Physical Sciences › Computer Science › Artificial Intelligence
Speech Recognition and Synthesis
567 papiers indexés
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- DAST: A Dual-Stream Voice Anonymization Attacker with Staged Training
Ridwan Arefeen, Xiaoxiao Miao, Rong Tong, Aik Beng Ng, Simon See, Timothy Liu · 16 mars 2026
Voice anonymization masks vocal traits while preserving linguistic content, which may still leak speaker-specific patterns. To assess and strengthen privacy evaluation, we propose a dual-stream attacker that fuses spectral and self-supervised learning features via parallel encoders with a three-stag…
- Mitigating Latent Mismatch in cVAE-Based Singing Voice Synthesis via Flow Matching
Minhyeok Yun, Yong-Hoon Choi · 16 mars 2026
Singing voice synthesis (SVS) aims to generate natural and expressive singing waveforms from symbolic musical scores. In cVAE-based SVS, however, a mismatch arises because the decoder is trained with latent representations inferred from target singing signals, while inference relies on latent repres…
- Self-Supervised Speech Models Encode Phonetic Context via Position-dependent Orthogonal Subspaces
Kwanghee Choi, Eunjung Yeo, Cheol Jun Cho, David R. Mortensen, David Harwath · 16 mars 2026
Transformer-based self-supervised speech models (S3Ms) are often described as contextualized, yet what this entails remains unclear. Here, we focus on how a single frame-level S3M representation can encode phones and their surrounding context. Prior work has shown that S3Ms represent phones composit…
- LiLAW: Lightweight Learnable Adaptive Weighting to Meta-Learn Sample Difficulty, Improve Noisy Training, Increase Fairness, and Effectively Use Synthetic Data
Abhishek Moturu, Muhammad Muzammil, Anna Goldenberg, Babak Taati · 16 mars 2026
Training deep neural networks with noise and data heterogeneity is a major challenge. We introduce Lightweight Learnable Adaptive Weighting (LiLAW), a method that dynamically adjusts the loss weight of each training sample based on its evolving difficulty, categorized as easy, moderate, or hard. Usi…
- Mask2Flow-TSE: Two-Stage Target Speaker Extraction with Masking and Flow Matching
Junwon Moon, Hyunjin Choi, Hansol Park, Heeseung Kim, Kyuhong Shim · 16 mars 2026
Target speaker extraction (TSE) extracts the target speaker's voice from overlapping speech mixtures given a reference utterance. Existing approaches typically fall into two categories: discriminative and generative. Discriminative methods apply time-frequency masking for fast inference but often ov…
- Continued Pretraining for Low-Resource Swahili ASR: Achieving State-of-the-Art Performance with Minimal Labeled Data
Hillary Mutisya, John Mugane · 13 mars 2026
We investigate continued pretraining (CPT) for adapting wav2vec2-bert-2.0 to Swahili automatic speech recognition (ASR). Our approach combines unlabeled audio with limited labeled data through pseudo-labeled CPT followed by supervised finetuning. With 20,000 labeled samples, we achieve 3.24% WER on …
- Toward Complex-Valued Neural Networks for Waveform Generation
Hyung-Seok Oh, Deok-Hyeon Cho, Seung-Bin Kim, Seong-Whan Lee · 13 mars 2026
Neural vocoders have recently advanced waveform generation, yielding natural and expressive audio. Among these approaches, iSTFT-based vocoders have recently gained attention. They predict a complex-valued spectrogram and then synthesize the waveform via iSTFT, thereby avoiding learned upsampling st…
- Duration Aware Scheduling for ASR Serving Under Workload Drift
Darshan Makwana, Yash Jogi, Harsh Kotta, Aayush Kubba · 13 mars 2026
Scheduling policies in large-scale Automatic Speech Recognition (ASR) serving pipelines play a key role in determining end-to-end (E2E) latency. Yet, widely used serving engines rely on first-come-first-served (FCFS) scheduling, which ignores variability in request duration and leads to head-of-line…
- AlphaFlowTSE: One-Step Generative Target Speaker Extraction via Conditional AlphaFlow
