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.