language:
- gl
license: apache-2.0
base_model: openai/whisper-tiny
tags:
- whisper-event
- generated_from_trainer
datasets:
- mozilla-foundation/common_voice_13_0
metrics:
- wer
model-index:
- name: Whisper Tiny Galician
results:
- task:
name: Automatic Speech Recognition
type: automatic-speech-recognition
dataset:
name: mozilla-foundation/common_voice_13_0 gl
type: mozilla-foundation/common_voice_13_0
config: gl
split: test
args: gl
metrics:
- name: Wer
type: wer
value: 26.13307119205298
Whisper Tiny Galician
Model summary
Whisper Tiny Galician is an automatic speech recognition (ASR) model for Galician (gl) speech. It is fine-tuned from [openai/whisper-tiny] on the Galician portion of Mozilla Common Voice 13.0, achieving a Word Error Rate (WER) of 26.13% on the Common Voice evaluation split.
This model provides lightweight transcription capabilities for Galician speech, suitable for low-resource applications or devices with limited computational capacity.
Model description
- Architecture: Transformer-based encoder–decoder (Whisper)
- Base model: openai/whisper-tiny
- Language: Galician (gl)
- Task: Automatic Speech Recognition (ASR)
- Output: Text transcription in Galician
- Decoding: Autoregressive sequence-to-sequence decoding
This tiny model leverages Whisper’s multilingual pretraining and is fine-tuned on Galician speech data to provide basic transcription functionality for a low-resource language, ideal for experimentation and lightweight applications.
Intended use
Primary use cases
- Lightweight transcription of Galician audio recordings
- Low-resource or offline ASR pipelines
- Educational and research purposes
Intended users
- Researchers working on Galician or low-resource ASR
- Developers building Galician speech applications
- Academic or institutional users
Out-of-scope use
- High-accuracy transcription requirements
- Real-time or low-latency ASR without optimization
- Speech translation tasks
Limitations and known issues
- Performance may degrade on:
- Noisy or low-quality recordings
- Conversational or spontaneous speech
- Accents underrepresented in Common Voice
- Transcription errors are expected due to the small model size
- Dataset biases from Common Voice may be reflected in outputs
Users are encouraged to evaluate the model on their own data before deployment.
Training and evaluation data
Training data
- Dataset: Mozilla Common Voice 13.0 (Galician 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 data
- Dataset: Mozilla Common Voice 13.0 (Galician evaluation split)
- Metric: Word Error Rate (WER)
Evaluation results
| Metric | Value |
|---|---|
| WER (eval) | 26.13% |
This reflects the expected performance of a tiny Whisper model fine-tuned for Galician.
Training procedure
Training hyperparameters
- Learning rate: 3.75e-5
- Optimizer: Adam (β1=0.9, β2=0.999, ε=1e-8)
- LR scheduler: Linear
- Warmup steps: 500
- Training steps: 5,000
- Train batch size: 256
- Evaluation batch size: 128
- Seed: 42
- Mixed precision training: Native AMP
Training results (summary)
| Training Loss | Epoch | Step | Validation Loss | WER |
|---|---|---|---|---|
| 0.3626 | 20.0 | 1000 | 0.5407 | 30.8464 |
| 0.1103 | 40.0 | 2000 | 0.5370 | 27.0402 |
| 0.0473 | 60.0 | 3000 | 0.5769 | 26.7263 |
| 0.03 | 80.0 | 4000 | 0.5936 | 26.1382 |
| 0.0244 | 100.0 | 5000 | 0.6003 | 26.1331 |
Framework versions
- Transformers 4.37.2
- PyTorch 2.2.0+cu121
- Datasets 2.16.1
- Tokenizers 0.15.1
How to use
from transformers import pipeline
hf_model = "HiTZ/whisper-tiny-gl" # 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.