Engineering & Signals · Interdisciplinary & Society
Audio & Speech Processing
The newest papers in Audio & Speech Processing that ReadSearch has read, newest first. Each links to its own page, with its authors, date, where it was published and the original.
334 papers, from 8 September 2026 to 4 November 2026: 334 arXiv preprints.
Get the few papers worth your time in Audio & Speech Processing every morning.
Follow Audio & Speech Processing and ReadSearch reads its new papers each day, then brings you the few worth reading, each with its key finding in a sentence.
Newest papers, page 8
Controlled Acquisition and Abstention in Three-Channel Score Conflicts
When audio, video, and text disagree, accuracy alone does not show whether to acquire another source or abstain.Disentangling Linguistic and Paralinguistic Information with Routed Sparse Autoencoders
Self-supervised speech encoders contain linguistic and paralinguistic information in a shared, entangled representation space.Conversational Voice Aesthetic Model with Reinforcement Learning from Human Listeners
We introduce Conversational Voice Aesthetic Model, a speech large language model for describing the voice aesthetics of real or synthetic speech responses in natural conversational contexts.A Novel Sentence Stress Detection Framework Leveraging Auxiliary Word-Stress Modeling and Loss Optimization
Prosodic stress is a crucial aspect of automatic pronunciation assessment (APA), encompassing both sentence stress detection (SSD) and word stress detection (WSD).HINTT Submission to the 2nd MLC-SLM Challenge: Comparing Cascaded and Unified Approaches to Diarization and ASR
This paper presents the HINTT system submitted to the 2nd Challenge and Workshop on Multilingual Conversational Speech Language Model (MLC-SLM).Conversation Is a Two-Body Problem: Dyadic Evaluation of Full-Duplex Dialogue Models
Full-duplex spoken dialogue models listen and speak at the same time, enabling voice agents to have natural, low-latency interactions that turn-based systems cannot offer.WorldSonus: Bringing Sound to Worlds
Recent advances in world models have enabled increasingly realistic visual synthesis.Phoneme-Guided Initialization for LLM-based Speech Recognition
Speech large language models (speech LLMs) perform well on automatic speech recognition (ASR) when sufficient paired speech-text data is available, but their performance degrades in low-resource…Towards AI-Generated Music Plagiarism Detection as a Version Identification Problem
The rapid expansion of text-to-music generative models challenges traditional paradigms of music creation and intellectual property.Relational Synthesis: Structure-Mediated Concatenative Synthesis for Foley and Retrieval-Augmented Audio Generation
We ask: given a retrieved source audio $S$ and a separate reference audio $R$, can we synthesize novel audio $Y$ out of this pair $(S,R)$ such that $Y$ remains acoustically consistent with $S$…Paradee: Distilling Kokoro-82M into an 8M-Parameter Single-Voice Text-to-Speech Model
We distill Kokoro-82M, a widely used open text-to-speech model with 54 voices, into Paradee, an 8.07M-parameter model that speaks one of them.GIVE-KWS: Gated Injection of Visual Evidence for Noise-Robust Query-by-Example Keyword Spotting
Visual speech promises noise-robust keyword spotting, yet a visual stream is not necessarily used.SEAL: Mixture-Closed Additive Reconstruction and Refinement-Aware Expert Routing for Efficient Speech Separation
Compact time-frequency separators that mask the mixture and refine through a shared cell face two limits.Logbook: Extremely Long-form Audio Event Understanding
Audio benchmarks are built around short, pre-segmented clips, limiting model design to brief inputs or fixed vocabularies.AccentCL: Robust Accent Classification with Incremental Expansion
Accent classifiers are typically trained with a fixed label inventory and cannot accommodate new accent categories as new data becomes available.Transferable Adversarial Robustness for Speech Foundation Models via Hierarchical Stabilization
Frozen speech foundation models (SFMs) make downstream adaptation efficient: the backbone can stay fixed while a task learns layer fusion and a lightweight classifier.Joint Estimation of Common-Slope Decay Rates and Spatial Amplitudes Using Parameterized Nonnegative Matrix Factorization
We formulate joint estimation of common-slope decay rates and amplitudes from room impulse responses (RIRs) as parameterized nonnegative matrix factorization with the Itakura--Saito divergence as…Factorized Delayed Streams Modeling for LLM-based Streaming ASR
Delayed Streams Modeling (DSM) enables LLM-based streaming automatic speech recognition (ASR) by aligning acoustic and text streams on a common timeline.Can LLM Agents Automate Reinforcement Learning for Text-to-Speech?
Although reinforcement learning (RL) post-training repairs the localized segmental errors of zero-shot text-to-speech (TTS), arriving at a working recipe still relies on tedious manual tuning, and…GS-Codec: A Gaussian-Splatting Bottleneck for Neural Audio Coding
Neural audio codecs compress waveforms into compact discrete tokens that underpin speech language models, real-time communication, and large-scale audio storage.