Voice style conversion (VSC) aims to transform a source utterance to match the timbre, accent, and emotion of a target voice while preserving linguistic content. StyleStream 1.0 introduced the first streamable zero-shot VSC system with state-of-the-art conversion quality, but remained limited in two ways: approximately 1s end-to-end latency on consumer hardware, and the lack of a text-based interface for designing or editing target voices. We present StyleStream 2.0, a fast and controllable streaming VSC framework that addresses these limitations. To reduce latency, we train a few-step pixel MeanFlow model and further fine-tune it with autoregressive feedback inspired by Self Forcing, improving robustness under streaming inference. This reduces end-to-end latency to 520~ms on a consumer GPU, a 2x speedup over StyleStream 1.0, while maintaining conversion quality. To enable controllable style manipulation, we remove the mel-spectrogram context and instead route target style information through a compact style embedding. In this embedding space, we train a unified text-conditioned flow matching model that supports both text-based voice design, which maps natural language prompts to voice styles, and instruction-based voice editing, which modifies a specific attribute of a source utterance while preserving the rest. Experiments show that StyleStream 2.0 achieves the strongest target style fidelity on VSC, voice design, and voice editing, while remaining competitive on intelligibility and non-edited attribute preservation.
Figures & tables
Figure 1: Left : Stylizer architecture. [r,t] is the MeanFlow interval, ω is the CFG strength. Right : Voice Controller architecture. When used for voice design, the auxiliary input is null ∅ ; for voice editing, source style embedding is also part of the input.
Model
NFE ↓
WER ( % ) ↓
S-SIM ↑
A-SIM ↑
E-SIM ↑
NMOS ↑
S-SMOS ↑
A-SMOS ↑
E-SMOS ↑
Ground Truth
—
4.42
—
—
—
3.72 ± .12
—
—
—
Qwen3-TTS
—
4.76
0.927
0.833
0.941
—
—
—
—
CosyVoice 2.0
10
8.07
0.814
0.484
0.674
3.54 ± .12
3.64 ± .13
3.25 ± .15
2.61 ± .15
CosyVoice 3.0
10
6.56
0.832
0.508
0.668
3.48 ± .13
3.78 ± .12
3.21 ± .14
2.80 ± .15
Vevo
32
13.33
0.853
0.659
0.745
3.42 ± .13
4.00 ± .14
3.94 ± .13
3.63 ± .13
Vevo 2.0
32
5.79
0.843
0.676
0.723
3.44 ± .13
4.06 ± .13
3.98 ± .13
3.11 ± .15
Table 1: Voice style conversion results (SS denotes StyleStream). The best results are shown in bold , and the second-best results are underlined . Ground-truth NMOS is reported as an upper bound and excluded from ranking.
Chunk (ms)
Model
Proc (ms) ↓
WER ( % ) ↓
S-SIM ↑
A-SIM ↑
E-SIM ↑
160
SS 1.0
500.1
16.59
0.821
0.624
0.724
160
SS 2.0
130.5
14.02
0.860
0.669
0.763
200
SS 1.0
528.8
15.25
0.825
0.632
0.732
200
SS 2.0
132.0
13.22
0.863
0.675
0.764
300
SS 1.0
559.3
13.34
0.833
0.644
0.748
300
SS 2.0
140.7
12.03
0.866
0.684
0.765
Table 2: Streaming chunk-size analysis on RTX 4060. We compare StyleStream 1.0 (SS 1.0, 12 NFE) and StyleStream 2.0 (SS 2.0, 4 NFE) across chunk sizes from 160 ms to 600 ms, reporting per-chunk processing time and quality. Best per chunk-size group is in bold .
Objective
Subjective
Model
WER ↓
Age-acc ↑
G-acc ↑
A-acc ↑
E-acc ↑
NMOS ↑
SC-MOS ↑
CapTTS
7.36
41.11
94.33
52.00
29.47
3.27 ± .13
3.54 ± .13
Qwen3-TTS
4.86
44.33
92.39
14.73
36.33
3.35 ± .14
2.97 ± .15
SS 2.0 (streaming)
10.42
67.94
98.94
46.73
30.13
3.23 ± .13
3.78 ± .12
SS 2.0 (offline)
6.89
70.67
99.44
52.73
32.20
3.45 ± .12
3.96 ± .10
Table 3: Voice design results. WER and VoxProfile accuracies are percentages. We use Age-acc, G-acc, A-acc, and E-acc to denote age, gender, accent and emotion classification accuracy. NMOS measures naturalness and SC-MOS measures style consistency with the prompt.
Style Consistency
MOS
Style Preservation
Model
WER ↓
Age-acc ↑
G-acc ↑
A-acc ↑
E-acc ↑
SC-MOS ↑
SP-MOS ↑
NMOS ↑
S-pres. ↑
A-pres. ↑
E-pres. ↑
Vevo
12.35
28.33
94.83
16.89
6.56
2.64 ± .16
3.64 ± .12
3.44 ± .13
0.9092
0.6303
0.8902
Vevo2
7.82
24.67
65.17
6.78
7.11
2.26 ± .15
3.28 ± .14
3.32 ± .14
0.8853
0.5948
0.8832
SS 2.0 (streaming)
9.62
67.17
99.67
46.56
13.67
3.64 ± .13
3.50 ± .13
3.21 ± .15
0.7644
0.4864
0.8794
SS 2.0 (offline)
6.43
71.67
99.17
54.33
14.67
3.76 ± .13
3.58 ± .12
3.48 ± .13
0.7899
0.5102
0.8851
Table 4: Voice editing results. WER and VoxProfile accuracies are percentages. We use Age-acc, G-acc, A-acc, and E-acc to denote age, gender, accent and emotion classification accuracy. SC-MOS measures instruction consistency, SP-MOS measures preservation of non-edited attributes, and NMOS measures naturalness.
