Voice style conversion aims to transform an input utterance to match a target speaker's timbre, accent, and emotion, with a central challenge being the disentanglement of linguistic content from style. While prior work has explored this problem, conversion quality remains limited, and real-time voice style conversion has not been addressed. We propose StyleStream, the first streamable zero-shot voice style conversion system that achieves state-of-the-art performance. StyleStream consists of two components: a Destylizer, which removes style attributes while preserving linguistic content, and a Stylizer, a diffusion transformer (DiT) that reintroduces target style conditioned on reference speech. Robust content-style disentanglement is enforced through text supervision and a highly constrained information bottleneck. This design enables a fully non-autoregressive architecture, achieving real-time voice style conversion with an end-to-end latency of 1 second. Samples and real-time demo: https://berkeley-speech-group.github.io/StyleStream/.
Figures & tables
Figure 1: System overview of StyleStream. The Destylizer extracts content features disentangled from style, and the Stylizer generates speech that preserves the source linguistic content while adopting the target timbre, accent, and emotion.
Figure 2: Destylizer architecture. The Destylizer, as part of the ASR encoder, is trained with a sequence-to-sequence ASR loss. The continuous representations immediately before the FSQ module are taken as content features.
Figure 3: Stylizer architecture. The Stylizer contains a style encoder and a diffusion transformer.
Model
WER ( % ) ↓
S-SIM ↑
A-SIM ↑
E-SIM ↑
NMOS ↑
A-SMOS ↑
E-SMOS ↑
S-SMOS ↑
Ground Truth
3.8
—
—
—
3.71 ± .11
—
—
—
FACodec [ 10 ]
15.5
0.763
0.408
0.668
2.55 ± .13
2.02 ± .13
2.76 ± .14
2.98 ± .14
CosyVoice 2.0 [ 7 ]
9.5
0.794
0.450
0.655
3.47 ± .11
2.26 ± .13
2.58 ± .15
3.25 ± .14
SeedVC v2 6
21.7
0.766
0.549
0.688
2.65 ± .12
3.34 ± .12
3.21 ± .13
3.11 ± .13
Vevo [ 11 ]
17.5
0.818
0.596
0.712
3.38 ± .12
3.93 ± .11
3.49 ± .13
3.76 ± .13
Vevo 1.5 [ 52 ]
20.5
0.798
0.554
0.683
3.10 ± .12
3.15 ± .14
3.15 ± .14
3.34 ± .14
Table 1: Results for zero-shot voice style conversion. The best results are shown in bold , and the second best results are underlined .
Chunk size (ms)
WER ( % ) ↓
S-SIM ↑
A-SIM ↑
E-SIM ↑
UTMOS ↑
200
19.8
0.797
0.586
0.704
2.33 ± .02
400
15.0
0.806
0.597
0.720
2.69 ± .03
600
13.6
0.811
0.602
0.734
2.85 ± .03
800
12.7
0.816
0.607
0.745
2.94 ± .03
1000
12.6
0.820
0.613
0.751
3.00 ± .03
Table 2: Chunk size vs. quality tradeoff
Chunk size (ms)
RTX 4060 (s)
RTX A6000 (s)
100
0.537 ± .053
0.427 ± .013
200
0.574 ± .059
0.429 ± .009
300
0.615 ± .057
0.432 ± .010
400
0.647 ± .090
0.441 ± .008
500
0.653 ± .051
0.431 ± .009
600
0.668 ± .048
0.429 ± .006
Table 3: Chunk size vs. processing time on RTX 4060 and RTX A6000. Entries show mean ± standard deviation over runs.
Content Features
WER ( % ) ↓
S-SIM ↑
A-SIM ↑
E-SIM ↑
UTMOS ↑
HuBERT-Large-ASR 18th
12.5
0.731
0.389
0.627
3.28 ± .020
Vevo Continuous
12.6
0.767
0.420
0.660
2.92 ± .029
Vevo Indices
25.6
0.757
0.564
0.657
2.65 ± .036
Destylizer (offline)
10.7
0.837
0.626
0.733
3.22 ± .029
Table 4: Stylizer performance trained on different content features.
Content Features
Accent Acc. ↓
Emotion Acc. ↓
Speaker Acc. ↓
HuBERT-Large-ASR 18th
78.00%
85.60%
86.60%
Vevo Continuous
68.93%
78.73%
64.76%
Vevo Indices
55.60%
53.40%
23.00%
CosyVoice 2.0 Indices
50.60%
56.20%
25.00%
Destylizer (offline)
43.50%
47.60%
3.50%
Destylizer (streaming)
33.80%
37.90%
3.01%
Table 5: Style classification accuracy across different content features. “Acc.” stands for accuracy.
