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Electrical Engineering and Systems Science

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Thursday 8 October

Audio and Speech Processing

Monaural Continuous-Radius Regional Speech Extraction with Cross-Radius Consistency Learning

Source-to-microphone distance enables speaker-independent speech extraction without prior enrollment. We propose a monaural continuous-radius regional speech extraction method that directly models the cumulative speech target within a queried radius. To realize continuous region control, the query radius is encoded as a continuous scalar and injected into a time-frequency extraction network, allowing a single model to operate over arbitrary radii within the trained range. Exploiting the nested structure of target-speaker sets across query radii, we introduce cross-radius consistency learning to stabilize predictions for adjacent radii sharing the same nonempty target-speaker set. Experiments on measured RIRs show that continuous-radius conditioning improves selective extraction over discrete conditioning. The proposed method achieves 29.25~dB SI-SDR, 4.40~dB SI-SDRi, 62.01~dB attenuation, and 0.46% RCE, outperforming fixed-threshold and local-range baselines. It also remains effective with more speakers and additive noise.
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All authors Biao Dong, Jie Chen, Jianwei Fang, Wei Xiao, Jiqing Han, Yongjun He

Audio and Speech Processing · Cross-listed from cs.SD

Beyond Speech Captions: Speech-Rewarded Style Planning for Conversational Text-to-Speech

Natural-language style descriptions provide an interpretable interface between large language models (LLMs) and controllable text-to-speech (TTS). However, using descriptions as pseudo-labels compresses target acoustics into text, and descriptive fidelity need not imply effective control of a particular synthesizer. We empirically show that speech-text alignment only weakly predicts downstream acoustic similarity among candidate instructions for the same utterance. We therefore propose Speech-Rewarded Style Planning (SRSP), which trains a text-based style planner through a frozen downstream TTS model. Given dialogue history and response text, the planner generates candidate instructions and is optimized with group-relative policy optimization (GRPO), using the teacher-forced likelihood of target speech tokens as the reward. On an English subset of the ISCSLP 2026 CoT-TTS corpus, SRSP achieves higher speech-style and emotion similarity to target speech and lower mel-cepstral distortion than the Base LLM and target-audio-informed captioning baselines. LLM-based expressive speech evaluation further shows gains over all baselines in contextual appropriateness and reference consistency.
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All authors Shiao Zhu, Lianbo Liu, Sizhen Lyu, Yuzhe Wang, Sheng Li, Takahiro Shinozaki

Audio and Speech Processing · Cross-listed from cs.SD

Edit Who Speaks, Control How They Speak: Global Timbre Editing and Local Instruction Control for TTS

Instruction-based text-to-speech (TTS) offers control over voice characteristics and speech expression through interfaces including voice cloning and text-based voice design. Voice cloning reproduces a reference voice, whereas text-based voice design creates a voice from a natural-language description. However, neither interface directly enables users to modify the timbre of a given reference and synthesize speech with the modified voice. Meanwhile, utterance-level expressive instructions leave changes across individual text segments underspecified. We introduce EDICT, a framework that unifies global timbre editing and local expressive control by using an edited acoustic reference to anchor voice identity across segments. To enable synthesis with an instruction-edited voice, EDICT combines reference audio with structured timbre edits to generate an edited reference in codec-token space. This representation serves as a shared voice anchor for a frozen TTS backbone, allowing segment-specific natural-language instructions to guide expression. To accommodate instruction changes while supporting acoustic continuity, EDICT rebuilds the KV cache at each segment boundary, refreshing instruction conditioning while retaining bounded acoustic context from previously generated speech. Evaluations on our proposed TimbreEdit-Bench and IntraTTS-Bench demonstrate improved timbre editing and a favorable balance between local instruction adherence, speaker consistency, and transition quality. Audio demos are available.
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All authors Junchuan Zhao, Chenglin Xu, Wei Zeng, Haoyang Li, Yiwen Guo, Ye Wang

Signal Processing

An Embodied Multiagent Framework Based on Token Communications for Cooperative ISAC

The emerging low-altitude economy demands unmanned aerial vehicle (UAV)-enabled integrated sensing and communication (ISAC) for reliable connectivity and environmental awareness. In particular, embodied UAV agents offer a promising means of supporting autonomous operations through a closed loop linking perception, decision-making, and physical actions. However, each UAV has access only to local observations, and effective cooperation requires exchanging local states and intentions. Directly sharing such information can incur substantial signaling overhead and hinder timely coordination in dynamic environments. To deal with this problem, this paper investigates a cooperative ISAC network of embodied UAV agents and formulates a joint token communications (TokCom) and physical control problem to minimize total propulsion energy subject to communication and sensing rate requirements. Then, we propose a state–intent TokCom (SI-TokCom) framework driven by multi-agent embodied policy learning. Specifically, separate pretrained codebooks enable compact exchanges of local states and intentions, while UAV agents jointly learn to select and compose tokens and determine physical actions based on local observations and received tokens. Simulation results show that SI-TokCom achieves 98.9% and 99.1% of the centralized baseline's communication and sensing rates, respectively. Compared with the local baseline, it improves the corresponding rates by 5.0% and 43.6%, respectively, with essentially unchanged propulsion energy. These results highlight the potential of TokCom for communication-efficient cooperation among embodied UAV agents in ISAC systems.
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All authors Jiahe Guo, Jun Du, Chunxiao Jiang, Jintao Wang

