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	---
	language:
	- en
	thumbnail: null
	pipeline_tag: automatic-speech-recognition
	tags:
	- automatic-speech-recognition
	- CTC
	- Attention
	- Transformer
	- pytorch
	- speechbrain
	- hf-asr-leaderboard
	license: apache-2.0
	datasets:
	- librispeech
	metrics:
	- wer
	- cer
	model-index:
	- name: wav2vec2+CTC by SpeechBrain
	results:
	- task:
	name: Automatic Speech Recognition
	type: automatic-speech-recognition
	dataset:
	name: LibriSpeech (clean)
	type: librispeech_asr
	config: clean
	split: test
	args:
	language: en
	metrics:
	- name: Test WER
	type: wer
	value: 1.90
	- task:
	name: Automatic Speech Recognition
	type: automatic-speech-recognition
	dataset:
	name: LibriSpeech (other)
	type: librispeech_asr
	config: other
	split: test
	args:
	language: en
	metrics:
	- name: Test WER
	type: wer
	value: 3.96
	---

	<iframe src="https://ghbtns.com/github-btn.html?user=speechbrain&repo=speechbrain&type=star&count=true&size=large&v=2" frameborder="0" scrolling="0" width="170" height="30" title="GitHub"></iframe>
	<br/><br/>

	# wav2vec 2.0 with CTC trained on LibriSpeech

	This repository provides all the necessary tools to perform automatic speech
	recognition from an end-to-end system pretrained on LibriSpeech (English Language) within
	SpeechBrain. For a better experience, we encourage you to learn more about
	[SpeechBrain](https://speechbrain.github.io).

	The performance of the model is the following:

	\| Release \| Test clean WER \| Test other WER \| GPUs \|
	\|:-------------:\|:--------------:\|:--------------:\|:--------:\|
	\| 24-03-22 \| 1.90 \| 3.96 \| 1xA100 40GB \|

	## Pipeline description

	This ASR system is composed of 2 different but linked blocks:
	- Tokenizer (unigram) that transforms words into characters and trained with
	the train transcriptions (EN).
	- Acoustic model (wav2vec2.0 + CTC). A pretrained wav2vec 2.0 model ([wav2vec2-large-960h-lv60-self](https://huggingface.co/facebook/wav2vec2-large-960h-lv60-self)) is combined with two DNN layers and finetuned on LibriSpeech.
	The obtained final acoustic representation is given to the CTC.

	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 tranformers and SpeechBrain with the following command:

	```
	pip install speechbrain transformers
	```

	Please notice that we encourage you to read our tutorials and learn more about
	[SpeechBrain](https://speechbrain.github.io).

	### Transcribing your own audio files (in English)

	```python
	from speechbrain.inference.ASR import EncoderASR

	asr_model = EncoderASR.from_hparams(source="speechbrain/asr-wav2vec2-librispeech", savedir="pretrained_models/asr-wav2vec2-librispeech")
	asr_model.transcribe_file("speechbrain/asr-wav2vec2-commonvoice-en/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](https://colab.research.google.com/drive/1hX5ZI9S4jHIjahFCZnhwwQmFoGAi3tmu?usp=sharing) 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.
	To train it from scratch follow these steps:
	1. Clone SpeechBrain:
	```bash
	git clone https://github.com/speechbrain/speechbrain/
	```
	2. Install it:
	```bash
	cd speechbrain
	pip install -r requirements.txt
	pip install -e .
	```

	3. Run Training:
	```bash
	cd recipes/LibriSpeech/ASR/CTC
	python train_with_wav2vec.py hparams/train_en_with_wav2vec.yaml --data_folder=your_data_folder
	```

	You can find our training results (models, logs, etc) [here](https://drive.google.com/drive/folders/1pg0QzW-LqAISG8Viw_lUTGjXwOqh7gkl?usp=sharing).

	### Limitations
	The SpeechBrain team does not provide any warranty on the performance achieved by this model when used on other datasets.


	# About SpeechBrain
	- Website: https://speechbrain.github.io/
	- Code: https://github.com/speechbrain/speechbrain/
	- HuggingFace: https://huggingface.co/speechbrain/


	# Citing SpeechBrain
	Please, cite SpeechBrain if you use it for your research or business.

	```bibtex
	@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 François 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}
	}
	```