |
| data_collator=data_collator, | |
| dataloader_num_workers=dataloader_num_workers, | |
| ) | |
| for batch in tqdm(eval_loader, desc=f"Evaluating {split}..."): | |
| # Model forward | |
| labels = batch["labels"] | |
| if data_args.log_audio: | |
| eval_audios.extend(batch.pop("audio")) | |
| metrics = pad_shard_unpad(p_eval_step, static_return=True)( | |
| params, batch.data, min_device_batch=per_device_eval_batch_size | |
| ) | |
| eval_metrics.append(metrics) | |
| # generation | |
| if training_args.predict_with_generate: | |
| generated_ids = pad_shard_unpad(p_generate_step)( | |
| params, batch.data, min_device_batch=per_device_eval_batch_size | |
| ) | |
| eval_preds.extend(jax.device_get(generated_ids.reshape(-1, gen_kwargs["max_length"]))) | |
| eval_labels.extend(labels) | |
| eval_time = time.time() - eval_start | |
| # normalize eval metrics | |
| eval_metrics = get_metrics(eval_metrics) | |
| eval_metrics = jax.tree_util.tree_map(jnp.mean, eval_metrics) | |
| # compute WER metric | |
| wer_desc = "" | |
| if training_args.predict_with_generate: | |
| wer_metric, pred_str, label_str, norm_pred_str, norm_label_str = compute_metrics(eval_preds, eval_labels) | |
| eval_metrics.update(wer_metric) | |
| wer_desc = " ".join([f"Eval {key}: {value} |" for key, value in wer_metric.items()]) | |
| # Print metrics | |
| logger.info(f"Eval Loss: {eval_metrics['loss']} | {wer_desc})") | |
| # Save metrics | |
| if has_tensorboard and jax.process_index() == 0 and "tensorboard" in training_args.report_to: | |
| write_metric(summary_writer, eval_metrics, model_args.step, prefix=split) | |
| if has_wandb and jax.process_index() == 0 and "wandb" in training_args.report_to: | |
| write_wandb_metric(wandb_logger, eval_metrics, eval_time, prefix=split) | |
| if training_args.predict_with_generate: | |
| write_wandb_pred( | |
| wandb_logger, eval_audios, pred_str, label_str, norm_pred_str, norm_label_str, prefix=split | |
| ) | |
| logger.info("***** Running Eval *****") | |
| logger.info(" Instantaneous batch size per device =" f" {training_args.per_device_eval_batch_size}") | |
| logger.info(f" Total eval batch size (w. parallel & distributed) = {eval_batch_size}") | |
| for split in vectorized_datasets: | |
| eval_step(split=split) | |
| if __name__ == "__main__": | |
| main() | |