Model reference · open weights

stupase

Available as managed deployment Audio cisco-ai Audio→audio 1 variants 552 dl/mo

stupase is an open-weight audio or speech model from cisco-ai. AxForge deploys and operates it for you on dedicated EU-owned hardware — with the licence handled where one is required.

Available as managed deployment — configured and operated for you on dedicated EU hardware, quoted per deployment.

What it is

Released bycisco-ai
TypeAudio & music
TaskAudio→audio
Released2026-06-22
Popularity552 downloads / month
LicenceOpen weights

About

What stupase is

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.


Read the full model card

Model Details

Model Description

StuPASE contains three main components:

  • DeWavLM-R: Performs low-hallucination phonetic enhancement, fine‑tuned from DeWavLM using dry targets for improved dereverberation.

    • Inputs: Noisy speech
    • Outputs: Enhanced phonetic representations
  • CFM: Performs phonetic-guided acoustic enhancement.

    • Inputs: Noisy Mel features + Enhanced phonetic representations
    • Outputs: Enhanced Mel features
  • Mel Vocoder: Reconstructs enhanced wavforms.

    • Inputs: Enhanced Mel features
    • Outputs: Enhanced speech

Developed by: Copyright © 2026 by Cisco Systems, Inc. All rights reserved. Cisco product group: Collaboration AI: Xiaobin Rong, Mansur Yesilbursa, Kamil Wojcicki Model type: Generative Speech Enhancement License: Apache 2.0 Finetuned from: WavLM-Large, DeWavLM


Model Sources

  • Repository: https://github.com/cisco-open/pase
  • Paper: https://arxiv.org/abs/2603.09234
  • Demo: https://xiaobin-rong.github.io/stupase_demo/

Uses

Direct Use

  • Enhance noisy or reverberant speech recordings
  • Improve perceptual quality and intelligibility
  • Preserve speaker identity and linguistic content
  • Supports 16 kHz mono audio

Out-of-Scope Use

  • Medical, legal, or safety‑critical decisions
  • Voice conversion or identity manipulation
  • Non‑speech audio enhancement

How to Get Started

Refer to the repository for quick-start code and examples: https://github.com/cisco-open/pase


Training Details

Training Data

We release a StuPASE checkpoint that has been trained on an updated list of datasets. For this release, training used:

  • Clean speech:
  • Noise:
    • DNS5 Challenge noise resources
  • Room impulse responses:

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.

Dataset Attribution

  • DNS5 Challenge clean speech (LibriVox subset): clean-speech material prepared from LibriVox through the DNS Challenge. The LibriVox recordings used for this portion are public domain and were used as clean-speech training data for the released checkpoint.
  • LibriSpeech: LibriSpeech by Vassil Panayotov et al., licensed under CC BY 4.0. It was used as clean-speech training data for the released checkpoint.
  • LibriTTS: LibriTTS by Heiga Zen et al., licensed under CC BY 4.0. It was used as clean-speech training data for the released checkpoint.
  • VCTK Corpus: the VCTK dataset from the Centre for Speech Technology Research, University of Edinburgh, licensed under CC BY 4.0. It was used as clean-speech training data for the released checkpoint.
  • DNS5 Challenge noise resources: noise data prepared through the DNS Challenge and used to synthesize noisy training mixtures for the released checkpoint. For this release, the DNS5 noise resources draw on AudioSet material licensed under CC BY 4.0, selected Freesound files licensed under CC0 1.0, and DEMAND environmental recordings licensed under CC BY-SA 3.0.
  • OpenSLR26 and OpenSLR28: OpenSLR26 and OpenSLR28 room impulse response resources, both licensed under Apache 2.0, were used to add reverberation during training.

All audio was resampled to 16 kHz.

Training Procedure

Preprocessing
  • Mixtures generated dynamically
  • SNR sampled from –5 to 15 dB
  • Reverberation applied with 50% probability
Training Hyperparameters
  • DeWavLM-R: 50k steps, LR 2e-5, batch size 20
  • CFM: 100k steps, LR 1e‑4, batch size 60
  • Mel Vocoder: 200k steps, LR 2e-4, batch size 60
  • Optimizer: AdamW with warmup + cosine decay
  • Hardware: 2 × NVIDIA RTX 4090 GPUs
Speeds, Sizes, Times
  • Total parameters: ~561M
  • Inference compute: ~104 GMAC/s

Evaluation

Testing Data

Metrics

Results

The performance of the retrained version compared to the original one:

ModelDNSMOSUTMOSSBSLPSSpkSimWER (%)

From the published model card. Full card on the HuggingFace links in the sidebar.

How it works

How audio & music work

Audio or textinputAudio modelrecognise / synthesiseText or audiooutputSpeech-to-text turns audio into text; text-to-speech and music models turn text into audio.

Using it via the API

Call it like any OpenAI endpoint

Once AxForge deploys stupase for you, it answers on the OpenAI-compatible API — the same base URL and keys as every other model. (stupase below is illustrative; you get the exact model name on deployment.)

$ curl -sS https://api.axforge.ai/v1/audio/transcriptions \
  -H "Authorization: Bearer $AXFORGE_API_KEY" \
  -F model="stupase" -F file=@audio.mp3

Create an account — your API key is available in the console. 3M free tokens every 30 days with every new account.

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