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FIR Digital Filter Design

FFT processors implement long FIR filters more efficiently than any other method, thanks to the speed of the FFT algorithm and the Fourier convolution theorem2.3.5). The convolution theorem states that the convolution of an input signal $ x$ with a filter impulse response $ h$ is given by the inverse DTFT of the signal's spectrum $ X(\omega)$ and the filter's frequency response $ H(\omega)$. Of course, in practice the DTFT is used in sampled form, replacing it with a (zero-padded) FFT. To make the most of FFT processors for FIR filter implementation, we need flexible ways to design all kinds of FIR filters for use in such FFT processors.

This appendix provides a starting point in the area of FIR digital filter design. The so-called ``window method'' for FIR filter design is discussed in some detail, and it is compared with the optimal Chebyshev method. Other methods, such as least-squares, are discussed briefly to provide some perspective. Tools for FIR filter design in both Octave and the Matlab Signal Processing Tool Box are listed where applicable. For further information on digital filter design, see the documentation for the Matlab Signal Processing Toolbox and/or [193,258,212,187,241].



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About the Author: Julius Orion Smith III
Julius Smith's background is in electrical engineering (BS Rice 1975, PhD Stanford 1983). He is presently Professor of Music and Associate Professor (by courtesy) of Electrical Engineering at Stanford's Center for Computer Research in Music and Acoustics (CCRMA), teaching courses and pursuing research related to signal processing applied to music and audio systems. See http://ccrma.stanford.edu/~jos/ for details.


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