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GetmanY1/wav2vec2-base-sami-22k
wav2vec2-base-sami-22k is a automatic speech recognition model from GetmanY1. Use it when you need speech turned into text. It is set up for transformers. The card lists the license as apache-2.0.
The base model pre-trained on 16kHz sampled speech audio. When using the model make sure that your speech input is also sampled at 16Khz.
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From the Hugging Face model README
The base model pre-trained on 16kHz sampled speech audio. When using the model make sure that your speech input is also sampled at 16Khz.
Note: This model does not have a tokenizer as it was pre-trained on audio alone. In order to use this model speech recognition, a tokenizer should be created and the model should be fine-tuned on labeled text data. Check out this blog for a more in-depth explanation of how to fine-tune the model.
Note: Fine-tuned version is available at GetmanY1/wav2vec2-base-sami-22k-finetuned
The Sámi Wav2Vec2 Base has the same architecture and uses the same training objective as the English one described in Paper. It is pre-trained on 22.4k hours of unlabeled Sámi speech from KAVI radio and television archive materials.
You can read more about the pre-trained model from this paper.
You can use this model for Sámi ASR (speech-to-text) and SER (Spoken Emotion Recognition) tasks.
See this notebook for more information on how to fine-tune the model.
This model was pre-trained with audio samples whose maximum length was 30 seconds so this model most likely works the best for quite short audios of similar length. However, you can try this model with a lot longer audios too and see how it works. If you encounter out-of-memory errors with very long audio files you can use the audio chunking method introduced in this blog post.
The data used for pre-training was from the KAVI archives so this model might have biases towards the voices of radio hosts. The pre-training data was filtered via neural VAD, but some non-speech events like music might be still present in the training data, which might cause issues when fine-tuned on clear (no background noise) speech.
This model was pre-trained with 22.4k hours of Sámi speech data from the following sources:
| Dataset | Hours | % of total hours |
|---|---|---|
| YleSámiRadio | 22415 h | 100 % |
Datasets were filtered to include a maximum length of 30 seconds long audio samples.
Training was done on 256 AMD MI250x GPU modules (512 GPUs from the software perspective), using LUMI.
The training script was provided by Fairseq and it is available here.
The following hyperparameters were used during training:
The pre-trained model was initialized with the following hyperparameters:
| Training Loss | Epoch | Validation Loss |
|---|---|---|
| 4.888 | 1 | 4.919 |
| 2.699 | 25 | 2.485 |
| 2.532 | 50 | 2.369 |
| 2.446 | 75 | 2.240 |
| 2.385 | 100 | 2.229 |
| 2.354 | 116 | 2.202 |
If you use our models or scripts, please cite our article as:
@inproceedings{getman-etal-2025-towards,
title = "Towards large-scale speech foundation models for a low-resource minority language",
author = "Getman, Yaroslav and
Gr{\'o}sz, Tam{\'a}s and
Hiovain-Asikainen, Katri and
Lehtonen, Tommi and
Kurimo, Mikko",
editor = "Johansson, Richard and
Stymne, Sara",
booktitle = "Proceedings of the Joint 25th Nordic Conference on Computational Linguistics and 11th Baltic Conference on Human Language Technologies (NoDaLiDa/Baltic-HLT 2025)",
month = mar,
year = "2025",
address = "Tallinn, Estonia",
publisher = "University of Tartu Library",
url = "https://aclanthology.org/2025.nodalida-1.19/",
pages = "192--200",
ISBN = "978-9908-53-109-0"
}
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