
Signal Enhancement with Variable Span Linear Filters
Springer (Publisher)
Published on 9. December 2018
Book
Paperback/Softback
IX, 172 pages
978-981-13-5709-1 (ISBN)
Description
This book introduces readers to the novel
concept of variable span speech enhancement filters, and demonstrates how it
can be used for effective noise reduction in various ways. Further, the book
provides the accompanying Matlab code, allowing readers to easily implement the
main ideas discussed. Variable span filters combine the ideas of optimal linear
filters with those of subspace methods, as they involve the joint
diagonalization of the correlation matrices of the desired signal and the
noise. The book shows how some well-known filter designs, e.g. the minimum
distortion, maximum signal-to-noise ratio, Wiener, and tradeoff filters (including
their new generalizations) can be obtained using the variable span filter
framework. It then illustrates how the variable span filters can be applied in
various contexts, namely in single-channel STFT-based enhancement, in
multichannel enhancement in both the time and STFT domains, and, lastly, in
time-domain binaural enhancement. In these contexts, the properties of these
filters are analyzed in terms of their noise reduction capabilities and desired
signal distortion, and the analyses are validated and further explored in
simulations.
More details
Series
Edition
Softcover reprint of the original 1st ed. 2016
Language
English
Place of publication
Singapore
Singapore
Target group
Professional and scholarly
Illustrations
25 s/w Abbildungen
IX, 172 p. 25 illus.
Dimensions
Height: 235 mm
Width: 155 mm
Thickness: 11 mm
Weight
289 gr
ISBN-13
978-981-13-5709-1 (9789811357091)
DOI
10.1007/978-981-287-739-0
Schweitzer Classification
Other editions
Additional editions

Jacob Benesty | Mads G. Christensen | Jesper R. Jensen
Signal Enhancement with Variable Span Linear Filters
Book
02/2016
Springer
€106.99
Shipment within 15-20 days
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.- General Concept with Filtering Vectors.- General Concept with Filtering Matrices.- Single-Channel Signal Enhancement in the STFT Domain.- Multichannel Signal Enhancement in the Time Domain.- Multichannel Signal Enhancement in the STFT Domain.- Binaural Signal Enhancement in the Time Domain.