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---
license: apache-2.0
language: fi
tags:
- automatic-speech-recognition
- fi
- finnish
model-index:
- name: wav2vec2-base-fi-lp-from-scratch-100h
results:
- task:
name: Automatic Speech Recognition
type: automatic-speech-recognition
dataset:
name: Lahjoita puhetta (Donate Speech)
type: lahjoita-puhetta
args: fi
metrics:
- name: Dev WER
type: wer
value: 26.40
- name: Dev CER
type: cer
value: 6.86
- name: Test WER
type: wer
value: 28.92
- name: Test CER
type: cer
value: 8.09
---
# Colloquial Finnish Wav2vec2-Base ASR
[GetmanY1/wav2vec2-base-fi-lp-from-scratch](https://huggingface.co/GetmanY1/wav2vec2-base-fi-lp-from-scratch) fine-tuned on 100 hours of [Lahjoita puhetta (Donate Speech)](https://link.springer.com/article/10.1007/s10579-022-09606-3) on 16kHz sampled speech audio. When using the model make sure that your speech input is also sampled at 16Khz.
## Model description
The Finnish Wav2Vec2 Base has the same architecture and uses the same training objective as the English and multilingual one described in [Paper](https://arxiv.org/abs/2006.11477).
You can read more about the pre-trained model from [this paper](TODO). The training scripts are available on [GitHub](https://github.com/aalto-speech/colloquial-Finnish-wav2vec2)
## Intended uses & limitations
You can use this model for Finnish ASR (speech-to-text).
### How to use
To transcribe audio files the model can be used as a standalone acoustic model as follows:
```
from transformers import Wav2Vec2Processor, Wav2Vec2ForCTC
from datasets import load_dataset
import torch
# load model and processor
processor = Wav2Vec2Processor.from_pretrained("GetmanY1/wav2vec2-base-fi-lp-from-scratch-100h")
model = Wav2Vec2ForCTC.from_pretrained("GetmanY1/wav2vec2-base-fi-lp-from-scratch-100h")
# load dummy dataset and read soundfiles
ds = load_dataset("mozilla-foundation/common_voice_16_1", "fi", split='test')
# tokenize
input_values = processor(ds[0]["audio"]["array"], return_tensors="pt", padding="longest").input_values # Batch size 1
# retrieve logits
logits = model(input_values).logits
# take argmax and decode
predicted_ids = torch.argmax(logits, dim=-1)
transcription = processor.batch_decode(predicted_ids)
```
### Limitations and bias
This model was fine-tuned with audio samples whose maximum length was 50 seconds so this model most likely works the best for 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](https://huggingface.co/blog/asr-chunking).
The model was fine-tuned on the data from the [Lahjoita puhetta (Donate Speech) corpus](https://link.springer.com/article/10.1007/s10579-022-09606-3) so this model might have biases towards colloquial Finnish.
## Citation
If you use our models or scripts, please cite our article as:
```bibtex
@inproceedings{getman24a_interspeech,
author={Yaroslav Getman and Tamas Grosz and Mikko Kurimo},
title={{What happens in continued pre-training? Analysis of self-supervised speech
models with continued pre-training for colloquial Finnish ASR}},
year=2024,
booktitle={Proc. INTERSPEECH 2024},
pages={XX--XX},
doi={XXXX},
issn={XXXX-XXXX}
}
```
## Team Members
- Yaroslav Getman, [Hugging Face profile](https://huggingface.co/GetmanY1), [LinkedIn profile](https://www.linkedin.com/in/yaroslav-getman/)
- Tamas Grosz, [Hugging Face profile](https://huggingface.co/Grosy), [LinkedIn profile](https://www.linkedin.com/in/tam%C3%A1s-gr%C3%B3sz-950a049a/)
Feel free to contact us for more details 🤗 |