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README.md
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---
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language:
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- en
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- de
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tags:
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- English
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- German
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- bilingual
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- KKY
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- FAV
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license: cc-by-nc-sa-4.0
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---
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# wav2vec2-base-en-de-50k
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This is a bilingual Wav2Vec 2.0 base model pre-trained from 100 thousand hours of speech (50 thousand hours of English and 50 thousand hours of German).
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It has been released along with a paper **A Comparative Analysis of Bilingual and Trilingual Wav2Vec Models for
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Automatic Speech Recognition in Multilingual Oral History Archives** accepted to INTERSPEECH2024 conference.
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## Paper
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The pre-print of our paper is available at http://arxiv.org/abs/2407.17160.
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### All pre-trained models released along with the paper
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- [fav-kky/wav2vec2-base-cs-50k](https://huggingface.co/fav-kky/wav2vec2-base-cs-50k) (monolingual Czech)
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- [fav-kky/wav2vec2-base-de-50k](https://huggingface.co/fav-kky/wav2vec2-base-de-50k) (monolingual German)
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- [fav-kky/wav2vec2-base-cs-en-100k](https://huggingface.co/fav-kky/wav2vec2-base-cs-en-100k) (bilingual Czech+English)
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- [fav-kky/wav2vec2-base-cs-de-100k](https://huggingface.co/fav-kky/wav2vec2-base-cs-de-100k) (bilingual Czech+German)
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- [fav-kky/wav2vec2-base-en-de-100k](https://huggingface.co/fav-kky/wav2vec2-base-en-de-100k) (bilingual English+German)
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- [fav-kky/wav2vec2-base-cs-en-de-150k](https://huggingface.co/fav-kky/wav2vec2-base-cs-en-de-150k) (trilingual Czech+English+German)
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## Citation
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If you find this model useful, please cite our paper:
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```
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@inproceedings{lehecka2024bitrilingual,
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title = {{A Comparative Analysis of Bilingual and Trilingual Wav2Vec Models for Automatic Speech Recognition in Multilingual Oral History Archives}},
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author = {
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Jan Lehe\v{c}ka and
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Josef V. Psutka and
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Lubo\v{s} \v{S}m\'{i}dl and
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Pavel Ircing and
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Josef Psutka
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},
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booktitle={Proc. Interspeech 2024},
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note={In Press},
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year={2024},
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url={https://arxiv.org/abs/2407.17160},
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}
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```
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## Usage
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This model does not have a tokenizer as it was pretrained on audio alone.
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In order to use this model for speech recognition, a tokenizer should be created
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and the model should be [fine-tuned](https://huggingface.co/blog/fine-tune-wav2vec2-english) on labeled ASR data.
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Inputs must be 16kHz mono audio files.
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This model can be used e.g., to extract per-frame contextual embeddings from audio:
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```python
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from transformers import Wav2Vec2Model, Wav2Vec2FeatureExtractor
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import torchaudio
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feature_extractor = Wav2Vec2FeatureExtractor.from_pretrained("fav-kky/wav2vec2-base-en-de-100k")
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model = Wav2Vec2Model.from_pretrained("fav-kky/wav2vec2-base-en-de-100k")
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speech_array, sampling_rate = torchaudio.load("/path/to/audio/file.wav")
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inputs = feature_extractor(
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speech_array,
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sampling_rate=16_000,
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return_tensors="pt"
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)["input_values"][0]
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output = model(inputs)
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embeddings = output.last_hidden_state.detach().numpy()[0]
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```
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## Related works
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