Systems and Control

Causal-fate dynamics of unrealized influence

Many dynamical systems generate influences whose consequences are not fully exhausted in the realized trajectory at the moment they arise. Such consequences are often treated as absent, delayed or statically stored, leaving unclear how unrealized influence retains future relevance as the system evolves. Here we formulate causal-fate dynamics, in which generated influence may be realized, remain latent, or be transformed by subsequent dynamics, and give an exact finite-transport representation when the relevant maps are specified. A connectome-constrained Caenorhabditis elegans model first motivates the biological hypothesis that unresolved inter-neuronal influence may persist and contribute to later propagation; it does not establish such a mechanism in living animals. We next examine operational Internet routing, where a dynamically updated cross-observer history retains predictive information beyond the current local route state. We then use the representation to construct a Transformer architecture that explicitly transports and selectively realizes latent contextual influence while retaining language-modeling function. The three studies distinguish a model-motivated scientific hypothesis, an observational phenomenon compatible with future-relevant history and an executable construction for carrying unrealized influence through subsequent computation.
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Systems and Control · Cross-listed from physics.app-ph

Time-Domain Analysis of Surface Acoustic Wave Magnetoelectric Sensors

The transient response of surface acoustic wave sensors is important for understanding their dynamic behavior, propagation delay, and response to time-varying signals. In this work, the time-domain propagation characteristics of a magnetoelectric SAW sensor based on a multilayer Love-wave delay line are investigated using complementary numerical and experimental approaches. Time-domain analysis is employed to determine the delay time and to extract the phase and group velocities, while frequency-domain analysis provides an independent characterization of the wave dispersion. Both numerically and experimentally, the phase and group velocities are found to differ, reflecting the dispersion of the Love wave in the multilayer structure. At the operating frequency of 146 MHz, the simulated group velocity obtained from wave-field tracking agrees within 0.4% with the value determined from the frequency-domain dispersion relation. The simulated delay time and group velocity also show good agreement with experiment, with deviations of 2.7% in delay time and 0.9% in group velocity. These results demonstrate the consistency of the time- and frequency-domain analyses and provide a quantitative characterization of the transient propagation behavior of SAW sensors.
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All authors Mohsen Samadi, Henrik Wolframm, Felix Weisheit, Dirk Meyners, Eckhard Quandt, Michael Höft, Martina Gerken

Signal Processing

Unrolled Time-Varying Graph Signal Restoration under Spatiotemporal Smoothness Priors

In this paper, we propose restoration methods for time-varying graph signals using deep algorithm unrolling (DAU). Time-varying graph signals, such as signals obtained from sensor networks, are nonuniformly distributed in space and observed as time series. Since these observed signals often contain noise and missing values, their restoration needs to consider both spatial and temporal relationships. Our approach is based on an optimization problem that models signal properties using a spatiotemporal regularizer that combines a Sobolev operator for spatial smoothness and a multi-tap FIR filter for temporal smoothness. We then unroll the iterative conjugate gradient method to solve this problem and learn the regularization parameters and filter coefficients in each iteration. Our method can be applied to supervised batch, supervised online, and unsupervised batch settings to learn these parameters. Experiments on several synthetic and real-world datasets show that the supervised batch method achieves the lowest RMSEs in almost all cases. The supervised online and the unsupervised batch methods outperform the existing methods of their own settings, and are often competitive with the existing batch methods.
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All authors Hayate Kojima, Hikari Noguchi, Koki Yamada, Yuichi Tanaka

Signal Processing

gr-ntnisac: An Open GNU Radio Testbed for NTN-ISAC

A passive receiver can sense nearby objects from the echoes of a 5G downlink. When the transmitter is a low-Earth-orbit (LEO) satellite, one transmitter covers a wide area, but the receiver must handle tens of kHz of Doppler and a direct-path delay that changes within every observation interval. Such a receiver should be tested where the correct answer is known before it is tried on a satellite. This paper presents gr-ntnisac, an open GNU Radio testbed that applies an emulated LEO channel to a standard 5G New Radio downlink, sends it through two cabled USRPs, and scores the receiver against the delay and Doppler commanded for every path. The same receiver code runs in simulation, on recorded captures, and on hardware. As a first use, we measure cancellation of the direct path, the strong satellite signal that masks weak echoes. Depth is 34.8 dB on hardware and 37.6 dB in a matched simulation without radios, so the radios are not the main limit. Controlled simulation shows that the emulator's 8-tap fractional-delay filter and a simulated rotating blade at almost the direct path's delay each limit depth to about 40 dB; without either, depth reaches 71.0 dB. A diagnostic canceller that also removes the per-subcarrier response and per-symbol phase reaches 72.7 dB on the same captures. On a conducted capture, a payload-aided receiver detects a -25 dB moving path in 76.9 percent of dwells.
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Audio and Speech Processing

