Low-latency Speech Enhancement via Speech Token Generation
Huaying Xue, Xiulian Peng, Yan Lu
TL;DR
The paper tackles robustness and latency in speech enhancement under unseen real-world noises by formulating enhancement as conditional generation of clean-speech tokens. It introduces a conditional generative framework that encodes clean speech into discrete TF-Codec tokens and autoregressively generates these tokens from noisy input, guided by an explicit-alignment scheme that enables pure next-token prediction. Leveraging a single-stage, group-quantized TF-Codec-based generator, the approach achieves low latency while maintaining high speech quality. Experiments on synthetic and real DNS data show improved perceptual quality and intelligibility over a TFNet baseline, with ablations confirming the benefits of explicit alignment and autoregressive design for robustness and temporal coherence.
Abstract
Existing deep learning based speech enhancement mainly employ a data-driven approach, which leverage large amounts of data with a variety of noise types to achieve noise removal from noisy signal. However, the high dependence on the data limits its generalization on the unseen complex noises in real-life environment. In this paper, we focus on the low-latency scenario and regard speech enhancement as a speech generation problem conditioned on the noisy signal, where we generate clean speech instead of identifying and removing noises. Specifically, we propose a conditional generative framework for speech enhancement, which models clean speech by acoustic codes of a neural speech codec and generates the speech codes conditioned on past noisy frames in an auto-regressive way. Moreover, we propose an explicit-alignment approach to align noisy frames with the generated speech tokens to improve the robustness and scalability to different input lengths. Different from other methods that leverage multiple stages to generate speech codes, we leverage a single-stage speech generation approach based on the TF-Codec neural codec to achieve high speech quality with low latency. Extensive results on both synthetic and real-recorded test set show its superiority over data-driven approaches in terms of noise robustness and temporal speech coherence.
