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--- |
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library_name: transformers |
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license: mit |
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language: fr |
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datasets: |
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- Cnam-LMSSC/vibravox |
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metrics: |
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- per |
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tags: |
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- audio |
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- automatic-speech-recognition |
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- speech |
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- phonemize |
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- phoneme |
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model-index: |
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- name: Wav2Vec2-base French finetuned for Speech-to-Phoneme by LMSSC |
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results: |
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- task: |
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name: Speech-to-Phoneme |
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type: automatic-speech-recognition |
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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 PER, in-domain training | |
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type: per |
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value: 4.0 |
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--- |
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# Model Card |
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- **Developed by:** [Cnam-LMSSC](https://huggingface.co/Cnam-LMSSC) |
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- **Model type:** [Wav2Vec2ForCTC](https://huggingface.co/transformers/v4.9.2/model_doc/wav2vec2.html#transformers.Wav2Vec2ForCTC) |
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- **Language:** French |
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- **License:** MIT |
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- **Finetuned from model:** [facebook/wav2vec2-base-fr-voxpopuli-v2](https://huggingface.co/facebook/wav2vec2-base-fr-voxpopuli-v2) |
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- **Finetuned dataset:** `soft_in_ear_microphone` audio of the `speech_clean` subset of [Cnam-LMSSC/vibravox](https://huggingface.co/datasets/Cnam-LMSSC/vibravox) |
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- **Samplerate for usage:** 16kHz |
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<p align="center"> |
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<img src="https://cdn-uploads.huggingface.co/production/uploads/6390fc80e6d656eb421bab69/KkZoQQmrn53U6BTLmr0XK.png" /> |
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</p> |
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## Output |
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As this model is specifically trained for a speech-to-phoneme task, the output is sequence of [IPA-encoded](https://en.wikipedia.org/wiki/International_Phonetic_Alphabet) words, without punctuation. |
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If you don't read the phonetic alphabet fluently, you can use this excellent [IPA reader website](http://ipa-reader.xyz) to convert the transcript back to audio synthetic speech in order to check the quality of the phonetic transcription. |
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## Link to other phonemizer models trained on other body conducted sensors : |
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An entry point to all **phonemizers** (speech-to-phoneme ASR) models trained on different sensor data from the trained on different sensor data from the [Vibravox dataset](https://huggingface.co/datasets/Cnam-LMSSC/vibravox) is available at [https://huggingface.co/Cnam-LMSSC/vibravox_phonemizers](https://huggingface.co/Cnam-LMSSC/vibravox_phonemizers). |
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### Disclaimer |
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Each of these models has been trained for a **specific non-conventional speech sensor** and is intended to be used with **in-domain data**. The only exception is the headset microphone phonemizer, which can certainly be used for many applications using audio data captured by airborne microphones. |
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Please be advised that using these models outside their intended sensor data may result in suboptimal performance. |
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## Training procedure |
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The model has been finetuned for 10 epochs with a constant learning rate of *1e-5*. To reproduce experiment please visit [jhauret/vibravox](https://github.com/jhauret/vibravox). |
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## Inference script : |
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```python |
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import torch, torchaudio |
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from transformers import AutoProcessor, AutoModelForCTC |
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from datasets import load_dataset |
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processor = AutoProcessor.from_pretrained("Cnam-LMSSC/phonemizer_soft_in_ear_microphone") |
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model = AutoModelForCTC.from_pretrained("Cnam-LMSSC/phonemizer_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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inputs = processor(audio_16kHz, sampling_rate=16_000, return_tensors="pt") |
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logits = model(inputs.input_values).logits |
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predicted_ids = torch.argmax(logits,dim = -1) |
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transcription = processor.batch_decode(predicted_ids) |
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print("Phonetic transcription : ", transcription) |
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``` |
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