New transform-domain adaptive algorithms for acoustic echo cancellation

Abstract This paper proposes new transform-domain (TD) adaptive algorithms for long order adaptive filters. The scheme is based on decomposing the long adaptive filter into smaller subfilters. Each subfilter uses a different error signal in its update equation thus providing reduced coupling between subfilters and consequent improvement in convergence speed. We consider in this paper least mean-square adaptation of each subfilter along with power normalization, and derive an analytical model for the mean-square performance of the algorithm. Analysis assumes correlated Gaussian input data and will show that gains in convergence speed come at the expense of high misadjustment. A hybrid of the proposed algorithm and the TDLMS is described that offers significant performance improvements since it can be tuned to achieve very fast convergence while allowing low misadjustment. Performance advantages of the hybrid algorithm and theoretical results are verified by simulation examples.

[1]  George-Othon Glentis,et al.  Efficient least squares adaptive algorithms for FIR transversal filtering , 1999, IEEE Signal Process. Mag..

[2]  C. Richard Johnson,et al.  Parameter drift in LMS adaptive filters , 1986, IEEE Trans. Acoust. Speech Signal Process..

[3]  Jae Chon Lee,et al.  Performance of transform-domain LMS adaptive digital filters , 1986, IEEE Trans. Acoust. Speech Signal Process..

[4]  Dai I. Kim,et al.  Performance analysis of signed self-orthogonalizing adaptive lattice filter , 2000 .

[5]  Steven L. Gay,et al.  The fast affine projection algorithm , 1995, 1995 International Conference on Acoustics, Speech, and Signal Processing.

[6]  Scott C. Douglas Analysis of the multiple-error and block least-mean-square adaptive algorithms , 1995 .

[7]  A. Peterson,et al.  Transform domain LMS algorithm , 1983 .

[8]  K. Senne,et al.  Performance advantage of complex LMS for controlling narrow-band adaptive arrays , 1981 .

[9]  T. Kailath,et al.  Numerically stable fast transversal filters for recursive least squares adaptive filtering , 1991, IEEE Trans. Signal Process..

[10]  Gerhard Schmidt,et al.  Acoustic echo control. An application of very-high-order adaptive filters , 1999, IEEE Signal Process. Mag..

[11]  Françoise Beaufays,et al.  Transform-domain adaptive filters: an analytical approach , 1995, IEEE Trans. Signal Process..

[12]  Iven M. Y. Mareels,et al.  Quantifying the effects of dimension on the convergence rate of the LMS adaptive FIR estimator , 1998, IEEE Trans. Signal Process..

[13]  Masashi Tanaka,et al.  A block exact fast affine projection algorithm , 1999, IEEE Trans. Speech Audio Process..

[14]  Arie Feuer,et al.  Variable length stochastic gradient algorithm , 1991, IEEE Trans. Signal Process..

[15]  Ehud Weinstein,et al.  Convergence analysis of LMS filters with uncorrelated Gaussian data , 1985, IEEE Trans. Acoust. Speech Signal Process..

[16]  W. K. Jenkins,et al.  The use of orthogonal transforms for improving performance of adaptive filters , 1989 .

[17]  J. Shynk Frequency-domain and multirate adaptive filtering , 1992, IEEE Signal Processing Magazine.

[18]  A. K. Chaturvedi,et al.  A new family of concurrent algorithms for adaptive Volterra and linear filters , 1999, IEEE Trans. Signal Process..