DocumentCode
1246545
Title
Slow asymptotic convergence of LMS acoustic echo cancelers
Author
Morgan, Dennis R.
Author_Institution
Dept. of Acoust. Res., AT&T Bell Labs., Murray Hill, NJ, USA
Volume
3
Issue
2
fYear
1995
fDate
3/1/1995 12:00:00 AM
Firstpage
126
Lastpage
136
Abstract
In most acoustic echo canceler (AEC) applications, an adaptive finite impulse response (FIR) filter is employed with coefficients that are computed using the LMS algorithm. The paper establishes a theoretical basis for the slow asymptotic convergence that is often noted in practice for such applications. The analytical approach expresses the mean-square error trajectory in terms of eigenmodes and then applies the asymptotic theory of Toeplitz matrices to obtain a solution that is based on a general characterization of the actual room impulse response. The method leads to good approximations even for a moderate number of taps (N>16) and applies to both full-band and subband designs. Explicit mathematical expressions of the mean-square error convergence are derived for bandlimited white noise, a first-order Markov process, and, more generally, pth-order rational spectra and a direct power-law model, which relates to lowpass FIR filters. These expressions show that the asymptotic convergence is generally slow, being at best of order 1/t for bandlimited white noise. It is argued that input filter design cannot do much to improve slow convergence. However, the theory suggests postfiltering as a remedy that would be useful for the full-band LMS AEC and may also be applicable to subband designs
Keywords
FIR filters; Markov processes; Toeplitz matrices; acoustic signal processing; adaptive filters; convergence of numerical methods; echo suppression; eigenvalues and eigenfunctions; least mean squares methods; low-pass filters; white noise; LMS acoustic echo cancelers; Toeplitz matrices; adaptive finite impulse response filter; asymptotic convergence; bandlimited white noise; direct power-law model; eigenmodes; first-order Markov process; full-band designs; lowpass FIR filters; mean-square error convergence; mean-square error trajectory; postfiltering; pth-order rational spectra; room impulse response; subband designs; Acoustic applications; Adaptive filters; Convergence; Finite impulse response filter; Least squares approximation; Loudspeakers; Markov processes; Microphones; Speech processing; White noise;
fLanguage
English
Journal_Title
Speech and Audio Processing, IEEE Transactions on
Publisher
ieee
ISSN
1063-6676
Type
jour
DOI
10.1109/89.366547
Filename
366547
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