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Automatic recognition of disordered speech remains a highly challenging task to date. Sources of variability commonly found in normal speech including accent, age or gender, when further compounded with the underlying causes of speech…

声音 · 计算机科学 2022-01-20 Mengzhe Geng , Shansong Liu , Jianwei Yu , Xurong Xie , Shoukang Hu , Zi Ye , Zengrui Jin , Xunying Liu , Helen Meng

Adapting speaker recognition systems to new environments is a widely-used technique to improve a well-performing model learned from large-scale data towards a task-specific small-scale data scenarios. However, previous studies focus on…

声音 · 计算机科学 2022-11-21 Zhenyu Wang , John H. L. Hansen

Text mismatch between pre-collected data, either training data or enrollment data, and the actual test data can significantly hurt text-dependent speaker verification (SV) system performance. Although this problem can be solved by carefully…

音频与语音处理 · 电气工程与系统科学 2025-08-07 Yexin Yang , Shuai Wang , Xun Gong , Yanmin Qian , Kai Yu

The prevalence of the powerful multilingual models, such as Whisper, has significantly advanced the researches on speech recognition. However, these models often struggle with handling the code-switching setting, which is essential in…

音频与语音处理 · 电气工程与系统科学 2024-01-15 Bobbi Aditya , Mahdin Rohmatillah , Liang-Hsuan Tai , Jen-Tzung Chien

In this paper, a novel architecture for speaker recognition is proposed by cascading speech enhancement and speaker processing. Its aim is to improve speaker recognition performance when speech signals are corrupted by noise. Instead of…

计算与语言 · 计算机科学 2020-05-25 Yanpei Shi , Qiang Huang , Thomas Hain

Data-intensive fine-tuning of speech foundation models (SFMs) to scarce and diverse dysarthric and elderly speech leads to data bias and poor generalization to unseen speakers. This paper proposes novel structured speaker-deficiency…

音频与语音处理 · 电气工程与系统科学 2024-12-30 Shujie Hu , Xurong Xie , Mengzhe Geng , Jiajun Deng , Zengrui Jin , Tianzi Wang , Mingyu Cui , Guinan Li , Zhaoqing Li , Helen Meng , Xunying Liu

In this paper, we propose a novel approach for the transcription of speech conversations with natural speaker overlap, from single channel speech recordings. The proposed model is a combination of a speaker diarization system and a hybrid…

音频与语音处理 · 电气工程与系统科学 2022-08-30 Srikanth Raj Chetupalli , Sriram Ganapathy

This study investigates the impact of integrating a dataset of disordered speech recordings ($\sim$1,000 hours) into the fine-tuning of a near state-of-the-art ASR baseline system. Contrary to what one might expect, despite the data being…

音频与语音处理 · 电气工程与系统科学 2025-12-22 Jimmy Tobin , Katrin Tomanek , Subhashini Venugopalan

Automatic speech recognition systems have undoubtedly advanced with the integration of multilingual and multitask models such as Whisper, which have shown a promising ability to understand and process speech across a wide range of…

计算与语言 · 计算机科学 2025-04-14 Xabier de Zuazo , Eva Navas , Ibon Saratxaga , Inma Hernáez Rioja

In the realm of automatic speech recognition (ASR), robustness in noisy environments remains a significant challenge. Recent ASR models, such as Whisper, have shown promise, but their efficacy in noisy conditions can be further enhanced.…

声音 · 计算机科学 2024-06-28 Yehoshua Dissen , Shiry Yonash , Israel Cohen , Joseph Keshet

Personalizing automatic speech recognition (ASR) systems for non-normative speech, such as dysarthric and aphasic speech, is challenging. While speaker-specific fine-tuning (SS-FT) is widely used, it is typically initialized directly from a…

声音 · 计算机科学 2026-03-17 Shan Jiang , Jiawen Qi , Chuanbing Huo , Yingqiang Gao , Qinyu Chen

Compared with automatic speech recognition (ASR), the human auditory system is more adept at handling noise-adverse situations, including environmental noise and channel distortion. To mimic this adeptness, auditory models have been widely…

计算与语言 · 计算机科学 2016-09-16 Peng Dai , Xue Teng , Frank Rudzicz , Ing Yann Soon

This paper presents, in the context of multi-channel ASR, a method to adapt a mask based, statistically optimal beamforming approach to a speaker of interest. The beamforming vector of the statistically optimal beamformer is computed by…

计算与语言 · 计算机科学 2018-06-21 Tobias Menne , Ralf Schlüter , Hermann Ney

In this paper, we address the problem of speaker verification in conditions unseen or unknown during development. A standard method for speaker verification consists of extracting speaker embeddings with a deep neural network and processing…

声音 · 计算机科学 2021-08-18 Luciana Ferrer , Mitchell McLaren , Niko Brummer

Despite the rapid progress of automatic speech recognition (ASR) technologies targeting normal speech in recent decades, accurate recognition of dysarthric and elderly speech remains highly challenging tasks to date. Sources of…

音频与语音处理 · 电气工程与系统科学 2022-03-18 Mengzhe Geng , Xurong Xie , Zi Ye , Tianzi Wang , Guinan Li , Shujie Hu , Xunying Liu , Helen Meng

Domain mismatch between training and testing can lead to significant degradation in performance in many machine learning scenarios. Unfortunately, this is not a rare situation for automatic speech recognition deployments in real-world…

计算与语言 · 计算机科学 2017-09-25 Wei-Ning Hsu , Yu Zhang , James Glass

Traditionally, research in automated speech recognition has focused on local-first encoding of audio representations to predict the spoken phonemes in an utterance. Unfortunately, approaches relying on such hyper-local information tend to…

音频与语音处理 · 电气工程与系统科学 2022-09-19 David M. Chan , Shalini Ghosh , Debmalya Chakrabarty , Björn Hoffmeister

In this work, we propose a novel variational Bayesian adaptive learning approach for cross-domain knowledge transfer to address acoustic mismatches between training and testing conditions, such as recording devices and environmental noise.…

音频与语音处理 · 电气工程与系统科学 2025-01-28 Hu Hu , Sabato Marco Siniscalchi , Chao-Han Huck Yang , Chin-Hui Lee

Speech enhancement in hearing aids remains a difficult task in nonstationary acoustic environments, mainly because current signal processing algorithms rely on fixed, manually tuned parameters that cannot adapt in situ to different users or…

This paper presents Conformer-1, an end-to-end Automatic Speech Recognition (ASR) model trained on an extensive dataset of 570k hours of speech audio data, 91% of which was acquired from publicly available sources. To achieve this, we…