Title :
Frequency domain non-linear adaptive filter
Author :
Mansour, David ; Gray, Augustine H., Jr.
Author_Institution :
University of California, Santa Barbara, CA
Abstract :
A new non-linear adaptive filter is presented. The algorithm is based on the Volterra series implemented in the frequency domain. For a finite memory of length N, the algorithm converges to the equivalent time domain non-linear adaptive filter as proposed by Roy and Sherman [1]. The frequency domain implementation offers a significant reduction in computation. For the second order Volterra series, the proposed algorithm requires O(N2) multiply-adds for N output points as opposed to an O(N3) for the time domain algorithm. For a large number of taps (≥ 32), the linear version of the proposed algorithm also offers a significant reduction in computation. Another advantage is its fast convergence to the Wiener solution due to the "pseudo-orthogonality" obtained by adapting in the frequency domain.
Keywords :
Acoustic signal processing; Adaptive filters; Biomedical engineering; Equalizers; Frequency domain analysis; Least squares approximation; Least squares methods; Linearity; Senior members; Signal processing algorithms;
Conference_Titel :
Acoustics, Speech, and Signal Processing, IEEE International Conference on ICASSP '81.
DOI :
10.1109/ICASSP.1981.1171263