UniverSR

Unified and Versatile Audio Super-Resolution via Vocoder-Free Flow Matching

This is the official PyTorch implementation of UniverSR: Unified and Versatile Audio Super-Resolution via Vocoder-Free Flow Matching (ICASSP 2026): paper, demo page.

UniverSR performs audio super-resolution directly in the complex STFT domain using flow matching, without requiring a separate neural vocoder. A single model handles multiple input sample rates (8 / 12 / 16 / 24 kHz → 48 kHz) across speech, music, and sound effects.

<p align="center"> <img src="assets/overview.png" width="720" /> </p>

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⚙️ Installation

pip install git+https://github.com/woongzip1/UniverSR.git

Or install from source:

git clone https://github.com/woongzip1/UniverSR.git
cd UniverSR
pip install -e .

For training, install extra dependencies:

pip install -e ".[train]"
# or: pip install -r requirements.txt

Requirements: Python ≥ 3.10, PyTorch ≥ 2.0, CUDA ≥ 11.8

---

🚀 Quick Start

import torchaudio
from universr import UniverSR

model = UniverSR.from_pretrained("woongzip1/universr-audio", device="cuda")

# Enhance a low-resolution audio file to 48 kHz
output = model.enhance("low_res_8k.wav", input_sr=8000)
torchaudio.save("output_48k.wav", output.cpu(), 48000)

Inference Options

output = model.enhance(
    "low_res.wav",
    input_sr=16000,          # 8000 / 12000 / 16000 / 24000
    ode_method="midpoint",   # 'euler', 'midpoint', 'rk4'
    ode_steps=4,             # number of ODE integration steps
    guidance_scale=1.5,      # classifier-free guidance (None = disabled)
)

Handling 48 kHz Input with Limited Bandwidth

If your audio is already at 48 kHz but has limited bandwidth, set input_sr to the sample rate that matches the content range (Nyquist frequency extends up to input_sr / 2 Hz). For instance, if content exists only up to 8 kHz, use input_sr=16000. UniverSR will automatically apply low-pass filtering to match the training pipeline before performing super-resolution.

# File is 48 kHz, but content only up to 8 kHz → use input_sr=16000
output = model.enhance("fullband_file.wav", input_sr=16000)

---

🤗 Pretrained Models

| Model | Domain | HuggingFace | |---|---|---| | universr-audio | General audio (recommended) | woongzip1/universr-audio | | universr-speech | Speech only | woongzip1/universr-speech |

To download a pretrained model locally:

# via CLI (requires: pip install huggingface_hub)
huggingface-cli download woongzip1/universr-audio --local-dir ./ckpts/universr-audio

# or in Python
from huggingface_hub import snapshot_download
snapshot_download("woongzip1/universr-audio", local_dir="./ckpts/universr-audio")

Loading from a Local Checkpoint

model = UniverSR.from_local(
    ckpt_path="ckpts/best_model.pth",
    config_path="configs/config.yaml",
    device="cuda",
)

---

🔊 Batch Inference

Enhance all .wav files in a folder:

python scripts/inference.py \
    --input_dir [YOUR INPUT DIR] \
    --output_dir results/enhanced/ \
    --model woongzip1/universr-audio \
    --input-sr 8000 \
    --ode-method midpoint \
    --ode-steps 4

From a local checkpoint:

python scripts/inference.py \
    --input_dir [YOUR INPUT DIR] \
    --output_dir results/enhanced/ \
    --ckpt ckpts/best_model.pth \
    --config configs/config.yaml \
    --input-sr 8000 \
    --ode-method midpoint \
    --ode-steps 4

---

📊 Evaluation

Compute Metrics (Generated vs. Ground Truth)

Compare enhanced outputs against ground-truth references using Log-Spectral Distance (LSD):

python scripts/compute_metrics.py \
    --reference_dir data/gt_48k/ \
    --output_dir results/enhanced/ \
    --cutoff-sr 8 \
    --output_json results/metrics.json

Full Pipeline Evaluation

Run evaluation using the training data pipeline with on-the-fly LR generation:

python evaluate.py \
    -c configs/config.yaml \
    --ckpt ckpts/best_model.pth \
    --sampling-rate 8 \
    --wandb true

---

🏋️ Training

1. Prepare File Lists

UniverSR expects 48 kHz mono .wav files listed in a text file (one absolute path per line). Use the provided script to generate these lists from your audio directories:

python scripts/prepare_file_list.py \
    --dirs /path/to/dataset1 /path/to/dataset2 \
    --output data/train.txt

Or use find directly in your terminal:

find /path/to/dataset -name "*.wav" > data/train.txt

2. Configure Training

Point the config to your file lists:

# configs/config.yaml
dataset:
  common:
    sr: 48000
    num_samples: 32767
  train:
    file_list: "./data/train.txt"
  val:
    file_list: "./data/val.txt"

Low-resolution inputs are generated on-the-fly during training via low-pass filtering and downsampling. The input sample rate distribution is controlled by collator.sampling_rates_probs:

collator:
  sampling_rates_probs:
    8: 0.7    # 70% of batches at 8 kHz
    12: 0.1   # 10% at 12 kHz
    16: 0.1   # 10% at 16 kHz
    24: 0.1   # 10% at 24 kHz

3. Start Training

python train.py -c configs/config.yaml --wandb true

Note: Please refer to the paper for dataset details. The pipeline works with any 48 kHz audio collection.

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⚠️ Known Limitations & Tips

Performance across input rates

The default training distribution allocates 70% of batches to 8 kHz and 10% each to 12/16/24 kHz, so 8 kHz→48 kHz results tend to be strongest. Higher input rates (especially 24 kHz) may show weaker high-frequency reconstruction due to spectral aliasing near the cutoff introduced by the naive downsample-upsample augmentation pipeline.

To improve higher input rates, consider:

• Retraining with a more balanced sampling_rates_probs distribution.
• Using slightly lower sr_to_lr_bins values for 12/16/24 kHz (e.g. 120/160/240 instead of 128/170/256) to give the model a small overlap margin at the spectral boundary.

Guidance scale recommendations

The classifier-free guidance (CFG) scale trades off objective fidelity with perceptual richness:

| Domain | Recommended guidance_scale | |---|---| | Speech | 1.0 – 1.5 | | Music | 1.5 – 2.0 | | Sound effects | 1.5 |

Higher values produce denser high-frequency structures but deviate more from the ground-truth reference. Setting guidance_scale=None or 0 disables CFG entirely.

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📖 Citation and License

We've released our code under the MIT License. If you find UniverSR useful in your research, please consider citing:

@article{choi2025universr,
  title={{UniverSR}: Unified and Versatile Audio Super-Resolution via Vocoder-Free Flow Matching},
  author={Choi, Woongjib and Lee, Sangmin and Lim, Hyungseob and Kang, Hong-Goo},
  journal={arXiv preprint arXiv:2510.00771},
  year={2025}
}

@inproceedings{choi2026universr,
  title     = {{UniverSR}: Unified and Versatile Audio Super-Resolution via Vocoder-Free Flow Matching},
  author    = {Choi, Woongjib and Lee, Sangmin and Lim, Hyungseob and Kang, Hong-Goo},
  booktitle = {IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP)},
  year      = {2026}
}

🙏 Acknowledgments

This project was developed at DSPAI Lab, Yonsei University. We thank the following open-source projects that inspired parts of our codebase:

• MIT 6.S184: Introduction to Flow Matching and Diffusion Models
• AudioSR
• FlashSR
• FLowHigh
• AP-BWE
• ConvNeXt V2
• torchdiffeq