✨ Add evalutaion scripts.
Browse files- run_eval.sh +7 -0
- run_eval_whisper_streaming.py +150 -0
run_eval.sh
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python run_eval_whisper_streaming.py \
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--model_id="jstoone/whisper-medium-da-cv11" \
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--dataset="google/fleurs" \
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--config="da_dk" \
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--language="da" \
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--device=0
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run_eval_whisper_streaming.py
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import argparse
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from transformers import pipeline
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from transformers.models.whisper.english_normalizer import BasicTextNormalizer
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from datasets import load_dataset, Audio
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import evaluate
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wer_metric = evaluate.load("wer")
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def is_target_text_in_range(ref):
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if ref.strip() == "ignore time segment in scoring":
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return False
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else:
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return ref.strip() != ""
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def get_text(sample):
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if "text" in sample:
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return sample["text"]
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elif "sentence" in sample:
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return sample["sentence"]
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elif "normalized_text" in sample:
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return sample["normalized_text"]
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elif "transcript" in sample:
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return sample["transcript"]
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elif "transcription" in sample:
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return sample["transcription"]
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else:
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raise ValueError(
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f"Expected transcript column of either 'text', 'sentence', 'normalized_text' or 'transcript'. Got sample of "
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".join{sample.keys()}. Ensure a text column name is present in the dataset."
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)
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whisper_norm = BasicTextNormalizer()
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def normalise(batch):
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batch["norm_text"] = whisper_norm(get_text(batch))
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return batch
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def data(dataset):
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for i, item in enumerate(dataset):
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yield {**item["audio"], "reference": item["norm_text"]}
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def main(args):
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batch_size = args.batch_size
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whisper_asr = pipeline(
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"automatic-speech-recognition", model=args.model_id, device=args.device
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)
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whisper_asr.model.config.forced_decoder_ids = (
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whisper_asr.tokenizer.get_decoder_prompt_ids(
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language=args.language, task="transcribe"
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)
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)
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dataset = load_dataset(
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args.dataset,
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args.config,
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split=args.split,
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streaming=args.streaming,
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use_auth_token=True,
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)
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# Only uncomment for debugging
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dataset = dataset.take(args.max_eval_samples)
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dataset = dataset.cast_column("audio", Audio(sampling_rate=16000))
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dataset = dataset.map(normalise)
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dataset = dataset.filter(is_target_text_in_range, input_columns=["norm_text"])
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predictions = []
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references = []
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# run streamed inference
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for out in whisper_asr(data(dataset), batch_size=batch_size):
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predictions.append(whisper_norm(out["text"]))
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references.append(out["reference"][0])
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wer = wer_metric.compute(references=references, predictions=predictions)
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wer = round(100 * wer, 2)
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print("WER:", wer)
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if __name__ == "__main__":
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parser = argparse.ArgumentParser()
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parser.add_argument(
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"--model_id",
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type=str,
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required=True,
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help="Model identifier. Should be loadable with 🤗 Transformers",
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)
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parser.add_argument(
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"--dataset",
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type=str,
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default="mozilla-foundation/common_voice_11_0",
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help="Dataset name to evaluate the `model_id`. Should be loadable with 🤗 Datasets",
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)
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parser.add_argument(
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"--config",
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type=str,
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required=True,
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help="Config of the dataset. *E.g.* `'en'` for the English split of Common Voice",
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)
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parser.add_argument(
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"--split",
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type=str,
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default="test",
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help="Split of the dataset. *E.g.* `'test'`",
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)
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parser.add_argument(
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"--device",
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type=int,
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default=-1,
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help="The device to run the pipeline on. -1 for CPU (default), 0 for the first GPU and so on.",
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)
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parser.add_argument(
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"--batch_size",
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type=int,
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default=16,
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help="Number of samples to go through each streamed batch.",
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)
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parser.add_argument(
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"--max_eval_samples",
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type=int,
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default=None,
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help="Number of samples to be evaluated. Put a lower number e.g. 64 for testing this script.",
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)
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parser.add_argument(
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"--streaming",
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type=bool,
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default=True,
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help="Choose whether you'd like to download the entire dataset or stream it during the evaluation.",
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)
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parser.add_argument(
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"--language",
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type=str,
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required=True,
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help="Two letter language code for the transcription language, e.g. use 'en' for English.",
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)
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args = parser.parse_args()
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main(args)
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