RLS AND IIR ADAPTIVE FILTERS

Chair: John Treichler, Applied Signal Technology (USA)

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New Fast Inverse QR Least Squares Adaptive Algorithms

Authors:

A.A. Rontogiannis, University of Athens
S. Theodoridis, Computer Technology Institute (GREECE)

Volume 2, Page 1412

Abstract:

This paper presents two new, closely related adaptive algorithms for LS system identification. The starting point for the derivation of the algorithms is the inverse Cholesky factor of the data correlation matrix, obtained via a QR decomposition (QRD). Both are of $O(p)$ computational complexity with $p$ being the order of the system. The first algorithm is a fixed order QRD scheme with enhanced parallelism. The second is a lattice type algorithm based on Givens rotations, with lower omplexity compared to previously derived ones.

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Fast Recursive Eigensubspace Adaptive Filters

Authors:

Peter Strobach, Fachhochschule Furtwangen (GERMANY)

Volume 2, Page 1416

Abstract:

A class of adaptive filters based on sequential eigen decomposition of the data covariance matrix is presented. These new algorithms are completely rank-revealing and hence they can perfectly handle the following two relevant data cases where conventional RLS methods fail to provide satisfactory results: (1) Highly oversampled "smooth" data with rank deficient or almost rank deficient covariance matrix. (2) Noise-corrupted data where a signal must be separated effectively from superimposed noise. This paper corrects the widely held belief that eigen-based algorithms must be computationally more demanding than conventional RLS techniques. A spatial RLS adaptive filter has a principal complexity of O(N*N) operations per time step. Somewhat ironically, though, the corresponding new eigen subspace adaptive filter requires only O(N*r) operations per time stepwhere r

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A Highly Modular Normalized Adaptive Lattice Algorithm for Multichannel Least Squares Filtering

Authors:

George-Othon Glentis, University of Twente (THE NETHERLANDS)
Cornelis H. Slump, University of Twente (THE NETHERLANDS)

Volume 2, Page 1420

Abstract:

In this paper a highly modular normalized adaptive lattice algorithm for multichannel Least Squares FIR filtering and multivariable system identification, is presented. Multichannel filters with different number of delay elements per input channel are allowed. The main features of the proposed multichannel adaptive lattice least squares algorithm is the use of scalar only operations, multiplications/divisions and square roots, and the local communication which enables the development of a fully pipelinable architecture.

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Adaptive RLS Filters with Linear and Quadratic Constraints

Authors:

J. Tuthill, Curtin University of Technology (AUSTRALIA)
Y.H. Leung, Curtin University of Technology (AUSTRALIA)
I. Thng, Curtin University of Technology (AUSTRALIA)

Volume 2, Page 1424

Abstract:

In many adaptive signal processing applications, it is often desired to impose linear and quadratic constraints on the adaptive filter weights in order to meet certain performance criteria. This paper presents a modification of the well known adaptive RLS or FLS algorithm to achieve this. By way of illustration, the paper considers an adaptive narrowband beamformer with first and second order spatial derivative constraints. The performance of the algorithm is studied via computer simulations.

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Analysis of the QR-RLS Algorithm for Colored- Input Signals

Authors:

Paulo S.R. Diniz, Federal University of Rio de Janeiro (BRAZIL)
Marcio G. Siqueira, Federal University of Rio de Janeiro (BRAZIL)

Volume 2, Page 1428

Abstract:

A detailed analysis of the QR-RLS algorithm in finite and infinite precision implementations is presented, emphasizing the case where the input signal samples are correlated. The expressions for the mean square values of all internal variables in steady state are first derived. These expressions are key to determine the dynamic range of the internal signals, and to derive the analytical expressions for the mean square values of the deviations in the output variables of the algorithm in finite wordlength implementations. Previous works address this problem considering the input signal a white noise, a situation not so often encountered in practice. The accuracy of all analytical results are verified through a number of computer simulations.

