We present OmniVoice, a massively multilingual zero-shot text-to-speech (TTS) model that scales to over 600 languages. At its core is a novel diffusion language model-style discrete non-autoregressive (NAR) architecture. Unlike conventional discrete NAR models that suffer from performance bottlenecks in complex two-stage (text-to-semantic-to-acoustic) pipelines, OmniVoice directly maps text to multi-codebook acoustic tokens. This simplified approach is facilitated by two key technical innovations: (1) a full-codebook random masking strategy for efficient training, and (2) initialization from a pre-trained LLM to ensure superior intelligibility. By leveraging a 581k-hour multilingual dataset curated entirely from open-source data, OmniVoice achieves the broadest language coverage to date and delivers state-of-the-art performance across Chinese, English, and diverse multilingual benchmarks. Our code and pre-trained models are publicly available at https://github.com/k2-fsa/OmniVoice.
Figures & tables
Figure 1: Illustration of OmniVoice architecture.
Figure 2: Comparison of per-layer masking and full-codebook random masking. The x-axis denotes the time dimension, the y-axis the codebook dimension, and white blocks indicate masked tokens.
Figure 3: Statistics of the multilingual training dataset.
Model
Params.
Training Data (hours)
LibriSpeech-PC test-clean
SIM-o ↑
WER ↓
UTMOS ↑
Ground-truth
-
-
0.690
1.87
4.10
AR Models
IndexTTS2
1.7B
55k Multi.
0.700
2.35
4.06
CosyVoice3
1.1B
1000k Multi.
0.694
1.59
4.28
VoxCPM
0.7B
1800k Multi.
0.717
1.74
4.18
Table 1: Objective evaluation results on Chinese and English test sets. Baseline results are obtained using official checkpoints. Params. (Parameters) denotes the total parameter size of voice cloning TTS systems (including audio tokenizer, vocoder, and other related components). Best results are highlighted in bold . Top: Results on LibriSpeech-PC. Bottom: Results on Seed-TTS test sets.
Model
CMOS ↑
SMOS ↑
Ground-truth
0.00
3.02 ± 0.20
Qwen3-TTS
0.40 ± 0.16
3.65 ± 0.18
ZipVoice
-0.30 ± 0.16
3.35 ± 0.19
MaskGCT
-0.38 ± 0.17
3.20 ± 0.18
OmniVoice-Emilia
0.42 ± 0.15
3.58 ± 0.18
OmniVoice
0.44 ± 0.16
3.80 ± 0.17
Table 2: Subjective evaluation results on Chinese and English test sets. CMOS and SMOS are used for evaluation. Best results are highlighted in bold .
Language
WER ↓
SIM-o ↑
OmniVoice
MiniMax
ElevenLabs
OmniVoice
MiniMax
ElevenLabs
Arabic
1.392
1.665
1.666
0.776
0.736
0.706
Cantonese
17.709 *
34.111
51.513
0.838
0.778
0.670
Chinese
1.008
2.252
16.026
0.821
0.780
0.677
Czech
2.856
3.875
2.108
0.837
0.796
0.685
Dutch
1.358
1.143
0.803
0.813
0.738
0.680
Table 3: Evaluation on the MiniMax-Multilingual-24 test set.
Model
FLEURS-Multilingual-102
Avg SIM-o ↑
Avg CER ↓
Languages with CER ≤
5%
10%
Ground-truth
-
5.11
75
92
OmniVoice
0.788
4.00
82
95
Table 4: Evaluation on the FLEURS-Multilingual-102 test set.
Figure 4: CERs of OmniVoice vs. ground truth across languages with varying training data durations on FLEURS-Multilingual-102. Hollow circles indicate languages where OmniVoice achieves lower CER than ground-truth, Filled circles indicate higher CER.
Masking Strategy
Librispeech-PC test-clean
SIM-o ↑
WER ↓
UTMOS ↑
SoundStorm-style mask
0.661
3.00
4.12
MaskGCT-style mask
0.660
2.04
4.17
Full-codebook random mask
0.697
1.57
4.23
- compute loss on a single codebook
0.648
2.85
4.22
Table 5: Impact of different masking strategies.
Model
Learning Rate
WER ↓
Libri
Seed-en
Seed-zh
LLM initialization
1e-4
1.57
1.72
0.89
Random initialization
1e-4
2.79
2.34
1.11
2e-4
2.52
2.43
1.01
5e-4
2.56
2.07
1.01
1e-3
2.72
2.29
1.02
Table 6: Impact of LLM initialization on WER across datasets. Abbreviations: Libri = Librispeech-PC test-clean, Seed-en = Seed-TTS test-en, Seed-zh = Seed-TTS test-zh.
Model
Librispeech-PC test-clean
SIM-o ↑
WER ↓
UTMOS ↑
w/o prompt denoise
0.697
1.57
4.23
w/ prompt denoise
0.668
1.56
4.32
Table 7: Impact of prompt denoising task.
Inference Steps
Batch Size
1
2
4
8
16
0.0319
0.0263
0.0235
0.0224
32
0.0598
0.0486
0.0436
0.0414
Table 8: RTF with different inference steps and batch sizes.
Appendix figures & tables3 assets
Supplementary material from the paper’s appendix.
