--- language: - ca license: apache-2.0 base_model: openai/whisper-large-v2 tags: - whisper-event - generated_from_trainer datasets: - mozilla-foundation/common_voice_13_0 metrics: - wer model-index: - name: Whisper Large-V2 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: 4.671620462989425 --- # Whisper Large-V2 Catalan ## Model summary **Whisper Large-V2 Catalan** is an automatic speech recognition (ASR) model for **Catalan (ca)** speech. It is fine-tuned from [openai/whisper-large-v2] on the **Catalan subset of Mozilla Common Voice 13.0**, achieving a **Word Error Rate (WER) of 4.6716%** on the evaluation split. This variant provides improved transcription accuracy and model efficiency compared to the original large model, leveraging enhancements from the V2 architecture. --- ## Model description * **Architecture:** Transformer-based encoder–decoder (Whisper V2) * **Base model:** openai/whisper-large-v2 * **Language:** Catalan (ca) * **Task:** Automatic Speech Recognition (ASR) * **Output:** Text transcription in Catalan * **Decoding:** Autoregressive sequence-to-sequence decoding Fine-tuned to optimize 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 underrepresented in Common Voice * Occasional transcription errors on challenging 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) | **4.6716%** | --- ## 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 | |:-------------:|:-----:|:-----:|:---------------:|:------:| | 0.1072 | 1.02 | 1000 | 0.1637 | 7.0329 | | 0.0239 | 3.02 | 2000 | 0.1784 | 7.0277 | | 0.0507 | 5.02 | 3000 | 0.1754 | 6.5773 | | 0.0571 | 7.02 | 4000 | 0.1620 | 6.5047 | | 0.0193 | 9.02 | 5000 | 0.1821 | 6.4887 | | 0.0625 | 11.02 | 6000 | 0.1443 | 6.7585 | | 0.0752 | 13.02 | 7000 | 0.1653 | 5.9097 | | 0.0359 | 15.02 | 8000 | 0.1406 | 5.8760 | | 0.0565 | 17.01 | 9000 | 0.1496 | 5.9680 | | 0.0196 | 19.01 | 10000 | 0.1788 | 5.2746 | | 0.0215 | 21.01 | 11000 | 0.1539 | 5.3895 | | 0.0178 | 23.01 | 12000 | 0.1800 | 5.3764 | | 0.0114 | 25.01 | 13000 | 0.1709 | 5.2078 | | 0.0123 | 27.01 | 14000 | 0.1827 | 5.2003 | | 0.0337 | 29.01 | 15000 | 0.1553 | 5.3655 | | 0.0108 | 31.01 | 16000 | 0.1476 | 4.9151 | | 0.0194 | 33.01 | 17000 | 0.1396 | 4.8477 | | 0.0472 | 35.0 | 18000 | 0.1202 | 4.8717 | | 0.0401 | 37.0 | 19000 | 0.1494 | 4.6716 | | 0.0127 | 39.0 | 20000 | 0.1187 | 4.7276 | --- ## 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-v2-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.