Uni-Stream: A Unified Streaming Speech Language Model for Speech Synthesis and Recognition
Authors: Xuan Shi, Chang Zeng, Tiantian Feng, Ye Zhang, Jun Liu, Shih-Heng Wang, Lie Lu, Shrikanth Narayanan
Organizations: University of Southern California, USA · National Institute of Informatics, Japan · Massachusetts Institute of Technology, USA · Dolby Laboratories, USA
Unified streaming speech language models aim to support speech recognition and synthesis within a single model, enabling real-time human-computer interaction. However, jointly supporting these two directions of speech-text conversion while preserving task-specific performance remains challenging. We propose Uni-Stream, a unified streaming speech language model built on compact pretrained speech models. Uni-Stream processes interleaved text and speech in a streaming format and achieves performance comparable to task-specific models through two-stage training. We also develop a data annotation pipeline for streaming training and will publicly release the generated transcriptions and word-level timestamps. We further examine Uni-Stream's representations across tasks and modalities and find that speech and text residence in distinct latent feature space even when trained in a unified model architecture, while representations within each modality become more task-dependent in later layers.
Figures & tables
Figure 1: Overview of Uni-Stream. (a) Streaming ASR and time-aligned TTS sequences (frame counts illustrative). (b) Shared backbone with text and diffusion heads. (c) Two-stage training: task-specific streaming adaptation followed by joint training. The distillation branches show the objectives evaluated in our training ablations.
Model
WER (%) ↓
Emit delay (s) ↓
RTF ↓
clean
other
clean
other
VibeVoice-ASR-Streaming
3.54
9.45
1.68
1.65
0.17
Ours (Adapted ASR)
3.56
10.56
0.66
0.67
0.19
Table 1: Stage I ASR results on LibriSpeech test sets.
Model
WER(%) ↓
SIM ↑
UTMOS ↑
FPL(ms) ↓
RTF ↓
Human recording
2.14
0.734
3.527
–
–
VibeVoice-1.5B
3.04
0.689
3.971
–
–
Fixed-ratio
7.67
0.720
3.813
122
0.57
Time-aligned
3.54
0.712
4.005
130
0.66
Table 2: Stage I TTS results on Seed-TTS-Eval English.
Training configuration
ASR (LibriSpeech)
TTS (Seed-TTS-en)
WER (%) ↓
Emission delay (s) ↓
RTF ↓
WER (%) ↓
SIM ↑
UTMOS ↑
FPL (ms) ↓
RTF ↓
clean
other
clean
other
40k steps
SFT only
6.53
16.67
0.66
0.66
0.16
5.03
0.688
3.980
123
0.64
+ TTS teacher
6.76
16.88
0.66
0.66
0.18
4.31
0.702
3.995
125
0.71
+ TTS & ASR teachers
6.13
15.65
0.66
0.67
0.15
4.17
0.698
4.014
120
0.59
Table 3: Stage II ASR and TTS results on LibriSpeech and Seed-TTS-en, respectively, at 40k and 300k training steps.
Figure 2: Content-matched similarity across layers.
Model
Mode
Backbone
Data (h)
ASR (LibriSpeech)
TTS (Seed-TTS-en)
ASR
TTS
WER (%) ↓
WER (%) ↓
SIM ↑
ASR / TTS
clean
other
Unified models
HoliTok-Unite ( Li et al., 2026 )
O
O
0.5B
60–86k / 95k
5.48
12.65
7.20
0.55
Ming-UniAudio ( Yan et al., 2025 )
O
O
16B-A3B
400k / 800k
1.62
–
1.85
0.58
Streaming ASR models
Table 4: Comparison with unified and single-task streaming speech models on ASR (LibriSpeech) and TTS (Seed-TTS-Eval). S: streaming (ASR latency); O: offline. n/r: not reported.
Appendix figures & tables3 assets
Supplementary material from the paper’s appendix.
Appendix
Aligner
Mean
Start
End
CrisperWhisper
34.9
31.4
38.5
MMS-FA
37.1
43.4
30.8
Qwen3-FA
30.4
30.0
30.9
Three-way merge
24.4
24.9
23.9
Appendix
Table 5: Word-boundary mean absolute error (ms) on TIMIT. Mean averages start and end errors.
