--- language: en tags: - speech-enhancement - dereverberation - diffusion-models - generative-models - pytorch - audio-processing license: mit datasets: - VoiceBank-DEMAND - WSJ0-CHiME3 - WSJ0-REVERB - EARS-WHAM - EARS-Reverb model_name: speech-enhancement-dereverberation-diffusion model_type: diffusion-based-generative-model library_name: pytorch key_files: - model.py - train.py - inference.py inference: false pipeline_tag: audio-to-audio --- # Speech Enhancement and Dereverberation with Diffusion-based Generative Models Diffusion process on a spectrogram: In the forward process noise is gradually added to the clean speech spectrogram x0, while the reverse process learns to generate clean speech in an iterative fashion starting from the corrupted signal xT. This repository contains the official PyTorch implementations for the papers: - Simon Welker, Julius Richter, Timo Gerkmann, [*"Speech Enhancement with Score-Based Generative Models in the Complex STFT Domain"*](https://www.isca-speech.org/archive/interspeech_2022/welker22_interspeech.html), ISCA Interspeech, Incheon, Korea, Sept. 2022. [[bibtex]](#citations--references) - Julius Richter, Simon Welker, Jean-Marie Lemercier, Bunlong Lay, Timo Gerkmann, [*"Speech Enhancement and Dereverberation with Diffusion-Based Generative Models"*](https://ieeexplore.ieee.org/abstract/document/10149431), IEEE/ACM Transactions on Audio, Speech, and Language Processing, vol. 31, pp. 2351-2364, 2023. [[bibtex]](#citations--references) - Julius Richter, Yi-Chiao Wu, Steven Krenn, Simon Welker, Bunlong Lay, Shinji Watanabe, Alexander Richard, Timo Gerkmann, [*"EARS: An Anechoic Fullband Speech Dataset Benchmarked for Speech Enhancement and Dereverberation"*](https://arxiv.org/abs/2406.06185), ISCA Interspecch, Kos, Greece, Sept. 2024. [[bibtex]](#citations--references) Audio examples and supplementary materials are available on our [SGMSE project page](https://www.inf.uni-hamburg.de/en/inst/ab/sp/publications/sgmse) and [EARS project page](https://sp-uhh.github.io/ears_dataset/). ## Key Files The following files are essential for this model: - `model.py`: Defines the model architecture and training processes. - `train.py`: Script for training the model. - `enhancement.py`: Used for evaluating and enhancing audio files. - `requirements.txt`: Lists the necessary dependencies. - `calc_metrics.py`: Script for calculating evaluation metrics. ## Follow-up work Please also check out our follow-up work with code available: - Jean-Marie Lemercier, Julius Richter, Simon Welker, Timo Gerkmann, [*"StoRM: A Diffusion-based Stochastic Regeneration Model for Speech Enhancement and Dereverberation"*](https://ieeexplore.ieee.org/document/10180108), IEEE/ACM Transactions on Audio, Speech, Language Processing, vol. 31, pp. 2724 -2737, 2023. [[github]](https://github.com/sp-uhh/storm) - Bunlong Lay, Simon Welker, Julius Richter, Timo Gerkmann, [*"Reducing the Prior Mismatch of Stochastic Differential Equations for Diffusion-based Speech Enhancement"*](https://www.isca-archive.org/interspeech_2023/lay23_interspeech.html), ISCA Interspeech, Dublin, Ireland, Aug. 2023. [[github]](https://github.com/sp-uhh/sgmse-bbed) ## Installation - Create a new virtual environment with Python 3.11 (we have not tested other Python versions, but they may work). - Install the package dependencies via `pip install -r requirements.txt`. - Let pip resolve the dependencies for you. If you encounter any issues, please check `requirements_version.txt` for the exact versions we used. - If using W&B logging (default): - Set up a [wandb.ai](https://wandb.ai/) account - Log in via `wandb login` before running our code. - If not using W&B logging: - Pass the option `--nolog` to `train.py`. - Your logs will be stored as local CSVLogger logs in `lightning_logs/`. ## Pretrained checkpoints - For the speech enhancement task, we offer pretrained checkpoints for models that have been trained on the VoiceBank-DEMAND and WSJ0-CHiME3 datasets, as described in our journal paper [2]. You can download them [here](https://drive.google.com/drive/folders/1CSnkhUSoiv3RG0xg7WEcVapyLuwDaLbe?usp=sharing). - SGMSE+ trained on VoiceBank-DEMAND: `gdown 1_H3EXvhcYBhOZ9QNUcD5VZHc6ktrRbwQ` - SGMSE+ trained on WSJ0-CHiME3: `gdown 16K4DUdpmLhDNC7pJhBBc08pkSIn_yMPi` - For the dereverberation task, we offer a checkpoint trained on our WSJ0-REVERB dataset. You can download it [here](https://drive.google.com/drive/folders/1082_PSEgrqoVVrNsAkSIcpLF1AAtzGwV?usp=sharing). - SGMSE+ trained on WSJ0-REVERB: `gdown 1eiOy0VjHh9V9ZUFTxu1Pq2w19izl9ejD` - Note that this checkpoint works better with sampler settings `--N 50 --snr 0.33`. - For 48 kHz models [3], we offer pretrained checkpoints for speech enhancement, trained on the EARS-WHAM dataset, and for dereverberation, trained on the EARS-Reverb dataset. You can download them [here](https://drive.google.com/drive/folders/1Tn6pVwjxUAy1DJ8167JCg3enuSi0hiw5?usp=sharing). - SGMSE+ trained on EARS-WHAM: `gdown 1t_DLLk8iPH6nj8M5wGeOP3jFPaz3i7K5` - SGMSE+ trained on EARS-Reverb: `gdown 1PunXuLbuyGkknQCn_y-RCV2dTZBhyE3V` Usage: - For resuming training, you can use the `--ckpt` option of `train.py`. - For evaluating these checkpoints, use the `--ckpt` option of `enhancement.py` (see section **Evaluation** below). ## Training Training is done by executing `train.py`. A minimal running example with default settings (as in our paper [2]) can be run with ```bash python train.py --base_dir ``` where `your_base_dir` should be a path to a folder containing subdirectories `train/` and `valid/` (optionally `test/` as well). Each subdirectory must itself have two subdirectories `clean/` and `noisy/`, with the same filenames present in both. We currently only support training with `.wav` files. To see all available training options, run `python train.py --help`. Note that the available options for the SDE and the backbone network change depending on which SDE and backbone you use. These can be set through the `--sde` and `--backbone` options. **Note:** - Our journal preprint [2] uses `--backbone ncsnpp`. - For the 48 kHz model [3], use `--backbone ncsnpp_48k --n_fft 1534 --hop_length 384 --spec_factor 0.065 --spec_abs_exponent 0.667 --sigma-min 0.1 --sigma-max 1.0 --theta 2.0` - Our Interspeech paper [1] uses `--backbone dcunet`. You need to pass `--n_fft 512` to make it work. - Also note that the default parameters for the spectrogram transformation in this repository are slightly different from the ones listed in the first (Interspeech) paper (`--spec_factor 0.15` rather than `--spec_factor 0.333`), but we've found the value in this repository to generally perform better for both models [1] and [2]. ## Evaluation To evaluate on a test set, run ```bash python enhancement.py --test_dir --enhanced_dir --ckpt ``` to generate the enhanced .wav files, and subsequently run ```bash python calc_metrics.py --test_dir --enhanced_dir ``` to calculate and output the instrumental metrics. Both scripts should receive the same `--test_dir` and `--enhanced_dir` parameters. The `--cpkt` parameter of `enhancement.py` should be the path to a trained model checkpoint, as stored by the logger in `logs/`. ## Citations / References We kindly ask you to cite our papers in your publication when using any of our research or code: ```bib @inproceedings{welker22speech, author={Simon Welker and Julius Richter and Timo Gerkmann}, title={Speech Enhancement with Score-Based Generative Models in the Complex {STFT} Domain}, year={2022}, booktitle={Proc. Interspeech 2022}, pages={2928--2932}, doi={10.21437/Interspeech.2022-10653} } ``` ```bib @article{richter2023speech, title={Speech Enhancement and Dereverberation with Diffusion-based Generative Models}, author={Richter, Julius and Welker, Simon and Lemercier, Jean-Marie and Lay, Bunlong and Gerkmann, Timo}, journal={IEEE/ACM Transactions on Audio, Speech, and Language Processing}, volume={31}, pages={2351-2364}, year={2023}, doi={10.1109/TASLP.2023.3285241} } ``` ```bib @inproceedings{richter2024ears, title={{EARS}: An Anechoic Fullband Speech Dataset Benchmarked for Speech Enhancement and Dereverberation}, author={Richter, Julius and Wu, Yi-Chiao and Krenn, Steven and Welker, Simon and Lay, Bunlong and Watanabe, Shinjii and Richard, Alexander and Gerkmann, Timo}, booktitle={ISCA Interspeech}, year={2024} } ``` >[1] Simon Welker, Julius Richter, Timo Gerkmann. "Speech Enhancement with Score-Based Generative Models in the Complex STFT Domain", ISCA Interspeech, Incheon, Korea, Sep. 2022. > >[2] Julius Richter, Simon Welker, Jean-Marie Lemercier, Bunlong Lay, Timo Gerkmann. "Speech Enhancement and Dereverberation with Diffusion-Based Generative Models", IEEE/ACM Transactions on Audio, Speech, and Language Processing, vol. 31, pp. 2351-2364, 2023. > >[3] Julius Richter, Yi-Chiao Wu, Steven Krenn, Simon Welker, Bunlong Lay, Shinji Watanabe, Alexander Richard, Timo Gerkmann. "EARS: An Anechoic Fullband Speech Dataset Benchmarked for Speech Enhancement and Dereverberation", ISCA Interspeech, Kos, Greece, 2024.