Toward accurate speaker inventories for online speaker diarization

An online speaker diarizer maintains a growing inventory of speakers, and applications read it directly: a live transcript enumerates them as they appear. Existing systems either fix the inventory's size in advance, so speakers beyond that size are merged, or grow it on distance alone, so noise and turn boundaries split speakers. Scored on speaker count over nine public datasets, both failures appear where DER reports neither: on VoxSRC-23 the fixed-capacity systems return 44-46% too few speakers and the unbounded tracker with the lowest DER 63% too many. We build on that embedding-based tracker and change only what triggers a registration. A candidate speaker pool keeps unregistered embeddings apart and registers a speaker only once one candidate has gathered enough mutually compatible embeddings; commit-gated label assignment holds an output while its candidate is undecided, so no speaker appears in the output before it is confirmed. Across the nine datasets the macro speaker-count error falls from 225% to 21% of the reference and macro DER from 17.59% to 15.95%, at a mean added latency of 0.109s over all outputs; on VoxSRC-23 the tracker obtains both the lowest DER and the most accurate speaker count of the systems compared.
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All authors Youngki Kwon, Hee-Soo Heo, Minjae Lee, Bong-Jin Lee

Audio and Speech Processing

Using the enrollment gallery as evidence: affine-invariant score calibration for target speaker tagging

Target speaker tagging (TST) assigns enrolled speaker identities to diarized segments of multi-speaker recordings. The enrollment utterances of the other registered speakers are an attractive source of evidence for calibrating each verification decision: they share the deployment domain of the test material and require no external cohort set. We show, however, that they cannot serve as the cohort of conventional score normalization. Enrollments tend to cluster by recording session, so the per-speaker cohort statistics reflect enrollment proximity to the rest of the gallery rather than impostor behavior, and normalization then rejects entire speakers. We propose gallery affinity verification, a cohort-free calibration that exploits the gallery through two affine-invariant terms, score dispersion and enrollment-profile deviance, and is therefore immune to such per-speaker shifts. On a synthetic benchmark and an in-house meeting corpus, it improves tagging accuracy with and without conventional score normalization and adds further gains when combined with it.
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All authors Hee-Soo Heo, Minjae Lee, Youngki Kwon, Bong-Jin Lee

Systems and Control

Reference-Filter-Driven Transition Probabilities for IMM-Based Satellite Maneuver Detection

Maneuver detection of non-cooperative satellites is an essential task of space situational awareness. The interacting multiple model (IMM) filter has been applied to detect unannounced maneuvers, but the standard IMM assumes a fixed Markov transition probability matrix (TPM) that must be tuned without knowledge of the maneuver rate of the target. To avoid this tuning, adaptive-TPM approaches update the TPM online from statistics computed inside the IMM. However, these statistics already depend on the TPM through the mixing step, and this dependence forms a feedback loop. Under the dynamics mismatch of orbit tracking, the loop can keep the maneuver probability high in the absence of a maneuver or suppress its rise when a maneuver occurs. In this paper, the TPM is driven by the normalized innovation squared (NIS) of a reference coast filter that is not mixed with the IMM, and the NIS is mapped continuously to the coast-to-maneuver transition probability. The reference-driven IMM (RD-IMM) is evaluated with four space-based optical sensors tracking a geosynchronous target. RD-IMM detects small burns that the fixed-TPM IMM fails to detect, with few false declarations in maneuver-free runs, and avoids both failures of the closed-loop adaptation. Moreover, RD-IMM reduces the position error after a small in-track burn by more than 50% compared with the single-filter approaches.
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Image and Video Processing · Cross-listed from cond-mat.mtrl-sci

Localization of Candidate Kikuchi Regions in RHEED Images: Visibility and Annotation Boundaries

Kikuchi lines and bands in reflection high-energy electron diffraction (RHEED) carry information on crystal geometry and electron scattering, but are often obscured by intense diffraction streaks. We combine multiscale convolution with spatial detail from skip connections and independent supervision that allows overlapping regions to localize streaks and candidate Kikuchi regions separately. Using manual polygon annotations made before model-assisted editing, the Kikuchi intersection-over-union (IoU) on 29 laboratory test images from separate growth batches was $0.6290 \pm 0.0137$. We then performed supervised adaptation to public chalcogenide images and stratified images by Kikuchi visibility using the median ratings of three observers who rated the images separately, two of whom rated with predictions hidden. With all 244 training images, IoU for the clear-feature group of 10 test images was $0.5014 \pm 0.0336$, whereas ambiguous images gave lower values. With 100 training images, clear-group IoU was $0.5171 \pm 0.0073$; expanding the training set also reduced responses on images without target features. Reported uncertainties are sample standard deviations across four laboratory runs or three adaptation runs. Public-image adaptation used reference masks of mixed provenance, including prediction-derived drafts. The method converts visual cues into inspectable spatial regions, providing a basis for analysis of line positions, intersections, and local intensity.
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All authors Lumou Weng, Hanshan Huang, Zhenhan Zhang, Gan Wang

