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license: apache-2.0
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---
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license: apache-2.0
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pipeline_tag: audio-to-audio
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---
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# StuPASE: Studio-Quality PASE
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StuPASE is a state-of-the-art generative speech enhancement model trained to remove noise and reverberation while preserving linguistic content and speaker identity, and achieving studio-level perceptual quality. It operates on 16 kHz mono audio.
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---
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## Model Details
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### Model Description
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<img src="framework_stupase.png" alt="High-level system design" width="80%">
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StuPASE contains three main components:
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- **DeWavLM-R:** Performs low-hallucination phonetic enhancement, fine‑tuned from DeWavLM using dry targets for improved dereverberation.
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- **Inputs**: Noisy speech
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- **Outputs**: Enhanced phonetic representations
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- **CFM:** Performs phonetic-guided acoustic enhancement.
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- **Inputs**: Noisy Mel features + Enhanced phonetic representations
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- **Outputs**: Enhanced Mel features
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- **Mel Vocoder:** Reconstructs enhanced wavforms.
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- **Inputs**: Enhanced Mel features
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- **Outputs**: Enhanced speech
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**Developed by:** Copyright © 2026 by Cisco Systems, Inc. All rights reserved.
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**Cisco product group**: Collaboration AI: Xiaobin Rong, Mansur Yesilbursa, Kamil Wojcicki
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**Model type:** Generative Speech Enhancement
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**License:** Apache 2.0
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**Finetuned from:** [WavLM-Large](https://github.com/microsoft/unilm/tree/master/wavlm), [DeWavLM](https://github.com/cisco-open/pase)
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---
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### Model Sources
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- **Repository:** https://github.com/cisco-open/pase
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- **Paper:** https://arxiv.org/abs/2603.09234
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- **Demo:** https://xiaobin-rong.github.io/stupase_demo/
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---
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## Uses
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### Direct Use
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- Enhance noisy or reverberant speech recordings
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- Improve perceptual quality and intelligibility
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- Preserve speaker identity and linguistic content
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- Supports **16 kHz mono audio**
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### Out-of-Scope Use
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- Medical, legal, or safety‑critical decisions
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- Voice conversion or identity manipulation
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- Non‑speech audio enhancement
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---
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## How to Get Started
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Refer to the repository for quick-start code and examples:
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https://github.com/cisco-open/pase
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---
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## Training Details
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### Training Data
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We release a StuPASE checkpoint that has been trained on an updated list of datasets. For this release, training used:
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- Clean speech:
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- DNS5 Challenge clean-speech resources derived from the LibriVox public-domain subset
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- [LibriSpeech](https://www.openslr.org/12/)
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- [LibriTTS](https://www.openslr.org/60/)
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- [VCTK](https://datashare.ed.ac.uk/handle/10283/3443)
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- Noise:
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- DNS5 Challenge noise resources
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- Room impulse responses:
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- [OpenSLR26](https://www.openslr.org/26/)
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- [OpenSLR28](https://www.openslr.org/28/)
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These source datasets were used to prepare training mixtures and train the released model. The model card and repository do not redistribute the underlying dataset contents; please refer to the original dataset pages and licenses below.
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### Dataset Attribution
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- DNS5 Challenge clean speech (LibriVox subset): clean-speech material prepared from [LibriVox](https://librivox.org/) through the [DNS Challenge](https://github.com/microsoft/DNS-Challenge). The LibriVox recordings used for this portion are [public domain](https://librivox.org/pages/public-domain/) and were used as clean-speech training data for the released checkpoint.
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- LibriSpeech: [LibriSpeech](https://www.openslr.org/12/) by Vassil Panayotov et al., licensed under [CC BY 4.0](https://creativecommons.org/licenses/by/4.0/). It was used as clean-speech training data for the released checkpoint.
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- LibriTTS: [LibriTTS](https://www.openslr.org/60/) by Heiga Zen et al., licensed under [CC BY 4.0](https://creativecommons.org/licenses/by/4.0/). It was used as clean-speech training data for the released checkpoint.
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- VCTK Corpus: the [VCTK dataset](https://datashare.ed.ac.uk/handle/10283/3443) from the Centre for Speech Technology Research, University of Edinburgh, licensed under [CC BY 4.0](https://creativecommons.org/licenses/by/4.0/). It was used as clean-speech training data for the released checkpoint.
