Many speech enhancement (SE) methods rely on continuous representations. Recently, discrete audio tokens have been explored to enable autoregressive generation for SE. However, it remains unclear whether discretization itself consistently improves SE performance. In this paper, we introduce LL-SDR, a token-based speech enhancement framework that explicitly leverages discretization to better separate speech and noise. Our first contribution is a Variance-Ordered Residual Vector Quantizer (VO-RVQ), designed to disentangle speech and noise distributions during tokenization. Second, we propose a latent-space discriminator to better align enhanced embeddings with semantic embeddings. Experiments show that LL-SDR outperforms continuous baselines and matches the performance of autoregressive token-based approaches. Despite its strong enhancement performance, LL-SDR remains lightweight and efficient, requiring only 40G MACs for a single forward pass on a 10-second 16 kHz speech segment and achieving low-latency inference with an RTF of 0.01 on GPU and 0.24 on CPU. Demos and source code are available at our project websites.
Figures & tables
Fig. 1: LL-SDR speech enhancement framework: a noisy waveform is encoded and enhanced in the discrete space via our variance-ordered quantizer where Ne is four and Nn is one, then decoded back to waveform. The triangle structure here illustrates that different quantizers have different dimensional sizes; a HuBERT-based semantic discriminator to align the enhanced representation with clean speech semantics.
Fig. 2: Visualization of latent embeddings using UMAP and t-SNE. Blue points denote noisy embeddings (the output of last Nn codebooks), and red points denote enhanced embeddings (the mean of first Ne codebooks). The top row shows the baseline model, where noisy and enhanced representations largely overlap. The bottom row corresponds to the proposed triangle (variance-ordered) quantization structure, which exhibits clearer structural separation and more organized distribution patterns.
Model
Type
Discrete
Reverb
No Reverb
Real
OVRL ↑
SIG ↑
BAK ↑
OVRL ↑
SIG ↑
BAK ↑
OVRL ↑
SIG ↑
BAK ↑
Noisy
–
–
1.39
1.76
1.50
2.48
3.39
2.62
2.26
3.05
2.51
Conv-TasNet
NAR
✗
2.01
2.42
2.71
3.00
3.09
3.34
-
-
-
DEMUCS
NAR
✗
2.55
2.86
3.90
3.35
3.58
4.15
2.99
3.26
4.03
FRCRN
NAR
✗
2.28
2.93
2.92
3.34
3.58
4.13
3.04
3.37
3.98
VoiceFixer
NAR
✗
3.13
3.43
4.02
3.25
3.50
4.11
3.08*
3.37*
4.00*
TABLE I: DNSMOS (OVRL / SIG / BAK) under reverberant, non-reverberant, and real-recording conditions. Most values for compared methods are taken from prior work [ 6 ] and [ 23 ] . Values marked with * are obtained from our own evaluation. Best and second-best results are highlighted.
Model
Params (M) ↓
MACs (G) ↓
RTF (H100) ↓
RTF (CPU) ↓
FRCRN
53
77
0.04
0.99
VoiceFixer
117
777
0.01
0.52
LLaSE-G1 (single)
1319
145
0.37
2.43
LL-SDR (Ours)
74
40
0.01
0.24
TABLE II: Model complexity and runtime comparison. Best and second-best results are highlighted.
Model
Reverb
Non-reverb
STFT ↓
Mel ↓
LPS ↓
STFT ↓
Mel ↓
LPS ↓
Noisy
244.41
90.78
20.43
182.97
73.15
13.97
FRCRN
216.18
76.37
13.72
125.17
44.39
3.60
VoiceFixer
192.51
68.98
9.77
180.40
63.22
8.24
LLaSE-G1
258.73
99.10
20.32
216.47
83.54
12.79
LL-SDR (Ours)
208.44
74.47
11.85
161.95
53.26
6.40
TABLE III: In-distribution Test:Distortion Metrics Under Reverberant and Non-Reverberant Conditions in DNS-Challenge 2020 Non-Blind Test Set.
Method
RTF (H100) ↓
RTF (CPU) ↓
STFT ↓
Mel ↓
LPS ↓
Noisy
-
-
155.72
54.23
14.76
FRCRN
0.04
0.99
100.10
29.81
4.93
VoiceFixer
0.01
0.52
123.53
30.86
6.56
LLaSE
0.37
2.43
183.25
60.55
13.94
LL–SDR
0.01
0.24
136.84
36.77
8.15
TABLE IV: Out-of-distribution distortion metrics on the VoiceBank-DEMAND test set.
Method
Accuracy ↑
Macro Recall ↑
Macro F1 ↑
RVQ
50.46
50.46
34.37
VO-RVQ
71.33
71.33
71.25
TABLE V: Spectral clustering results. Enhanced and noisy embeddings are first combined without labels, clustered using spectral clustering, and then compared with the true labels.
Method
Reverb
No Reverb
OVRL ↑
SIG ↑
BAK ↑
OVRL ↑
SIG ↑
BAK ↑
Continuous
2.95
3.30
3.84
3.30
3.56
4.09
RVQ
3.13
3.43
4.01
3.36
3.59
4.16
VO-RVQ
3.14
3.45
3.99
3.39
3.62
4.17
VO-RVQ + HuBERT
3.22
3.51
4.03
3.40
3.63
4.16
TABLE VI: Ablation study One. Best and second-best results are highlighted.
