whisper-large-ca / README.md
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metadata
language:
  - ca
license: apache-2.0
base_model: openai/whisper-large
tags:
  - whisper-event
  - generated_from_trainer
datasets:
  - mozilla-foundation/common_voice_13_0
metrics:
  - wer
model-index:
  - name: Whisper Large Catalan
    results:
      - task:
          name: Automatic Speech Recognition
          type: automatic-speech-recognition
        dataset:
          name: mozilla-foundation/common_voice_13_0 ca
          type: mozilla-foundation/common_voice_13_0
          config: ca
          split: test
          args: ca
        metrics:
          - name: Wer
            type: wer
            value: 5.070020005715919

Whisper Large Catalan

Model summary

Whisper Large Catalan is an automatic speech recognition (ASR) model for Catalan (ca) speech. It is fine-tuned from [openai/whisper-large] on the Catalan subset of Mozilla Common Voice 13.0, achieving a Word Error Rate (WER) of 5.070% on the evaluation split.

This model is suitable for high-accuracy transcription and supports longer audio sequences with larger model capacity compared to the medium variant.


Model description

  • Architecture: Transformer-based encoder–decoder (Whisper)
  • Base model: openai/whisper-large
  • Language: Catalan (ca)
  • Task: Automatic Speech Recognition (ASR)
  • Output: Text transcription in Catalan
  • Decoding: Autoregressive sequence-to-sequence decoding

Fine-tuned to improve transcription quality on Catalan audio.


Intended use

Primary use cases

  • High-accuracy transcription of Catalan audio
  • Research and development in Catalan ASR
  • Media, educational, or accessibility applications

Out-of-scope use

  • Real-time transcription without optimization
  • Speech translation
  • Safety-critical applications without further validation

Limitations and known issues

  • Performance may degrade on:
    • Noisy or low-quality recordings
    • Conversational or spontaneous speech
    • Regional dialects not well represented in Common Voice
  • Occasional transcription errors on difficult audio

Training and evaluation data

  • Dataset: Mozilla Common Voice 13.0 (Catalan subset)

  • Data type: Crowd-sourced, read speech

  • Preprocessing:

    • Audio resampled to 16 kHz
    • Text normalized using Whisper tokenizer
    • Filtering of invalid or problematic samples
  • Evaluation metric: Word Error Rate (WER) on held-out evaluation set


Evaluation results

Metric Value
WER (eval) 5.070%

Training procedure

Training hyperparameters

  • Learning rate: 1e-5
  • Optimizer: Adam (β1=0.9, β2=0.999, ε=1e-8)
  • LR scheduler: Linear
  • Warmup steps: 500
  • Training steps: 20,000
  • Train batch size: 32
  • Eval batch size: 16
  • Gradient accumulation steps: 2
  • Seed: 42

Training results (summary)

Training Loss Epoch Step Validation Loss WER
0.1059 1.02 1000 0.1744 7.6342
0.0159 3.02 2000 0.1943 7.3850
0.0526 5.02 3000 0.1899 6.8522
0.058 7.02 4000 0.1782 6.7802
0.0161 9.02 5000 0.1995 6.6339
0.065 11.02 6000 0.1563 6.4544
0.082 13.02 7000 0.1789 6.0309
0.0339 15.02 8000 0.1509 5.7554
0.0581 17.01 9000 0.1573 6.0446
0.0181 19.01 10000 0.1838 5.5913
0.0188 21.01 11000 0.1610 5.4804
0.0134 23.01 12000 0.1821 5.3953
0.008 25.01 13000 0.1748 5.3804
0.0071 27.01 14000 0.1858 5.4701
0.0371 29.01 15000 0.1610 5.6599
0.0076 31.01 16000 0.1571 5.1655
0.0181 33.01 17000 0.1449 5.4558
0.0522 35.0 18000 0.1340 5.8388
0.0356 37.0 19000 0.1458 5.0700
0.0132 39.0 20000 0.1310 5.1941

Framework versions

  • Transformers 4.33.0.dev0
  • PyTorch 2.0.1+cu117
  • Datasets 2.14.4
  • Tokenizers 0.13.3

How to use

from transformers import pipeline

hf_model = "HiTZ/whisper-large-ca"  # replace with actual repo ID
device = 0  # set to -1 for CPU

pipe = pipeline(
    task="automatic-speech-recognition",
    model=hf_model,
    device=device
)

result = pipe("audio.wav")
print(result["text"])

Ethical considerations and risks

  • This model transcribes speech and may process personal data.
  • Users should ensure compliance with applicable data protection laws (e.g., GDPR).
  • The model should not be used for surveillance or non-consensual audio processing.

Citation

If you use this model in your research, please cite:

@misc{dezuazo2025whisperlmimprovingasrmodels,
  title={Whisper-LM: Improving ASR Models with Language Models for Low-Resource Languages},
  author={Xabier de Zuazo and Eva Navas and Ibon Saratxaga and Inma Hernáez Rioja},
  year={2025},
  eprint={2503.23542},
  archivePrefix={arXiv},
  primaryClass={cs.CL}
}

Please, check the related paper preprint in arXiv:2503.23542 for more details.


License

This model is available under the Apache-2.0 License. You are free to use, modify, and distribute this model as long as you credit the original creators.


Contact and attribution

  • Fine-tuning and evaluation: HiTZ/Aholab - Basque Center for Language Technology
  • Base model: OpenAI Whisper
  • Dataset: Mozilla Common Voice

For questions or issues, please open an issue in the model repository.