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Attention-based encoder-decoder architectures such as Listen, Attend, and Spell (LAS), subsume the acoustic, pronunciation and language model components of a traditional automatic speech recognition (ASR) system into a single neural…

Developing fast and efficient algorithms for retrieval of objects to a given user query is an area of active research. The present study investigates retrieval of time series objects from a phoneme database to a given user pattern or query.…

定量方法 · 定量生物学 2007-12-28 Radhakrishnan Nagarajan , Anand Nagarajan , Mariofanna Milanova

Typical ASR systems segment the input audio into utterances using purely acoustic information, which may not resemble the sentence-like units that are expected by conventional machine translation (MT) systems for Spoken Language…

Automatic speech recognition (ASR) has benefited from advances in pretrained speech and language models, yet most systems remain constrained to monolingual settings and short, isolated utterances. While recent efforts in context-aware ASR…

计算与语言 · 计算机科学 2026-03-09 Yuchen Zhang , Haralambos Mouratidis , Ravi Shekhar

Automatic Speech Recognition (ASR) is an imperfect process that results in certain mismatches in ASR output text when compared to plain written text or transcriptions. When plain text data is to be used to train systems for spoken language…

计算与语言 · 计算机科学 2021-04-02 Prashant Serai , Vishal Sunder , Eric Fosler-Lussier

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…

Identifying mistakes (i.e., miscues) made while reading aloud is commonly approached post-hoc by comparing automatic speech recognition (ASR) transcriptions to the target reading text. However, post-hoc methods perform poorly when ASR…

机器学习 · 计算机科学 2025-05-30 Griffin Dietz Smith , Dianna Yee , Jennifer King Chen , Leah Findlater

Sequence-to-sequence (seq2seq) models are competitive with hybrid models for automatic speech recognition (ASR) tasks when large amounts of training data are available. However, data sparsity and domain adaptation are more problematic for…

计算与语言 · 计算机科学 2021-06-16 Chak-Fai Li , Francis Keith , William Hartmann , Matthew Snover , Owen Kimball

This paper proposes a new strategy for learning powerful cross-modal embeddings for audio-to-video synchronization. Here, we set up the problem as one of cross-modal retrieval, where the objective is to find the most relevant audio segment…

计算机视觉与模式识别 · 计算机科学 2020-11-05 Soo-Whan Chung , Joon Son Chung , Hong-Goo Kang

In a noisy environment, a lossy speech signal can be automatically restored by a listener if he/she knows the language well. That is, with the built-in knowledge of a "language model", a listener may effectively suppress noise interference…

机器学习 · 计算机科学 2019-07-03 Chien-Feng Liao , Yu Tsao , Xugang Lu , Hisashi Kawai

Speech representation and modelling in high-dimensional spaces of acoustic waveforms, or a linear transformation thereof, is investigated with the aim of improving the robustness of automatic speech recognition to additive noise. The…

计算与语言 · 计算机科学 2015-03-31 Matthew Ager , Zoran Cvetkovic , Peter Sollich

Generative models have excelled in audio tasks using approaches such as language models, diffusion, and flow matching. However, existing generative approaches for speech enhancement (SE) face notable challenges: language model-based methods…

音频与语音处理 · 电气工程与系统科学 2025-05-28 Ziqian Wang , Zikai Liu , Xinfa Zhu , Yike Zhu , Mingshuai Liu , Jun Chen , Longshuai Xiao , Chao Weng , Lei Xie

In this paper, we investigate the benefit that off-the-shelf word embedding can bring to the sequence-to-sequence (seq-to-seq) automatic speech recognition (ASR). We first introduced the word embedding regularization by maximizing the…

计算与语言 · 计算机科学 2020-02-06 Alexander H. Liu , Tzu-Wei Sung , Shun-Po Chuang , Hung-yi Lee , Lin-shan Lee

Audio deepfake detection aims to detect real human voices from those generated by Artificial Intelligence (AI) and has emerged as a significant problem in the field of voice biometrics systems. With the ever-improving quality of synthetic…

声音 · 计算机科学 2026-05-12 Aishwarya Fursule , Shruti Kshirsagar , Anderson R. Avila

Despite the impressive performance recently achieved by automatic speech recognition (ASR), we observe two primary challenges that hinder its broader applications: (1) The difficulty of introducing scalability into the model to support more…

音频与语音处理 · 电气工程与系统科学 2023-06-29 Zhongzhi Yu , Yang Zhang , Kaizhi Qian , Yonggan Fu , Yingyan Lin

In most automatic speech recognition (ASR) systems, the audio signal is processed to produce a time series of sensor measurements (e.g., filterbank outputs). This time series encodes semantic information in a speaker-dependent way. An…

声音 · 计算机科学 2019-05-10 David N. Levin

Mapping two modalities, speech and text, into a shared representation space, is a research topic of using text-only data to improve end-to-end automatic speech recognition (ASR) performance in new domains. However, the length of speech…

声音 · 计算机科学 2023-10-10 Jiaxu Zhu , Weinan Tong , Yaoxun Xu , Changhe Song , Zhiyong Wu , Zhao You , Dan Su , Dong Yu , Helen Meng

Attention-based sequence-to-sequence modeling provides a powerful and elegant solution for applications that need to map one sequence to a different sequence. Its success heavily relies on the availability of large amounts of training data.…

计算与语言 · 计算机科学 2021-02-12 Yun Tang , Juan Pino , Changhan Wang , Xutai Ma , Dmitriy Genzel

Modern Automatic Speech Recognition (ASR) systems can achieve high performance in terms of recognition accuracy. However, a perfectly accurate transcript still can be challenging to read due to disfluency, filter words, and other errata…

计算与语言 · 计算机科学 2021-02-23 Junwei Liao , Yu Shi , Ming Gong , Linjun Shou , Sefik Eskimez , Liyang Lu , Hong Qu , Michael Zeng

Speech recognition system performance degrades in noisy environments. If the acoustic models are built using features of clean utterances, the features of a noisy test utterance would be acoustically mismatched with the trained model. This…

计算与语言 · 计算机科学 2015-07-16 D. S. Pavan Kumar