Modern Text-to-Speech (TTS) systems increasingly leverage Large Language Model (LLM) architectures to achieve scalable, high-fidelity, zero-shot generation. However, these systems typically rely on fixed-frame-rate acoustic tokenization, resulting in speech sequences that are significantly longer than, and asynchronous with their corresponding text. Beyond computational inefficiency, this sequence length disparity often triggers hallucinations in TTS and amplifies the modality gap in spoken language modeling (SLM). In this paper, we propose a novel tokenization scheme that establishes one-to-one synchronization between continuous acoustic features and text tokens, enabling unified, single-stream modeling within an LLM. We demonstrate that these synchronous tokens maintain high-fidelity audio reconstruction and can be effectively modeled in a latent space by a large language model with a flow matching head. Moreover, the ability to seamlessly toggle speech modality within the context enables text-only guidance--a technique that blends logits from text-only and text-speech modes to flexibly bridge the gap toward text-only LLM intelligence. Experimental results indicate that our approach achieves performance competitive with state-of-the-art TTS and SLM systems while virtually eliminating content hallucinations and preserving linguistic integrity, all at a significantly reduced inference cost.
Figures & tables
Figure 1: Word-level alignment via Viterbi decoding. An illustration of the forced alignment between a speech waveform and the transcript “That’s exactly what happened…”. The Viterbi algorithm identifies the most probable frame-level assignment to determine a position pi for each token wi in the encoded text sequence.
Figure 2: Operating under a Variational Autoencoder (VAE) framework, our model utilizes a symmetric encoder-decoder architecture. Each module integrates a CNN-based component for local acoustic feature extraction and reconstruction, complemented by a transformer-based backbone designed to capture the dynamic temporal range of synchronized speech-text sequences.
Figure 3: Attention mask of the encoder (left) and the decoder (right). Asterisks ( ∗ ) mark text-assigned temporal indices. In the encoder, non-text-assigned positions are restricted to intra-block attention, excluding boundary tokens; conversely, text-assigned positions are permitted to attend across both preceding and succeeding blocks. The decoder also utilizes a localized mechanism where each position attends to the current and immediately preceding blocks.
Figure 4: Each text token wi is paired with the speech representation at the K -shifted position, comprising token features si−K , the number of preceeding frames fi−Kbefore , the number of successive frames fi−Kafter , and processed by an autoregressive decoder. The decoder predicts the next text token and produces a conditioning vector, which the flow matching head uses to generate the next speech representation (si−K+1,fi−K+1before,fi−K+1before) .
Models
FPS
BPF
CER ↓
SIM ↑
oMOS ↑
Reference
24k
384
0.00
85.4
3.17
Encodec [ 3 , 55 ]
75
40
0.16
82.4
3.04
Encodec [ 3 , 55 ]
75
80
0.11
83.6
3.10
DAC-24khz [ 4 ]
75
80
0.17
83.4
2.94
Mimi [ 5 ]
12.5
352
0.14
83.9
3.04
Higgs V2 [ 6 ]
25
80
0.10
84.1
3.20
Table 1: Comparison of reconstruction performance across discrete and continuous tokenizers. We report the frame rate (fps), bits per frame (bpf), and key performance metrics (CER, SIM, and oMOS). Continuous baselines ( † ) are reported at 16-bit precision.
SeedTTS-Eval
LibriTTSR-Eval
Models
Mem
RTF
Hours
CER
SIM
oMOS
CER
SIM
oMOS
XTTS v2 [ 25 ]
2.1
0.19
30K
2.10
70.3
2.67
0.59
74.0
3.17
Index-TTS2 [ 17 ]
10.1
0.58
55K
0.31
79.8
2.95
0.23
83.3
3.34
Higgs Audio v2 [ 6 ]
16.3
0.44
10M
9.57
75.3
2.98
1.88
79.7
3.28
VibeVoice 1.5B [ 19 ]
5.2
0.51
0.5–1.6M
3.07
72.3
2.92
1.25
79.5
3.24
FireRedTTS-2 [ 18 ]
12.2
0.76
1.1–1.4M
0.81
75.1
2.96
1.44
81.2
3.28
Table 2: Voice cloning evaluation. Objective comparison between TADA and baseline models using character error rate (CER), speaker similarity (SIM) and Objective Mean Opinion Score (oMOS) metrics.
