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metadata
license: apache-2.0
tags:
  - generated_from_trainer
metrics:
  - wer
model-index:
  - name: whisper-large-v2-japanese-5k-steps
    results: []

whisper-large-v2-japanese-5k-steps

This model is a fine-tuned version of openai/whisper-large-v2 on the Japanese CommonVoice dataset (v11).. It achieves the following results on the evaluation set:

  • Loss: 0.4200
  • Wer: 0.7449

Model description

This model is finetuned for 5000 steps for research purposes which means that the transcriptions might not be that satisfactory for users.

Training and evaluation data

  • Training Data: CommonVoice (v11) train split
  • Validation Data: CommonVoice (v11) Validation split
  • Test Data: CommonVoice (v11) Test split

Training procedure

Training hyperparameters

The following hyperparameters were used during training:

  • learning_rate: 1e-05
  • train_batch_size: 50
  • eval_batch_size: 16
  • 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: 5000
  • mixed_precision_training: Native AMP

Training results

Training Loss Epoch Step Validation Loss Wer
0.0111 7.63 1000 0.3210 0.7888
0.0007 15.27 2000 0.3585 0.7478
0.0003 22.9 3000 0.3937 0.7432
0.0002 30.53 4000 0.4123 0.7443
0.0002 38.17 5000 0.4200 0.7449

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-large-v2-japanese-5k-steps")
model = WhisperForConditionalGeneration.from_pretrained("clu-ling/whisper-large-v2-japanese-5k-steps").to(device)
forced_decoder_ids = processor.get_decoder_prompt_ids(language="ja", task="transcribe")

# load the dataset
commonvoice_eval = load_dataset("mozilla-foundation/common_voice_11_0", "ja", 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=input_speech["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)
transcription = processor.batch_decode(predicted_ids, skip_special_tokens=True)

print(transcription)

Framework versions

  • Transformers 4.26.0.dev0
  • Pytorch 1.13.1
  • Datasets 2.8.1.dev0
  • Tokenizers 0.13.2