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Estimation of Nonstationary AR Model Using the Weighted Recursive Least Square Algorithm

Authors:

M. Milosavljevic, University of Belgrade (YUGOSLAVIA)
M. Dj. Veinovi, University of Belgrade (YUGOSLAVIA)
Branko Kovacevic, University of Belgrade (YUGOSLAVIA)

Volume 2, Page 1432

Abstract:

In this paper a new method of estimation time-varying AR models using weihted recursive least square algorithm with variable forgetting factor is described. The variable forgetting factor is adapted to a nonstationary signal by a generalized likelihood ratio algorithm through so-called discrimination function which gives good measure of nonstationarity. In this way we connect results from nonstationary signal estimation and jump detection area and obtain algorithm which exhibits good tracking performance together with parameter estimation accuracy. The feasibility of the approach is demonstrated with both simulation data and real speech signals.

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Structural Issues in Cascade-form Adaptive IIR Filters

Authors:

Geoffrey A. Williamson, Illinois Institute of Technology
James P. Ashley, Motorola Inc.
Majid Nayeri, Michigan State University (USA)

Volume 2, Page 1436

Abstract:

Adaptive IIR filters implemented in cascade-form are attractive due to the ease with which their stability may be monitored. In this paper, four cascade-form structures are compared for use in adaptive filtering with respect to complexity of implementation, error surface geometry, and adaptation speed. The four structures include a cascade of second order pole/zero sections, a cascade of second order all-pole sections followed by a tapped delay line and two new structures introduced by this paper. The latter pair includes a tapped cascade, which is a cascade of second order all-pole sections whose output is constructed as a weighted combination of signals tapped from the cascade. The second new structure is a modification of the tapped cascade that yields orthogonal signals at the taps of the cascade. It is shown that the tapped cascade provides the best overall performance in the respects noted above.

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Robust Parameter Tracking Through Regional Forgetting

Authors:

Robert Shorten, Daimler-Benz Research (GERMANY)
Andreas Schutte, Daimler-Benz Research (GERMANY)
A.D. Fagan, Daimler-Benz Research (GERMANY)

Volume 2, Page 1440

Abstract:

The recursive least squares (RLS) algorithm with exponential forgetting ((lambda)RLS) is perhaps the best known and most widely used algorithm for tracking the time varying parameters of a linear regression model. The implicit assumption in using the (lambda)RLS algorithm is that the information is uniformly distributed over the time horizon. Frequently this assumption does not hold and seriousdifficulties can be encountered when using many model structures. These include convergence of the parameters to local system or noise characteristics and output bursting, i.e. a large error when the operating point changes. In this paper several simple alternatives to the standard (lambda)RLS algorithm are proposed.The proposed algorithms extend the idea of a sliding window by quantising the input space.These algorithms considerably reduce the risk of forgetting useful information and eliminate the possibility of output bursting by relating the adaptation capabilities of the algorithm to the amount of input stimul

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Adaptive Line Enhancement Using a Second-Order IIR Filter

Authors:

H.J.W. Belt, Eindhoven University of Technology (THE NETHERLANDS)
A.C. den Brinker, Eindhoven University of Technology (THE NETHERLANDS)
F.P.A. Benders, Eindhoven University of Technology (THE NETHERLANDS)

Volume 2, Page 1444

Abstract:

A second-order IIR filter is considered as the basic component of an Adaptive Line enhancer (ALE). As a new feature, the bandwidth of the proposed ALE is adapted simultaneously with the center frequency. This leads to the possibility to combine convergence speed and accuracy. The adaptation of the filter poles is controlled by a sign algorithm. The stepsizes are chosen such that transients caused by the retuning of the filter are ensured to remain much smaller in amplitude than the response of the filter to the input signal. When the input signal consists of a sinusoid corrupted by wide-band noise, an accurate frequency parameter estimate can be obtained with an algorithm given in this paper.

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