Abstract
The theory and performance of adaptive frequency selective filters is examined. The frequency sampling filter is a realization of a FIR filter as the cascade of an all-zero FIR filter with a bank of IIR digital resonators. The result of such a realization is that each coefficient can be directly identified with an amplitude of the transfer function at a particular frequency. The update method is the LMS algorithm, with the desired signal as a delayed version of the input. A discussion of the application of the adaptive frequency sampling filters to sub-band coding is included.<>