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*asr-wav2vec2-commonvoice-15-fr* is an Automatic Speech Recognition model fine-tuned on CommonVoice 15.0 French set with *LeBenchmark/wav2vec2-FR-7K-large* as the pretrained wav2vec2 model.
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The fine-tuned model achieves the following performance :
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| 2023-09-08 | 9.14 | 11.21 | 4xV100 32GB |
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## Model Details
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- **License:** Apache-2.0
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- **Finetuned from model:** LeBenchmark/wav2vec2-FR-7K-large
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## Training Details
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#### Training Hyperparameters
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<!-- This section provides information about throughput, start/end time, checkpoint size if relevant, etc. -->
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[More Information Needed]
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## Evaluation
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<!-- This section describes the evaluation protocols and provides the results. -->
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### Testing Data, Factors & Metrics
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#### Testing Data
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<!-- This should link to a Dataset Card if possible. -->
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[More Information Needed]
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#### Metrics
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<!-- These are the evaluation metrics being used, ideally with a description of why. -->
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[More Information Needed]
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### Results
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#### Summary
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## Environmental Impact
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<!-- Total emissions (in grams of CO2eq) and additional considerations, such as electricity usage, go here. Edit the suggested text below accordingly -->
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Carbon emissions can be estimated using the [Machine Learning Impact calculator](https://mlco2.github.io/impact#compute) presented in [Lacoste et al. (2019)](https://arxiv.org/abs/1910.09700).
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- **Hardware Type:** [More Information Needed]
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- **Hours used:** [More Information Needed]
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- **Cloud Provider:** [More Information Needed]
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- **Compute Region:** [More Information Needed]
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- **Carbon Emitted:** [More Information Needed]
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## Technical Specifications [optional]
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### Model Architecture and Objective
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[More Information Needed]
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### Compute Infrastructure
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#### Hardware
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[More Information Needed]
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#### Software
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## Citation
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*asr-wav2vec2-commonvoice-15-fr* is an Automatic Speech Recognition model fine-tuned on CommonVoice 15.0 French set with *LeBenchmark/wav2vec2-FR-7K-large* as the pretrained wav2vec2 model.
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The fine-tuned model achieves the following performance :
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| Release | Valid WER | Test WER | GPUs | Epochs
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|:-------------:|:--------------:|:--------------:| :--------:|:--------:|
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| 2023-09-08 | 9.14 | 11.21 | 4xV100 32GB | 30 |
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## Model Details
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- **License:** Apache-2.0
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- **Finetuned from model:** LeBenchmark/wav2vec2-FR-7K-large
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## How to transcribe a file with the model
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### Install and import speechbrain
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```bash
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pip install speechbrain
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```
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```python
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from speechbrain.inference.ASR import EncoderASR
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```
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### Pipeline
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```python
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def transcribe(audio, model):
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return model.transcribe_file(audio).lower()
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def save_transcript(transcript, audio, output_file):
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with open(output_file, 'w', encoding='utf-8') as file:
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file.write(f"{audio}\t{transcript}\n")
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def main():
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model = EncoderASR.from_hparams(model_wav2vec2, savedir="tmp/")
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transcript = transcribe(audio, model)
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save_transcript(transcript, audio, "out.txt")
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```
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## Training Details
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#### Training Hyperparameters
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Refer to the hyperparams.yaml file to get the hyperparameters information.
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#### Training time
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With 4xV100 32GB, the training took ~ 81 hours.
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#### Software
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(Speechbrain)[https://speechbrain.github.io/]:
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```bibtex
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@misc{SB2021,
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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 },
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title = {SpeechBrain},
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year = {2021},
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publisher = {GitHub},
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journal = {GitHub repository},
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howpublished = {\\\\url{https://github.com/speechbrain/speechbrain}},
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}
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```
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## Citation
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