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Pre-trained speech encoders have been central to pushing state-of-the-art results across various speech understanding and generation tasks. Nonetheless, the capabilities of these encoders in low-resource settings are yet to be thoroughly…

计算与语言 · 计算机科学 2023-05-30 Hao Yang , Jinming Zhao , Gholamreza Haffari , Ehsan Shareghi

High-fidelity neural audio codecs in Text-to-speech (TTS) aim to compress speech signals into discrete representations for faithful reconstruction. However, prior approaches faced challenges in effectively disentangling acoustic and…

声音 · 计算机科学 2025-09-23 Ruonan Zhang , Xiaoyang Hao , Yichen Han , Junjie Cao , Yue Liu , Kai Zhang

Phonological features provide a language-general and linguistically grounded representation of speech. We present PhonoQ-2.0, a multilingual frame-level phonological feature recognizer built on self-supervised speech models. The system…

计算与语言 · 计算机科学 2026-05-26 Abner Hernandez , Tomás Arias-Vergara , Daiqi Liu , Andreas Maier , Paula Andrea Pérez-Toro

While many speakers of low-resource languages regularly code-switch between their languages and other regional languages or English, datasets of codeswitched speech are too small to train bespoke acoustic models from scratch or do language…

计算与语言 · 计算机科学 2023-11-28 Tolúlopé Ògúnrèmí , Christopher D. Manning , Dan Jurafsky

Scaling spoken language modeling requires speech tokens that are both efficient and universal. Recent work has proposed syllables as promising speech tokens at low temporal resolution, but existing models are constrained to English and fail…

音频与语音处理 · 电气工程与系统科学 2026-02-02 Cheol Jun Cho , Nicholas Lee , Alan W Black , Gopala K. Anumanchipalli

Efficiently representing audio signals in a compressed latent space is critical for latent generative modelling. However, existing autoencoders often force a choice between continuous embeddings and discrete tokens. Furthermore, achieving…

声音 · 计算机科学 2025-09-15 Marco Pasini , Stefan Lattner , George Fazekas

Neural Speech Codecs face a fundamental trade-off at low bitrates: preserving acoustic fidelity often compromises semantic richness. To address this, we introduce SACodec, a novel codec built upon an asymmetric dual-quantizer that employs…

声音 · 计算机科学 2025-12-25 Zhongren Dong , Bin Wang , Jing Han , Haotian Guo , Xiaojun Mo , Yimin Cao , Zixing Zhang

Speaker verification (SV) utilizing features obtained from models pre-trained via self-supervised learning has recently demonstrated impressive performances. However, these pre-trained models (PTMs) usually have a temporal resolution of 20…

音频与语音处理 · 电气工程与系统科学 2026-01-28 Jisoo Myoung , Sangwook Han , Kihyuk Kim , Jong Won Shin

This paper presents XLSR which learns cross-lingual speech representations by pretraining a single model from the raw waveform of speech in multiple languages. We build on wav2vec 2.0 which is trained by solving a contrastive task over…

计算与语言 · 计算机科学 2020-12-17 Alexis Conneau , Alexei Baevski , Ronan Collobert , Abdelrahman Mohamed , Michael Auli

Large Audio Language Models (LALMs) demonstrate impressive performance across diverse tasks, ranging from speech recognition to general audio understanding. However, their scalability is limited by the quadratic complexity of attention and…

音频与语音处理 · 电气工程与系统科学 2025-11-27 Saurabhchand Bhati , Samuel Thomas , Hilde Kuehne , Rogerio Feris , James Glass

We propose TQCodec, a neural audio codec designed for high-bitrate, high-fidelity music streaming. Unlike existing neural codecs that primarily target ultra-low bitrates (<= 16kbps), TQCodec operates at 44.1 kHz and supports bitrates from…

Speech-to-speech translation (S2ST) converts input speech to speech in another language. A challenge of delivering S2ST in real time is the accumulated delay between the translation and speech synthesis modules. While recently incremental…

计算与语言 · 计算机科学 2022-07-18 Danni Liu , Changhan Wang , Hongyu Gong , Xutai Ma , Yun Tang , Juan Pino

We introduce LMCodec, a causal neural speech codec that provides high quality audio at very low bitrates. The backbone of the system is a causal convolutional codec that encodes audio into a hierarchy of coarse-to-fine tokens using residual…

We propose a multi-stage framework for universal speech enhancement, designed for the Interspeech 2025 URGENT Challenge. Our system first employs a Sparse Compression Network to robustly separate sources and extract an initial clean speech…

声音 · 计算机科学 2025-06-03 Nabarun Goswami , Tatsuya Harada

We investigate a novel cross-lingual multi-speaker text-to-speech synthesis approach for generating high-quality native or accented speech for native/foreign seen/unseen speakers in English and Mandarin. The system consists of three…

音频与语音处理 · 电气工程与系统科学 2019-11-27 Zhaoyu Liu , Brian Mak

This paper proposes RefXVC, a method for cross-lingual voice conversion (XVC) that leverages reference information to improve conversion performance. Previous XVC works generally take an average speaker embedding to condition the speaker…

音频与语音处理 · 电气工程与系统科学 2024-06-25 Mingyang Zhang , Yi Zhou , Yi Ren , Chen Zhang , Xiang Yin , Haizhou Li

The accuracy of end-to-end (E2E) automatic speech recognition (ASR) models continues to improve as they are scaled to larger sizes, with some now reaching billions of parameters. Widespread deployment and adoption of these models, however,…

Self-supervised learning (SSL) has advanced speech processing. However, existing speech SSL methods typically assume a single sampling rate and struggle with mixed-rate data due to temporal resolution mismatch. To address this limitation,…

声音 · 计算机科学 2026-03-25 Zikang Huang , Meng Ge , Tianrui Wang , Xuanchen Li , Xiaobao Wang , Longbiao Wang , Jianwu Dang

We present a hybrid framework that leverages the trade-off between temporal and frequency precision in audio representations to improve the performance of speech enhancement task. We first show that conventional approaches using specific…

音频与语音处理 · 电气工程与系统科学 2018-12-24 Jang-Hyun Kim , Jaejun Yoo , Sanghyuk Chun , Adrian Kim , Jung-Woo Ha

We propose FlowDec, a neural full-band audio codec for general audio sampled at 48 kHz that combines non-adversarial codec training with a stochastic postfilter based on a novel conditional flow matching method. Compared to the prior work…