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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)
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