• 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