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README.md
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---
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datasets:
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- Cnam-LMSSC/vibravox
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language: fr
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library_name: transformers
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license: mit
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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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model-index:
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- name: EBEN(M
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- type: n-mos
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value: ???
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name: Test Noresqa-MOS, in-domain training
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---
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<p align="center">
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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
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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)
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## Overview
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This
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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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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/
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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.
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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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```
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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=4,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["headset_microphone"] to Vibravox["forehead_accelerometer"]
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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: 0.7488
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---
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<p align="center">
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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=4,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)
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## Overview
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This model, trained on [Vibravox](https://huggingface.co/datasets/Cnam-LMSSC/vibravox) body conduction sensor data, maps clean speech to body-conducted speech.
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## Inference script :
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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_reverse_forehead_accelerometer")
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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.headset_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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degraded_audio_16kHz, _ = model(cut_audio_16kHz)
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```
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