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README.md

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+# **<span style="color:#76b900;">🤫 RE-USE: Multilingual Universal Speech Enhancement</span>**
+# Model Overview
+
+
+## Description
+In universal speech enhancement, the goal is to restore the **quality** of diverse degraded speech while preserving **fidelity**, ensuring that all other factors remain unchanged, e.g., linguistic content, speaker identity, emotion, accent, and other paralinguistic attributes. Inspired by the **distortion–perception trade-off theory**, our proposed single model achieves a good balance between these two objectives and has the following desirable properties:
+
+- Robustness to **diverse degradations**, including additive noise, reverberation, clipping, bandwidth limitation, codec artifacts, packet loss and low-quality mics .
+- Support for **multiple input sampling rates**, including 8, 16, 22.05, 24, 32, 44.1, and 48 kHz.
+- Strong **language-agnostic** capability, enabling effective performance across different languages.
+
+This model is for research and development only.
+
+## Usage
+Directly try our [**Gradio Interactive Demo**](https://huggingface.co/spaces/nvidia/RE-USE) by uploading your noisy audio/video !!
+
+## Environment Setup
+1. (For **Mamba** setup)Pre-built Docker environments can be downloaded [here](https://github.com/RoyChao19477/SEMamba?tab=readme-ov-file#-docker-support) to simplify **Mamba** setup.
+
+2. If you need bandwidth extension:
+
+```bash
+pip install resampy 
+```
+3. Download and navigate to the HuggingFace repository:
+```
+huggingface-cli download nvidia/RE-USE --local-dir ./REUSE --local-dir-use-symlinks False
+cd ./REUSE
+```
+
+## Inference
+Follow the simple steps below to generate enhanced speech using our model:
+1. Place your noisy speech files in the folder `noisy_audio/`
+2. Run the following command:
+```bash
+sh inference.sh
+```
+3. The enhanced speech files will be saved in `enhanced_audio/`.
+
+That's all !
+
+**Note:**
+
+a. You can enable bandwidth extension by setting the target bandwidth using the `BWE argument` in the script.
+
 ---
-tags:
-- model_hub_mixin
-- pytorch_model_hub_mixin
+
+If your noisy speech files are **long and may cause GPU out-of-memory (OOM)** errors, please use the following procedure instead:
+1. Place your long noisy speech files in the folder `long_noisy_audio/`
+2. Run the following command:
+```bash
+sh inference_chunk_wise.sh
+```
+3. The enhanced speech files will be saved in `Long_enhanced_audio/`.
+
+**Note:**
+
+a. You can enable bandwidth extension by setting the target bandwidth using the `BWE argument` in the script.
+
+b. You can also configure the `chunk_size_in_seconds` and `hop_length_portion` directly in the script.
+
 ---
 
-This model has been pushed to the Hub using the [PytorchModelHubMixin](https://huggingface.co/docs/huggingface_hub/package_reference/mixins#huggingface_hub.PyTorchModelHubMixin) integration:
-- Code: [More Information Needed]
-- Paper: [More Information Needed]
-- Docs: [More Information Needed]
+## License/Terms of Use
+This model is released under the [NVIDIA One-Way Noncommercial License (NSCLv1)](https://github.com/NVlabs/HMAR/blob/main/LICENSE).
+
+## Deployment Geography
+Global.
+
+## Use Case
+Researchers and general users can use this model to enhance the quality of their speech data.
+
+## Release Date
+Hugging Face 2026/03/18
+
+## References
+[1] [Rethinking Training Targets, Architectures and Data Quality for Universal Speech Enhancement](https://arxiv.org/abs/2603.02641), 2025.
+(Note: The released model checkpoint differs from the one reported in the paper. It incorporates additional degradation types (e.g., microphone response and more codecs) and is fine-tuned on a smaller, high-quality clean subset.)