Method
Reverb
No Reverb
OVRL ↑
SIG ↑
BAK ↑
OVRL ↑
SIG ↑
BAK ↑
w/o LNCE
3.15
3.45
4.01
3.37
3.61
4.15
w/o Lsem
3.13
3.45
3.96
3.38
3.61
4.16
LL-SDR (Ours)
3.27
3.55
4.08
3.39
3.62
4.16
TABLE VII: Ablation study Two. Best results are highlighted.
Method
Reverb
No Reverb
Real
OVRL ↑
SIG ↑
BAK ↑
OVRL ↑
SIG ↑
BAK ↑
OVRL ↑
SIG ↑
BAK ↑
Shuffle One
3.19
3.50
4.00
3.38
3.61
4.15
3.12
3.44
3.99
Shuffle Two
3.12
3.44
3.97
3.38
3.62
4.15
3.12
3.43
4.00
Shuffle Three
3.18
3.48
4.00
3.36
3.60
4.14
3.12
3.44
3.99
LL-SDR (Ours)
3.27
3.55
4.08
3.39
3.62
4.16
3.16
3.45
4.05
TABLE VIII: Ablation study Three. Best results are highlighted.
Team/Model
Team Rank
OVRL
NISQA
UTMOS
Bobbsun
1
2.88
3.22
2.09
rc
2
2.83
2.92
2.03
FRCRN
NA
1.52
1.30
1.32
VoiceFixer
NA
2.83
3.40
1.98
LLaSE–G1
NA
2.62
3.02
1.94
LL-SDR (Ours)
NA
2.93
2.92
1.95
TABLE IX: Speech enhancement results on the 2025 URGENT Challenge blind test set.
Fig. 3: Left: DNSMOS OVRL scores for different Number of codebooks settings. Right: DNSMOS OVRL scores for different Ne settings.OVRL scores are averaged over three independent runs for each setting. The solid lines indicate the mean OVRL scores under reverberant and non-reverberant conditions, while the shaded regions represent the standard deviation across the three runs.
Speech enhancement improves intelligibility and listening quality by suppressing noise, reverberation, and competing speakers in recorded or live audio.
We propose DriftSE, a novel one-step generative framework for speech enhancement formulated as a latent distribution equilibrium problem. During training, the drifting field aligns the generator's pushforward distribution with the clean speech manifold through drifting in a latent domain. During inference, the drifting process is discarded, enabling one-step generation. We establish that its enhancement quality depends fundamentally on the choice of latent representation. Semantic latents preserve phonetic structure but fail to capture physical acoustic cues, whereas acoustic latents reconstruct the physical signal but risk linguistic hallucination. Therefore, we introduce dual-latent drifting, performing parallel drifting in both semantic and acoustic latents to simultaneously preserve phonetic intelligibility and acoustic fidelity. Additionally, we demonstrate that DriftSE enables fully unpaired training by aligning latent distributions rather than exact point-wise targets. Consequently, DriftSE facilitates cross-dataset learning in the absence of paired noisy-clean samples. Moreover, DriftSE exhibits broad architectural flexibility across different generator backbones. Extensive evaluations on additive denoising and convolutive dereverberation demonstrate robust one-step enhancement across both offline and real-time causal settings. Notably, DriftSE achieves state-of-the-art word error rates across all four evaluated datasets while strictly operating at 1 NFE. Code and audio examples are available online.
Liang Xu, Diego Caviedes-Nozal, W. Bastiaan Kleijn +2
Victoria University of Wellington, New Zealand · GN Advanced Science, Denmark · Lincoln University
Inspired by recent developments in neural speech coding and diffusion-based language modeling, we tackle speech enhancement by modeling the conditional distribution of clean speech codes given noisy speech codes using absorbing discrete diffusion. The proposed approach, which we call ADDSE, leverages both the expressive latent space of neural audio codecs and the non-autoregressive sampling procedure of diffusion models. To efficiently model the hierarchical structure of residual vector quantization codes, we propose RQDiT, which combines techniques from RQ-Transformer and diffusion Transformers for non-autoregressive modeling. Results show competitive performance in terms of non-intrusive objective metrics on two datasets, especially at low signal-to-noise ratios and with few sampling steps. Code and audio examples are available online.
Philippe Gonzalez
Department of Health Technology, Technical University of Denmark
Neural audio codecs have largely promoted the application of language models (LMs) for speech applications. However, the effectiveness of autoregressive LM-based models in unifying speech enhancement (SE) tasks remains underexplored. In this work, we propose UniSE, a unified decoder-only LM-based framework to handle different SE tasks including speech restoration, target speaker extraction, and speech separation. Conditioned on input speech features, it autoregressively generates target discrete tokens, facilitating compatibility between distinct learning patterns of multiple tasks. To further optimize speech quality, we introduce a progressive reinforcement learning strategy with multiple assessment criteria. Experiments on several benchmarks show that UniSE achieves competitive performance compared to discriminative and generative baselines, demonstrating the capacity of LMs in unifying SE tasks. The code and demo are available at: https://github.com/alibaba/unified-audio/tree/main/QuarkAudio-UniSE.
Haoyin Yan, Chengwei Liu, Shaofei Xue +4
Qwen Business Unit of Alibaba, China · Tongyi AI Lab of Alibaba, China