Appendix
LibriSpeech-PC test-clean
Seed-TTS test-en
Seed-TTS test-zh
Steps
SIM-o ↑
WER ↓
UTMOS ↑
SIM-o ↑
WER ↓
UTMOS ↑
SIM-o ↑
WER ↓
UTMOS ↑
64
0.729
1.28
4.30
0.742
1.60
3.92
0.777
0.81
3.13
32
0.729
1.30
4.28
0.741
1.53
3.91
0.777
0.84
3.11
16
0.728
1.50
4.23
0.735
1.72
3.92
0.773
0.99
3.00
8
0.713
2.02
4.02
0.716
1.94
3.59
0.756
1.58
2.72
Appendix
Table 9: Objective evaluation results of OmniVoice with different inference steps.
Language
Ground-truth
OmniVoice
Language
Ground-truth
OmniVoice
Afrikaans
3.54
3.08
Amharic
4.98
7.30
Armenian
1.38
1.45
Assamese
5.29
3.92
Asturian
3.79
2.07
Azerbaijani
2.60
3.20
Belarusian
1.80
1.00
Bengali
4.10
3.08
Bosnian
1.61
0.66
Bulgarian
1.66
0.87
Burmese
6.77
12.20
Cantonese
11.01
21.92
Appendix
Table 10: Per-language CERs of ground-truth and OmniVoice on FLEURS-Multilingual-102
Language
ID
ISO
Hours
Language
ID
ISO
Hours
Language
ID
ISO
Hours
Abadi
kbt
kbt
9.73
Abkhazian
ab
abk
57.27
Abron
abr
abr
9.22
Abua
abn
abn
10.27
Adamawa Fulfulde
fub
fub
13.12
Adyghe
ady
ady
32.6
Afade
aal
aal
10.19
Afrikaans
af
afr
4.4
Agwagwune
yay
yay
8.26
Aja (Benin)
ajg
ajg
5.63
Akebu
keu
keu
9.1
Alago
ala
ala
11.04
Albanian
sq
sqi
8.59
Algerian Arabic
arq
arq
9.64
Algerian Saharan Arabic
aao
aao
2.02
Ambo-Pasco Quechua
qva
qva
9.59
Ambonese Malay
abs
abs
10.03
Amdo Tibetan
adx
adx
56.94
Appendix
Table 11: Supported languages of OmniVoice along with its OmniVoice language ID, ISO 639-3 code, and training data duration (hours).
We introduce Voxtral TTS, an expressive multilingual text-to-speech model that generates natural speech from as little as 3 seconds of reference audio. Voxtral TTS adopts a hybrid architecture that combines auto-regressive generation of semantic speech tokens with flow-matching for acoustic tokens. These tokens are encoded and decoded with Voxtral Codec, a speech tokenizer trained from scratch with a hybrid VQ-FSQ quantization scheme. In human evaluations conducted by native speakers, Voxtral TTS is preferred for multilingual voice cloning due to its naturalness and expressivity, achieving a 68.4% win rate over ElevenLabs Flash v2.5. We release the model weights under a CC BY-NC license.
Autoregressive (AR) text-to-speech (TTS) models generate discrete speech tokens sequentially, which makes inference slow and can degrade robustness, since local errors propagate to later positions and can escalate into hallucination. This limitation stems from their left-to-right AR commitment: each token must be determined before future speech-token context is available. However, such ordering is not an inherent requirement for TTS, since the model receives the full input text before synthesis. In this paper, we introduce DELTA-TTS, a lightweight LoRA-based adaptation framework that converts a pretrained AR TTS model into a discrete diffusion language model (dLLM) for confidence-ordered speech-token decoding. To better capture the local structure of speech, DELTA-TTS incorporates a convolution module that injects local acoustic context, together with a 1/t-weighted training objective and a time-shifted inference schedule that together defer low-confidence positions to later steps. Trained on only 585 hours of LibriTTS, DELTA-TTS achieves a 1.75% WER on Seed-TTS test-en, outperforming its AR backbone while generating tokens 3.3 times faster. Further analysis shows that DELTA-TTS produces sharper text-speech alignment, increases overall decoding confidence, and mitigates the hallucinations observed in AR generation. Audio samples, inference code, and model weights are available at https://slpr-lab.github.io/DELTA-TTS.
Recently, zero-shot text-to-speech (TTS) has enabled high-fidelity and expressive speech synthesis, but it often fails to imitate unseen speaking styles from uncommon scenarios (e.g., crosstalk, dialects). Moreover, fine-tuning pretrained models requires large, high-quality datasets, limiting rapid personalization. We propose VoiceTTA, a reinforcement learning-based test-time adaptation (TTA) method that improves voice imitation of pretrained zero-shot TTS models. VoiceTTA introduces two style rewards based on coefficient-of-variation differences of F0 and energy, combined with speaker similarity and intelligibility (WER from a pretrained Whisper model), and optimizes learnable prefixes via group relative preference optimization (GRPO) in a flow matching-based model at inference time. Extensive experiments demonstrate substantial improvements on uncommon speech prompts, outperforming state-of-the-art baselines. Audio samples are available at https://voicetta.pages.dev/
Tianxin Xie, Chenxing Li, Dong Yu +1
The Hong Kong University of Science and Technology (Guangzhou) · Tencent