Audio and Speech Processing

Randomized Scores and Diverse Timbres: Augmenting Automatic Music Transcription with Online-Generated Data

Automatic music transcription (AMT) is limited by the scarcity of audio recordings paired with precise symbolic annotations. Synthetic data can provide supervision at scale, but it remains unclear whether effective transfer depends on realistic score structure or broad timbral coverage. We study these factors separately through an online sampler–renderer pipeline. A unified corruption sampler ranges from unmodified MIDI clips through partial corruption to deeply randomized note-event distributions. The renderer converts these events to audio while independently controlling instrument and timbral coverage. A fixed transcription model is trained jointly on offline recordings and newly rendered examples. Controlled ablations reveal an asymmetry between the two factors: moderate corruption of the note-event distribution does not impair transfer and can improve it, whereas broader renderer-side timbral support consistently improves out-of-domain generalization under a fixed note-event distribution. Finally, online-rendered examples complement real and existing synthetic data in a strong combined-data regime. These results suggest that synthetic AMT data should prioritize coverage of note-level attributes and their timbral realizations over realistic joint score structure.
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Signal Processing

Integrity and Credibility in Navigation: From Error Characterization to Operational Assuran

Navigation integrity and estimator credibility use error and uncertainty information to address different questions. This perspective distinguishes actual positioning error from reported statistical characterizations, formulates integrity as control of hazardous unwarned-use risk under the requirement of a specified operation, and describes credibility as the justification of an uncertainty or risk claim by its evidence usage and modeling process. Three hypothetical scenarios show how deteriorated accuracy, a contradicted uncertainty report, and a hazardous event can lead to different credibility and integrity judgments. The resulting assessment asks what happened, why the reported claim should be believed, and whether the information is sufficient for the intended operation.
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All authors Penggao Yan, Baoshan Song, Yuan Li, Ronghe Jin, Li-Ta Hsu

Signal Processing

2D Coprime Pilots for Delay-Doppler Sensing in OFDM-ISAC Systems

Integrated Sensing and Communication (ISAC) is envisioned to endow future 6G systems with seamless sensing capabilities. To support efficient sensing with minimum communication overhead, sparse pilots embedded within communication frames have emerged as a promising solution. Along this line of research, existing studies have achieved engaging results in maximizing the unambiguous sensing region. However, jointly maximizing the sensing region and sensing accuracy remains challenging due to the lack of a unified performance metric and an effective pilot design framework. This paper jointly optimizes the unambiguous sensing region and sensing accuracy for estimating delay-Doppler (DD) parameters in Orthogonal Frequency Division Multiplexing (OFDM)-ISAC systems, where sensing mutual information (SMI) is adopted as a unified performance metric to characterize the overall sensing capability. Specifically, the joint optimization is formulated as an SMI maximization problem by systematically resolving sensing ambiguity and optimizing sensing accuracy. In particular, based on the generalized Bezout identity, we derive a 2D (timefrequency) coprime condition, which, as far as the authors know, is the first necessary and sufficient condition to achieve the unique estimation of DD parameters in the literature. Under this unambiguous condition, we further propose an Adam-Guided Iterative Refinement (AGIR) algorithm to optimize the sensing accuracy. Numerical results demonstrate the advantage of the proposed framework over existing designs, owing to the freedom offered by the 2D coprime condition in optimizing the sensing accuracy.
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All authors Guanping Shang, Zeyan Zhuang, Anzheng Tang, Shenghui Song, Chi-Ying Tsui

