Small conformer (13M parameters) for LibriSpeech (with Transformer LM)
This repository provides all the necessary tools to perform automatic speech
recognition from an end-to-end system pretrained on LibriSpeech (EN) within
SpeechBrain. For a better experience, we encourage you to learn more about
SpeechBrain.
The performance of the model is the following:
Release
Test clean WER
Test other WER
GPUs
24-03-22
2.49
6.03
1xV100 32GB
Pipeline description
This ASR system is composed of 3 different but linked blocks:
Tokenizer (unigram) that transforms words into subword units and trained with
the train transcriptions of LibriSpeech.
Neural language model (Transformer LM) trained on the full 10M words dataset.
Acoustic model made of a small conformer encoder and a joint decoder with CTC +
transformer. Hence, the decoding also incorporates the CTC probabilities.
The system is trained with recordings sampled at 16kHz (single channel).
The code will automatically normalize your audio (i.e., resampling + mono channel selection) when calling transcribe_file if needed.
Install SpeechBrain
First of all, please install SpeechBrain with the following command:
pip install speechbrain
Please notice that we encourage you to read our tutorials and learn more about
SpeechBrain.
Please, cite SpeechBrain if you use it for your research or business.
bibtex
1@misc{speechbrain,
2 title={{SpeechBrain}: A General-Purpose Speech Toolkit},
3 author={Mirco Ravanelli and Titouan Parcollet and Peter Plantinga and Aku Rouhe and Samuele Cornell and Loren Lugosch and Cem Subakan and Nauman Dawalatabad and Abdelwahab Heba and Jianyuan Zhong and Ju-Chieh Chou and Sung-Lin Yeh and Szu-Wei Fu and Chien-Feng Liao and Elena Rastorgueva and François Grondin and William Aris and Hwidong Na and Yan Gao and Renato De Mori and Yoshua Bengio},
4 year={2021},
5 eprint={2106.04624},
6 archivePrefix={arXiv},
7 primaryClass={eess.AS},
8 note={arXiv:2106.04624}
9}