Overview
- Research area: CV
- Authors: Feiyan Zhou, Luyuan Wang, Shoufa Chen
- Published: 2026-05-19
- arXiv: 2505.14308
- Generates audio directly in raw waveform space, avoiding latent compression losses.
- Waveform patchify + amplitude lifting enable stable flow matching optimization.
- Trained on 5M curated video-text-audio triplets from an automated data pipeline.
- Competitive with or better than latent-space approaches on VGGSound and AudioCaps.
Abstract (translated)
Modern audio generation predominantly relies on latent-space compression, introducing additional complexity and potential information loss. In this work, we challenge this paradigm with WavFlow, a framework that generates high-fidelity audio directly in raw waveform space without intermediate representations.
To overcome the inherent difficulties of modeling high-dimensional and low-energy signals, we reshape audio into 2D token grids through waveform patchify and introduce amplitude lifting to align signal scales, enabling stable optimization via direct x-prediction in flow matching.
To capture complex semantic alignment and temporal synchronization, we leverage an automated data pipeline to curate 5 million high-quality video-text-audio triplets, allowing the model to learn fine-grained acoustic patterns from scratch.
Experimental results show that WavFlow achieves competitive performance on the video-to-audio benchmark VGGSound (FD_PaSST: 59.98, IS_PANNs: 17.40, DeSync: 0.44) and the text-to-audio benchmark AudioCaps (FD_PANNs: 10.63, IS_PANNs: 12.62), matching or even surpassing existing latent-based methods. Our work demonstrates that intermediate compression is not a prerequisite for high-quality synthesis, offering a simpler and more scalable alternative for multimodal audio generation.
Key Results
| Benchmark | Task | Metrics | |---|---|---| | VGGSound | Video-to-audio | FD_PaSST: 59.98, IS_PANNs: 17.40, DeSync: 0.44 | | AudioCaps | Text-to-audio | FD_PANNs: 10.63, IS_PANNs: 12.62 |