A continuous-time adaptive filter structure

• Published in: 1995 International Conference on Acoustics, Speech, and Signal Processing Vol. 2; pp. 1061 - 1064 vol.2
• Main Authors: Ortiz-Balbuena, L., Martinez-Gonzalez, A., Perez-Meana, H., Nino de Rivera, L., Ramirez-Angulo, J.
• Format: Conference Proceeding
• Language: English
• Published: IEEE 1995
• Subjects: Adaptive filters; Bandwidth; Computational complexity; Convergence; Digital signal processing; Digital systems; Least squares approximation; Signal processing algorithms; Signal synthesis; Transfer functions
• ISBN: 0780324315; 9780780324312
• ISSN: 1520-6149
• DOI: 10.1109/ICASSP.1995.480417

Adaptive filters have been traditionally developed in a digital environment which involves a large number of computations to derive the coefficients of the desired approximation. Most of the time, these calculations required a machine of great capacity and that is not practical for some applications like channel equalization in cellular systems. This paper proposes a continuous-time adaptive filter which is based on representing the impulse response of an adaptive filter as a linear combination of a set of orthogonal exponentials. An important practical advantage is that if a satisfactory representation can be obtained by exponentials simple filter structures can be synthesized. An analog adaptive filter structure that improve the time convergence of conventional realizations using a continuos time LMS algorithm to reduce the error between the reference system and the adaptive filter is shown.