A Square Root Normalized LMS Algorithm for Adaptive Identification with Non-Stationary Inputs

  • Alouane Monia Turki-Hadj (Department of Technologies del'Information et des Communications (TIC) at Ecole Nationale d'Ingenieurs de Tunis(ENIT))
  • Published : 2007.03.31

Abstract

The conventional normalized least mean square (NLMS) algorithm is the most widely used for adaptive identification within a non-stationary input context. The convergence of the NLMS algorithm is independent of environmental changes. However, its steady state performance is impaired during input sequences with low dynamics. In this paper, we propose a new NLMS algorithm which is, in the steady state, insensitive to the time variations of the input dynamics. The square soot (SR)-NLMS algorithm is based on a normalization of the LMS adaptive filter input by the Euclidean norm of the tap-input. The tap-input power of the SR-NLMS adaptive filter is then equal to one even during sequences with low dynamics. Therefore, the amplification of the observation noise power by the tap-input power is cancelled in the misadjustment time evolution. The harmful effect of the low dynamics input sequences, on the steady state performance of the LMS adaptive filter are then reduced. In addition, the square root normalized input is more stationary than the base input. Therefore, the robustness of LMS adaptive filter with respect to the input non stationarity is enhanced. A performance analysis of the first- and the second-order statistic behavior of the proposed SR-NLMS adaptive filter is carried out. In particular, an analytical expression of the step size ensuring stability and mean convergence is derived. In addition, the results of an experimental study demonstrating the good performance of the SR-NLMS algorithm are given. A comparison of these results with those obtained from a standard NLMS algorithm, is performed. It is shown that, within a non-stationary input context, the SR-NLMS algorithm exhibits better performance than the NLMS algorithm.

Keywords

References

  1. A. Gilloire, D. S. E. Mou1ines, and P. Duhamel, 'State of the art in acoustic echo cancellation,' in Digital Signal Processing in Telecommunications, Anibal R. Figueiras-Vidal(Ed.), 1996, pp. 45-91
  2. J. Benesty and Y. Huang, Adaptive Signal Processing: Applications to Real-World Problems. Springer-Verlag, 2003
  3. H.-C. Shin and A. H. Sayed, 'Mean-square performance of a family of affine projection algorithms,' IEEE Trans. Signal Process., vol. 52, pp. 90-102, Jan. 2004 https://doi.org/10.1109/TSP.2003.820077
  4. S. Haykin, Adaptive Filter Theory. Englewood Cliffs, NJ: Prentice-Hall, 2nd ed., 1991
  5. M. Tarrab and A. Feuer, 'Convergence and performance analysis of the normalized LMS algorithm with uncorrelated Gaussian data,' IEEE Trans. Inf. Theory, vol. 34, pp. 680-691, July 1988 https://doi.org/10.1109/18.9768
  6. M. T.-H. Alouane, R. Taleb, H. Arfa, and M. Jaidane-Saidanc, 'Analysis of the normalized stochastic gradient adaptive algorithm through modeling of speech non stationarities: Case study of acoustic echo cancellation,' in Proc. International Symp. on Computational Intelligence and Intelligent Informatics, Tunisia, May 2003
  7. T. Aboulnasr and K. Mayyas, 'A robust variable step-size LMS-type algorithm: Analysis and simulations,' IEEE Trans. Signal Process., vol. 45, pp. 631-639, Mar. 1997 https://doi.org/10.1109/78.558478
  8. J. Kim and A. D. Poularikas, 'Performance anaysis of the adjusted step size NLMS algorithm,' in Proc. 36th Southeastern Symp. on System Theory, 2004, pp. 467-471
  9. K. Dogancay and O. Tanrikulu, 'Adaptive filtering algorithms with selective partial updates,' IEEE Trans. Circuits Syst. II, Analog Digit. Signal Process., vol. 48, pp. 762-769, Aug. 2005 https://doi.org/10.1109/82.959866
  10. O. Macchi, Adaptive Signal Processing: The Least Mean Squares Approach with Application in Transmissions. New York: Wiley, 1995
  11. T. Hsia, 'Convergence analysis of LMS and NLMS adaptive algorithms,' in Proc. ICASSP, vol. 8, Feb. 1983, pp. 667-670
  12. W. Fong, J. Godzill, A. Doucet, and M. West, 'Monte carlo smoothing with application to audio signal enhancement,' IEEE Trans. Signal Process., vol. 50, pp. 438-449, Feb. 2002 https://doi.org/10.1109/78.978397