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---
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
```python
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:
```bibtex
@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](https://arxiv.org/abs/2503.23542)
for more details.
---
## License
This model is available under the
[Apache-2.0 License](https://www.apache.org/licenses/LICENSE-2.0).
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.
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