Duojia Li, Shuhan Zhang, Zihan Qian, Wenxuan Wu, Shuai Wang, Qingyang Hong, Lin Li, Haizhou Li · 12 mars 2026
In target speaker extraction (TSE), we aim to recover target speech from a multi-talker mixture using a short enrollment utterance as reference. Recent studies on diffusion and flow-matching generators have improved target-speech fidelity. However, multi-step sampling increases latency, and one-step…
- Speaker Verification with Speech-Aware LLMs: Evaluation and Augmentation
Thomas Thebaud, Yuzhe Wang, Laureano Moro-Velazquez, Jesus Villalba-Lopez, Najim Dehak · 12 mars 2026
Speech-aware large language models (LLMs) can accept speech inputs, yet their training objectives largely emphasize linguistic content or specific fields such as emotions or the speaker's gender, leaving it unclear whether they encode speaker identity. First, we propose a model-agnostic scoring prot…
- Probabilistic Verification of Voice Anti-Spoofing Models
Evgeny Kushnir, Alexandr Kozodaev, Dmitrii Korzh, Mikhail Pautov, Oleg Kiriukhin, Oleg Y. Rogov · 12 mars 2026
Recent advances in generative models have amplified the risk of malicious misuse of speech synthesis technologies, enabling adversaries to impersonate target speakers and access sensitive resources. Although speech deepfake detection has progressed rapidly, most existing countermeasures lack formal …
- When Fine-Tuning Fails and when it Generalises: Role of Data Diversity and Mixed Training in LLM-based TTS
Anupam Purwar, Aditya Choudhary · 12 mars 2026
Large language models are increasingly adopted as semantic backbones for neural text-to-speech systems. However, frozen LLM representations are insufficient for modeling speaker specific acoustic and perceptual characteristics. Our experiments involving fine tuning of the Language Model backbone of …
- Are Deep Speech Denoising Models Robust to Adversarial Noise?
Will Schwarzer, Neel Chaudhari, Philip S. Thomas, Andrea Fanelli, Xiaoyu Liu · 12 mars 2026
Deep noise suppression (DNS) models enjoy widespread use throughout a variety of high-stakes speech applications. However, we show that four recent DNS models can each be reduced to outputting unintelligible gibberish through the addition of psychoacoustically hidden adversarial noise, even in low-b…
- G-STAR: End-to-End Global Speaker-Tracking Attributed Recognition
Jing Peng, Ziyi Chen, Haoyu Li, Yucheng Wang, Duo Ma, Mengtian Li, Yunfan Du, Dezhu Xu, Kai Yu, Shuai Wang · 12 mars 2026
We study timestamped speaker-attributed ASR for long-form, multi-party speech with overlap, where chunk-wise inference must preserve meeting-level speaker identity consistency while producing time-stamped, speaker-labeled transcripts. Previous Speech-LLM systems tend to prioritize either local diari…
- SENS-ASR: Semantic Embedding injection in Neural-transducer for Streaming Automatic Speech Recognition
Youness Dkhissi (LIUM), Valentin Vielzeuf (LIUM), Elys Allesiardo (LIUM), Anthony Larcher (LIUM) · 12 mars 2026
Many Automatic Speech Recognition (ASR) applications require streaming processing of the audio data. In streaming mode, ASR systems need to start transcribing the input stream before it is complete, i.e., the systems have to process a stream of inputs with a limited (or no) future context. Compared …
- HyWA: Hypernetwork Weight Adapting Personalized Voice Activity Detection
Mahsa Ghazvini Nejad, Hamed Jafarzadeh Asl, Amin Edraki, Mohammadreza Sadeghi, Masoud Asgharian, Yuanhao Yu, Vahid Partovi Nia · 12 mars 2026
Personalized Voice Activity Detection (PVAD) systems activate only in response to a specific target speaker. Speaker-conditioning methods are employed to inject information about the target speaker into a VAD pipeline, to achieve personalization. Existing speaker-conditioning methods typically modif…
- Latent Speech-Text Transformer
Yen-Ju Lu, Yashesh Gaur, Wei Zhou, Benjamin Muller, Jesus Villalba, Najim Dehak, Luke Zettlemoyer, Gargi Ghosh, Mike Lewis, Srinivasan Iyer, Duc Le · 11 mars 2026
