--- language: - cs - en - de tags: - Czech - English - German - trilingual - KKY - FAV license: cc-by-nc-sa-4.0 --- # wav2vec2-base-cs-en-de-150k This is a trilingual Wav2Vec 2.0 base model pre-trained from 150 thousand hours of speech (50 thousand hours of Czech, 50 thousand hours of English and 50 thousand hours of German). It has been released along with a paper **A Comparative Analysis of Bilingual and Trilingual Wav2Vec Models for Automatic Speech Recognition in Multilingual Oral History Archives** accepted to INTERSPEECH2024 conference. ## Paper https://www.isca-archive.org/interspeech_2024/lehecka24_interspeech.pdf Pre-print: http://arxiv.org/abs/2407.17160. ### All pre-trained models released along with the paper - [fav-kky/wav2vec2-base-cs-50k](https://huggingface.co/fav-kky/wav2vec2-base-cs-50k) (monolingual Czech) - [fav-kky/wav2vec2-base-de-50k](https://huggingface.co/fav-kky/wav2vec2-base-de-50k) (monolingual German) - [fav-kky/wav2vec2-base-cs-en-100k](https://huggingface.co/fav-kky/wav2vec2-base-cs-en-100k) (bilingual Czech+English) - [fav-kky/wav2vec2-base-cs-de-100k](https://huggingface.co/fav-kky/wav2vec2-base-cs-de-100k) (bilingual Czech+German) - [fav-kky/wav2vec2-base-en-de-100k](https://huggingface.co/fav-kky/wav2vec2-base-en-de-100k) (bilingual English+German) - [fav-kky/wav2vec2-base-cs-en-de-150k](https://huggingface.co/fav-kky/wav2vec2-base-cs-en-de-150k) (trilingual Czech+English+German) ## Citation If you find this model useful, please cite our paper: ``` @inproceedings{lehecka24_interspeech, title = {A Comparative Analysis of Bilingual and Trilingual Wav2Vec Models for Automatic Speech Recognition in Multilingual Oral History Archives}, author = {Jan Lehečka and Josef V. Psutka and Lubos Smidl and Pavel Ircing and Josef Psutka}, year = {2024}, booktitle = {Interspeech 2024}, pages = {1285--1289}, doi = {10.21437/Interspeech.2024-472}, issn = {2958-1796}, } ``` ## Usage This model does not have a tokenizer as it was pretrained on audio alone. In order to use this model for speech recognition, a tokenizer should be created and the model should be [fine-tuned](https://huggingface.co/blog/fine-tune-wav2vec2-english) on labeled ASR data. Inputs must be 16kHz mono audio files. This model can be used e.g., to extract per-frame contextual embeddings from audio: ```python from transformers import Wav2Vec2Model, Wav2Vec2FeatureExtractor import torchaudio feature_extractor = Wav2Vec2FeatureExtractor.from_pretrained("fav-kky/wav2vec2-base-cs-en-de-150k") model = Wav2Vec2Model.from_pretrained("fav-kky/wav2vec2-base-cs-en-de-150k") speech_array, sampling_rate = torchaudio.load("/path/to/audio/file.wav") inputs = feature_extractor( speech_array, sampling_rate=16_000, return_tensors="pt" )["input_values"][0] output = model(inputs) embeddings = output.last_hidden_state.detach().numpy()[0] ``` ## Related works