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--- |
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language: fr |
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license: mit |
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library_name: transformers |
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tags: |
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- audio |
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- audio-to-audio |
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- speech |
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datasets: |
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- Cnam-LMSSC/vibravox |
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model-index: |
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- name: EBEN(M=4,P=2,Q=4) |
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results: |
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- task: |
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name: Bandwidth Extension |
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type: speech-enhancement |
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dataset: |
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name: Vibravox["soft_in_ear_microphone"] |
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type: Cnam-LMSSC/vibravox |
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args: fr |
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metrics: |
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- name: Test STOI, in-domain training |
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type: stoi |
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value: 0.8676 |
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- name: Test Noresqa-MOS, in-domain training |
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type: n-mos |
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value: 4.331 |
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--- |
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<p align="center"> |
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<img src="https://cdn-uploads.huggingface.co/production/uploads/65302a613ecbe51d6a6ddcec/zhB1fh-c0pjlj-Tr4Vpmr.png" style="object-fit:contain; width:280px; height:280px;" > |
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</p> |
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# Model Card |
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- **Developed by:** [Cnam-LMSSC](https://huggingface.co/Cnam-LMSSC) |
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- **Model:** [EBEN(M=4,P=2,Q=4)](https://github.com/jhauret/vibravox/blob/main/vibravox/torch_modules/dnn/eben_generator.py) (see [publication in IEEE TASLP](https://ieeexplore.ieee.org/document/10244161) - [arXiv link](https://arxiv.org/abs/2303.10008)) |
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- **Language:** French |
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- **License:** MIT |
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- **Training dataset:** `speech_clean` subset of [Cnam-LMSSC/vibravox](https://huggingface.co/datasets/Cnam-LMSSC/vibravox) (see [VibraVox paper on arXiV](https://arxiv.org/abs/2407.11828)) |
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- **Samplerate for usage:** 16kHz |
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## Overview |
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This bandwidth extension model, trained on [Vibravox](https://huggingface.co/datasets/Cnam-LMSSC/vibravox) body conduction sensor data, enhances body-conducted speech audio by denoising and regenerating mid and high frequencies from low-frequency content. |
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## Disclaimer |
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This model, trained for **a specific non-conventional speech sensor**, is intended to be used with **in-domain data**. Using it with other sensor data may lead to suboptimal performance. |
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## Link to BWE models trained on other body conducted sensors : |
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The entry point to all EBEN models for Bandwidth Extension (BWE) is available at [https://huggingface.co/Cnam-LMSSC/vibravox_EBEN_models](https://huggingface.co/Cnam-LMSSC/vibravox_EBEN_models). |
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## Training procedure |
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Detailed instructions for reproducing the experiments are available on the [jhauret/vibravox](https://github.com/jhauret/vibravox) Github repository. |
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## Inference script : |
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```python |
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import torch, torchaudio |
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from vibravox.torch_modules.dnn.eben_generator import EBENGenerator |
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from datasets import load_dataset |
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model = EBENGenerator.from_pretrained("Cnam-LMSSC/EBEN_soft_in_ear_microphone") |
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test_dataset = load_dataset("Cnam-LMSSC/vibravox", "speech_clean", split="test", streaming=True) |
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audio_48kHz = torch.Tensor(next(iter(test_dataset))["audio.soft_in_ear_microphone"]["array"]) |
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audio_16kHz = torchaudio.functional.resample(audio_48kHz, orig_freq=48_000, new_freq=16_000) |
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cut_audio_16kHz = model.cut_to_valid_length(audio_16kHz[None, None, :]) |
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enhanced_audio_16kHz, enhanced_speech_decomposed = model(cut_audio_16kHz) |
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``` |
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