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.
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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
N. Wiener: Extrapolation, Interpolation, and Smoothing of Stationary Time Series (Wiley, New York 1949)
N. Wiener, E. Hopf: On a class of singular integral equations, Proc. Prussian Acad. Math.-Phys. Ser. (1931) p. 696
N. Levinson: The Wiener rms (root-mean-square) error criterion in filter design and prediction, J. Math. Phys. 25, 261-278 (1947)
T. Kailath: A view of three decades of linear filtering theory, IEEE Trans. Inf. Theory IT-20, 146-181 (1974)
S. Haykin: Adaptive Filter Theory, 4th edn. (Prentice Hall, Upper Saddle River 2002)
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)
J. Benesty, T. Gänsler: New insights into the RLS algorithm, EURASIP J. Appl. Signal Process. 2004, 331-339 (2004)
G.H. Golub, C.F. Van Loan: Matrix Computations (The Johns Hopkins Univ. Press, Baltimore 1996)
J.M.B. Dias, J.M.N. Leitão: Efficient computation of trTR −1 for Toeplitz matrices, IEEE Signal Process. Let. 9, 54-56 (2002)
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)
B. Widrow, M.E. Hoff Jr.: Adaptive switching circuits, IRE WESCON Conv. Rec. 4, 96-104 (1960)
A. Feuer, E. Weinstein: Convergence analysis of LMS filters with uncorrelated Gaussian data, IEEE Trans. Acoust. Speech ASSP 33, 222-230 (1985)
B. Widrow, S.D. Stearns: Adaptive Signal Processing (Prentice Hall, Englewood Cliffs 1985)
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)
R.H. Kwong, E.W. Johnston: A variable step size LMS algorithm, IEEE Trans. Signal Process. 40, 1633-1642 (1992)
V.J. Mathews, Z. Xie: A stochastic gradient adaptive filter with gradient adaptive step size, IEEE Trans. Signal Process. 41, 2075-2087 (1993)
J.B. Evans, P. Xue, B. Liu: Analysis and implementation of variable step size adaptive algorithms, IEEE Trans. Signal Process. 41, 2517-2535 (1993)
T. Aboulnasr, K. Mayyas: A robust variable step-size LMS-type algorithm: analysis and simulations, IEEE Trans. Signal Process. 45, 631-639 (1997)
D.I. Pazaitis, A.G. Constantinides: A novel kurtosis driven variable step-size adaptive algorithm, IEEE Trans. Signal Process. 47, 864-872 (1999)
A. Mader, H. Puder, G.U. Schmidt: Step-size control for acoustic echo cancellation filters - An overview, Signal Process. 80, 1697-1719 (2000)
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)
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)
J. Benesty, H. Rey, L.R. Vega, S. Tressens: A non-parametric VSS-NLMS algorithm, IEEE Signal Process. Lett. 13, 581-584 (2006), .
D.L. Duttweiler: Proportionate normalized least-mean-square adaptation in echo cancelers, IEEE Trans. Audio Speech 8, 508-518 (2000)
J. Benesty, S.L. Gay: An improved PNLMS algorithm, Proc. IEEE ICASSP (2002) pp. 1881-1884
S.L. Gay: An efficient fast converging adaptive filter for network echo cancellation, Proc. Assilomar Conf. 1, 394-398 (1998)
A. Gersho: Adaptive filtering with binary reinforcement, IEEE Trans. Inf. Theory IT-30, 191-199 (1984)
M.G. Bellanger: Adaptive Digital Filters and Signal Analysis (Marcel Dekker, New York 1987)
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)
N.J. Bershad: On the optimum data non-linearity in LMS adaptation, IEEE Trans. Acoust. Speech ASSP 34, 69-76 (1986)
R. Gray: On the asymptotic eigenvalue distribution of Toeplitz matrices, IEEE Trans. Inform. Theory IT-18, 725-730 (1972)
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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
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DOI: https://doi.org/10.1007/978-3-540-49127-9_6
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