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