Skip to main content

Part of the book series: Springer Handbooks ((SHB))

Abstract

The Wiener filter, named after its inventor, has been an extremely useful tool since its invention in the early 1930s. This optimal filter is not only popular in different aspects of speech processing but also in many other applications. This chapter presents the most fundamental results of the Wiener theory with an emphasis on the Wiener-Hopf equations, which are not convenient to solve in practice. An alternative approach to solving these equations directly is the use of an adaptive filter, which is why this work also describes the most classical adaptive algorithms that are able to converge, in a reasonable amount of time, to the optimal Wiener filter.

This is a preview of subscription content, log in via an institution to check access.

Access this chapter

Chapter
USD 29.95
Price excludes VAT (USA)
  • Available as PDF
  • Read on any device
  • Instant download
  • Own it forever
eBook
USD 579.00
Price excludes VAT (USA)
  • Available as EPUB and PDF
  • Read on any device
  • Instant download
  • Own it forever
Hardcover Book
USD 729.00
Price excludes VAT (USA)
  • Durable hardcover edition
  • Dispatched in 3 to 5 business days
  • Free shipping worldwide - see info

Tax calculation will be finalised at checkout

Purchases are for personal use only

Institutional subscriptions

Abbreviations

DFT:

discrete Fourier transform

FIR:

finite impulse response

IIR:

infinite impulse response

IPNLMS:

improved PNLMS

LMS:

least mean square

MIMO:

multiple-input multiple-output

MMSE:

minimum mean-square error

MSE:

mean-square error

NLMS:

normalized least-mean-square

PNLMS:

proportionate NLMS

SIMO:

single-input multiple-output

SISO:

single-input single-output

References

  1. N. Wiener: Extrapolation, Interpolation, and Smoothing of Stationary Time Series (Wiley, New York 1949)

    MATH  Google Scholar 

  2. N. Wiener, E. Hopf: On a class of singular integral equations, Proc. Prussian Acad. Math.-Phys. Ser. (1931) p. 696

    Google Scholar 

  3. N. Levinson: The Wiener rms (root-mean-square) error criterion in filter design and prediction, J. Math. Phys. 25, 261-278 (1947)

    Article  MathSciNet  Google Scholar 

  4. T. Kailath: A view of three decades of linear filtering theory, IEEE Trans. Inf. Theory IT-20, 146-181 (1974)

    Article  MathSciNet  MATH  Google Scholar 

  5. S. Haykin: Adaptive Filter Theory, 4th edn. (Prentice Hall, Upper Saddle River 2002)

    MATH  Google Scholar 

  6. J. Benesty, T. Gänsler: Computation of the condition number of a non-singular symmetric Toeplitz matrix with the Levinson-Durbin algorithm, IEEE Trans. Signal Process. 54, 2362-2364 (2006)

    Article  Google Scholar 

  7. J. Benesty, T. Gänsler: New insights into the RLS algorithm, EURASIP J. Appl. Signal Process. 2004, 331-339 (2004)

    Article  MathSciNet  MATH  Google Scholar 

  8. G.H. Golub, C.F. Van Loan: Matrix Computations (The Johns Hopkins Univ. Press, Baltimore 1996)

    MATH  Google Scholar 

  9. J.M.B. Dias, J.M.N. Leitão: Efficient computation of trTR −1 for Toeplitz matrices, IEEE Signal Process. Let. 9, 54-56 (2002)

    Article  Google Scholar 

  10. B. Widrow: Adaptive filters. In: Aspects of Network and System Theory, ed. by R.E. Kalman, N. DeClaris (Holt Rinehart and Winston, New York 1970)

    Google Scholar 

  11. B. Widrow, M.E. Hoff Jr.: Adaptive switching circuits, IRE WESCON Conv. Rec. 4, 96-104 (1960)

    Google Scholar 

  12. A. Feuer, E. Weinstein: Convergence analysis of LMS filters with uncorrelated Gaussian data, IEEE Trans. Acoust. Speech ASSP 33, 222-230 (1985)

    Article  Google Scholar 

  13. B. Widrow, S.D. Stearns: Adaptive Signal Processing (Prentice Hall, Englewood Cliffs 1985)

    MATH  Google Scholar 

  14. R.W. Harris, D.M. Chabries, F.A. Bishop: A variable step (VS) adaptive filter algorithm, IEEE Trans. Acoust. Speech ASSP 34, 309-316 (1986)