Models
CER
SIM
oMOS
sSIM
sMOS
Index-TTS [ 17 ]
1.90
76.9
2.84
4.25
3.61
VibeVoice 1.5B [ 19 ]
2.51
73.3
2.54
3.92
3.91
FireRedTTS-2 [ 18 ]
21.6
73.8
2.84
3.98
3.58
TADA-3B
2.34
67.0
2.86
-
-
+ Text-Free Guidance
4.30
72.4
2.87
-
-
+ Online RS
2.74
74.7
2.84
4.18
3.78
Table 3: Performance comparison for long-form, expressive speech generation using EARS dataset samples.
SI PPL ↓
sSC ↑
tSC ↑
Models
T
TS
T
TS
T
TS
Llama-1B-Ins
23.7
-
73.8
-
97.4
-
Llama-3B-Ins
20.9
-
79.4
-
98.0
-
TWIST-7B [ 54 ]
-
-
-
55.3
-
74.1
TWIST-13B [ 54 ]
-
-
-
55.4
-
76.4
SpiritLM-7B [ 26 ]
-
-
79.4
61.0
98.0
82.9
Table 4: Language capability evaluation on the Seamless Interaction (SI), sSC, and tSC datasets, comparing text-only (T) and text-speech (TS) inference modes with and without Text-Only Guidance.
Figure 5: Reconstruction quality (CER, SS, and oMOS) vs. token density for uniformly spaced tokens.
Nfl
RTF
TLLMstep
TFMstep
CER
SIM
oMOS
2 steps
0.05
31
8
1.82
77.3
2.82
4 steps
0.05
31
14
0.85
79.1
2.96
10 steps
0.09
31
23
0.55
80.2
3.11
20 steps
0.13
31
54
0.63
79.8
3.11
Table 5: Analysis of inference latency and TTS performance of TADA-1B across varying flow matching sampling steps.
Models
PPL
CER
SIM
oMOS
Llama-1B-Instruct
23.7
-
-
-
TADA-1B ( λCE=0.1 , Base)
36.5
0.87
78.4
3.02
TADA-1B ( λCE=1.0 )
31.2
0.89
77.6
3.00
TADA-1B ( λCE=λKD=0.1 )
26.6
0.84
77.7
2.98
Table 6: Abalation results of TADA-1B with different language preservation loss config.
Zero-shot text-to-speech (TTS) relies on robust speech representations. However, current speech tokenizers face a fundamental trade-off: acoustic codecs preserve high-fidelity audio but lack linguistic constraints, causing content errors during generation, whereas semantic tokens from self-supervised learning (SSL) models ensure precise text alignment but discard some acoustic information. To bridge this gap, we propose SARA, a dual-stream VAE that directly fuses a frozen SSL semantic anchor with a dedicated residual acoustic encoder. This effectively mitigates the dilemma, creating an efficient and compact latent space without relying on complex regularizers. SARA achieves superior reconstruction quality over strong baselines. Furthermore, in downstream zero-shot TTS tasks, it yields highly natural and expressive synthesis quality, and maintains robust generation performance even under accelerated inference, offering a favorable trade-off between synthesis speed and computational cost.
Peijie Chen, Wenhao Guan, Weijie Wu +7
School of Informatics, Xiamen University, China · School of Electronic Science and Engineering, Xiamen University, China · DiDi Global Inc., Beijing, China
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.
Large language model (LLM)-based text-to-speech (TTS) systems achieve remarkable naturalness via autoregressive (AR) decoding, but require N sequential steps to generate N speech tokens. We present LLaDA-TTS, which replaces the AR LLM with a masked diffusion model that completes generation in a fixed number of parallel steps, decoupling inference latency from sequence length. Remarkably, using only 50 hours of fine-tuning data, we successfully transfer a pretrained AR checkpoint to the masked diffusion paradigm via bidirectional attention. At 64 steps, LLaDA-TTS achieves 0.98% CER (zh) and 1.96% WER (en) on Seed-TTS-Eval, matching the original CosyVoice 3 baseline performance while delivering a 2x LLM-stage speedup--a notable acceleration achieved despite the absence of KV cache, an optimization the AR baseline heavily relies on. Beyond acceleration, the bidirectional architecture naturally enables zero-shot speech editing--including word-level insertion, deletion, and substitution--without any additional training. Theoretically, we prove that AR-pretrained weights are near-optimal for bidirectional masked prediction under the locality property of acoustic tokens, explaining this rapid convergence. This general method modifies only the attention mask and objective, applying seamlessly to any LLM-based AR TTS system. Code and audio samples will be available at https://deft-piroshki-b652b5.netlify.app/.