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Zero-shot voice conversion (VC) aims to convert the original speaker's timbre to any target speaker while keeping the linguistic content. Current mainstream zero-shot voice conversion approaches depend on pre-trained recognition models to…

声音 · 计算机科学 2024-12-04 Yuke Li , Xinfa Zhu , Hanzhao Li , JiXun Yao , WenJie Tian , XiPeng Yang , YunLin Chen , Zhifei Li , Lei Xie

This paper presents a method of sequence-to-sequence (seq2seq) voice conversion using non-parallel training data. In this method, disentangled linguistic and speaker representations are extracted from acoustic features, and voice conversion…

音频与语音处理 · 电气工程与系统科学 2020-01-14 Jing-Xuan Zhang , Zhen-Hua Ling , Li-Rong Dai

Frame stacking is broadly applied in end-to-end neural network training like connectionist temporal classification (CTC), and it leads to more accurate models and faster decoding. However, it is not well-suited to conventional neural…

计算与语言 · 计算机科学 2017-05-18 Xu Tian , Jun Zhang , Zejun Ma , Yi He , Juan Wei

Autoregressive sequence models, such as Transformer-based vision-language action (VLA) policies, can be tremendously effective for capturing complex and generalizable robotic behaviors. However, such models require us to choose a…

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

Real-time speech synthesis requires balancing inference latency and acoustic fidelity for interactive applications. Conventional continuous text-to-speech pipelines require computationally intensive neural vocoders to reconstruct phase…

音频与语音处理 · 电气工程与系统科学 2026-04-15 Tianhui Su , Tien-Ping Tan , Salima Mdhaffar , Yannick Estève , Aghilas Sini

This paper introduces a cross-lingual dubbing system that translates speech from one language to another while preserving key characteristics such as duration, speaker identity, and speaking speed. Despite the strong translation quality of…

计算与语言 · 计算机科学 2025-12-30 Jeongsoo Choi , Jaehun Kim , Joon Son Chung

Understanding and reasoning over long contexts is a crucial capability for language models (LMs). Although recent models support increasingly long context windows, their accuracy often deteriorates as input length grows. In practice, models…

计算与语言 · 计算机科学 2026-04-17 Xi Ye , Wuwei Zhang , Fangcong Yin , Howard Yen , Danqi Chen

Speech codecs are traditionally optimized for waveform fidelity, allocating bits to preserve acoustic detail even when much of it can be inferred from linguistic structure. This leads to inefficient compression and suboptimal performance on…

声音 · 计算机科学 2025-12-29 Liuyang Bai , Weiyi Lu , Li Guo

The fusion of speech and language in the era of large language models has garnered significant attention. Discrete speech token is often utilized in text-to-speech tasks for speech compression and portability, which is convenient for joint…

声音 · 计算机科学 2025-04-01 Yixing Li , Ruobing Xie , Xingwu Sun , Yu Cheng , Zhanhui Kang

In this paper we aim to automatically discover high quality frame-level speech features and acoustic tokens directly from unlabeled speech data. A Multi-granular Acoustic Tokenizer (MAT) was proposed for automatic discovery of multiple sets…

计算与语言 · 计算机科学 2017-07-19 Cheng-Tao Chung , Cheng-Yu Tsai , Chia-Hsiang Liu , Lin-Shan Lee

Modern text-to-speech systems are able to produce natural and high-quality speech, but speech contains factors of variation (e.g. pitch, rhythm, loudness, timbre)\ that text alone cannot contain. In this work we move towards a speech…

音频与语音处理 · 电气工程与系统科学 2020-10-29 Giorgio Fabbro , Vladimir Golkov , Thomas Kemp , Daniel Cremers

With the rise of Speech Large Language Models (SpeechLLMs), two dominant approaches have emerged for speech processing: discrete tokens and continuous features. Each approach has demonstrated strong capabilities in audio-related processing…

计算与语言 · 计算机科学 2025-08-26 Dingdong Wang , Junan Li , Mingyu Cui , Dongchao Yang , Xueyuan Chen , Helen Meng

Discrete speech tokenization is a fundamental component in speech codecs. However, in large-scale speech-to-speech systems, the complexity of parallel streams from multiple quantizers and the computational cost of high-time-dimensional…

声音 · 计算机科学 2025-07-28 Rongkun Xue , Yazhe Niu , Shuai Hu , Zixin Yin , Yongqiang Yao , Jing Yang

Recent years have witnessed a trend that large language model (LLM) based text-to-speech (TTS) emerges into the mainstream due to their high naturalness and zero-shot capacity. In this paradigm, speech signals are discretized into token…

Integrating audio comprehension and generation into large language models (LLMs) remains challenging due to the continuous nature of audio and the resulting high sampling rates. Here, we introduce a novel approach that combines Variational…

音频与语音处理 · 电气工程与系统科学 2025-03-31 Shivam Mehta , Nebojsa Jojic , Hannes Gamper

Most text-to-speech (TTS) methods use high-quality speech corpora recorded in a well-designed environment, incurring a high cost for data collection. To solve this problem, existing noise-robust TTS methods are intended to use noisy speech…

声音 · 计算机科学 2022-06-30 Takaaki Saeki , Kentaro Tachibana , Ryuichi Yamamoto

Current continuous sign language recognition (CSLR) methods struggle with handling diverse samples. Although dynamic convolutions are ideal for this task, they mainly focus on spatial modeling and fail to capture the temporal dynamics and…

计算机视觉与模式识别 · 计算机科学 2025-07-08 Sheng Liu , Yiheng Yu , Yuan Feng , Min Xu , Zhelun Jin , Yining Jiang , Tiantian Yuan

Data-driven approaches hold promise for audio captioning. However, the development of audio captioning methods can be biased due to the limited availability and quality of text-audio data. This paper proposes a SynthAC framework, which…

声音 · 计算机科学 2023-09-19 Feiyang Xiao , Qiaoxi Zhu , Jian Guan , Xubo Liu , Haohe Liu , Kejia Zhang , Wenwu Wang

It is highly desirable that speech enhancement algorithms can achieve good performance while keeping low latency for many applications, such as digital hearing aids, acoustically transparent hearing devices, and public address systems. To…

音频与语音处理 · 电气工程与系统科学 2022-06-01 Chengshi Zheng , Wenzhe Liu , Andong Li , Yuxuan Ke , Xiaodong Li