+
+## Model Architecture
+**Architecture Type:** Convolutional encoder, Convolutional decoder, and Mamba for time–frequency modeling <br>
+**Network Architecture:** Bi-directional Mamba with 30 layers <br>
+**Number of model parameters:** 9.6M <br>
+ 
+## Input
+Input Type(s): Audio <br>
+Input Format(s): .wav files <br>
+Input Parameters: One-Dimensional (1D) <br>
+Other Properties Related to Input: 8000 Hz - 48000 Hz Mono-channel Audio <br>
+
+## Output
+Output Type(s): Audio <br>
+Output Format: .wav files <br>
+Output Parameters: One-Dimensional (1D) <br>
+Other Properties Related to Output: 8000 Hz - 48000 Hz Mono-channel Audio <br>
+
+Our AI models are designed and/or optimized to run on NVIDIA GPU-accelerated systems. By leveraging NVIDIA’s hardware (e.g. GPU cores) and software frameworks (e.g., CUDA libraries), the model achieves faster training and inference times compared to CPU-only solutions.
+
+## Software Integration
+**Runtime Engine(s):**
+* Not Applicable (N/A)
+
+**Supported Hardware Microarchitecture Compatibility:**
+* NVIDIA Ampere (A100)
+
+**Preferred Operating System(s):**
+* Linux
+
+The integration of foundation and fine-tuned models into AI systems requires additional testing using use-case-specific data to ensure safe and effective deployment. Following the V-model methodology, iterative testing and validation at both unit and system levels are essential to mitigate risks, meet technical and functional requirements, and ensure compliance with safety and ethical standards before deployment.
+
+## Model Version(s)
+Current version: 30USEMamba_peak+GAN_tel_mic_1134k
+
+## Training Datasets
+**Data Modality:**
+Audio
+
+**Audio Training Data Size:**
+Less than 10,000 Hours
+
+* [LibriVox data from DNS5 challenge (EN)](https://github.com/microsoft/DNS-Challenge/tree/master) (~350 hours of speech data)
+* [LibriTTS (EN)](https://openslr.org/60/) (~200 hours of speech data)
+* [VCTK (EN)](https://datashare.ed.ac.uk/handle/10283/3443) (~80 hours of speech data)
+* [WSJ (EN)](https://catalog.ldc.upenn.edu/LDC93S6A) (~85 hours of speech data)
+* [EARS (EN)](https://sp-uhh.github.io/ears_dataset/) (~100 hours of speech data)
+* [Multilingual Librispeech (De, En, Es, Fr)](https://www.openslr.org/94/) (~450 hours of speech data)
+* [CommonVoice 19.0 (De, En, Es, Fr, zh-CN)](https://huggingface.co/datasets/fsicoli/common_voice_19_0) (~1300 hours of speech data)
+* [Audioset+FreeSound noise in DNS5 challenge](https://github.com/microsoft/DNS-Challenge/tree/master) (~180 hours of noise data)
+* [WHAM! Noise](http://wham.whisper.ai/) (~80 hours of noise data)
+* [FSD50K (human voice filtered)](https://huggingface.co/datasets/Fhrozen/FSD50k) (~100 hours of non-speech data)
+* [(Part of) Free Music Archive (medium)](https://github.com/mdeff/fma) (~200 hours of non-speech data)
+* [Simulated RIRs from DNS5 challenge](https://github.com/microsoft/DNS-Challenge/tree/master) (~60k samples of room impulse response)
+* [MicIRP](https://micirp.blogspot.com/p/about-micirp.html) (~70 samples of microphone impulse response)
+
+## Inference
+**Acceleration Engine:** None <br>
+**Test Hardware:** NVIDIA A100
+
+## Ethical Considerations
+NVIDIA believes Trustworthy AI is a shared responsibility and we have established policies and practices to enable development for a wide array of AI applications. When downloaded or used in accordance with our terms of service, developers should work with their internal model team to ensure this model meets requirements for the relevant industry and use case and addresses unforeseen product misuse.
+Please report model quality, risk, security vulnerabilities or NVIDIA AI Concerns [here](https://app.intigriti.com/programs/nvidia/nvidiavdp/detail).
+
+## Citation
+Please consider to cite our paper and this framework, if they are helpful in your research.
+
+```bibtex
+@article{fu2026rethinking,
+  title={Rethinking Training Targets, Architectures and Data Quality for Universal Speech Enhancement},
+  author={Fu, Szu-Wei and Chao, Rong and Yang, Xuesong and Huang, Sung-Feng and Zezario, Ryandhimas E and Nasretdinov, Rauf and Juki{\'c}, Ante and Tsao, Yu and Wang, Yu-Chiang Frank},
+  journal={arXiv preprint arXiv:2603.02641},
+  year={2026}
+}
+```