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Mandarin Chinese is characterized by being a tonal language; the pitch (or $F_0$) of its utterances carries considerable linguistic information. However, speech samples from different individuals are subject to changes in amplitude and…

In this paper, we present our overall efforts to improve the performance of a code-switching speech recognition system using semi-supervised training methods from lexicon learning to acoustic modeling, on the South East Asian…

计算与语言 · 计算机科学 2018-06-19 Pengcheng Guo , Haihua Xu , Lei Xie , Eng Siong Chng

Whispering is a distinct form of speech known for its soft, breathy, and hushed characteristics, often used for private communication. The acoustic characteristics of whispered speech differ substantially from normally phonated speech and…

音频与语音处理 · 电气工程与系统科学 2024-02-08 Zhaofeng Lin , Tanvina Patel , Odette Scharenborg

Fine-tuning pretrained ASR models for specific domains is challenging for small organizations with limited labeled data and computational resources. Here, we explore different data selection pipelines and propose a robust approach that…

Recently, cycle-consistent adversarial network (Cycle-GAN) has been successfully applied to voice conversion to a different speaker without parallel data, although in those approaches an individual model is needed for each target speaker.…

音频与语音处理 · 电气工程与系统科学 2018-06-26 Ju-chieh Chou , Cheng-chieh Yeh , Hung-yi Lee , Lin-shan Lee

Recent advances in audio generation led to an increasing number of deepfakes, making the general public more vulnerable to financial scams, identity theft, and misinformation. Audio deepfake detectors promise to alleviate this issue, with…

Training machine learning algorithms for speech applications requires large, labeled training data sets. This is problematic for clinical applications where obtaining such data is prohibitively expensive because of privacy concerns or lack…

音频与语音处理 · 电气工程与系统科学 2018-04-30 Yishan Jiao , Ming Tu , Visar Berisha , Julie Liss

Paralinguistic sounds, like laughter and sighs, are crucial for synthesizing more realistic and engaging speech. However, existing methods typically depend on proprietary datasets, while publicly available resources often suffer from…

音频与语音处理 · 电气工程与系统科学 2025-09-30 Bingsong Bai , Qihang Lu , Wenbing Yang , Zihan Sun , Yueran Hou , Peilei Jia , Songbai Pu , Ruibo Fu , Yingming Gao , Ya Li , Jun Gao

Dominant researches adopt supervised training for speaker extraction, while the scarcity of ideally clean corpus and channel mismatch problem are rarely considered. To this end, we propose speaker-aware mixture of mixtures training (SAMoM),…

音频与语音处理 · 电气工程与系统科学 2022-04-18 Zifeng Zhao , Rongzhi Gu , Dongchao Yang , Jinchuan Tian , Yuexian Zou

Speech separation aims to separate multiple speech sources from a speech mixture. Although speech separation is well-solved on some existing English speech separation benchmarks, it is worthy of more investigation on the generalizability of…

声音 · 计算机科学 2022-03-14 Kuan-Po Huang , Yuan-Kuei Wu , Hung-yi Lee

We present the OC16-CE80 Chinese-English mixlingual speech database which was released as a main resource for training, development and test for the Chinese-English mixlingual speech recognition (MixASR-CHEN) challenge on O-COCOSDA 2016.…

计算与语言 · 计算机科学 2016-09-28 Dong Wang , Zhiyuan Tang , Difei Tang , Qing Chen

We introduce a sophisticated multi-speaker speech data simulator, specifically engineered to generate multi-speaker speech recordings. A notable feature of this simulator is its capacity to modulate the distribution of silence and overlap…

音频与语音处理 · 电气工程与系统科学 2023-10-20 Tae Jin Park , He Huang , Coleman Hooper , Nithin Koluguri , Kunal Dhawan , Ante Jukic , Jagadeesh Balam , Boris Ginsburg

In this work, we propose a novel data augmentation method for clinical audio datasets based on a conditional Wasserstein Generative Adversarial Network with Gradient Penalty (cWGAN-GP), operating on log-mel spectrograms. To validate our…

声音 · 计算机科学 2025-02-11 Matthias Seibold , Armando Hoch , Mazda Farshad , Nassir Navab , Philipp Fürnstahl

In this paper, we focus on improving the performance of the text-dependent speaker verification system in the scenario of limited training data. The speaker verification system deep learning based text-dependent generally needs a large…

声音 · 计算机科学 2020-11-24 Xiaoyi Qin , Yaogen Yang , Lin Yang , Xuyang Wang , Junjie Wang , Ming Li

Speech language models (SpeechLMs) accept speech input and produce speech output, allowing for more natural human-computer interaction compared to text-based large language models (LLMs). Traditional approaches for developing SpeechLMs are…

计算与语言 · 计算机科学 2024-12-03 Aohan Zeng , Zhengxiao Du , Mingdao Liu , Lei Zhang , Shengmin Jiang , Yuxiao Dong , Jie Tang

Models that can handle a wide range of speakers and acoustic conditions are essential in speech emotion recognition (SER). Often, these models tend to show mixed results when presented with speakers or acoustic conditions that were not…

计算与语言 · 计算机科学 2022-01-11 Norbert Braunschweiler , Rama Doddipatla , Simon Keizer , Svetlana Stoyanchev

Recently, there has been an increasing focus on audio-text cross-modal learning. However, most of the existing audio-text datasets contain only simple descriptions of sound events. Compared with classification labels, the advantages of such…

声音 · 计算机科学 2024-03-08 Xuenan Xu , Xiaohang Xu , Zeyu Xie , Pingyue Zhang , Mengyue Wu , Kai Yu

This study presents a domain adaptation approach for speaker diarization targeting conversational Indonesian audio. We address the challenge of adapting an English-centric diarization pipeline to a low-resource language by employing…

Speech synthesis systems can now produce highly realistic vocalisations that pose significant authenticity challenges. Despite substantial progress in deepfake detection models, their real-world effectiveness is often undermined by evolving…

声音 · 计算机科学 2026-02-12 Qizhou Wang , Hanxun Huang , Guansong Pang , Sarah Erfani , Christopher Leckie

Recent research using pre-trained transformer models suggests that just 10 minutes of transcribed speech may be enough to fine-tune such a model for automatic speech recognition (ASR) -- at least if we can also leverage vast amounts of text…