Audio and Speech Processing · Cross-listed from cs.SD

MiDashengLM-Spatial: Unifying General Audio Understanding and Spatial Awareness

Large audio-language models (LALMs) have achieved strong performance in general audio understanding, yet most are designed for monaural input and discard the inter-channel cues essential for spatial perception. In contrast, existing spatial audio-language models are purpose-built for spatial tasks and fail to capitalize on the general understanding capabilities of monaural LALMs. We present MiDashengLM-Spatial, the first open-source end-to-end unified audio-language model, to our knowledge, which supports both general audio understanding and spatial awareness within a single architecture. It extends MiDashengLM with a spatial audio encoder, Spatial-Dasheng, integrated through a hierarchical semantic-to-spatial conditioning module that injects intermediate semantic representations into the spatial branch at multiple depths while preserving the original semantic pathway. To provide spatial audio-language supervision at scale, we develop a data synthesis pipeline that renders diverse spatial acoustic scenes with scene-level spatial descriptions and question-answer pairs. Experiments show that Spatial-Dasheng achieves strong performance on sound event localization and detection in real-world scenes, and that MiDashengLM-Spatial substantially outperforms existing LALMs on spatial understanding and reasoning benchmarks. Meanwhile, it remains competitive with state-of-the-art 8B-scale LALMs on diverse monaural benchmarks, demonstrating that spatial awareness can be acquired without compromising general audio understanding. The source code and model checkpoint are available at https://github.com/xiaomi-research/midashenglm-spatial and https://huggingface.co/mispeech/midashenglm-spatial.
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All authors Jinbo Hu, Hang Su, Lichun Fan, Heinrich Dinkel, Gang Li, Zhanchen Dai, Yiru Zhang, Chang Liu, Peng Wang, Junnan Wu, Jian Luan, Cong Zou, Heng Qu

Audio and Speech Processing

SmoothConv and DuplexConv: Complementary Mandarin Multi-Party Conversational Speech Corpora for Speech Interaction

Recent advances in large audio language models (LALMs) have driven the development of natural and intelligent speech interaction systems. Such systems need to model complex conversational behaviors, including turn-taking, overlapping speech, and speaker coordination. Multi-party conversations provide a realistic setting for studying these behaviors, yet existing Mandarin conversational corpora often lack synchronized participant-level speech tracks and comprehensive annotations. In this work, we introduce SmoothConv and DuplexConv, two complementary Mandarin multi-party conversational speech corpora totaling 2,100 hours. SmoothConv provides human-verified conversations for reliable analysis and evaluation, while DuplexConv offers large-scale automatically annotated conversations through a scalable pipeline for model training. Both corpora provide synchronized participant-level speech tracks and multi-dimensional fine-grained annotations. We further release the SmoothConv Benchmark and evaluate these resources on speech separation, multi-speaker automatic speech recognition (MSASR), and turn detection tasks. Experimental results demonstrate the utility of the proposed resources for multi-party speech interaction modeling. The datasets, benchmark, and related resources are publicly available.
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All authors Chengyou Wang, Mingchen Shao, Chunjiang He, Zeyu Zhu, Jierui Guo, Bingshen Mu, Zikai Liu, Hanke Xie, Yuhang Dai, Zhou Zhu, Lei Xie

Image and Video Processing · Cross-listed from cs.CV

A Unified Score Matching Paradigm for Video Anomaly Detection and Anticipation

Video anomaly detection (VAD) is a fundamental and safety-critical task in computer vision. Recent generative approaches detect anomalies from a distributional perspective, but remain limited by local anomaly modes. Meanwhile, video anomaly anticipation (VAA), as a proactive extension beyond post-hoc detection, introduces additional challenges. In particular, the contrastive inference paradigm in VAD, which relies on ground-truth frames, is not applicable to VAA, hindering its development. To address these challenges, we propose a unified score-driven framework, termed Uni-DSM, based on denoising score matching (DSM), which models anomaly patterns through likelihood estimation and score functions over the learned data distribution. Within this unified framework, we adopt a shared noise-conditioned score transformer backbone with scene-dependent embeddings and motion-aware weighting for distribution-level modeling. Instead of introducing separate architectures, Uni-DSM unifies VAD and VAA through different inference and supervision paradigms built upon the same score-based formulation. For VAD, we instantiate an autoregressive denoising score matching (ADSM) mechanism, which progressively accumulates anomalous evidence via autoregressive denoising, enabling enhanced perception of local modes beyond visual cues. For VAA, we extend the same architecture by incorporating a lightweight auxiliary decoder and a novel self-distilled denoising score matching (SDSM) mechanism. By constructing supervision from output discrepancies instead of relying on unavailable future ground truth, our method achieves efficient training suitable or early anomaly anticipation. Extensive experiments on multiple benchmark datasets demonstrate state-of-the-art performance in both VAD and VAA while maintaining high efficiency, establishing a unified and scalable pipeline from anomaly detection to anticipation.
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All authors Congqi Cao, Zhenhe Liang, Hanwen Zhang, Yifan Zhao, Qinyi Lv, Lingtong Min, Yanning Zhang

Image and Video Processing · Cross-listed from cs.CV

Improving Image-Based Nutrition Estimation Through Multimodal Food-Item Verification and Recovery

