
Optimal Time-Domain Noise Reduction Filters
A Theoretical Study
Springer (Publisher)
Published on 20. April 2011
Book
Paperback/Softback
VII, 79 pages
978-3-642-19600-3 (ISBN)
Description
Additive noise is ubiquitous in acoustics environments and can affect the intelligibility and quality of speech signals. Therefore, a so-called noise reduction algorithm is required to mitigate the effect of the noise that is picked up by the microphones. This work proposes a general framework in the time domain for the single and multiple microphone cases, from which it is very convenient to derive, study, and analyze all kind of optimal noise reduction filters. Not only that all known algorithms can be deduced from this approach, shedding more light on how they function, but new ones can be discovered as well.
Reviews / Votes
From the reviews:
"The book is devoted to the problem of recovering speech signals that are corrupted by additive noise. . The exposition is excellent. The authors work actively in the domain and recently issued also a book on microphone array signal processing. . the book represents a very useful guide to the noise reduction area for high-level researchers." (Dumitru Stanomir, Zentralblatt MATH, Vol. 1218, 2011)More details
Series
Edition
2011 ed.
Language
English
Place of publication
Berlin
Germany
Publishing group
Springer Berlin
Target group
Professional and scholarly
Research
Illustrations
1 farbige Abbildung
VII, 79 p. 1 illus. in color.
Dimensions
Height: 235 mm
Width: 155 mm
Thickness: 6 mm
Weight
149 gr
ISBN-13
978-3-642-19600-3 (9783642196003)
DOI
10.1007/978-3-642-19601-0
Schweitzer Classification
Other editions
Additional editions

E-Book
04/2011
1st Edition
Springer
€53.49
Available for download
Persons
Jacob Benesty received a Master's degree in microwaves from Pierre & Marie Curie University, France, in 1987, and a Ph.D. degree in control and signal processing from Orsay University, France, in April 1991. During his Ph.D. (from Nov. 1989 to Apr. 1991), he worked on adaptive filters and fast algorithms at the Centre National d'Etudes des Telecommunications (CNET), Paris, France. From January 1994 to July 1995, he worked at Telecom Paris University on multichannel adaptive filters and acoustic echo cancellation. From October 1995 to May 2003, he was first a Consultant and then a Member of the Technical Staff at Bell Laboratories, Murray Hill, NJ, USA. In May 2003, he joined the University of Quebec, INRS-EMT, in Montreal, Quebec, Canada, as a Professor. He is also a Visiting Professor at the Technion, Haifa, in Israel, an Adjunct Professor at Aalborg University, in Denmark, and a Guest Professor at Northwestern Polytechnical University, Xi'an, Shaanxi, in China.
Jingdong Chen received the Ph.D. degree in pattern recognition and intelligence control from the Chinese Academy of Sciences in 1998. From 1998 to 1999, he was with ATR Interpreting Telecommunications Research Laboratories, Kyoto, Japan, where he conducted research on speech synthesis, speech analysis, as well as objective measurements for evaluating speech synthesis. He then joined the Griffith University, Brisbane, Australia, where he engaged in research on robust speech recognition and signal processing. From 2000 to 2001, he worked at ATR Spoken Language Translation Research Laboratories on robust speech recognition and speech enhancement. From 2001 to 2009, he was a Member of Technical Staff at Bell Laboratories, Murray Hill, New Jersey, working on acoustic signal processing for telecommunications. He subsequently joined WeVoice Inc. in New Jersey, serving as the Chief Scientist. He is currently a professor at the Northwestern Polytechnical University in Xi'an, China. His research interests include acoustic signal processing, adaptive signal processing, speech enhancement, adaptive noise/echo control, microphone array signal processing, signal separation, and speech communication.
Chao Pan received the Bachelor degree in electronics and information engineering from the Northwestern Polytechnical University (NPU) in 2011. He is currently a Ph.D. student in information and communication engineering at NPU and also a visiting Ph.D. student at INRS-EMT, University of Quebec. His research interests are in speech enhancement, noise reduction, and microphone array signal processing for hands-free speech communications.
Content
Introduction.- Single-Channel Noise Reduction with a Filtering Vector.- Single-Channel Noise Reduction with a Rectangular Filtering Matrix.- Multichannel Noise Reduction with a Filtering Vector.- Multichannel Noise Reduction with a Rectangular Filtering Matrix.