The analysis of the continuous-time LMS algorithm

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.< >

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Bibliographic Details
Published inIEEE transactions on acoustics, speech, and signal processing Vol. 37; no. 4; pp. 595 - 597
Main Authors Karni, S., Zeng, G.
Format Journal Article
LanguageEnglish
Published IEEE 01.04.1989
Subjects
Adaptive arrays
Adaptive filters
Algorithm design and analysis
Delay
Digital filters
Finite impulse response filter
Least squares approximation
Prototypes
Speech analysis
Transversal filters
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ISSN0096-3518
DOI10.1109/29.17546

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Summary: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.< >
ISSN:0096-3518
DOI:10.1109/29.17546