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- DNS5 Challenge noise resources: noise data prepared through the [DNS Challenge](https://github.com/microsoft/DNS-Challenge) and used to synthesize noisy training mixtures for the released checkpoint. For this release, the DNS5 noise resources draw on [AudioSet](https://research.google.com/audioset/index.html) material licensed under [CC BY 4.0](https://creativecommons.org/licenses/by/4.0/), selected [Freesound](https://freesound.org/) files licensed under [CC0 1.0](https://creativecommons.org/publicdomain/zero/1.0/), and [DEMAND](https://zenodo.org/record/1227121#.XRKKxYhKiUk) environmental recordings licensed under [CC BY-SA 3.0](https://creativecommons.org/licenses/by-sa/3.0/deed.en_CA).
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- OpenSLR26 and OpenSLR28: [OpenSLR26](https://www.openslr.org/26/) and [OpenSLR28](https://www.openslr.org/28/) room impulse response resources, both licensed under Apache 2.0, were used to add reverberation during training.
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All audio was resampled to 16 kHz.
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### Training Procedure
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#### Preprocessing
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- Mixtures generated dynamically
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- SNR sampled from –5 to 15 dB
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- Reverberation applied with 50% probability
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#### Training Hyperparameters
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- **DeWavLM-R:** 50k steps, LR 2e-5, batch size 20
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- **CFM:** 100k steps, LR 1e‑4, batch size 60
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- **Mel Vocoder:** 200k steps, LR 2e-4, batch size 60
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- Optimizer: AdamW with warmup + cosine decay
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- Hardware: 2 × NVIDIA RTX 4090 GPUs
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#### Speeds, Sizes, Times
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- Total parameters: ~561M
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- Inference compute: ~104 GMAC/s
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---
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## Evaluation
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### Testing Data
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- Simulated [LibriSpeech](https://www.openslr.org/12/) test set (using test split)
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- [DNS1 test set](https://github.com/microsoft/DNS-Challenge/tree/interspeech2020/master/datasets/test_set/synthetic) with/without reverberation
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### Metrics
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- DNSMOS, UTMOS
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- LPS, SpeechBERTScore (SBS)
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- Speaker Similarity, using [WavLM-Large based TCAPA-TDNN (finetuned)](https://github.com/microsoft/UniSpeech/tree/main/downstreams/speaker_verification)
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- WER, using Whisper-Large-v3
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### Results
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The performance of the retrained version compared to the original one:
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| Model | DNSMOS | UTMOS | SBS | LPS | SpkSim | WER (%) |
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|:-----:|:------:|:-----:|:---:|:---:|:------:|:-------:|
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| DeWavLM-R (orig.) | 3.35 | 3.94 | 0.84 | 0.88 | 0.49 | 13.22 |
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| **DeWavLM-R (retrained)** | 3.38 | 3.62 | 0.85 | 0.89 | 0.41 | 12.27 |
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| StuPASE (orig.) | 3.37 | 4.08 | 0.85 |0.90 |0.68 | 11.57|
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| **StuPASE (retrained)** | 3.36 | 4.02 | 0.85 | 0.89 | 0.66 | 12.06 |
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It can be seen that the retrained version achieves performance very close to that of the original version on our simulated test set.
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Overall, StuPASE achieves:
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- Lowest WER among evaluated generative and discriminative baselines
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- Highest speaker similarity (SpkSim)
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- Highest perceptual quality with low hallucination rates
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- Consistent performance across noisy and reverberant conditions
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---
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## Bias, Risks, and Limitations
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- Model trained primarily on English speech; performance may degrade for other languages.
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- Very strong noise or mismatched reverberation conditions can introduce artifacts.
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- Speaker characteristics are preserved but not guaranteed perfectly.
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---
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### Recommendations
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Evaluate outputs for your specific use case. Avoid deployments where misunderstanding enhanced speech could have safety or legal consequences.
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---
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## Citation
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If you use StuPASE in your research, please cite:
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```bibtex
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@misc{StuPASE,
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title={{StuPASE: Towards Low-Hallucination Studio-Quality Generative Speech Enhancement}},
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author={Xiaobin Rong and Jun Gao and Zheng Wang and Mansur Yesilbursa and Kamil Wojcicki and Jing Lu},
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year={2026},
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eprint={2603.09234},
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archivePrefix={arXiv},
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primaryClass={eess.AS},
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url={https://arxiv.org/abs/2603.09234},
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}
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```
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Copyright © 2026 by Cisco Systems, Inc. All rights reserved.
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## Model Card Authorship & Contact
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- Mansur Yesilbursa: myesilbu@cisco.com
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