Papers for

call center technology providers

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.

VoxTurn improves voice agent turn-taking decisions with one model

VoxTurn: Answering Several Turn-Taking Questions in One Pass with One Model

Abstract: A cascaded voice agent must decide, throughout each silence of its user, whether the user has finished their turn: answering too early cuts the user off, answering too late makes the agent slow. It often needs related decisions too: will the user soon go on, is speech during its own turn an interruption or a backchannel, is a sound speech at all. With open detectors, each extra decision costs another model or another call, and on the 14-language eot-bench they cut users off in more than a quarter of mid-turn pauses. Because they are trained on segmented utterances, this mismatch is the largest single source of error. We present VoxTurn, a 0.8B speech LLM that casts these decisions as multiple-choice questions, scores the hidden state of each option, and answers any set of them in one forward pass with a block-isolating attention mask, exactly as if asked one at a time. Trained on real conversational pauses with labels derived from their timing, and re-queried while a silence lasts, VoxTurn reaches 11.5% false cut-offs in English at a 300 ms latency budget, comparable to two commercial detectors, and a 14-language mean of 19.9%, the lowest among open models, at 8.3 ms per decision. In Pipecat and LiveKit it cuts off 17% and 9% of turns, and it ranks fifth of 24 in TurnBench end-of-turn detection without any TurnBench training data. The model, code and framework plugins are released.

Thu 15 OctSound
The gist
Voice assistants must decide when users have finished speaking to respond quickly without interruptions. Current methods use multiple models for different related decisions, causing mistakes and delays. The authors present VoxTurn, a single large speech model that answers several turn-taking questions all at once. It performs well across many languages and is faster and more accurate than open alternatives, cutting user interruptions by a notable margin.
Open → 2610.21150v1

Flow matching and policy optimization reduce speech enhancement hallucinations

AuraSE: Low-Hallucination Generative Speech Enhancement via Multimodal Flow Matching and Inference Policy Optimization

Abstract: Generative speech enhancement models can produce cleaner and more natural-sounding speech than conventional discriminative approaches, but may hallucinate by changing speech content or speaker identity, even with transcript conditioning. We present AuraSE, a flow-matching framework that addresses hallucination through complementary modality and inference designs. First, a double-stream-to-single-stream multimodal Diffusion Transformer (MMDiT) allows transcript and acoustic representations to interact while preserving a dedicated pathway for the degraded input. Second, we find that the best decoder configuration, governed by guidance scale, sampling temperature, and step count, varies substantially across utterances. This observation motivates Inference Policy Optimization (IPO), an online, on-policy preference optimization method. IPO generates multiple candidates from the current model under different inference configurations, ranks them with a multi-objective reward, and learns from their relative preferences. AuraSE-IPO ranks first on 11 of 12 metrics across the synthetic test sets and obtains the highest DNSMOS and blind-listening scores among the evaluated systems on the real DNS blind test set. At deployment, it uses a fixed $10$-step ODE decoder without classifier-free guidance (CFG) or per-utterance configuration search.

Mon 5 OctSound
The gist
Speech enhancement systems try to make noisy recorded speech sound clearer and more natural, but sometimes they accidentally change what is being said or who is speaking. The authors introduce AuraSE, which uses a special type of model that carefully blends the noisy speech sound with a matching transcript to avoid these mistakes. They also found that adjusting the way the system processes each sentence during listening improves results, so they developed a way for the system to learn how best to do this on the fly. This approach consistently improved speech quality and reduced errors on various tests, including real-world noisy recordings.
Open → 2610.06632v1

Arabic speech tasks advance with new multidialectal NADI 2026 benchmark

NADI 2026: The Second Multidialectal Arabic Speech Processing Shared Task

Abstract: NADI 2026 is the seventh edition of the Nuanced Arabic Dialect Identification (NADI) shared task series and the second dedicated to multidialectal Arabic speech processing. This edition comprises five tasks and eight subtasks spanning Automatic Speech Recognition (ASR), Spoken Dialect Identification (SDID), Text-to-Speech (TTS), Spoken Language Translation (SLT), and Spoken Language Understanding (SLU). NADI 2026 emphasizes realistic evaluation through low-bandwidth, mixed-dialect, code-switched, out-of-domain, and zero-shot settings, while introducing TTS, SLT, and SLU to the series for the first time. The shared task attracted 21 participating teams from at least 13 countries, with 48 test-phase submissions and 14 submitted system-description papers. Results show that out-of-domain generalization remains a major bottleneck and highlight the effectiveness of recent Arabic-specialized speech models, multimodal dialect identification approaches, and ensemble methods. Overall, NADI 2026 provides a broader and more challenging benchmark for robust Arabic dialect speech processing.

Tue 22 SeptComputation and Language
The gist
Understanding and processing different Arabic dialects in speech is challenging because of many variations and code-switching. The authors present NADI 2026, a competition with five main language processing tasks, including recognizing speech, identifying dialects, text-to-speech, translating spoken language, and understanding spoken content. They tested systems under tough conditions like mixed dialects and limited data quality. The results show it is still hard for systems to work well outside familiar data, but specialized Arabic models and combining multiple methods help performance. This benchmark gives a tougher test for future work in Arabic speech technology.
Open → 2609.27086v1