Abstract
The adaptation process in digital filters requires extensive calculation. This computation makes adaptation a slow and time consuming process. Simple, but versatile, parallel algorithms for adaptive filters, suitable for VLSI implementation, are in demand. In this paper a regular and modular parallel algorithm for an adaptive filter is presented. This parallel structure is based on the Gradient Vector Estimation Algorithm, which minimizes the Mean Square Error. In the parallel method, the tap weights of the adaptive filter are updated everys steps, whereas in the recursive algorithms, the tap weights are updated at each step. Fors step update, bit strings of lengths are used to derive the terms with which the tap weights of the adaptive filter are calculated. The algorithm presented computes the tap weights at timen+s as a function of the tap weights at timen, the inputs from timen+1 ton+s−1, and the desired output from timen+1 ton+s−1. The algorithm also can be mapped to a VLSI architecture that is both regular and modular and allows either expansion of the order of the filter or the degree of parallelism obtainable. A comparison between the performance of the sequential LMS algorithm, Fast Exact LMS algorithm, and the parallel binary structured LMS algorithm is presented.
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References
M. Ishikawa, T. Tsukahara, and T. Kimura, “VLSI Architecture for An Adaptive Equalizer,”Proc. Intl. Conf. on Acoust., Speech, Signal Processing, Tokyo, Japan, 1986.
N. Kobayashi, H. Gambe, K. Aoki, M. Koshikawa, and T. Ikezawa, “An Adaptive Transversal Filter VLSI,”Proc. Intl. Conf. on Acoust., Speech, Signal Processing, Tokyo, Japan, 1986.
K.H. Cheng and S. Sahni, “A New VLSI System for Adaptive Recursive Filtering,”Parallel Computing, Vol. 10, No. 1, March 1989.
M. Yaminysharif and T.S. Durrani, “Adaptive Signal Processing Using a Modified Gradient Estimation Technique,”Proc. Intl. Conf. on Acoust., Speech, Signal Processing, Tokyo, Japan, 1986.
G. Carayannis, D.G. Manolakis, and N. Kalouptsidis, “A Fast Sequential Algorithm for Least-Squares Filtering and Prediction,”IEEE Trans. Acoust., Speech, Signal Processing, Vol. ASSP-31, 1983.
G.A. Clark, S.K. Mitra, and S.R. Parker, “Block Implementation of Adaptive Digital Filters,”IEEE Trans. Acoust., Speech, Signal Processing, Vol. ASSP-29, 1981.
K. Kurosawa and S. Tsujii, “A New Type Adaptive Algorithm of Parallel Type Structure,”Proc. Intl. Conf. on Acoust., Speech, Signal Processing, Tokyo, Japan, 1986.
K.K. Parhi and D.G. Messerschmitt, “Concurrent Cellular VLSI Adaptive Filter Architecture,”IEEE Trans. Circuits Syst., Oct. 1987.
K. Takahashi, “Performance improvement of LMS algorithm,”Electronics and Communication in Japan, Part 3, Vol. 75, No. 5, 1991.
S. Park, “A VLSI Architecture For Adaptive Signal Processing,”IEEE Proceedings of the Southeastcon, 1991.
J.A. Starzyk and M. Eshghi, “Highly Parallel Adaptive Filter,”Proc. Intl. Symp. on Circuits and Systems, Portland, Oregon, May 1989.
J. Benesty and P. Duhaamel, “A Fast Exact Least Mean Square Adaptive Algorithm,”IEEE Transactions on Signal Processing, Vol. 40, No. 12, 1992.
D.E. Borth, I.A. Grson, J.R. Haug, and C.D. Thompson, “Flexible Adaptive FIR filter VLSI IC,”IEEE Journal on Selected Areas in Communications, April 1988.
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Eshghi, M., DeGroat, J. A parallel binary structured LMS algorithm for transversal adaptive filters. Journal of VLSI Signal Processing 10, 127–140 (1995). https://doi.org/10.1007/BF02407031
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DOI: https://doi.org/10.1007/BF02407031