--- language: - es 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 Spanish results: - task: name: Automatic Speech Recognition type: automatic-speech-recognition dataset: name: mozilla-foundation/common_voice_13_0 es type: mozilla-foundation/common_voice_13_0 config: es split: test args: es metrics: - name: Wer type: wer value: 4.89488506963824 --- # Whisper Large-V2 Spanish ## Model summary **Whisper Large-V2 Spanish** is a state-of-the-art automatic speech recognition (ASR) model for **Spanish (es)**, fine-tuned from [openai/whisper-large-v2] on the **Spanish subset of Mozilla Common Voice 13.0**. It achieves a **Word Error Rate (WER) of 4.8949%** on the evaluation set. This model is optimized for applications that demand **very high transcription accuracy** while benefiting from improvements in Large-V2 architecture over the original Large model. --- ## Model description * **Architecture:** Transformer-based encoder–decoder (Whisper Large-V2) * **Base model:** openai/whisper-large-v2 * **Language:** Spanish (es) * **Task:** Automatic Speech Recognition (ASR) * **Output:** Text transcription in Spanish * **Decoding:** Autoregressive sequence-to-sequence decoding Large-V2 introduces enhanced pretraining for multilingual and robust speech recognition, yielding lower WERs than Large in most languages. --- ## Intended use ### Primary use cases * High-accuracy Spanish speech transcription * Long-form audio content like podcasts, interviews, or lectures * ASR research and benchmarking in Spanish ### Limitations * Performance may degrade on highly noisy audio, heavy regional accents, or spontaneous speech * High computational cost for real-time inference * Not validated for legal, medical, or safety-critical transcription without human review --- ## Training and evaluation data * **Dataset:** Mozilla Common Voice 13.0 (Spanish subset) * **Data type:** Crowd-sourced read speech * **Preprocessing:** * Audio resampled to 16 kHz * Text tokenized using Whisper tokenizer * Removal of invalid or corrupted samples * **Evaluation metric:** Word Error Rate (WER) on held-out evaluation set --- ## Evaluation results | Metric | Value | | ---------- | ---------- | | WER (eval) | **4.8949%** | --- ## 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: 20000 * Train batch size: 32 (gradient accumulation 2 → effective batch size 64) * Eval batch size: 16 * Seed: 42 ### Training results (summary) | Training Loss | Epoch | Step | Validation Loss | WER | |:-------------:|:-----:|:-----:|:---------------:|:------:| | 0.0869 | 2.0 | 1000 | 0.1754 | 6.1516 | | 0.0913 | 4.0 | 2000 | 0.1652 | 5.7500 | | 0.051 | 6.0 | 3000 | 0.1643 | 5.7757 | | 0.0391 | 8.0 | 4000 | 0.1881 | 5.6589 | | 0.0104 | 10.0 | 5000 | 0.2026 | 5.6211 | | 0.0806 | 12.01 | 6000 | 0.1741 | 5.7398 | | 0.0077 | 14.01 | 7000 | 0.2119 | 5.6038 | | 0.0357 | 16.01 | 8000 | 0.1776 | 5.6147 | | 0.1087 | 18.01 | 9000 | 0.1868 | 5.5172 | | 0.0401 | 20.01 | 10000 | 0.2014 | 5.4428 | | 0.0334 | 22.01 | 11000 | 0.1751 | 5.2824 | | 0.0071 | 24.01 | 12000 | 0.2295 | 5.2490 | | 0.0374 | 26.01 | 13000 | 0.2098 | 5.2574 | | 0.0023 | 28.01 | 14000 | 0.2498 | 5.0418 | | 0.0025 | 30.01 | 15000 | 0.2311 | 4.9385 | | 0.0006 | 32.01 | 16000 | 0.2544 | 4.8949 | | 0.0009 | 34.02 | 17000 | 0.2691 | 5.1246 | | 0.003 | 36.02 | 18000 | 0.2249 | 5.0277 | | 0.0009 | 38.02 | 19000 | 0.2603 | 5.0373 | | 0.0008 | 40.02 | 20000 | 0.2657 | 5.0225 | --- ## Framework versions - Transformers 4.33.0.dev0 - PyTorch 2.0.1+cu117 - Datasets 2.14.4 - Tokenizers 0.13.3 --- ## Example usage ```python from transformers import pipeline hf_model = "HiTZ/whisper-large-v2-es" # replace with actual repo ID device = 0 # -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.