Tuyển tập báo cáo các nghiên cứu khoa học quốc tế ngành hóa học dành cho các bạn yêu hóa học tham khảo đề tài: Research Article Efficient Multichannel NLMS Implementation for Acoustic Echo Cancellation | Hindawi Publishing Corporation EURASIP Journal on Audio Speech and Music Processing Volume 2007 Article ID 78439 6 pages doi 2007 78439 Research Article Efficient Multichannel NLMS Implementation for Acoustic Echo Cancellation Fredric Lindstrom 1 Christian Schuldt 2 and Ingvar Claesson2 1KonftelAB Research and Development Box 268 90106 Umea Sweden 2 Department of Signal Processing Blekinge Institute of Technology 37225 Ronneby Sweden Received 31 May 2006 Revised 9 November 2006 Accepted 14 November 2006 Recommended by Kutluyil Dogancay An acoustic echo cancellation structure with a single loudspeaker and multiple microphones is from a system identification perspective generally modelled as a single-input multiple-output system. Such a system thus implies specific echo-path models adaptive filter for every loudspeaker to microphone path. Due to the often large dimensionality of the filters which is required to model rooms with standard reverberation time the adaptation process can be computationally demanding. This paper presents a selective updating normalized least mean square NLMS -based method which reduces complexity to nearly half in practical situations while showing superior convergence speed performance as compared to conventional complexity reduction schemes. Moreover the method concentrates the filter adaptation to the filter which is most misadjusted which is a typically desired feature. Copyright 2007 Fredric Lindstrom et al. This is an open access article distributed under the Creative Commons Attribution License which permits unrestricted use distribution and reproduction in any medium provided the original work is properly cited. 1. INTRODUCTION Acoustic echo cancellation AEC 1 2 is used in teleconferencing equipment in order to provide high quality fullduplex communication. The core of an AEC solution is an adaptive filter which estimates the impulse response of the loudspeaker enclosure microphone LEM system. Typical adaptive algorithms for
