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#!/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)
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