初始化项目,由ModelHub XC社区提供模型
Model: dbdmg/wav2vec2-xls-r-300m-italian-robust Source: Original Platform
This commit is contained in:
244
eval.py
Normal file
244
eval.py
Normal file
@@ -0,0 +1,244 @@
|
||||
#!/usr/bin/env python3
|
||||
import argparse
|
||||
import re
|
||||
from typing import Dict
|
||||
from sklearn import feature_extraction
|
||||
|
||||
import torch
|
||||
from src.data.normalization import normalize_string
|
||||
from datasets import Audio, Dataset, load_dataset, load_metric
|
||||
|
||||
from transformers import (
|
||||
AutoFeatureExtractor,
|
||||
pipeline,
|
||||
AutoTokenizer,
|
||||
Wav2Vec2Processor,
|
||||
Wav2Vec2ProcessorWithLM,
|
||||
Wav2Vec2ForCTC,
|
||||
AutoConfig,
|
||||
)
|
||||
|
||||
|
||||
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 = normalize_string(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
|
||||
# feature_extractor = AutoFeatureExtractor.from_pretrained(args.model_id)
|
||||
# sampling_rate = feature_extractor.sampling_rate
|
||||
|
||||
if args.ctcdecode:
|
||||
processor = Wav2Vec2ProcessorWithLM.from_pretrained(args.model_id)
|
||||
decoder = processor.decoder
|
||||
else:
|
||||
processor = Wav2Vec2Processor.from_pretrained(args.model_id)
|
||||
decoder = None
|
||||
|
||||
feature_extractor = processor.feature_extractor
|
||||
tokenizer = processor.tokenizer
|
||||
sampling_rate = feature_extractor.sampling_rate
|
||||
|
||||
config = AutoConfig.from_pretrained(args.model_id)
|
||||
model = Wav2Vec2ForCTC.from_pretrained(args.model_id)
|
||||
|
||||
# resample audio
|
||||
dataset = dataset.cast_column("audio", Audio(sampling_rate=sampling_rate))
|
||||
|
||||
# load eval pipeline
|
||||
if args.device is None:
|
||||
args.device = 0 if torch.cuda.is_available() else -1
|
||||
|
||||
asr = pipeline(
|
||||
"automatic-speech-recognition",
|
||||
model=model,
|
||||
config=config,
|
||||
feature_extractor=feature_extractor,
|
||||
decoder=decoder,
|
||||
tokenizer=tokenizer,
|
||||
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,
|
||||
#decoder_kwargs={"beam_width": args.beam_width},
|
||||
)
|
||||
|
||||
batch["prediction"] = prediction["text"]
|
||||
batch["target"] = normalize_text(
|
||||
batch["sentence"], invalid_chars_regex, normalize_to_lower
|
||||
)
|
||||
return batch
|
||||
|
||||
def map_and_decode(batch):
|
||||
inputs = processor(
|
||||
batch["audio"]["array"],
|
||||
sampling_rate=batch["audio"]["sampling_rate"],
|
||||
return_tensors="pt",
|
||||
)
|
||||
with torch.no_grad():
|
||||
logits = model(**inputs).logits
|
||||
transcription = processor.batch_decode(logits.numpy()).text
|
||||
batch["prediction"] = transcription
|
||||
batch["target"] = normalize_text(
|
||||
batch["sentence"], invalid_chars_regex, normalize_to_lower
|
||||
)
|
||||
return batch
|
||||
|
||||
# transcription = .lower()
|
||||
# run inference on all examples
|
||||
result = dataset.map(map_to_pred, remove_columns=dataset.column_names)
|
||||
|
||||
# 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 5 seconds.",
|
||||
)
|
||||
parser.add_argument(
|
||||
"--stride_length_s",
|
||||
type=float,
|
||||
default=None,
|
||||
help="Stride of the audio chunks. Defaults to 1 second.",
|
||||
)
|
||||
parser.add_argument(
|
||||
"--log_outputs",
|
||||
action="store_true",
|
||||
help="If defined, write outputs to log file for analysis.",
|
||||
)
|
||||
parser.add_argument(
|
||||
"--ctcdecode",
|
||||
action="store_true",
|
||||
help="Apply the ctc decoder to the output (only if present in the model card).",
|
||||
)
|
||||
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.",
|
||||
)
|
||||
parser.add_argument(
|
||||
"--beam_width",
|
||||
type=int,
|
||||
default=1,
|
||||
help="Beam width used by the pyctc decoder.",
|
||||
)
|
||||
args = parser.parse_args()
|
||||
|
||||
main(args)
|
||||
Reference in New Issue
Block a user