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Adaptive identification of acoustic multichannel systems using sparse representations[electronic resource] /

紀錄類型:	書目-語言資料,印刷品 : Monograph/item
杜威分類號:	621.3822
書名/作者:	Adaptive identification of acoustic multichannel systems using sparse representations/ by Karim Helwani.
作者:	Helwani, Karim.
出版者:	Cham : : Springer International Publishing :, 2015.
面頁冊數:	xiv, 113 p. : : ill. (some col.), digital ;; 24 cm.
Contained By:	Springer eBooks
標題:	Adaptive signal processing.
標題:	MIMO systems.
標題:	Engineering.
標題:	Signal, Image and Speech Processing.
標題:	Input/Output and Data Communications.
標題:	Communications Engineering, Networks.
標題:	Acoustics.
ISBN:	9783319089546 (electronic bk.)
ISBN:	9783319089539 (paper)
內容註:	Introduction -- Fundamentals of Adaptive Filter Theory -- Spatio-Temporal Regularized Recursive Least Squares Algorithm -- Sparse Representation of Multichannel Acoustic Systems -- Unique System Identification from Projections -- Geometrical Constraints -- Acoustic Echo Suppression -- Conclusion.
摘要、提要註:	This book treats the topic of extending the adaptive filtering theory in the context of massive multichannel systems by taking into account a priori knowledge of the underlying system or signal. The starting point is exploiting the sparseness in acoustic multichannel system in order to solve the non-uniqueness problem with an efficient algorithm for adaptive filtering that does not require any modification of the loudspeaker signals. The book discusses in detail the derivation of general sparse representations of acoustic MIMO systems in signal or system dependent transform domains. Efficient adaptive filtering algorithms in the transform domains are presented and the relation between the signal- and the system-based sparse representations is emphasized. Furthermore, the book presents a novel approach to spatially preprocess the loudspeaker signals in a full-duplex communication system. The idea of the preprocessing is to prevent the echoes from being captured by the microphone array in order to support the AEC system. The preprocessing stage is given as an exemplarily application of a novel unified framework for the synthesis of sound figures. Finally, a multichannel system for the acoustic echo suppression is presented that can be used as a postprocessing stage for removing residual echoes. As first of its kind, it extracts the near-end signal from the microphone signal with a distortionless constraint and without requiring a double-talk detector.
電子資源:	http://dx.doi.org/10.1007/978-3-319-08954-6

Adaptive identification of acoustic multichannel systems using sparse representations[electronic resource] /
Helwani, Karim.

Adaptive identification of acoustic multichannel systems using sparse representations[electronic resource] /by Karim Helwani. - Cham :Springer International Publishing :2015. - xiv, 113 p. :ill. (some col.), digital ;24 cm. - T-labs series in telecommunication services,2192-2810. - T-labs series in telecommunication services..

Introduction -- Fundamentals of Adaptive Filter Theory -- Spatio-Temporal Regularized Recursive Least Squares Algorithm -- Sparse Representation of Multichannel Acoustic Systems -- Unique System Identification from Projections -- Geometrical Constraints -- Acoustic Echo Suppression -- Conclusion.

This book treats the topic of extending the adaptive filtering theory in the context of massive multichannel systems by taking into account a priori knowledge of the underlying system or signal. The starting point is exploiting the sparseness in acoustic multichannel system in order to solve the non-uniqueness problem with an efficient algorithm for adaptive filtering that does not require any modification of the loudspeaker signals. The book discusses in detail the derivation of general sparse representations of acoustic MIMO systems in signal or system dependent transform domains. Efficient adaptive filtering algorithms in the transform domains are presented and the relation between the signal- and the system-based sparse representations is emphasized. Furthermore, the book presents a novel approach to spatially preprocess the loudspeaker signals in a full-duplex communication system. The idea of the preprocessing is to prevent the echoes from being captured by the microphone array in order to support the AEC system. The preprocessing stage is given as an exemplarily application of a novel unified framework for the synthesis of sound figures. Finally, a multichannel system for the acoustic echo suppression is presented that can be used as a postprocessing stage for removing residual echoes. As first of its kind, it extracts the near-end signal from the microphone signal with a distortionless constraint and without requiring a double-talk detector.

ISBN: 9783319089546 (electronic bk.)

Standard No.: 10.1007/978-3-319-08954-6doiSubjects--Topical Terms:

184330
Adaptive signal processing.

LC Class. No.: TK5102.9

Dewey Class. No.: 621.3822

Adaptive identification of acoustic multichannel systems using sparse representations[electronic resource] /
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