Single-image nutrition estimation can fail silently when visible foods are missed. Even when a food is correctly identified, its proposed region may not support portion estimation. We propose a framework that uses multimodal large language models (MLLMs) to inventory visible foods and separately verify food identity and whether each proposed 2D region supports portion estimation. One whole-image review uses these verification results to identify unresolved gaps and omitted foods, triggering at most one targeted recovery pass. Recovered regions are re-verified without access to the recovery prompt, then reconciled into a final item set for nutrition estimation. The framework requires no task-specific fine-tuning. Matched evaluation on common valid-output samples shows that item-level grounding improves mass accuracy across all tested settings and energy accuracy relative to an adapted retrieval baseline, with item-identity precision and recall also improving, while post-recovery visual coverage is assessed separately at inference time without ground-truth annotations.
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All authors Jingbo Yue, Bruce Coburn, Jinge Ma, Jui-Feng Chi, Fengqing Zhu

Signal Processing

SEER: Source-Conditioned Emotion Enhancement via Retrieval for Cochlear-Implant Speech

Cochlear implants (CIs) restore speech access but weaken cues needed for vocal emotion recognition. Prior CI-oriented enhancement requires parallel normal/strong recordings and intensity labels. We propose SEER, a retrieval-based framework that learns which same-emotion reference helps each source remain recognizable after CI processing. A source-conditioned retriever learns CI-aware utility from sampled emotional voice conversion outcomes, while uncertainty-guided exploration avoids exhaustive pair evaluation; neither parallel recordings nor intensity labels are required. SEER improves Source macro-F1 at N8 by 7.30 points on RAVDESS and 11.66 points on ESD, with significant ESD gains across N4/N8/N16. Sixteen-listener RAVDESS gains are significant across all conditions. Exhaustive analysis finds an aggregate benefit from stronger references but little effect from matching gender or content.
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All authors Hsing-Hang Chou, Yun-Shao Lin, Ching-Chin Sung, Chi-Chun Lee

Systems and Control

A User-Triggered UAV Dispatching System for Precise and Timely Mountain Search Missions

Smartphones carried by hikers routinely record routes, locations, and requests for assistance, and these records can provide precise clues for mountain search. However, they are rarely transformed in a direct and timely way into unmanned aerial vehicle (UAV) missions that support search and rescue. This paper presents a user-triggered UAV dispatching system that integrates mobile information collection, ground station mission management, and UAV search into a single workflow. In our system, a mobile application is designed to continuously track user records and synchronize these records to the cloud database, thereby initiating all subsequent search events and UAV-dispatch missions. A control platform is developed to retrieve records from the cloud database for operator verification and mission scheduling. A UAV prototype is implemented to execute the approved mission, follow waypoints, report status, relay phone records when available, and record downward-facing video for later inspection. Simulations are conducted for UAV-dispatching missions using a terrain-following flight altitude of $80$~m above ground level and demonstrate successful route generation, ordered waypoint traversal, and return to the launch site for landing of the UAV dispatching. Local software tests verified timeout detection with a 1s threshold and phone-record transfer from the mobile application through the UAV to the ground station. Outdoor trials showed that the UAV could fly autonomously and capture downward-facing imagery in which a person, a nearby footpath, and surrounding vegetation were clearly discernible, providing visual cues for operator assessment and further inspection of the search area.
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All authors Mingyang Wang, Yi Hong, Tungchak Lee, Ashwin Sundar, Kevin Hung, Qubeijian Wang, Yalin Liu

Audio and Speech Processing

Towards Automated Clinical Behavioral Coding with Large Language Models: A Case study Using BOSCC recordings of Children

Autism spectrum disorder (ASD) is a neurodevelopmental condition characterized by differences in social communication and by restricted interests and repetitive behaviors. Treatment interventions often target social-communication skills, creating a need for reliable measures of behavioral change. The Brief Observation of Social Communication Change (BOSCC) is a validated treatment-response measure based on brief play and social-communication interactions between a child and trained examiner. The BOSCC coding process is resource-intensive and requires trained experts, motivating the automation of coding in order to improve scalability and accessibility. In this work, we evaluate general-purpose large language models (LLMs) for predicting speech-related BOSCC codes from different input representations. We compare transcript, diarized-transcript, and targeted audio conditions across 163 in-house recordings. LLMs are able to perform well in assessing verbal exchange, but do not perform as well when identifying atypical speech patterns. Additionally, performance varies considerably across scoring decisions, with no consistent pattern across diagnosis groups. An audit of model predictions indicates that applying the BOSCC coding criteria and interpreting ambiguous speech evidence remain challenges.
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All authors Jordan Prescott, Aditya Kommineni, Tiantian Feng, Megan Micheletti, Alyssa Viggiano, Luis Angeles, Anfeng Xu, Lynn Perry, Catherine Lord, Daniel Messinger, Shrikanth Narayanan

