DocumentCode :
813060
Title :
The analysis of the continuous-time LMS algorithm
Author :
Karni, Shlomo ; Zeng, Gengsheng
Author_Institution :
Dept. of Electr. & Comput. Eng., New Mexico Univ., Albuquerque, NM, USA
Volume :
37
Issue :
4
fYear :
1989
fDate :
4/1/1989 12:00:00 AM
Firstpage :
595
Lastpage :
597
Abstract :
A continuous-time analog adaptive filter is suggested using the digital prototype. The continuous-time LMS (least-mean squares) algorithm is then described by a set of simultaneous first-order equations. The adaptive gain is shown to be unbounded theoretically.<>
Keywords :
adaptive filters; filtering and prediction theory; least squares approximations; analogue adaptive filter; continuous-time LMS algorithm; least-mean squares; simultaneous first-order equations; Adaptive arrays; Adaptive filters; Algorithm design and analysis; Delay; Digital filters; Finite impulse response filter; Least squares approximation; Prototypes; Speech analysis; Transversal filters;
fLanguage :
English
Journal_Title :
Acoustics, Speech and Signal Processing, IEEE Transactions on
Publisher :
ieee
ISSN :
0096-3518
Type :
jour
DOI :
10.1109/29.17546
Filename :
17546
Link To Document :
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