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README.md
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Work in Progress
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Work in Progress
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---
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language: "en"
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thumbnail:
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tags:
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- speechbrain
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- VAD
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- SAD
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- Voice Activity Detection
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- Speech Activity Detection
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- Speaker Diarization
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- pytorch
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- CRDNN
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- LibriSpeech
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- LibryParty
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datasets:
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- Urbansound8k
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metrics:
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- Accuracy
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---
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<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>
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<br/><br/>
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# Voice Activity Detection with a (small) CRDNN model trained on Libriparty
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This repository provides all the necessary tools to perform voice activity detection with SpeechBrain using a model pretrained on Libriparty
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The pre-trained system can process short and long speech recordings and outputs the segments where speech activity is detected.
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The output of the system looks like this:
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```
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segment_001 0.00 2.57 NON_SPEECH
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segment_002 2.57 8.20 SPEECH
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segment_003 8.20 9.10 NON_SPEECH
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segment_004 9.10 10.93 SPEECH
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segment_005 10.93 12.00 NON_SPEECH
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segment_006 12.00 14.40 SPEECH
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segment_007 14.40 15.00 NON_SPEECH
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segment_008 15.00 17.70 SPEECH
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```
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For a better experience, we encourage you to learn more about
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[SpeechBrain](https://speechbrain.github.io).
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# Results
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The model performance on the LibriParty test set is:
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| Release | hyperparams file | Test Precision | Test Recall | Test F-Score | Model link | GPUs |
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|:-------------:|:---------------------------:| -----:| -----:| --------:| :-----------:| :-----------:|
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| 2021-09-09 | train.yaml | 0.939 | 0.951 | 0.945 | https://drive.google.com/drive/folders/1Z7J3Zd7M5M9VYoNtbrbpbxSoKWUpjhzp?usp=sharing | 1xV100 16GB
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## Pipeline description
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This system is composed of a CRDNN that outputs posteriors probabilities with a value close to one for speech frames and close to zero for non-speech segments.
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A threshold is applied on top of the posteriors to detect candidate speech boundaries.
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Depending on the active options, these boundaries can be post-processed (e.g, merging close segments, removing short segments, etc) to further improve the performance. See more details below.
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## Install SpeechBrain
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First of all, please install SpeechBrain with the following command:
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```
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pip install speechbrain
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```
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Please notice that we encourage you to read our tutorials and learn more about
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[SpeechBrain](https://speechbrain.github.io).
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### Perform Voice Activity Detection
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```python
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from speechbrain.pretrained import VAD
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VAD = VAD.from_hparams(source="speechbrain/vad_crdnn_libriparty", savedir="pretrained_models/vad_crdnn_libriparty")
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boundaries = VAD.get_speech_segments("speechbrain/vad_example.wav")
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# Print the output
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VAD.save_boundaries(boundaries)
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```
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The output is a tensor that contains the beginning/end second of each
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detected speech segment. You can save the boundaries on a file with:
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```python
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VAD.save_boundaries(boundaries, save_path='VAD_file.txt')
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```
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Sometimes it is useful to jointly visualize the VAD output with the input signal itself. This is helpful to quickly figure out if the VAD is doing or not a good job.
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To do it:
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```python
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upsampled_boundaries = VAD.upsample_boundaries(boundaries, audio_file)
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torchaudio.save('vad_final.wav', upsampled_boundaries.cpu(), sample_rate)
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```
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This creates a "VAD signal" with the same dimensionality as the original signal.
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You can now open *vad_final.wav* and *speechbrain/vad_example.wav* with software like audacity to visualize them jointly.
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### VAD pipeline details
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The pipeline for detecting the speech segments is the following:
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1. Compute posteriors probabilities at the frame level.
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2. Apply a threshold on the posterior probability.
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3. Derive candidate speech segments on top of that.
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4. Apply energy VAD within each candidate segment (optional). This might break down long sentences into short one based on the energy content.
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5. Merge segments that are too close.
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6. Remove segments that are too short.
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7. Double-check speech segments (optional). This could is a final check to make sure the detected segments are actually speech ones.
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We designed the VAD such that you can have access to all of these steps (this might help to debug):
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```python
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# 1- Let's compute frame-level posteriors first
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prob_chunks = VAD.get_speech_prob_file(audio_file)
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# 2- Let's apply a threshold on top of the posteriors
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prob_th = VAD.apply_threshold(prob_chunks).float()
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# 3- Let's now derive the candidate speech segments
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boundaries = VAD.get_boundaries(prob_th)
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# 4- Apply energy VAD within each candidate speech segment (optional)
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boundaries = VAD.energy_VAD(audio_file,boundaries)
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# 5- Merge segments that are too close
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boundaries = VAD.merge_short_segments(boundaries, close_th=0.250)
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# 6- Re,ove segments that are too short
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boundaries = VAD.remove_short_segments(boundaries, len_th=0.250)
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# 7- Double-check speech segments (optional).
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boundaries = VAD.double_check_speech_segments(boundaries, audio_file, speech_th=0.5)
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```
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### Inference on GPU
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To perform inference on the GPU, add `run_opts={"device":"cuda"}` when calling the `from_hparams` method.
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### Training
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The model was trained with SpeechBrain (ea17d22).
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To train it from scratch follows these steps:
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1. Clone SpeechBrain:
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```bash
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git clone https://github.com/speechbrain/speechbrain/
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```
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2. Install it:
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```
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cd speechbrain
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pip install -r requirements.txt
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pip install -e .
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```
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3. Run Training:
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Training heavily relies on data augmentation. Make sure you have downloaded all the datasets needed:
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- LibriParty: https://drive.google.com/file/d/1--cAS5ePojMwNY5fewioXAv9YlYAWzIJ/view?usp=sharing
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- Musan: https://www.openslr.org/resources/17/musan.tar.gz
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- CommonLanguage: https://zenodo.org/record/5036977/files/CommonLanguage.tar.gz?download=1
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```
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cd recipes/LibriParty/VAD
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python train.py hparams/train.yaml --data_folder=/path/to/LibriParty --musan_folder=/path/to/musan/ --commonlanguage_folder=/path/to/common_voice_kpd
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```
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### Limitations
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The SpeechBrain team does not provide any warranty on the performance achieved by this model when used on other datasets.
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# **Citing SpeechBrain**
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Please, cite SpeechBrain if you use it for your research or business.
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```bibtex
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@misc{speechbrain,
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title={{SpeechBrain}: A General-Purpose Speech Toolkit},
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author={Mirco Ravanelli and Titouan Parcollet and Peter Plantinga and Aku Rouhe and Samuele Cornell and Loren Lugosch and Cem Subakan and Nauman Dawalatabad and Abdelwahab Heba and Jianyuan Zhong and Ju-Chieh Chou and Sung-Lin Yeh and Szu-Wei Fu and Chien-Feng Liao and Elena Rastorgueva and François Grondin and William Aris and Hwidong Na and Yan Gao and Renato De Mori and Yoshua Bengio},
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year={2021},
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eprint={2106.04624},
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archivePrefix={arXiv},
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primaryClass={eess.AS},
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note={arXiv:2106.04624}
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}
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```
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