Auto-regressive speech-text models pre-trained on interleaved text tokens and discretized speech tokens demonstrate strong speech understanding and generation, yet remain substantially less compute-efficient than text LLMs, partly due to the much longer sequences of speech tokens relative to text. T…
- GDR-learners: Orthogonal Learning of Generative Models for Potential Outcomes
Valentyn Melnychuk, Stefan Feuerriegel · 10 mars 2026
Various deep generative models have been proposed to estimate potential outcomes distributions from observational data. However, none of them have the favorable theoretical property of general Neyman-orthogonality and, associated with it, quasi-oracle efficiency and double robustness. In this paper,…
- The Talking Robot: Distortion-Robust Acoustic Models for Robot-Robot Communication
Hanlong Li, Karishma Kamalahasan, Jiahui Li, Kazuhiro Nakadai, Shreyas Kousik · 10 mars 2026
We present Artoo, a learned acoustic communication system for robots that replaces hand-designed signal processing with end-to-end co-trained neural networks. Our system pairs a lightweight text-to-speech (TTS) transmitter (1.18M parameters) with a conformer-based automatic speech recognition (ASR) …
- Benchmarking Language Modeling for Lossless Compression of Full-Fidelity Audio
Phillip Long, Zachary Novack, Chris Donahue · 10 mars 2026
Autoregressive "language" models (LMs) trained on raw waveforms can be repurposed for lossless audio compression, but prior work is limited to 8-bit audio, leaving open whether such approaches work for practical settings (16/24-bit) and can compete with existing codecs. We benchmark LM-based compres…
- Koopman Regularized Deep Speech Disentanglement for Speaker Verification
Nikos Chazaridis, Mohammad Belal, Rafael Mestre, Timothy J. Norman, Christine Evers · 9 mars 2026
Human speech contains both linguistic content and speaker dependent characteristics making speaker verification a key technology in identity critical applications. Modern deep learning speaker verification systems aim to learn speaker representations that are invariant to semantic content and nuisan…
- StreamVoiceAnon+: Emotion-Preserving Streaming Speaker Anonymization via Frame-Level Acoustic Distillation
Nikita Kuzmin, Kong Aik Lee, Eng Siong Chng · 9 mars 2026
We address the challenge of preserving emotional content in streaming speaker anonymization (SA). Neural audio codec language models trained for audio continuation tend to degrade source emotion: content tokens discard emotional information, and the model defaults to dominant acoustic patterns rathe…
- Whisper-CD: Accurate Long-Form Speech Recognition using Multi-Negative Contrastive Decoding
Hoseong Ahn, Jeongyun Chae, Yoonji Park, Kyuhong Shim · 9 mars 2026
Long-form speech recognition with large encoder-decoder models such as Whisper often exhibit hallucinations, repetition loops, and content omissions. These errors can accumulate and be further amplified when the previous segment's transcription is used as decoding context. We propose Whisper-CD, a t…
- New Insights into Optimal Alignment of Acoustic and Linguistic Representations for Knowledge Transfer in ASR
Xugang Lu, Peng Shen, Hisashi Kawai · 6 mars 2026
Aligning acoustic and linguistic representations is a central challenge to bridge the pre-trained models in knowledge transfer for automatic speech recognition (ASR). This alignment is inherently structured and asymmetric: while multiple consecutive acoustic frames typically correspond to a single l…
- BabyHuBERT: Multilingual Self-Supervised Learning for Segmenting Speakers in Child-Centered Long-Form Recordings
Th\'eo Charlot, Tarek Kunze, Maxime Poli, Alejandrina Cristia, Emmanuel Dupoux, Marvin Lavechin · 6 mars 2026
Child-centered daylong recordings are essential for studying early language development, but existing speech models trained on clean adult data perform poorly due to acoustic and linguistic differences. We introduce BabyHuBERT, a self-supervised speech model trained on 13,000 hours of multilingual c…
