๐ Transformer for LibriSpeech (with Transformer LM)
This repository offers all the essential tools for automatic speech recognition using an end - to - end system pretrained on LibriSpeech (EN) within SpeechBrain. For a better experience, explore SpeechBrain.
๐ Quick Start
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 as follows:
Release |
Test clean WER |
Test other WER |
GPUs |
24 - 03 - 22 |
2.27 |
5.53 |
4xV100 32GB |
โจ Features
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 transformer 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.
๐ฆ Installation
Install SpeechBrain
First of all, please install SpeechBrain with the following command:
pip install speechbrain
๐ก Usage Tip
Please notice that we encourage you to read our tutorials and learn more about SpeechBrain.
๐ป Usage Examples
Transcribing your own audio files (in English)
from speechbrain.inference.ASR import EncoderDecoderASR
asr_model = EncoderDecoderASR.from_hparams(source="speechbrain/asr-transformer-transformerlm-librispeech", savedir="pretrained_models/asr-transformer-transformerlm-librispeech")
asr_model.transcribe_file("speechbrain/asr-transformer-transformerlm-librispeech/example.wav")
Inference on GPU
To perform inference on the GPU, add run_opts={"device":"cuda"}
when calling the from_hparams
method.
Parallel Inference on a Batch
Please, see this Colab notebook to figure out how to transcribe in parallel a batch of input sentences using a pre - trained model.
Training
The model was trained with SpeechBrain (Commit hash: 'f73fcc35').
To train it from scratch follow these steps:
- Clone SpeechBrain:
git clone https://github.com/speechbrain/speechbrain/
- Install it:
cd speechbrain
pip install -r requirements.txt
pip install -e .
- Run Training:
cd recipes/LibriSpeech/ASR/transformer
python train.py hparams/transformer.yaml --data_folder=your_data_folder
You can find our training results (models, logs, etc) here.
Limitations
The SpeechBrain team does not provide any warranty on the performance achieved by this model when used on other datasets.
๐ Documentation
About SpeechBrain
- Website: https://speechbrain.github.io/
- Code: https://github.com/speechbrain/speechbrain/
- HuggingFace: https://huggingface.co/speechbrain/
๐ License
This project is licensed under the apache - 2.0 license.
๐ Citing SpeechBrain
Please, cite SpeechBrain if you use it for your research or business.
@misc{speechbrain,
title={{SpeechBrain}: A General-Purpose Speech Toolkit},
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 Franรงois Grondin and William Aris and Hwidong Na and Yan Gao and Renato De Mori and Yoshua Bengio},
year={2021},
eprint={2106.04624},
archivePrefix={arXiv},
primaryClass={eess.AS},
note={arXiv:2106.04624}
}