--- language: - es license: apache-2.0 base_model: openai/whisper-large-v3 tags: - whisper-event - generated_from_trainer datasets: - mozilla-foundation/common_voice_13_0 metrics: - wer model-index: - name: Whisper Large-V3 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: 4.9295277686894154 --- # Whisper Large-V3 Spanish ## Model summary **Whisper Large-V3 Spanish** is a cutting-edge automatic speech recognition (ASR) model for **Spanish (es)**, fine-tuned from [openai/whisper-large-v3] on the **Spanish subset of Mozilla Common Voice 13.0**. It achieves a **Word Error Rate (WER) of 4.9295%** on the evaluation set, making it one of the most accurate Whisper models for Spanish. This model incorporates improvements from the Large-V3 architecture, including **better noise robustness, enhanced multilingual pretraining, and mixed precision training for efficiency**. --- ## Model description * **Architecture:** Transformer-based encoder–decoder (Whisper Large-V3) * **Base model:** openai/whisper-large-v3 * **Language:** Spanish (es) * **Task:** Automatic Speech Recognition (ASR) * **Output:** Text transcription in Spanish * **Decoding:** Autoregressive sequence-to-sequence decoding Large-V3 builds upon Large-V2, offering lower WER and improved generalization across accents and audio conditions. --- ## Intended use ### Primary use cases * High-accuracy transcription of Spanish audio * Podcasts, interviews, lectures, and long-form audio * Research or commercial applications requiring top-tier ASR performance in Spanish ### Limitations * Performance may drop on heavily accented or extremely noisy audio * High memory and compute requirements, particularly for real-time use * Not suitable for critical domains (medical, legal) without human verification --- ## 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 * Filtering of corrupted or invalid samples * **Evaluation metric:** Word Error Rate (WER) on held-out evaluation set --- ## Evaluation results | Metric | Value | | ---------- | ---------- | | WER (eval) | **4.9295%** | --- ## 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 * Mixed precision training: Native AMP ### Training results (summary) | Training Loss | Epoch | Step | Validation Loss | Wer | |:-------------:|:-----:|:-----:|:---------------:|:------:| | 0.058 | 2.04 | 1000 | 0.1540 | 4.6851 | | 0.0124 | 4.07 | 2000 | 0.1829 | 4.6787 | | 0.0052 | 6.11 | 3000 | 0.2190 | 4.8096 | | 0.0024 | 8.15 | 4000 | 0.2289 | 4.8776 | | 0.0024 | 10.18 | 5000 | 0.2341 | 4.8923 | | 0.0015 | 12.22 | 6000 | 0.2459 | 4.9340 | | 0.0021 | 14.26 | 7000 | 0.2558 | 4.9276 | | 0.0011 | 16.29 | 8000 | 0.2540 | 5.1015 | | 0.0013 | 18.33 | 9000 | 0.2611 | 5.1855 | | 0.0005 | 20.37 | 10000 | 0.2720 | 4.9379 | | 0.0028 | 22.4 | 11000 | 0.2614 | 5.0110 | | 0.0004 | 24.44 | 12000 | 0.2652 | 4.9898 | | 0.0004 | 26.48 | 13000 | 0.2850 | 4.9776 | | 0.0006 | 28.51 | 14000 | 0.2736 | 4.9732 | | 0.0002 | 30.55 | 15000 | 0.2944 | 5.1566 | | 0.0002 | 32.59 | 16000 | 0.2949 | 5.0007 | | 0.0001 | 34.62 | 17000 | 0.3094 | 4.9552 | | 0.0 | 36.66 | 18000 | 0.3185 | 4.9622 | | 0.0 | 38.7 | 19000 | 0.3229 | 4.9462 | | 0.0 | 40.73 | 20000 | 0.3245 | 4.9295 | --- ## Framework versions - Transformers 4.37.2 - PyTorch 2.2.0+cu121 - Datasets 2.16.1 - Tokenizers 0.15.1 --- ## Example usage ```python from transformers import pipeline hf_model = "HiTZ/whisper-large-v3-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.