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Titlebook: Speech Enhancement; Jacob Benesty,Shoji Makino,Jingdong Chen Book 2005 Springer-Verlag Berlin Heidelberg 2005 Interface.Signal.Universal M

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發(fā)表于 2025-3-21 17:56:04 | 只看該作者 |倒序?yàn)g覽 |閱讀模式
書(shū)目名稱(chēng)Speech Enhancement
編輯Jacob Benesty,Shoji Makino,Jingdong Chen
視頻videohttp://file.papertrans.cn/875/874002/874002.mp4
概述Describes the newest developments of this exciting field.Includes supplementary material:
叢書(shū)名稱(chēng)Signals and Communication Technology
圖書(shū)封面Titlebook: Speech Enhancement; Jacob Benesty,Shoji Makino,Jingdong Chen Book 2005 Springer-Verlag Berlin Heidelberg 2005 Interface.Signal.Universal M
描述.We live in a noisy world! In all applications (telecommunications, hands-free communications, recording, human-machine interfaces, etc.) that require at least one microphone, the signal of interest is usually contaminated by noise and reverberation. As a result, the microphone signal has to be "cleaned" with digital signal processing tools before it is played out, transmitted, or stored...This book is about speech enhancement. Different well-known and state-of-the-art methods for noise reduction, with one or multiple microphones, are discussed. By speech enhancement, we mean not only noise reduction but also dereverberation and separation of independent signals. These topics are also covered in this book. However, the general emphasis is on noise reduction because of the large number of applications that can benefit from this technology...The goal of this book is to provide a strong reference for researchers, engineers, and graduate students who are interested in the problem of signal and speech enhancement. To do so, we invited well-known experts to contribute chapters covering the state of the art in this focused field..
出版日期Book 2005
關(guān)鍵詞Interface; Signal; Universal Mobile Telecommunications System (UMTS); blind source separation; communica
版次1
doihttps://doi.org/10.1007/3-540-27489-8
isbn_softcover978-3-642-06317-6
isbn_ebook978-3-540-27489-6Series ISSN 1860-4862 Series E-ISSN 1860-4870
issn_series 1860-4862
copyrightSpringer-Verlag Berlin Heidelberg 2005
The information of publication is updating

書(shū)目名稱(chēng)Speech Enhancement影響因子(影響力)




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Sven Nordholm,Hai Quang Dam,Nedelko Grbi?,Siow Yong Low
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Tomohiro Nakatani,Masato Miyoshi,Keisuke Kinoshita
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1860-4862 udents who are interested in the problem of signal and speech enhancement. To do so, we invited well-known experts to contribute chapters covering the state of the art in this focused field..978-3-642-06317-6978-3-540-27489-6Series ISSN 1860-4862 Series E-ISSN 1860-4870
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Speech Enhancement: Application of the Kalman Filter in the Estimate-Maximize (EM) Framework,ntly estimated by applying a nonlinear extension to the Kalman filter, namely the .. Extensive experimental study, using real speech and noise signals is provided to compare the proposed methods with alternative speech enhancement algorithms. Kalman filter based algorithms are shown to maintain the
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發(fā)表于 2025-3-23 00:35:18 | 只看該作者
Adaptive Microphone Array Employing Spatial Quadratic Soft Constraints and Spectral Shaping,emes utilize the spatial, spectral, and temporal domains in an efficient and concise manner allowing a computational effective processing while maintaining high performance speech enhancement. Evaluations on the same data set, gathered from a car, show that the proposed schemes achieve good noise su
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發(fā)表于 2025-3-23 04:52:00 | 只看該作者
Single-Microphone Blind Dereverberation,atic speech recognition performance even when the reverberation time is as long as 1.0 sec provided a sufficiently large number of observed signals are available. Further discussions on several future directions are also provided with a view to extending HERB so that it can cope with more realistic
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發(fā)表于 2025-3-23 07:32:04 | 只看該作者
Separation and Dereverberation of Speech Signals with Multiple Microphones,e challenge lies in the coexistence of spatial interference from competing sources and temporal echoes due to room reverberation in the observed microphone signals. Focusing only on optimizing the signal-to-interference ratio is inadequate for most speech processing systems where source separation a
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