A novel normalized wavelet domain least-mean-square (LMS) algorithm is described. The faster convergence rate of this algorithm as compared with time-domain LMS is established. The wavelet domain LMS algorithm requires only real arithmetic. In its most basic form it has a computational complexity that is higher than that of the traditional LMS technique by a factor of cN, where N is the length of the transformed vector (or sliding analysis window) and c is the length of the analysis wavelet. Other preconditioning strategies that yield a faster convergence rate for a given fixed excess mean squared error are discussed. The authors also describe low-complexity implementations of the wavelet domain LMS algorithm. These implementations exploit the structure of the wavelet transform of the underlying stochastic process.<<ETX>>
[1]
Martin Vetterli,et al.
Adaptive filtering in sub-bands
,
1988,
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing.
[2]
Stéphane Mallat,et al.
Multifrequency channel decompositions of images and wavelet models
,
1989,
IEEE Trans. Acoust. Speech Signal Process..
[3]
J. Meijerink,et al.
An iterative solution method for linear systems of which the coefficient matrix is a symmetric -matrix
,
1977
.
[4]
Ronald R. Coifman,et al.
Signal processing and compression with wavelet packets
,
1994
.
[5]
A. Peterson,et al.
Transform domain LMS algorithm
,
1983
.