    Article  Google Scholar 

  15. R.H. Kwong, E.W. Johnston: A variable step size LMS algorithm, IEEE Trans. Signal Process. 40, 1633-1642 (1992)

    Article  MATH  Google Scholar 

  16. V.J. Mathews, Z. Xie: A stochastic gradient adaptive filter with gradient adaptive step size, IEEE Trans. Signal Process. 41, 2075-2087 (1993)

    Article  MATH  Google Scholar 

  17. J.B. Evans, P. Xue, B. Liu: Analysis and implementation of variable step size adaptive algorithms, IEEE Trans. Signal Process. 41, 2517-2535 (1993)

    Article  MATH  Google Scholar 

  18. T. Aboulnasr, K. Mayyas: A robust variable step-size LMS-type algorithm: analysis and simulations, IEEE Trans. Signal Process. 45, 631-639 (1997)

    Article  Google Scholar 

  19. D.I. Pazaitis, A.G. Constantinides: A novel kurtosis driven variable step-size adaptive algorithm, IEEE Trans. Signal Process. 47, 864-872 (1999)

    Article  MATH  Google Scholar 

  20. A. Mader, H. Puder, G.U. Schmidt: Step-size control for acoustic echo cancellation filters - An overview, Signal Process. 80, 1697-1719 (2000)

    Article  MATH  Google Scholar 

  21. H.-C. Shin, A.H. Sayed, W.-J. Song: Variable step-size NLMS and affine projection algorithms, IEEE Signal Process. Lett. 11, 132-135 (2004)

    Article  Google Scholar 

  22. D.R. Morgan, S.G. Kratzer: On a class of computationally efficient, rapidly converging, generalized NLMS algorithms, IEEE Signal Process. Lett. 3, 245-247 (1996)

    Article  Google Scholar 

  23. J. Benesty, H. Rey, L.R. Vega, S. Tressens: A non-parametric VSS-NLMS algorithm, IEEE Signal Process. Lett. 13, 581-584 (2006), .

    Article  Google Scholar 

  24. D.L. Duttweiler: Proportionate normalized least-mean-square adaptation in echo cancelers, IEEE Trans. Audio Speech 8, 508-518 (2000)

    Article  Google Scholar 

  25. J. Benesty, S.L. Gay: An improved PNLMS algorithm, Proc. IEEE ICASSP (2002) pp. 1881-1884

    Google Scholar 

  26. S.L. Gay: An efficient fast converging adaptive filter for network echo cancellation, Proc. Assilomar Conf. 1, 394-398 (1998)

    Google Scholar 

  27. A. Gersho: Adaptive filtering with binary reinforcement, IEEE Trans. Inf. Theory IT-30, 191-199 (1984)

    Article  MATH  Google Scholar 

  28. M.G. Bellanger: Adaptive Digital Filters and Signal Analysis (Marcel Dekker, New York 1987)

    MATH  Google Scholar 

  29. T. Claasen, W. Mecklenbrauker: Comparison of the convergence of two algorithms for adaptive FIR digital filters, IEEE Trans. Acoust. Speech ASSP 29, 670-678 (1981)

    Article  MATH  Google Scholar 

  30. N.J. Bershad: On the optimum data non-linearity in LMS adaptation, IEEE Trans. Acoust. Speech ASSP 34, 69-76 (1986)

    Article  Google Scholar 

  31. R. Gray: On the asymptotic eigenvalue distribution of Toeplitz matrices, IEEE Trans. Inform. Theory IT-18, 725-730 (1972)

    Article  MathSciNet  MATH  Google Scholar 

Download references

Author information

Authors and Affiliations

Authors

Corresponding authors

Correspondence to Jacob Benesty Prof. , Yiteng (Arden) Huang Dr. or Jingdong Chen Dr. .

Editor information

Editors and Affiliations

Rights and permissions

Reprints and permissions

Copyright information

© 2008 Springer-Verlag Berlin Heidelberg

About this chapter

Cite this chapter

Benesty, J., Huang, Y.(., Chen, J. (2008). Wiener and Adaptive Filters. In: Benesty, J., Sondhi, M.M., Huang, Y.A. (eds) Springer Handbook of Speech Processing. Springer Handbooks. Springer, Berlin, Heidelberg. https://doi.org/10.1007/978-3-540-49127-9_6

Download citation

  • DOI: https://doi.org/10.1007/978-3-540-49127-9_6

  • Publisher Name: Springer, Berlin, Heidelberg

  • Print ISBN: 978-3-540-49125-5

  • Online ISBN: 978-3-540-49127-9

  • eBook Packages: EngineeringEngineering (R0)

Publish with us

Policies and ethics