Image and Video Processing · Cross-listed from cs.CV

Skeleton-Guided Progressive Test-Time Adaptation for Thin Curvilinear Structures

Accurate segmentation of thin curvilinear structures is vital for various real-world applications, from vessel analysis to road extraction. Yet their intricate geometry makes even minor pixel-wise errors enough to break the global topology, and this structural fragility turns severe domain shifts into catastrophic failures. The difficulty is most acute under cross-modality gaps, where the imaging process itself differs fundamentally between source and target. While test-time adaptation (TTA) offers a practical source-free remedy, existing methods adapt feature statistics and confidence, neither of which constrains connectivity, and thus degrade under such extreme gaps. To address this, we propose Skeleton-Guided Progressive Test-Time Adaptation (SGP-TTA). Progressive Batch Normalization (ProgBN) shifts normalization from frozen source statistics toward current target estimates under a sample-count schedule, so that the source-target balance follows the stage of adaptation rather than a fixed coefficient. Consensus Skeleton Recall (CSR) then derives a structural target from geometrically aligned multi-view predictions and updates only the BN affine parameters to preserve connected structures. Extensive experiments show that SGP-TTA consistently outperforms existing TTA methods in topological connectivity, with the largest margins under cross-modality shift. The project page is available at https://boa-jang.github.io/SGP-TTA.
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All authors Boa Jang, JunGyu Lee, Gwanho Lee, Jinwook Choi, Young-Gon Kim

Audio and Speech Processing · Cross-listed from cs.SD

Cross-Lingual Speaker Verification with Self-Supervised Pre-Trained Models

Speaker verification (SV) performance degrades under language mismatch due to the entanglement of speaker identity with language-specific acoustic cues. To address this problem, we leverage large-scale self-supervised pre-trained models (PTMs) to learn language-agnostic speaker representations. We utilize PTMs as robust front-end feature extractors, capitalizing on their rich acoustic and linguistic knowledge acquired from vast, diverse audio data. These generalized features are then used to train a downstream speaker embedding network, effectively disentangling speaker identity from language-specific characteristics. We validate our approach on the TidyVoice2026 benchmark, which benchmarks SV under language mismatch. Our proposed system (team T02) achieves equal error rates (EERs) of 2.21% on tv26_eval-A and 2.99% on tv26_eval-U.
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All authors Jinghan Peng, Yu Zheng, Weiqiang Wang, Jian Liu

Systems and Control

A Latent Space Optimization Approach for Symbolic Discovery of Dynamical Models

In this paper, we propose a hybrid latent-space Bayesian Optimization (BO) and global optimization framework for solving symbolic regression tasks to discover dynamical models from data. Symbolic regression can discover equations from data without fixing the functional form of the expression a priori. The proposed framework uses the fact that if the functional form of the expression is fixed, the symbolic regression task reduces to a parameter estimation problem, which must be solved to global optimality. Motivated by this structure, first, we train a VAE to map the discrete space of expression trees into a continuous latent space. Then, we use BO to search this latent space while assessing a candidate expression's prediction error by solving the parameter estimation problem to global optimality. We evaluate our framework across two case studies: static reaction rate law discovery and dynamic concentration identification in a continuous stirred-tank reactor (CSTR). The results show that our framework identifies the true governing equations faster than MINLP formulations without solver timeouts and achieves higher sample efficiency than evolutionary baselines under tight evaluation budgets.
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Image and Video Processing

Adapting Appearance-Based Gaze Estimation to Narrow-Range, Long-Duration Screen Viewing

Appearance-based gaze estimation offers a low-cost alternative to infrared eye tracking for screen-based behavioral and clinical applications, however existing models are typically developed for wide ranges of gaze angle and head pose. Prolonged screen viewing presents a distinct regime in which gaze remains near the screen center, head motion is limited, and calibration drift accumulates over time. In this work, we benchmark six published estimators along with a proposed differential-gaze model on 140 long-duration facial video recordings with synchronized eye tracking under subject-disjoint evaluation and a common budget of calibration frames. Differential estimation was the only static method that significantly improved upon a baseline predictor of each recording's mean gaze, and was further boosted by addition of temporal context. It also yielded the strongest agreement with reference fixations and saccades, demonstrating that low angular error alone could not validate eye movement reconstruction. These findings establish differential estimation as a promising foundation for reliable gaze tracking in long-duration, narrow-range settings.
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Wednesday 7 October

Systems and Control

Centrality-based Structural-Electrical Contingency Screening and Transmission Reinforcement Prioritization

Rapid contingency screening is essential for secure transmission system operation and planning. Although full AC contingency analysis provides accurate system assessment, it requires high computational effort for large contingency sets. Topology-based screening methods reduce this burden but may not accurately represent operating-point-dependent electrical stress. In addition, conventional single-contingency screening does not quantify interactions among simultaneous outages or provide a direct criterion for reinforcement prioritization. This paper presents a hybrid structural-electrical screening approach that combines a computationally efficient, dispatch-independent structural analysis based on reactance-weighted centrality measures with a flow-based severity assessment using line outage distribution factors. Two indices are evaluated using the flow-based severity: a synergy index to quantify super-additive interactions among simultaneous line outages and a tolerance index to rank reinforcement candidates based on their reduction in aggregate post-contingency severity. The correspondence between structural indicators and electrical severity is evaluated quantitatively. Tests on the IEEE 39-bus system identify 115 super-additive line pairs beyond a 5% threshold, with the most severe interaction exceeding seven times the linear combination of individual outage effects. The highest-ranked reinforcement reduces the mean post-contingency thermal severity by 7.5%. The structural rankings show limited correlation with flow-based severity, with a Spearman coefficient of approximately 0.2. The proposed approach provides an efficient pre-screening method to reduce contingency and reinforcement search spaces before detailed AC and economic evaluations.
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All authors Abanish Tiwari, Chandan Chaudhary, Mekh Raj Joshi, Mohammed Ben-Idris, Joydeep Mitra