Stage I
Stage II
Setting
TTS-TA
TTS-FR
ASR
Uni-Stream
Training steps
80k
80k
20k
300k
GPUs
2
4
4
4
Tokens per GPU
5,600
2,000
6,400
4,800
Warm-up
2%
2%
2%
0.22%
Appendix
Table 6: Training settings on H100 80 GB GPUs. TA/FR: time-aligned/fixed-ratio TTS. Uni-Stream: TTS-teacher-only configuration.
Metric
Uni-Stream − specialist
Both teachers − TTS teacher only
ASR WER
+0.13
− 0.07
[ − 0.03, 0.29]
[ − 0.22, 0.09]
TTS WER
− 0.33
+0.19
[ − 1.68, 0.63]
[ − 0.75, 1.41]
SIM
+0.002
− 0.001
[0.000, 0.005]
[ − 0.003, 0.002]
Appendix
Table 7: Paired Stage II differences with 95% confidence intervals. The specialist comparison uses Uni-Stream trained with the TTS teacher alone.
We present UNISON, a latent diffusion framework that unifies speech generation, sound generation, and audio editing within a single model. A single model handles text-to-audio, text-to-speech, zero-shot speaker cloning, mixed speech-and-sound generation, scene-level audio editing, speech-in-scene editing, and timed temporal composition, all of which share a single set of weights. Our architecture features two core designs: (1) Layer-wise deep LLM fusion, which injects hidden states from uniformly sampled layers of a frozen MLLM into corresponding MM-DiT blocks via learned projections, providing depth-matched semantic conditioning that improves instruction following over single-layer baselines; and (2) a unified multi-task architecture where task identity is encoded solely by a channel-wise mask and source audio is provided through VAE-encoded channel concatenation. Training is stabilized by an online GPU-side multi-task data synthesis pipeline with task-homogeneous batching and a two-stage curriculum. With 621M--732M trainable parameters, UNISON achieves results competitive with or exceeding task-specialist models across evaluated domains, while being roughly 4× smaller than comparable unified systems.
Zhaoqing Li, Haoning Xu, Jingran Su +9
The Chinese University of Hong Kong · The Hong Kong Polytechnic University · City University of Hong Kong +3
Text-speech joint spoken language modeling (SLM) aims at natural and intelligent speech-based interactions, but developing such a system may suffer from modality mismatch: speech unit sequences are much longer than text tokens. Prior work reduces this gap with text-aligned tokenization and embedding (TASTE), producing speech tokens that align in lengths with their textual counterparts. However, the dependence on an external ASR system and the use of a non-causal decoder limits streaming use. To address this limitation, we propose TASTE-S, a streamable extension of TASTE suitable for real-time usage. TASTE-S integrates a CTC-based ASR module into the encoder for instant dual-modality encoding. We also redesign the unit decoder to enable on-the-fly decoding. With joint training, we show that TASTE-S matches TASTE's performance while significantly reducing latency. Further investigations reveal that TASTE-S remains robust to transcriptions and enables long-form encoding and decoding.
Liang-Hsuan Tseng, Hung-yi Lee
Graduate Institute of Communication Engineering, National Taiwan University
Unification of automatic speech recognition (ASR) systems reduces development and maintenance costs, but training a single model to perform well in both offline and low-latency streaming settings remains challenging. We present a Unified ASR framework for Transducer (RNNT) training that supports both offline and streaming decoding within a single model, using chunk-limited attention with right context and dynamic chunked convolutions. To further close the gap between offline and streaming performance, we introduce an efficient Triton implementation of mode-consistency regularization for RNNT (MCR-RNNT), which encourages agreement across training modes. Experiments show that the proposed approach improves streaming accuracy at low latency while preserving offline performance and scaling to larger model sizes and training datasets. The proposed Unified ASR framework and the English model checkpoint are open-sourced.
Andrei Andrusenko, Vladimir Bataev, Lilit Grigoryan +3