Automatic Speech Recognition
Transformers
PyTorch
JAX
French
wav2vec2
audio
hf-asr-leaderboard
mozilla-foundation/common_voice_6_0
robust-speech-event
speech
xlsr-fine-tuning-week
Eval Results (legacy)
Instructions to use bonvent/test2 with libraries, inference providers, notebooks, and local apps. Follow these links to get started.
- Libraries
- Transformers
How to use bonvent/test2 with Transformers:
# Use a pipeline as a high-level helper from transformers import pipeline pipe = pipeline("automatic-speech-recognition", model="bonvent/test2")# Load model directly from transformers import AutoProcessor, AutoModelForCTC processor = AutoProcessor.from_pretrained("bonvent/test2") model = AutoModelForCTC.from_pretrained("bonvent/test2") - Notebooks
- Google Colab
- Kaggle
| #!/usr/bin/env python3 | |
| from datasets import load_dataset, load_metric, Audio, Dataset | |
| from transformers import pipeline, AutoFeatureExtractor, AutoTokenizer, AutoConfig, AutoModelForCTC, Wav2Vec2Processor, Wav2Vec2ProcessorWithLM | |
| import re | |
| import torch | |
| import argparse | |
| from typing import Dict | |
| def log_results(result: Dataset, args: Dict[str, str]): | |
| """ DO NOT CHANGE. This function computes and logs the result metrics. """ | |
| log_outputs = args.log_outputs | |
| dataset_id = "_".join(args.dataset.split("/") + [args.config, args.split]) | |
| # load metric | |
| wer = load_metric("wer") | |
| cer = load_metric("cer") | |
| # compute metrics | |
| wer_result = wer.compute(references=result["target"], predictions=result["prediction"]) | |
| cer_result = cer.compute(references=result["target"], predictions=result["prediction"]) | |
| # print & log results | |
| result_str = ( | |
| f"WER: {wer_result}\n" | |
| f"CER: {cer_result}" | |
| ) | |
| print(result_str) | |
| with open(f"{dataset_id}_eval_results.txt", "w") as f: | |
| f.write(result_str) | |
| # log all results in text file. Possibly interesting for analysis | |
| if log_outputs is not None: | |
| pred_file = f"log_{dataset_id}_predictions.txt" | |
| target_file = f"log_{dataset_id}_targets.txt" | |
| with open(pred_file, "w") as p, open(target_file, "w") as t: | |
| # mapping function to write output | |
| def write_to_file(batch, i): | |
| p.write(f"{i}" + "\n") | |
| p.write(batch["prediction"] + "\n") | |
| t.write(f"{i}" + "\n") | |
| t.write(batch["target"] + "\n") | |
| result.map(write_to_file, with_indices=True) | |
| def normalize_text(text: str, invalid_chars_regex: str, to_lower: bool) -> str: | |
| """ DO ADAPT FOR YOUR USE CASE. this function normalizes the target text. """ | |
| text = text.lower() if to_lower else text.upper() | |
| text = re.sub(invalid_chars_regex, " ", text) | |
| text = re.sub("\s+", " ", text).strip() | |
| return text | |
| def main(args): | |
| # load dataset | |
| dataset = load_dataset(args.dataset, args.config, split=args.split, use_auth_token=True) | |
| # for testing: only process the first two examples as a test | |
| # dataset = dataset.select(range(10)) | |
| # load processor | |
| if args.greedy: | |
| processor = Wav2Vec2Processor.from_pretrained(args.model_id) | |
| decoder = None | |
| else: | |
| processor = Wav2Vec2ProcessorWithLM.from_pretrained(args.model_id) | |
| decoder = processor.decoder | |
| feature_extractor = processor.feature_extractor | |
| tokenizer = processor.tokenizer | |
| # resample audio | |
| dataset = dataset.cast_column("audio", Audio(sampling_rate=feature_extractor.sampling_rate)) | |
| # load eval pipeline | |
| if args.device is None: | |
| args.device = 0 if torch.cuda.is_available() else -1 | |
| config = AutoConfig.from_pretrained(args.model_id) | |
| model = AutoModelForCTC.from_pretrained(args.model_id) | |
| #asr = pipeline("automatic-speech-recognition", model=args.model_id, device=args.device) | |
| asr = pipeline("automatic-speech-recognition", config=config, model=model, tokenizer=tokenizer, | |
| feature_extractor=feature_extractor, decoder=decoder, device=args.device) | |
| # build normalizer config | |
| tokenizer = AutoTokenizer.from_pretrained(args.model_id) | |
| tokens = [x for x in tokenizer.convert_ids_to_tokens(range(0, tokenizer.vocab_size))] | |
| special_tokens = [ | |
| tokenizer.pad_token, tokenizer.word_delimiter_token, | |
| tokenizer.unk_token, tokenizer.bos_token, | |
| tokenizer.eos_token, | |
| ] | |
| non_special_tokens = [x for x in tokens if x not in special_tokens] | |
| invalid_chars_regex = f"[^\s{re.escape(''.join(set(non_special_tokens)))}]" | |
| normalize_to_lower = False | |
| for token in non_special_tokens: | |
| if token.isalpha() and token.islower(): | |
| normalize_to_lower = True | |
| break | |
| # map function to decode audio | |
| def map_to_pred(batch, args=args, asr=asr, invalid_chars_regex=invalid_chars_regex, normalize_to_lower=normalize_to_lower): | |
| prediction = asr(batch["audio"]["array"], chunk_length_s=args.chunk_length_s, stride_length_s=args.stride_length_s) | |
| batch["prediction"] = prediction["text"] | |
| batch["target"] = normalize_text(batch["sentence"], invalid_chars_regex, normalize_to_lower) | |
| return batch | |
| # run inference on all examples | |
| result = dataset.map(map_to_pred, remove_columns=dataset.column_names) | |
| # filtering out empty targets | |
| result = result.filter(lambda example: example["target"] != "") | |
| # compute and log_results | |
| # do not change function below | |
| log_results(result, args) | |
| 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, required=True, 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 Common Voice" | |
| ) | |
| parser.add_argument( | |
| "--split", type=str, required=True, help="Split of the dataset. *E.g.* `'test'`" | |
| ) | |
| parser.add_argument( | |
| "--chunk_length_s", type=float, default=None, help="Chunk length in seconds. Defaults to None. For long audio files a good value would be 5.0 seconds." | |
| ) | |
| parser.add_argument( | |
| "--stride_length_s", type=float, default=None, help="Stride of the audio chunks. Defaults to None. For long audio files a good value would be 1.0 seconds." | |
| ) | |
| parser.add_argument( | |
| "--log_outputs", action='store_true', help="If defined, write outputs to log file for analysis." | |
| ) | |
| parser.add_argument( | |
| "--greedy", action='store_true', help="If defined, the LM will be ignored during inference." | |
| ) | |
| parser.add_argument( | |
| "--device", | |
| type=int, | |
| default=None, | |
| help="The device to run the pipeline on. -1 for CPU (default), 0 for the first GPU and so on.", | |
| ) | |
| args = parser.parse_args() | |
| main(args) | |