Model
WER ( % ) ↓
S-SIM ↑
A-SIM ↑
E-SIM ↑
UTMOS ↑
StyleStream (offline)
10.7
0.837
0.626
0.733
3.22 ± .029
w/o Style Emb
15.3
0.775
0.509
0.653
3.47 ± .024
w/ FSQ Indices
123.5
0.829
0.573
0.717
2.41 ± .024
Table 6: Ablation studies on the use of style encoder and the choice of Destylizer content features.
FSQ Levels
WER ( % ) ↓
S-SIM ↑
A-SIM ↑
E-SIM ↑
UTMOS ↑
[7,5,5,5,5]
13.5
0.745
0.439
0.638
3.58 ± .022
[5,5,3,3]
14.9
0.822
0.600
0.725
3.17 ± .031
[5,3,3]*
10.7
0.837
0.626
0.733
3.22 ± .029
[3,3,3]
19.0
0.825
0.618
0.732
3.13 ± .032
[5,3]
101.4
0.834
0.582
0.714
2.70 ± .029
Table 7: Ablation on Destylizer FSQ bottleneck size. “*” denotes the architecture used in all main experiments.
Training data
WER (%) ↓
S-SIM ↑
A-SIM ↑
E-SIM ↑
LMG (1.3k h)
10.7
0.837
0.626
0.733
Emilia (50k h)
10.3
0.816
0.594
0.687
Table 8: Ablation on the Destylizer training data configurations.
Streaming zero-shot voice conversion (VC) has become increasingly popular due to its potential for real-time applications. The recently proposed MeanVC achieves lightweight streaming zero-shot VC, but it has several limitations: its chunk-wise autoregressive denoising doubles the effective training sequence length, conversion quality degrades under small-chunk settings, and its timbre encoder directly relies on reference mel-spectrograms, making it sensitive to reference audio quality. To address these limitations we propose MeanVC 2. We introduce future-receptive chunking (FRC), which explicitly schedules past and future receptive fields across diffusion transformer decoder layers and removes clean-chunk teacher forcing. By incorporating bounded future context, FRC enables stable conversion with a 40 ms chunk size. We further introduce a universal timbre token encoder, which constructs a timbre representation from a global speaker embedding and retrieves fine-grained timbre cues via cross-attention, improving robustness to low-quality references and enhancing zero-shot speaker similarity. Experimental results show that MeanVC 2 significantly outperforms MeanVC, while reducing latency from 211 ms to 110 ms. Audio samples are publicly available. The source code will be publicly released.
Guobin Ma, Yuxuan Xia, Yuepeng Jiang +6
The University of New South Wales, Australia · WeNet Open Source Community, China
Streaming zero-shot voice conversion struggles to disentangle timbre from linguistic content without degrading utility or inflating latency. Current methods rely on information bottleneck (IB) or speaker perturbation. While IB filters out timbre, it discards prosody, forcing models to explicitly inject features like fundamental frequency. This often requires buffering future frames, creating algorithmic lookahead latency. On the other hand, existing perturbation methods largely overlook the crucial trade-off between timbre leakage and utility preservation. Recognizing this neglected trade-off, we find that the inherent objective of Speaker Anonymization (SA) aligns well with balancing these factors. Thus, we introduce SA as a novel perturbation mechanism to explicitly mitigate timbre leakage while retaining prosodic utility. Crucially, SA's robust representations significantly alleviate the generator's reliance on future context, enabling our strictly causal, zero-lookahead network. Audio samples are available at https://amphionteam.github.io/Zero-VC-demo/.
Yudong Li, Zihao Fang, Junwen Qiu +4
The Chinese University of Hong Kong, Shenzhen · Shenzhen Transsion Holdings Co., Ltd. · Shenzhen Loop Area Institute +1
Flow-matching approaches to voice conversion (VC) have gained attention owing to their high speech quality and strong speaker similarity. Among them, one-step models such as MeanVoiceFlow are particularly attractive because they enable efficient inference; however, their reliance on a computationally intensive content encoder remains a bottleneck. We therefore propose MeanVoiceFlow2, a framework that jointly optimizes a flow-based conversion module and a computationally efficient content encoder. The model is trained through conversion distillation using MeanVoiceFlow and the reconstruction of real data. We further incorporate diffusion-GAN training with sample mixing and teacher-guided conditioning augmentation to enhance realism and disentanglement. Experiments on zero-shot VC showed that MeanVoiceFlow2 achieved higher perceptual quality and approximately 9× faster inference than MeanVoiceFlow while maintaining comparable speaker similarity. Audio samples are available at https://www.kecl.ntt.co.jp/people/kaneko.takuhiro/projects/meanvoiceflow2/.