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相关论文: WESPER: Zero-shot and Realtime Whisper to Normal V…

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Recent progress in large-scale zero-shot speech synthesis has been significantly advanced by language models and diffusion models. However, the generation process of both methods is slow and computationally intensive. Efficient speech…

音频与语音处理 · 电气工程与系统科学 2024-10-25 Zhen Ye , Zeqian Ju , Haohe Liu , Xu Tan , Jianyi Chen , Yiwen Lu , Peiwen Sun , Jiahao Pan , Weizhen Bian , Shulin He , Wei Xue , Qifeng Liu , Yike Guo

Traditional control interfaces for robotic-assisted minimally invasive surgery impose a significant cognitive load on surgeons. To improve surgical efficiency, surgeon-robot collaboration capabilities, and reduce surgeon burden, we present…

机器人学 · 计算机科学 2025-03-25 Ana Davila , Jacinto Colan , Yasuhisa Hasegawa

End-to-end automatic speech recognition (ASR) and large language models, such as Whisper and GPT-2, have recently been scaled to use vast amounts of training data. Despite the large amount of training data, infrequent content words that…

计算与语言 · 计算机科学 2023-06-06 Guangzhi Sun , Xianrui Zheng , Chao Zhang , Philip C. Woodland

A Spoken dialogue system for an unseen language is referred to as Zero resource speech. It is especially beneficial for developing applications for languages that have low digital resources. Zero resource speech synthesis is the task of…

音频与语音处理 · 电气工程与系统科学 2020-09-11 Karthik Pandia D S , Anusha Prakash , Mano Ranjith Kumar , Hema A Murthy

Neural transducers have achieved human level performance on standard speech recognition benchmarks. However, their performance significantly degrades in the presence of cross-talk, especially when the primary speaker has a low…

音频与语音处理 · 电气工程与系统科学 2023-03-31 Desh Raj , Junteng Jia , Jay Mahadeokar , Chunyang Wu , Niko Moritz , Xiaohui Zhang , Ozlem Kalinli

In this work, we propose a method to create domain-sensitive speech recognition models that utilize textual domain information by conditioning its generation on a given text prompt. This is accomplished by fine-tuning a pre-trained,…

音频与语音处理 · 电气工程与系统科学 2023-10-09 Feng-Ting Liao , Yung-Chieh Chan , Yi-Chang Chen , Chan-Jan Hsu , Da-shan Shiu

Automatic speech recognition (ASR) for dysarthric speech remains challenging due to data scarcity, particularly in non-English languages. To address this, we fine-tune a voice conversion model on English dysarthric speech (UASpeech) to…

Contemporary conversational systems often present a significant limitation: their responses lack the emotional depth and disfluent characteristic of human interactions. This absence becomes particularly noticeable when users seek more…

计算与语言 · 计算机科学 2024-04-03 Rohan Chaudhury , Mihir Godbole , Aakash Garg , Jinsil Hwaryoung Seo

In real-world scenarios, speech signals are inevitably corrupted by various types of interference, making speech enhancement (SE) a critical task for robust speech processing. However, most existing SE methods only handle a limited range of…

声音 · 计算机科学 2025-12-12 Fei Liu , Yang Ai , Ye-Xin Lu , Rui-Chen Zheng , Hui-Peng Du , Zhen-Hua Ling

Zero-shot voice conversion aims to transform a source speech utterance to match the timbre of a reference speech from an unseen speaker. Traditional approaches struggle with timbre leakage, insufficient timbre representation, and mismatches…

声音 · 计算机科学 2024-11-18 Songting Liu

OpenAI's Whisper Automated Speech Recognition model excels in generalizing across diverse datasets and domains. However, this broad adaptability can lead to diminished performance in tasks requiring recognition of specific vocabularies.…

人工智能 · 计算机科学 2025-08-12 Vishakha Lall , Yisi Liu

A speaker extraction algorithm seeks to extract the target speaker's speech from a multi-talker speech mixture. The prior studies focus mostly on speaker extraction from a highly overlapped multi-talker speech mixture. However, the…

音频与语音处理 · 电气工程与系统科学 2022-09-01 Zexu Pan , Meng Ge , Haizhou Li

Given a piece of speech and its transcript text, text-based speech editing aims to generate speech that can be seamlessly inserted into the given speech by editing the transcript. Existing methods adopt a two-stage approach: synthesize the…

声音 · 计算机科学 2021-09-14 Chuanxin Tang , Chong Luo , Zhiyuan Zhao , Dacheng Yin , Yucheng Zhao , Wenjun Zeng

Universal Speech Enhancement (USE) aims to restore speech quality under diverse degradation conditions while preserving signal fidelity. Despite recent progress, key challenges in training target selection, the distortion--perception…

In real-world settings, speech signals are almost always affected by reverberation produced by the working environment; these corrupted signals need to be \emph{dereverberated} prior to performing, e.g., speech recognition, speech-to-text…

音频与语音处理 · 电气工程与系统科学 2021-10-06 Diego León , Felipe Tobar

Unsupervised representation learning of speech has been of keen interest in recent years, which is for example evident in the wide interest of the ZeroSpeech challenges. This work presents a new method for learning frame level…

音频与语音处理 · 电气工程与系统科学 2020-08-18 Mingjie Chen , Thomas Hain

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

Automated Speech Recognition shows superhuman performance for adult English speech on a range of benchmarks, but disappoints when fed children's speech. This has long sat in the way of child-robot interaction. Recent evolutions in…

计算与语言 · 计算机科学 2024-11-20 Ruben Janssens , Eva Verhelst , Giulio Antonio Abbo , Qiaoqiao Ren , Maria Jose Pinto Bernal , Tony Belpaeme

Real-time Automatic Speech Recognition (ASR) is a fundamental building block for many commercial applications of ML, including live captioning, dictation, meeting transcriptions, and medical scribes. Accuracy and latency are the most…

声音 · 计算机科学 2025-07-16 Atila Orhon , Arda Okan , Berkin Durmus , Zach Nagengast , Eduardo Pacheco

We introduce DiffuseST, a low-latency, direct speech-to-speech translation system capable of preserving the input speaker's voice zero-shot while translating from multiple source languages into English. We experiment with the synthesizer…

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