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A family of proportionate normalized subband adaptive filter algorithms. (English) Zbl 1208.94026
Summary: The concept of proportionate adaptation is extended to the normalized subband adaptive filter (NSAF), and seven proportionate normalized subband adaptive filter algorithms are established. The proposed algorithms are proportionate normalized subband adaptive filter (PNSAF), \(\mu \)-law PNSAF (MPNSAF), improved PNSAF (IPNSAF), the improved IPNSAF (IIPNSAF), the set-membership IPNSAF (SM-IPNSAF), the selective partial update IPNSAF (SPU-IPNSAF), and SM-SPU-IPNSAF which are suitable for sparse system identification in network echo cancellation. When the impulse response of the echo path is sparse, the PNSAF has initial faster convergence than NSAF but slows down dramatically after initial convergence. The MPNSAF algorithm has fast convergence speed during the whole adaptation. The IPNSAF algorithm is suitable for both sparse and dispersive impulse responses. The SM-IPNSAF exhibits good performance with significant reduction in the overall computational complexity compared with the ordinary IPNSAF. In SPU-IPNSAF, the filter coefficients are partially updated rather than the entire filter at every adaptation. In SM-SPU-IPNSAF algorithm, the concepts of SM and SPU are combined which leads to a reduction in computational complexity. The simulation results show good performance of the proposed algorithms.

MSC:
94A12 Signal theory (characterization, reconstruction, filtering, etc.)
93E11 Filtering in stochastic control theory
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[1] Widrow, B.; Stearns, S.D., Adaptive signal processing, (1985), Prentice-Hall Englewood Cliffs, NJ · Zbl 0593.93063
[2] Spriet, A.; Rombouts, G.; Moonen, M.; Wouters, J., Adaptive feedback cancellation in hearing aids, Journal of the franklin institute, 343, 6, 545-573, (2006) · Zbl 1104.93032
[3] Haykin, S., Adaptive filter theory, (2002), Prentice-Hall NJ
[4] Diniz, P.S.R., Adaptive filtering: algorithms and practical implementation, (2002), Kluwer · Zbl 1145.93300
[5] Sayed, A.H., Fundamentals of adaptive filtering, (2003), Wiley
[6] Muneyasu, M.; Hinamoto, T., A realization of TD adaptive filters using affine projection algorithm, Journal of the franklin institute, 335, 7, 1185-1193, (1998) · Zbl 0996.93051
[7] Basin, M.; Calderon-Alvarez, D., Optimal filtering over linear observations with unknown parameters, Journal of the franklin institute, 347, 6, 988-1000, (2010) · Zbl 1201.93121
[8] Pai, M.C., Design of adaptive sliding mode controller for robust tracking and model following, Journal of the franklin institute, 347, 10, 1837-1849, (2010) · Zbl 1214.93029
[9] Alanis, A.Y.; Sanchez, E.N.; Loukianov, A.G.; Hernandez, E.A., Discrete-time recurrent high order neural networks for nonlinear identification, Journal of the franklin institute, 347, 7, 1253-1256, (2010) · Zbl 1202.93082
[10] Tellez, F.O.; Loukianov, A.G.; Sanchez, E.N.; Corrochano, E.J.B., Decentralized neural identification and control for uncertain nonlinear systems: application to planar robot, Journal of the franklin institute, 347, 7, 1015-1034, (2010) · Zbl 1201.93128
[11] Zhengdao, H.S.Z., A new method for fault prediction of model-unknown nonlinear system, Journal of the franklin institute, 345, 2, 136-153, (2008) · Zbl 1167.93331
[12] Duttweiler, D.L., Proportionate normalized least-Mean-squares adaptation in echo cancelers, IEEE transactions on speech and audio processing, 8, 508-518, (2000)
[13] S.L. Gay, An efficient, fast converging adaptive filter for network echo cancellation, in: Proceedings of the Asilomar Conference on Signals, Systems, and Computers, Pacific Grove, California, USA, November 1998, pp. 394-398.
[14] J. Benesty, S.L. Gay, An improved PNLMS algorithm, in: Proceedings of the ICASSP-02, Orlando, USA, May 2002, pp. 1881-1884.
[15] Deng, H.; Doroslovacki, M., Improving convergence of the PNLMS algorithm for sparse impluse response identification, IEEE signal processing letters, 12, 181-184, (2005)
[16] Deng, H.; Doroslovacki, M., Proportionate adaptive algorithms for network echo cancellation, IEEE transactions on signal processing, 54, 1794-1803, (May 2006)
[17] Naylor, P.A.; Cui, J.; Brookes, M., Adaptive algorithms for sparse echo cancellation, Signal processing, 86, 1182-1192, (2006) · Zbl 1163.94366
[18] O. Hoshuyama, R.A. Goubran, A. Sugiyama, A generalized proportionate variable step-size algorithm for fast changing acoustic environments, in: Proceedings of the ICASSP-04, Montreal, Quebec, Canada, May 2004, pp. 161-164.
