Frontend-Assisted Large Audio-Language Models for SLT2026 SmartGlasses Challenge
Authors: Yushi Wang, Zeyan Song, Xiaobin Rong, Chengwei Liu, Jun Gao, Jing Lu
Organizations: Key Laboratory of Modern Acoustics, Nanjing University, Nanjing, China · NJU-Horizon Intelligent Audio Lab, Horizon Robotics, Beijing, China · Qwen Business Unit of Alibaba, Hangzhou, China
This paper describes our systems for the Track 1 time-stamped speaker-attributed ASR task of the SmartGlasses Challenge. To handle overlapping speech, background noise and far-field conditions in two-person conversations, we employ SoulX-Transcriber, a multimodal large language model that takes full-length audio as input and generates structured transcriptions. In addition, we investigate the effect of speech frontend processing, including a rule-based Guided Source Separation method and a data-driven speech separation model, both of which improve performance on the validation set. Considering inference stability, our final submission uses the fine-tuned SoulX-Transcriber system, achieving a tcpCER of 6.22 on the test set.
Figures & tables
Fig. 1: Overview of our systems, where the three systems are developed and evaluated independently.
System
tcpCER
cpCER
DER
Baseline VibeVoice-ASR
14.09
12.88
17.24
System 1 SoulX-pretrained
9.02
8.45
3.70
System 1 SoulX-finetuned
7.82
7.30
2.56
System 2 GSS-SoulX
w/ oracle VAD
7.39
6.65
1.14
w/ estimated VAD
6.97
6.12
2.62
TABLE I: Experimental results on the Track 1 TSA-ASR task official validation set.
Fig. 2: Chat-level average tcpCER across different speech overlap ratio levels (Low/Mid/High) for different systems on the Track 1 TSA-ASR task official validation set.
Recent advances in large language models (LLMs) and multimodal LLMs (MLLMs) have created new opportunities for wearable speech interfaces, with smart glasses providing an egocentric platform for continuous audio sensing and assistance. However, speech recognition and understanding in this setting remain challenging because of dynamic acoustic conditions, speaker overlap, and the spatial ambiguity introduced by wearer-centered recording geometry. To support systematic evaluation in this setting, we introduce the IEEE SLT 2026 SmartGlasses Challenge for egocentric multi-speaker speech processing. The challenge consists of two tracks, Dyadic Dialogue Understanding and Multi-party Meeting Understanding, and jointly evaluates Time-Stamped Speaker-Attributed Automatic Speech Recognition (TSA-ASR) and Spoken Language Understanding (SLU). It is built on a 106-hour four-channel egocentric speech dataset containing 714 sessions collected in real-world scenarios. This paper describes challenge tasks, dataset construction, submissions, and summarizes the main findings from the shared evaluation. The results show that heavy speaker overlap remains a major factor affecting TSA-ASR performance, while paralinguistic acoustic understanding continues to be difficult for current audio-language models in complex SLU settings. Further details can be found on the official challenge website.
Dehui Gao, Zhixian Zhao, Zhennan Lin +14
Huawei · AIShell Inc · Shanghai Jiao Tong University +4
Target speaker extraction (TSE) is a key frontend for enabling audio large language models (LLMs) to recognize and understand a designated speaker in overlapping speech. However, multilingual mixtures exhibit heterogeneous extraction demands, causing fixed-depth systems to either under-process challenging inputs or waste computation on inputs that have already converged. Existing TSE frontends generally apply uniform computation, while Audio-LLM adaptation methods neither explicitly model language-conditioned extraction demands nor preserve plug-and-play compatibility with frozen downstream models. We therefore propose LATTE, a lightweight 5.2M-parameter frontend that combines an acoustic Difficulty Estimator, a task-adaptive phonology-informed prior learned from multi-depth TSE gains, and an Adaptive Hourglass Extractor to route each mixture without requiring language identity at inference. On PolyExtract, LATTE achieves a macro SI-SNRi of 15.07 dB at 0.30 GFLOPs, retaining performance within 0.09 dB of a 0.53-GFLOP fixed-depth model while reducing computation by 43.4%, and lowers the average WER from 16.8% to 14.6% relative to TF-GridNet. By jointly modeling sample-level extraction demands and phonological structure, LATTE provides a practical framework for quality-aware, compute-efficient multilingual TSE and offers a scalable path toward robust Audio-LLM deployment.
Yanxiang Chen, Haoyuan Deng, Jinyu Wang +4
Sun Yat-Sen University, Zhuhai, China · Beijing Normal-Hong Kong Baptist University, Zhuhai, China · Fudan University, Shanghai, China
We study timestamped speaker-attributed automatic speech recognition (SA-ASR) for long-form, multi-party speech with overlap. In this setting, chunk-wise inference must preserve meeting-level speaker identity consistency while producing time-stamped, speaker-labeled transcripts. Prior Speech-LLM systems tend to prioritize either local diarization or global labeling, lacking the ability to jointly model fine-grained temporal boundaries and robust cross-chunk identity linking. We propose G-STAR, an end-to-end framework that couples a cache-conditioned speaker-tracking module with a Speech-LLM transcription backbone. The tracker provides structured speaker cues with temporal grounding, and the LLM generates attributed text conditioned on these cues. G-STAR supports component-wise optimization and joint end-to-end training, enabling flexible learning under heterogeneous supervision and domain shift. Under chunk-wise decoding protocols, experiments on both oracle-segmented local evaluation and full-meeting global evaluation show strong speaker-attributed transcription performance.
Jing Peng, Ziyi Chen, Haoyu Li +7
Shanghai Jiao Tong University · Shenzhen Research Institute of Big Data · Nanjing University +2