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asr-whisper-large-v2-commonvoice-mn

Automatic Speech Recognition speechbrain PyTorch
Mongolian whisper Transformer hf-asr-leaderboard Eval Results
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asr-whisper-large-v2-commonvoice-mn
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---
language:
- mn
thumbnail: null
pipeline_tag: automatic-speech-recognition
tags:
- whisper
- pytorch
- speechbrain
- Transformer
- hf-asr-leaderboard
license: apache-2.0
datasets:
- commonvoice
metrics:
- wer
- cer
model-index:
- name: asr-whisper-large-v2-commonvoice-mn
  results:
  - task:
      name: Automatic Speech Recognition
      type: automatic-speech-recognition
    dataset:
      name: CommonVoice 10.0 (Mongolian)
      type: mozilla-foundation/common_voice_10_0
      config: mn
      split: test
      args:
        language: mn
    metrics:
    - name: Test WER
      type: wer
      value: '64.92'
inference: false
---

<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/>

# whisper large-v2 fine-tuned on CommonVoice Mongolian

This repository provides all the necessary tools to perform automatic speech
recognition from an end-to-end whisper model fine-tuned on CommonVoice (Mongolian 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 CER | Test WER | GPUs |
|:-------------:|:--------------:|:--------------:| :--------:|
| 01-02-23 | 25.73 | 64.92 | 1xV100 16GB |

## Pipeline description

This ASR system is composed of whisper encoder-decoder blocks:
- The pretrained whisper-large-v2 encoder is frozen.
- The pretrained Whisper tokenizer is used.
- A pretrained Whisper-large-v2 decoder ([openai/whisper-large-v2](https://huggingface.co/openai/whisper-large-v2)) is finetuned on CommonVoice MN.
The obtained final acoustic representation is given to the greedy decoder. 

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==4.26.0
```

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 Mongolian)

```python

from speechbrain.inference.ASR import WhisperASR

asr_model = WhisperASR.from_hparams(source="speechbrain/asr-whisper-large-v2-commonvoice-mn", savedir="pretrained_models/asr-whisper-large-v2-commonvoice-mn")
asr_model.transcribe_file("speechbrain/asr-whisper-large-v2-commonvoice-mn/example-mn.wav")


```
### Inference on GPU
To perform inference on the GPU, add  `run_opts={"device":"cuda"}`  when calling the `from_hparams` method.

### 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/CommonVoice/ASR/transformer/
python train_with_whisper.py hparams/train_mn_hf_whisper.yaml --data_folder=your_data_folder
```

You can find our training results (models, logs, etc) [here](https://drive.google.com/drive/folders/10E2xclgNx_6BFxNmv9i1HorBNnsMveP_?usp=share_link).

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

#### Referencing SpeechBrain

```
@misc{SB2021,
    author = {Ravanelli, Mirco and Parcollet, Titouan and Rouhe, Aku and Plantinga, Peter and Rastorgueva, Elena and Lugosch, Loren and Dawalatabad, Nauman and Ju-Chieh, Chou and Heba, Abdel and Grondin, Francois and Aris, William and Liao, Chien-Feng and Cornell, Samuele and Yeh, Sung-Lin and Na, Hwidong and Gao, Yan and Fu, Szu-Wei and Subakan, Cem and De Mori, Renato and Bengio, Yoshua },
    title = {SpeechBrain},
    year = {2021},
    publisher = {GitHub},
    journal = {GitHub repository},
    howpublished = {\\\\url{https://github.com/speechbrain/speechbrain}},
  }
```

#### About SpeechBrain
SpeechBrain is an open-source and all-in-one speech toolkit. It is designed to be simple, extremely flexible, and user-friendly. Competitive or state-of-the-art performance is obtained in various domains.

Website: https://speechbrain.github.io/

GitHub: https://github.com/speechbrain/speechbrain