whisper-small-sp
This model is a fine-tuned version of openai/whisper-small on the commonvoice dataset v11
dataset.
It achieves the following results on the evaluation set:
- Loss: 0.4485
- Wer: 20.6842
Model description
More information needed
Intended uses & limitations
More information needed
Training and evaluation data
More information needed
Training procedure
Training hyperparameters
The following hyperparameters were used during training:
- learning_rate: 0.0005
- train_batch_size: 16
- eval_batch_size: 8
- seed: 42
- optimizer: Adam with betas=(0.9,0.999) and epsilon=1e-08
- lr_scheduler_type: linear
- lr_scheduler_warmup_steps: 500
- training_steps: 25000
- mixed_precision_training: Native AMP
Training results
Training Loss | Epoch | Step | Validation Loss | Wer |
---|---|---|---|---|
2.2671 | 0.13 | 1000 | 2.2108 | 76.2667 |
1.4465 | 0.26 | 2000 | 1.6057 | 67.8753 |
1.0997 | 0.39 | 3000 | 1.1928 | 54.2433 |
0.9389 | 0.52 | 4000 | 1.0020 | 47.8307 |
0.7881 | 0.65 | 5000 | 0.8933 | 46.0046 |
0.7596 | 0.78 | 6000 | 0.7721 | 38.5595 |
0.5678 | 0.91 | 7000 | 0.6903 | 36.2897 |
0.4412 | 1.04 | 8000 | 0.6476 | 32.7473 |
0.4239 | 1.17 | 9000 | 0.5973 | 30.8142 |
0.3935 | 1.3 | 10000 | 0.5444 | 29.0208 |
0.3307 | 1.43 | 11000 | 0.5024 | 27.0434 |
0.2937 | 1.56 | 12000 | 0.4608 | 24.7318 |
0.2471 | 1.69 | 13000 | 0.4259 | 22.8940 |
0.2357 | 1.82 | 14000 | 0.3936 | 21.6018 |
0.2292 | 1.95 | 15000 | 0.3776 | 20.8004 |
0.1493 | 2.08 | 16000 | 0.4599 | 24.0491 |
0.1708 | 2.21 | 17000 | 0.4370 | 23.3443 |
0.1385 | 2.34 | 18000 | 0.4277 | 22.3171 |
0.1288 | 2.47 | 19000 | 0.4050 | 21.0118 |
0.1627 | 2.6 | 20000 | 0.4507 | 23.4004 |
0.1675 | 2.73 | 21000 | 0.4346 | 22.8261 |
0.159 | 2.86 | 22000 | 0.4179 | 22.2949 |
0.1458 | 2.99 | 23000 | 0.3978 | 21.0810 |
0.0487 | 3.12 | 24000 | 0.4456 | 20.8617 |
0.0401 | 3.25 | 25000 | 0.4485 | 20.6842 |
Transcription:
from datasets import load_dataset, Audio
import torch
from transformers import WhisperProcessor, WhisperForConditionalGeneration
# device
device = torch.device("cuda" if torch.cuda.is_available() else "cpu")
# load the model
processor = WhisperProcessor.from_pretrained("clu-ling/whisper-small-spanish")
model = WhisperForConditionalGeneration.from_pretrained("clu-ling/whisper-small-spanish").to(device)
forced_decoder_ids = processor.get_decoder_prompt_ids(language="es", task="transcribe")
# load the dataset
commonvoice_eval = load_dataset("mozilla-foundation/common_voice_11_0", "es", split="validation", streaming=True)
commonvoice_eval = commonvoice_eval.cast_column("audio", Audio(sampling_rate=16000))
sample = next(iter(commonvoice_eval))["audio"]
# features and generate token ids
input_features = processor(sample["array"], sampling_rate=sample["sampling_rate"], return_tensors="pt").input_features
predicted_ids = model.generate(input_features.to(device), forced_decoder_ids=forced_decoder_ids)
# decode
transcription = processor.batch_decode(predicted_ids, skip_special_tokens=True)
print(transcription)
Evaluation:
Evaluates this model on mozilla-foundation/common_voice_11_0
test split.
from transformers.models.whisper.english_normalizer import BasicTextNormalizer
from datasets import load_dataset, Audio
import evaluate
import torch
import re
from transformers import WhisperProcessor, WhisperForConditionalGeneration
# device
device = torch.device("cuda" if torch.cuda.is_available() else "cpu")
# metric
wer_metric = evaluate.load("wer")
# model
processor = WhisperProcessor.from_pretrained("clu-ling/whisper-small-spanish")
model = WhisperForConditionalGeneration.from_pretrained("clu-ling/whisper-small-spanish")
# dataset
dataset = load_dataset("mozilla-foundation/common_voice_11_0", "es", split="test", )#cache_dir=args.cache_dir
dataset = dataset.cast_column("audio", Audio(sampling_rate=16000))
#for debuggings: it gets some examples
#dataset = dataset.shard(num_shards=10000, index=0)
#print(dataset)
def normalize(batch):
batch["gold_text"] = whisper_norm(batch['sentence'])
return batch
def map_wer(batch):
model.to(device)
forced_decoder_ids = processor.get_decoder_prompt_ids(language = "es", task = "transcribe")
inputs = processor(batch["audio"]["array"], sampling_rate=batch["audio"]["sampling_rate"], return_tensors="pt").input_features
with torch.no_grad():
generated_ids = model.generate(inputs=inputs.to(device), forced_decoder_ids=forced_decoder_ids)
transcription = processor.batch_decode(generated_ids, skip_special_tokens=True)[0]
batch["predicted_text"] = whisper_norm(transcription)
return batch
# process GOLD text
processed_dataset = dataset.map(normalize)
# get predictions
predicted = processed_dataset.map(map_wer)
# word error rate
wer = wer_metric.compute(references=predicted['gold_text'], predictions=predicted['predicted_text'])
wer = round(100 * wer, 2)
print("WER:", wer)
Framework versions
- Transformers 4.25.1
- Pytorch 1.13.0+cu117
- Datasets 2.7.1
- Tokenizers 0.13.2
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