[19] S.C. Douglas, Analysis and implementation of the max-NLMS adaptive filter, in: Proceedings of the 29th Asilomar Conference on Signals, Systems, and Computers, Pacific Grove, CA, October 1995, pp. 659-663.
[20] T. Aboulnasr, K. Mayyas, Selective coefficient update of gradient-based adaptive algorithms, in: Proceedings of the IEEE International Conference on Acoustics, Speech, and Signal Processing, Munich, Germany, April 1997, pp. 1929-1932.
[21] Aboulnasr, T.; Mayyas, K., Complexity reduction of the NLMS algorithm via selective coefficient update, IEEE transactions on signal processing, 47, 5, 1421-1424, (1999)
[22] T. Schertler, Selective block update NLMS type algorithms, in: Proceedings of the IEEE International Conference on Acoustics, Speech, and Signal Processing, Seattle, WA, May 1998, pp. 1717-1720.
[23] Doğançay, K.; Tanrıkulu, O., Adaptive filtering algorithms with selective partial updates, IEEE transactions on circuits and systems II: analog and digital signal processing, 48, 8, 762-769, (2001) · Zbl 1010.93540
[24] Werner, S.; de Campos, M.L.R.; Diniz, P.S.R., Partial-update NLMS algorithms with data-selective updating, IEEE transactions on signal processing, 52, 4, 938-948, (2004) · Zbl 1369.94315
[25] Doğançay, K., Complexity considerations for transform-domain adaptive filters, Signal processing, 83, 1177-1192, (2003) · Zbl 1144.93374
[26] Abadi, M.S.E.; Husøy, J.H., Selective partial update and set-membership subband adaptive filters, Signal processing, 88, 2463-2471, (2008) · Zbl 1151.94333
[27] Gollamudi, S.; Nagaraj, S.; Kapoor, S.; Huang, Y.F., Set-membership filtering and a set-membership normalized LMS algorithm with an adaptive step-size, IEEE signal processing letters, 5, 111-114, (1998)
[28] Werner, S.; Diniz, P.S.R., Set-membership affine projection algorithm, IEEE signal processing letters, 8, 231-235, (2001)
[29] P.S.R. Diniz, R.P. Braga, S. Werner, Set-membership affine projection algorithm for echo cancellation, in: Proceedings of the ISCAS, Island of Kos, Greece, May 2006, pp. 405-408.
[30] Diniz, P.S.R.; Werner, S., Set-membership binormalized data-reusing LMS algorithms, IEEE transactions on signal processing, 51, 124-134, (2003) · Zbl 1369.94130
[31] Werner, S.; de Campos, M.L.R.; Diniz, P.S.R., Partial-update NLMS algorithms with data-selective updating, IEEE transactions on signal processing, 52, 938-949, (2004) · Zbl 1369.94315
[32] S. Werner, J.A. Apolinario, P.S.R. Diniz, Set-membership proportionate affine projection algorithms, EURASIP Journal on Audio, Speech, and Music Processing (2007) 34242 (10pp). doi:10.1155/2007/34242.
[33] de Courville, M.; Duhamel, P., Adaptive filtering in subbands using a weighted criterion, IEEE transactions on signal processing, 46, 2359-2371, (1998)
[34] Lee, K.A.; Gan, W.S., Improving convergence of the NLMS algorithm using constrained subband updates, IEEE signal processing letters, 11, 736-739, (2004)
[35] Mayyas, K.; Aboulnasr, T., A fast exact weighted subband adaptive algorithm and its application to mono and stereo acoustic echo cancellation, Journal of the franklin institute, 342, 3, 235-253, (2005) · Zbl 1063.93504
[36] Pradhan, S.S.; Reddy, V.E., A new approach to subband adaptive filtering, IEEE transactions on signal processing, 47, 655-664, (1999)
[37] Shynk, J.J., Frequency domain and multirate adaptive filtering, IEEE signal processing magazine, 9, 14-37, (1992)
[38] Gilloire, A.; Vetterli, M., Adaptive filtering in subbands with critical sampling: analysis, experiments, and application to acoustic echo cancellation, IEEE transactions on signal processing, 40, 1862-1875, (1992) · Zbl 0775.93251
[39] Husøy, J.H.; Abadi, M.S.E., Unified approach to adaptive filters and their performance, IET signal processing, 2, 97-109, (2008)
[40] D.E. Knuth, Sorting and Searching of The Art of Computer Programming, second ed., vol. 3, Addison-Wesley, Reading, MA, 1973. · Zbl 0302.68010
[41] Malvar, H., Signal processing with lapped transforms, (1992), Artech House · Zbl 0948.94505
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