Setting
WER ( % ) ↓
S-SIM ↑
A-SIM ↑
E-SIM ↑
w/ AR-ft
12.03
0.866
0.684
0.765
w/o AR-ft
19.54
0.866
0.676
0.767
Table 5: Ablation studies. Left : Effect of with or without autoregressive fine-tuning (AR-ft) under streaming inference. Right : Different style encoder effect under offline inference.
Appendix figures & tables9 assets
Supplementary material from the paper’s appendix.
Appendix
Tier
Samples
Audio source
Voice design
2,000,005
Synthesized
Accent editing
3,000,028
Synthesized
Emotion editing
2,999,992
Synthesized
Age editing
1,499,961
Natural
Gender editing
1,481,314
Natural
Joint age+gender editing
1,000,001
Natural
Appendix
Table 6: Training tiers for the text-based StyleStream 2.0 corpus.
Design prompt type
Target axes
Count
Timbre
age, gender
600
Accent
accent
600
Emotion
emotion
600
Timbre+accent
age, gender, accent
300
Timbre+emotion
age, gender, emotion
300
All attributes
age, gender, accent, emotion
600
Appendix
Table 7: Voice design prompt mix for evaluation.
Figure 2: Naturalness MOS (NMOS) and Emotion SMOS (E-SMOS) evaluation interfaces. The S-SMOS and A-SMOS surveys are analogous to E-SMOS, with the emotion target replaced by speaker identity or accent and the listener instructed to ignore the other style axes.
Figure 3: Voice-design (VD) and voice-editing (VE) evaluation interfaces. Each item pairs a source and a test audio sample with a natural-language style prompt (VD) or edit instruction (VE), and asks for naturalness plus task-specific match (and preservation, for VE) ratings.
ω
WER ( % ) ↓
S-SIM ↑
A-SIM ↑
E-SIM ↑
1
9.53
0.8806
0.6852
0.7462
2
7.19
0.8835
0.7071
0.7711
3
7.03
0.8791
0.7055
0.7751
4
6.98
0.8741
0.7000
0.7766
5
6.97
0.8694
0.6929
0.7747
6
7.00
0.8653
0.6870
0.7722
Appendix
Table 8: CFG guidance-scale ablation on the offline StyleStream 2.0 model at NFE =4 .
NFE
WER ( % ) ↓
S-SIM ↑
A-SIM ↑
E-SIM ↑
1
7.06
0.8755
0.6976
0.7689
2
7.05
0.8791
0.7052
0.7740
4
7.02
0.8787
0.7058
0.7744
8
7.06
0.8785
0.7047
0.7768
16
7.09
0.8791
0.7056
0.7756
Appendix
Table 9: NFE ablation on the offline StyleStream 2.0 model at ω=3 .
Text conditioning
WER ↓
Age-acc ↑
G-acc ↑
A-acc ↑
E-acc ↑
Trainable embeddings
6.80
59.67
81.17
45.60
29.47
Flan-T5-small
7.02
61.06
97.94
47.87
33.60
Flan-T5-base
7.05
64.56
98.33
49.20
39.93
Flan-T5-XL
6.91
58.94
97.61
49.13
38.40
Appendix
Table 10: Voice-design text-encoder ablation after 150k controller training steps. WER and attribute accuracies are percentages. Flan-T5 encoders are frozen; trainable embeddings are randomly initialized. Bold / underline mark the best/second-best values within this ablation.
Text conditioning
WER ↓
Age-acc ↑
G-acc ↑
A-acc ↑
E-acc ↑
S-pres. ↑
A-pres. ↑
E-pres. ↑
Trainable embeddings
6.77
60.00
60.00
43.56
12.56
0.5873
0.4480
0.8723
Flan-T5-small
6.86
54.17
97.00
47.44
16.11
0.5795
0.4350
0.8689
Flan-T5-base
6.86
61.50
92.17
49.00
22.89
0.5960
0.4535
0.8721
Flan-T5-XL
6.95
55.00
97.33
49.44
24.33
0.5992
0.4433
0.8718
Appendix
Table 11: Voice-editing text-encoder ablation after 150k controller training steps, under the same conditions as Table 10 . WER and accuracies are percentages; preservation scores are similarities to the source. Best/second-best values include ties.
Asset
License / terms
Source
Emilia / Emilia-EN
CC BY-NC 4.0; audio copyright remains with original owners
https://huggingface.co/datasets/amphion/Emilia
CapSpeech
CC BY-NC 4.0
https://github.com/WangHelin1997/CapSpeech
ESD
Free for research use only
https://hltsingapore.github.io/ESD/download.html
GLOBE
CC0 1.0
https://huggingface.co/datasets/MushanW/GLOBE
LibriTTS
CC BY 4.0
https://www.openslr.org/60/
RAVDESS
CC BY-NC-SA 4.0
https://zenodo.org/records/1188976
Appendix
Table 12: Licenses and terms for existing assets used in this work.
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/.
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
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/.