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Whisfusion: Parallel ASR Decoding via a Diffusion Transformer

Sound 2025-08-12 v1 Artificial Intelligence Machine Learning Audio and Speech Processing

Abstract

Fast Automatic Speech Recognition (ASR) is critical for latency-sensitive applications such as real-time captioning and meeting transcription. However, truly parallel ASR decoding remains challenging due to the sequential nature of autoregressive (AR) decoders and the context limitations of non-autoregressive (NAR) methods. While modern ASR encoders can process up to 30 seconds of audio at once, AR decoders still generate tokens sequentially, creating a latency bottleneck. We propose Whisfusion, the first framework to fuse a pre-trained Whisper encoder with a text diffusion decoder. This NAR architecture resolves the AR latency bottleneck by processing the entire acoustic context in parallel at every decoding step. A lightweight cross-attention adapter trained via parameter-efficient fine-tuning (PEFT) bridges the two modalities. We also introduce a batch-parallel, multi-step decoding strategy that improves accuracy by increasing the number of candidates with minimal impact on speed. Fine-tuned solely on LibriSpeech (960h), Whisfusion achieves a lower WER than Whisper-tiny (8.3% vs. 9.7%), and offers comparable latency on short audio. For longer utterances (>20s), it is up to 2.6x faster than the AR baseline, establishing a new, efficient operating point for long-form ASR. The implementation and training scripts are available at https://github.com/taeyoun811/Whisfusion.

Keywords

Cite

@article{arxiv.2508.07048,
  title  = {Whisfusion: Parallel ASR Decoding via a Diffusion Transformer},
  author = {Taeyoun Kwon and Junhyuk Ahn and Taegeun Yun and Heeju Jwa and Yoonchae Choi and Siwon Park and Nam-Joon Kim and Jangchan Kim and Hyun Gon Ryu and Hyuk-Jae Lee},
  journal= {arXiv preprint arXiv:2508.07048},
  year   = {2025}
}

Comments

16 pages, 9 figures

R2 v1 2026-07-01T04:42:36.236Z