Signal Processing

Flexible Analog Self-Interference Cancellation for Waveform-Robust Monostatic ISAC

Monostatic integrated sensing and communication (ISAC) requires in-band full-duplex operation, where self-interference cancellation (SIC) must suppress the strong transmit leakage while preserving weak sensing echoes. In the analog domain, this task is complicated by transmitter impairments, which generate conjugate and nonlinear SI components, and by waveforms with widely different statistics. This paper develops a flexible analog self-interference cancellation (FASIC) framework for monostatic ISAC based on the widely linear normalized least-mean-square (WL-NLMS) algorithm. FASIC generates the cancellation signal in digital baseband, subtracts it in the radio-frequency domain, and enables flexible sensing-range control via a configurable cancellation span. Its normalized adaptation remains stable across waveforms with substantially different peak-to-average power ratios (PAPRs), for which fixed-step adaptation can become unstable. We further relate transmit-waveform statistics to analog SIC performance and predict the waveform-dependent cancellation floor. Simulations validate the analysis and demonstrate stable analog SIC of approximately $30$–$34$ dB, with the low-PAPR waveforms achieving a $3.3$ dB deeper floor owing to their smaller high-order amplitude moments. Near-field static reflections within the cancellation span are suppressed while delayed and Doppler-shifted echoes are preserved. These results establish FASIC as a waveform-robust analog SIC solution and identify waveform statistics as a key consideration for sixth-generation (6G) waveform selection.
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All authors Anh Tuyen Le, Xiaojing Huang, J. Andrew Zhang, Peiyuan Qin, Le Chung Tran, Nhan Thanh Nguyen, Y. Jay Guo

Systems and Control

Constructive Safety-Critical Control for a Class of Underactuated Systems: A Hierarchical Approach

Underactuated systems with nontrivial geometry are abundant in practical robotic problems. However, little is known about constructive safety-critical control design in the presence of underactuation. In this work, we provide a constructive framework for synthesizing safe control architectures and control barrier functions for a class of quadrotor-like underactuated systems. By studying the geometry of their underactuation, we characterize when these systems have natural layered architectures useful for safety-critical control design. Using these architectures, we devise a constructive, hierarchical CBF synthesis procedure and provide conditions under which it succeeds.
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Systems and Control

EnergyNet in Practice: Long Grid, Short Grids and Mobility-Based Energy Peering

Electrification is planned as a network problem: more demand, more upstream capacity, local resources waiting for reinforcement. We argue that a material part of that bill, in money and time, follows from architecture, not electrification itself, and propose the Short Grid as the deployment unit: a locally controlled building, block or campus whose demand, generation, storage and local market operate behind Energy Routers enforcing an approved connection envelope to the Long Grid. Because the envelope governs exchange, not installed capacity, local resources can grow and many sites can build in parallel at the pace of PV and EV installation, not grid reinforcement. PV and EV batteries dominate today, but the architecture is general: any generation or storage, fixed or mobile, fits behind the same boundary and interfaces. EnergyNet Operators coordinate exchange among Short Grids and with the Long Grid; protection stays local; Vehicle-to-EnergyNet (V2EN) and Mobility-Based Energy Peering (MEP) make mobility-financed batteries accessible within and between them; priority-based operation protects critical services in shortage. We test it in models, not yet the field: a ten-building neighbourhood in Lund, Sweden; a bus-linked microgrid pair at Stanford and Half Moon Bay; and a proposed 20-site network in Rockaway, New York. Households and services are the largest demand category in Sweden, Denmark, Norway, California and New York (484 TWh/yr); halving their grid imports would be about 242 TWh. Europe's network investment to 2040 is estimated above EUR 1.2 trillion; modelling shows it depends on system design. Negagrid value, following the negawatt, counts network investment avoided, downsized or deferred, net of local substitute costs. "Don't pay twice": build coordinated local capability first; reinforce the Long Grid for what remains.
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All authors Jonas Birgersson, Max Collins, Marc A. Weiss, Jimmy Chen, Daniel Kammen, Tomas Kåberger, Mark Z. Jacobson, Newsha K. Ajami, Franklin Carrero-Martínez, Michael Menser, Henrik Ny, Lou Riordan, Jill G. Ferguson