DocumentCode
2026620
Title
Wavelet-based adaptive filtering
Author
Doroslovacki, Milos ; Fan, Hong
Author_Institution
Cincinnati Univ., OH, USA
Volume
3
fYear
1993
fDate
27-30 April 1993
Firstpage
488
Abstract
Theoretical and experimental analysis and description of wavelet-based filtering are given in the case of a stationary desired signal. The impulse responses of the adaptive filter and the unknown system producing the desired signal are represented by discrete-time wavelet series. The authors have found the coefficients that minimize the mean square error and pointed out the time-frequency localized structure of the modeling error. An LMS (least mean square) adaptive filtering algorithm is derived. Its transform domain interpretation is shown, as are possibilities for faster convergence and better numerical properties. The authors have observed better modeling of desired signals in the time-frequency plane, faster convergence, and smaller error than in the case of FIR (finite impulse response) filters.<>
Keywords
adaptive filters; convergence of numerical methods; discrete time systems; least squares approximations; transient response; wavelet transforms; FIR filters; adaptive filtering; convergence; discrete-time wavelet series; impulse responses; least mean square; time-frequency localized structure; time-frequency plane; transform domain; wavelet-based filtering;
fLanguage
English
Publisher
ieee
Conference_Titel
Acoustics, Speech, and Signal Processing, 1993. ICASSP-93., 1993 IEEE International Conference on
Conference_Location
Minneapolis, MN, USA
ISSN
1520-6149
Print_ISBN
0-7803-7402-9
Type
conf
DOI
10.1109/ICASSP.1993.319541
Filename
319541
Link To Document