Abstract
Adaptive filters have been traditionally developed in a digital environment which involves large number of computations to get the coefficients that make the desired approximation. Most of the time, this calculations required a great capacity machines 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 adaptive filter as a linear combination of a set of orthogonal exponentials. An important practical advantage of it is that if a satisfactory representation can be obtained by exponentials and simple filter structures can be synthesized.
Original language | English |
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Pages (from-to) | 1061-1064 |
Number of pages | 4 |
Journal | ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings |
Volume | 2 |
State | Published - 1995 |
Externally published | Yes |
Event | Proceedings of the 1995 20th International Conference on Acoustics, Speech, and Signal Processing. Part 2 (of 5) - Detroit, MI, USA Duration: 9 May 1995 → 12 May 1995 |