Papers for

voice interface developers

Papers whose findings have a practical use for this group, as judged from the abstract. Open a paper to read what it means in practice.

Voice interfaces tested for reliability with computer-use agents

Talk2Agent: Benchmarking Voice Interfaces for Text Agents

Abstract: Large language model (LLM) computer-use agents are typically evaluated with clean written instructions, despite speech being an increasingly popular interface for interacting with such systems. Speech input introduces an additional failure point: transcription errors can alter task-critical entities, constraints, or targets before the agent begins reasoning, while conventional ASR metrics do not directly measure whether the information required for successful execution has been preserved. We introduce Talk2Agent, a benchmark for evaluating how effectively voice interfaces convey human-spoken instructions to LLM-based computer-use agents. Talk2Agent builds human-spoken versions of tasks from WildClawBench and OSWorld and evaluates a range of voice interfaces, including dedicated ASR models, audio-capable LLMs, contextual biasing, and LLM-based ontology repair. Because repeatedly executing long-horizon computer-use tasks is costly and stochastic, we further propose an execution-free, task-conditioned evaluation framework that projects the original task grader onto prompt-addressable intentions and measures how much task-relevant information is retained after the voice interface. On WildClawBench, Talk2Agent's execution-free native projection provides a practical, execution-grounded measure of voice-interface quality, correlating with downstream task completion and improving Pearson correlation by 0.246 over WER/CER on 32 hours of real human speech.

Wed 30 SeptArtificial Intelligence
The gist
Talking to computer programs using your voice is becoming more common, but speech recognition can mess up important details when converting spoken words into text. The authors created Talk2Agent, a way to check how well voice systems pass on spoken instructions to computer programs that use large language models. Instead of always running tasks to see if they work, their method evaluates whether the key information from speech was kept correctly, saving time and effort. Their approach better predicts how well the tasks will be done compared to traditional speech error metrics.
Open → 2609.38867v1

Calibrated order-aware method improves open-vocabulary keyword spotting accuracy

CORD-KWS: Calibrated, Order-Aware Detection for Open-Vocabulary Keyword Spotting

Abstract: Open-vocabulary keyword spotting (KWS) must detect arbitrary keywords without retraining. Cross-attention-based models achieve state-of-the-art performance but require pairwise interaction between the audio and each keyword, recomputing the fused representation for every audio--keyword pair. Embedding-based models avoid this computational cost through independent encoding and similarity scoring, yet remain inferior on challenging benchmarks such as LibriPhrase-hard. We show that this gap is primarily a training issue rather than a limitation of model capacity. While contrastive learning encourages correct keywords to rank above negatives, it does not explicitly calibrate absolute similarity scores, which are critical for applying a fixed threshold across keyword detection. We propose Calibrated, Order-aware Detection KWS (CORD-KWS), a training framework that introduces two complementary objectives while retaining the same encoders, cosine scoring, and O(1) keyword enrollment: a calibrated detection head that supervises absolute scores, and a frame-level CTC objective that preserves phonetic order information. CORD-KWS achieves EERs of 0.43% and 9.64% on LibriPhrase-easy and LibriPhrase-hard, respectively, outperforming existing embedding- and cross-attention-based models.

Fri 25 SeptSound
The gist
Detecting spoken keywords without retraining for each new word is hard, especially in difficult audio conditions. The authors found that existing models often fail because they do not calibrate their confidence scores properly, making it hard to apply a fixed detection threshold. They propose a new training method, called CORD-KWS, that calibrates these scores and also helps the model learn the correct order of sounds in words. This approach improves detection accuracy on challenging test sets without increasing the time it takes to recognize keywords.
Open → 2609.31869v1

Participatory speech dataset creation methods improve queer community inclusion

Towards participatory speech dataset curation: A queer case study and conceptual framework

Abstract: In this paper, we motivate the need for a participatory speech dataset creation framework through a case study of the LGBTQIA+, or queer, community - a community with documented concerns about AI and reported harms, including attempts to develop 'gaydar' technologies that purportedly identify individuals as queer. We review common speech data collection practices, why these methods may be unsuitable for engaging with queer speakers, and discuss previous efforts in participatory AI with queer community engagement, as well as participatory endeavours specific to speech data collection for other marginalized communities. From this review, we develop a conceptual framework for participatory speech data curation by, for, and with marginalized communities drawing on insights from co-design and knowledge sharing. We propose a framework comprising overlapping and two-way processes of defining a community, project formulation, modes of participation, and personal autonomy.

Mon 21 SeptArtificial IntelligenceComputers and Society
The gist
Collecting voice data often misses the needs and concerns of queer people, especially due to past harmful AI uses. The authors look at current speech data methods and why they may not work well for the LGBTQIA+ community. They review efforts to include queer voices in AI and speech data work. From this, they suggest a new way to create speech datasets that involves queer and marginalized communities as active participants who help shape the project and keep control over their data.
Open → 2609.25496v1

StepAudio 3 enables real time spoken dialogue with deep reasoning

StepAudio 3 Realtime Technical Report

Abstract: Realtime spoken interaction demands deep reasoning, prompt responses, and fluid turn-taking. We present StepAudio 3 Realtime, an audio-language foundation model organized around a continuous listen-converse-think-act loop. Deep Perception captures rich acoustic cues to interpret user intent, while Seamless Duplex models synchronized audio streams to handle pauses, backchannels, and interruptions naturally. Crucially, we resolve the tension between deep deliberation and latency via Think-While-Speaking, executing private reasoning in parallel with spoken delivery. In reasoning mode, StepAudio 3 reaches a 73.0 macro average on StepAudioChat. With Think-While-Speaking, it achieves dialogue and reasoning performance comparable to dedicated reasoning models while speaking in real time. Furthermore, an integrated Voice Agent handles asynchronous tool execution without disrupting the dialogue flow. StepAudio 3 Realtime achieves top-tier performance across key dimensions: an exceptional 90.6 on the MMSU benchmark, 98.9 Overall on the Artificial Analysis Full-Duplex Bench, and a 56.0% macro task-success rate on $τ$-Voice.

Sat 12 SeptSound
The gist
Holding natural conversations with machines is hard because they need to quickly understand speech, think deeply, and reply without awkward pauses. The authors present StepAudio 3, a system that listens and talks continuously while reasoning in the background, making conversations flow smoothly even during interruptions or back-and-forth talk. This approach lets the system think and speak at the same time, delivering responses as good as specialized reasoning models but in real time. They showed strong results on several benchmarks testing understanding, reasoning, and conversation skills.
Open → 2609.14005v1