This paper describes a study of the use of low order FIR filters for audio equalization. The filter frequency response is specified using a set of frequency samples and a fast algorithm is used to calculate its coefficients in real time. When the samples correspond to exactly equally spaced frequencies, a good filter is obtained. Audio equalization requires that the frequency samples are logarithmically spaces, and the direct approach gives rise to large errors. This effect is explained and an alternative approach considered. Results of simulation and real time performance analyses are also presented.
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