import argparse import logging import sys import datetime import os import pandas as pd from transformers import pipeline from transformers.models.whisper.english_normalizer import BasicTextNormalizer from datasets import load_dataset, Audio import evaluate from belarusian_text_normalizer import BelarusianTextNormalizer now_str = datetime.datetime.now().strftime('%Y%m%d-%H%M%S') logger = logging.getLogger(__name__) logging.basicConfig( format="%(asctime)s - %(levelname)s - %(name)s - %(message)s", datefmt="%m/%d/%Y %H:%M:%S", handlers=[ logging.StreamHandler(sys.stdout), logging.FileHandler(filename=f'eval_{now_str}.log', mode='w') ], ) logger.setLevel(logging.INFO) wer_metric = evaluate.load("wer") text_normalizer = BelarusianTextNormalizer() def is_target_text_in_range(ref): if ref.strip() == "ignore time segment in scoring": return False else: return ref.strip() != "" def normalise(sample, text_column: str): sample["reference_norm"] = text_normalizer(sample[text_column]) return sample def data(dataset,text_column: str): for i, item in enumerate(dataset): yield {**item["audio"], "reference_norm": item["reference_norm"], 'reference': item[text_column]} def clean_filename(filename: str): return filename.replace(os.path.sep, '_') def main(args): logger.info(f'running evaluation script with following parameters: {args}') logger.info(f'using following text normalizer: {text_normalizer}') batch_size = args.batch_size whisper_asr = pipeline("automatic-speech-recognition", model=args.model_id, device=args.device) whisper_asr.model.config.forced_decoder_ids = ( whisper_asr.tokenizer.get_decoder_prompt_ids( language=args.language, task="transcribe" ) ) logger.info('loading dataset') dataset = load_dataset( args.dataset, args.config, split=args.split, streaming=args.streaming, use_auth_token=True, ) # Only uncomment for debugging dataset = dataset.take(args.max_eval_samples) # TODO: probably no need in cast, because pipelien migh handle resampling internally. need to check dataset = dataset.cast_column("audio", Audio(sampling_rate=16000)) dataset = dataset.map(normalise, fn_kwargs=dict(text_column=args.text_column)) dataset = dataset.filter(is_target_text_in_range, input_columns=["reference_norm"]) predictions = [] predictions_norm = [] references = [] references_norm = [] audio_paths = [] logger.info('running inference') for out in whisper_asr(data(dataset, text_column=args.text_column), batch_size=batch_size): predictions.append(out["text"]) predictions_norm.append(text_normalizer(out["text"])) references.append(out["reference"][0]) references_norm.append(out["reference_norm"][0]) audio_paths.append(out['path'][0]) logger.info('computing metrics') wer = wer_metric.compute(references=references_norm, predictions=predictions_norm) wer = wer * 100 logger.info('metrics computed') logger.info(f'WER: {wer}') if args.save_predictions is True: preds_fp = f'preds_{args.dataset}_{args.config}_{args.split}_{now_str}.tsv' preds_fp = clean_filename(preds_fp) logger.info(f'saving predictions to: "{preds_fp}"') preds_df = pd.DataFrame({ 'audio_path': audio_paths, 'prediction_norm': predictions_norm, 'reference_norm': references_norm, 'prediction': predictions, 'reference': references, }) preds_df.to_csv(preds_fp, sep='\t', index=False) else: logger.info('save_predictions is False. will not save predictions to a file') if args.push_to_hub is True: logger.info(f'updating model card and pushing to HuggingFace Hub') evaluate.push_to_hub( model_id=args.model_id, metric_value=wer, metric_type="wer", metric_name="WER", dataset_name=args.dataset, dataset_type=args.dataset, dataset_config=args.config, dataset_split=args.split, task_type="automatic-speech-recognition", task_name="Automatic Speech Recognition" ) else: logger.info('push_to_hub is False. will not update model card and push to HuggingFace Hub') if __name__ == "__main__": parser = argparse.ArgumentParser() parser.add_argument( "--model_id", type=str, required=True, help="Model identifier. Should be loadable with 🤗 Transformers", ) parser.add_argument( "--dataset", type=str, default="mozilla-foundation/common_voice_11_0", help="Dataset name to evaluate the `model_id`. Should be loadable with 🤗 Datasets", ) parser.add_argument( "--config", type=str, required=True, help="Config of the dataset. *E.g.* `'en'` for the English split of Common Voice", ) parser.add_argument( "--split", type=str, default="test", help="Split of the dataset. *E.g.* `'test'`", ) parser.add_argument( "--text_column", type=str, required=True, help="Dataset column name containing target transcription of an audiofile" ) parser.add_argument( "--device", type=int, default=-1, help="The device to run the pipeline on. -1 for CPU (default), 0 for the first GPU and so on.", ) parser.add_argument( "--batch_size", type=int, default=16, help="Number of samples to go through each streamed batch.", ) parser.add_argument( "--max_eval_samples", type=int, default=None, help="Number of samples to be evaluated. Put a lower number e.g. 64 for testing this script.", ) parser.add_argument( "--streaming", type=bool, default=True, help="Choose whether you'd like to download the entire dataset or stream it during the evaluation.", ) parser.add_argument( "--language", type=str, required=True, help="Two letter language code for the transcription language, e.g. use 'en' for English.", ) parser.add_argument( '--push_to_hub', type=bool, default=True, help="Whether to update model card and push changes to HuggingFace Hub" ) parser.add_argument( '--save_predictions', type=bool, default=True, help="Whether to store predictions and target transcriptions to a file" ) args = parser.parse_args() main(args)