Simultaneous Speech Translation (SimulST) requires balancing high translation quality with low latency. Recent work introduced REINA, a method that trains a Read/Write policy based on estimating the information gain of reading more audio. However, we find that information-based policies often lack temporal context, leading the policy to bias itself toward reading most of the audio before starting to write. We improve REINA using two distinct strategies: a supervised alignment network (REINA-SAN) and a timestep-augmented network (REINA-TAN). Our results demonstrate that while both methods significantly outperform the baseline and resolve stability issues, REINA-TAN provides a slightly superior Pareto frontier for streaming efficiency, whereas REINA-SAN offers more robustness against 'read loops'. Applied to Whisper, both methods improve the pareto frontier of streaming efficiency as measured by Normalized Streaming Efficiency (NoSE) scores up to 7.1% over existing competitive baselines.
Figures & tables
Figure 1: Information Gain Given a fixed % of source audio, we plot information gain for each label token. There is a clearly visible Read - Write boundary at the token hull .
Figure 2: Read Loop Despite ingesting more and more audio over time, the policy continues to predict roughly the same value, corresponding to Read decisions and entering a Read Loop.
Figure 3: Latency–quality trade-off of Whisper policy ablations. We plot XComet-XL versus LAAL by sweeping the policy threshold and averaging over de/fr/es → en. REINA-TAN achieves the strongest Pareto frontier across both datasets.
Figure 4: Mean per-token latency (actual emission time minus monotonic alignment time boundaries) as a function of relative token position within the sentence, with 95% confidence intervals for 2000 samples on the dev split of CVSS-C.
FLEURS
Europarl
Model
LAAL(s)
BLEU
X-COMET
LAAL(s)
BLEU
X-COMET
De → En
REINA
1.33
25.2
.696
1.40
16.3
.608
REINA-All
1.21
26.9
.772
1.26
21.8
.760
REINA-SAN
1.03
28.6
.794
1.09
21.4
.755
REINA-TAN
1.03
27.9
.775
0.98
20.5
.721
Table 1: Streaming evaluation results on FLEURS and Europarl benchmarks. Operating points are selected to represent low and medium latency regimes across languages.
Metric / Model
de → en
fr → en
es → en
NoSE Bounds [x,y] (s)
[1.90,2.05]
[1.67,1.79]
[1.78,1.92]
Whisper-Large v3 Non-Streaming (BLEU)
33.40
31.02
22.70
SeamlessM4Tv2 (BLEU)
37.00
33.97
25.49
SeamlessStreaming (NoSE)
0.925
0.940
0.936
Whisper-REINA (NoSE)
0.921
0.955
0.944
Whisper-REINA-SAN (NoSE)
0.987
0.984
0.951
Table 2: Comparison of Normalized Streaming Efficiency (NoSE) and non-streaming baselines on FLEURS. NoSE values are calculated within the specified latency boundaries [x,y] .
Simultaneous speech-to-text translation (Simul-S2TT) consists of generating partial translations while the incoming audio frames are processed by the system. However, the streaming nature of this setup creates the challenge of deciding the best moment to perform an accurate translation while minimizing the delay. To address this challenge, we utilize the cross-attention mechanism of the encoder-decoder architecture to find the right alignment between the input speech frames and the target text tokens. In this paper, we propose the Recent Frame Attention Policy (RFAP) and the Dual-Condition Attention Policy (DCAP) that allow offline trained speech-to-text translation models to be used in streaming scenarios without requiring additional training. Results on three different language translation pairs over the CVSS-C corpus show that the RFAP is able to surpass other policies with gains of up to 4.0 BLEU while reducing the translation delay by almost 1 second. Moreover, the DCAP is able to preserve a high translation quality when the latency is very low.
Simultaneous speech translation (SimulST) produces translations incrementally while processing partial speech input. Although large language models (LLMs) have shown strong capabilities in offline translation tasks, applying them to SimulST poses notable challenges. Existing LLM-based SimulST approaches either incur significant computational overhead due to repeated encoding of bidirectional speech encoder, or they depend on a fixed read/write policy, limiting the efficiency and performance. In this work, we introduce Efficient and Adaptive Simultaneous Speech Translation (EASiST) with fully unidirectional architecture, including both speech encoder and LLM. EASiST includes a multi-latency data curation strategy to generate semantically aligned SimulST training samples and redefines SimulST as an interleaved generation task with explicit read/write tokens. To facilitate adaptive inference, we incorporate a lightweight policy head that dynamically predicts read/write actions. Additionally, we employ a multi-stage training strategy to align speech-text modalities and optimize both translation and policy behavior. Experiments on both in-domain (MuST-C) and out-of-domain (Europarl-ST) En-De and En-Es datasets demonstrate that EASiST offers superior latency-quality trade-offs compared to several strong baselines.
Biao Fu, Donglei Yu, Minpeng Liao +5
School of Informatics, Xiamen University · Key Laboratory of Digital Protection and Intelligent Processing of Intangible Cultural Heritage of Fujian and Taiwan (Xiamen University), Ministry of Culture and Tourism · Alibaba Group Tongyi Lab
Simultaneous speech translation (SimulST) aims to translate ongoing speech with low latency while preserving translation quality. Recent studies have adopted large language models (LLMs) for SimulST and formulated it as a multi-turn dialogue task for efficient generation, achieving state-of-the-art performance. However, existing dialogue-style SimulST approaches typically rely on specialized training schemes to acquire simultaneous translation behavior, either using synthesized read/write trajectories, which may provide suboptimal training signals, or jointly optimizing translation quality and latency, whose competing objectives may cause optimization instability. In this paper, we introduce SimPOTrans, a reinforcement learning (RL) framework that enables offline speech translation models to self-adapt to dialogue-style SimulST. Through a simple two-stage RL post-training pipeline, SimPOTrans rewards an offline model for exhibiting translation behaviors beneficial to simultaneous scenarios when given partial speech inputs, thereby enabling offline-to-simultaneous self-adaptation without synthesized read/write trajectories or explicit latency optimization. At inference time, SimPOTrans requires no external decision policy and enables flexible quality--latency control by simply adjusting the chunk size of speech input. Experiments on the ACL 60/60 and MCIF benchmarks across three language directions (En→De, En→Zh, and En→Ja) demonstrate that SimPOTrans outperforms several strong baselines, achieving a superior quality--latency trade-off.