charlie@oakhill.UUCP (Charlie Thompson) (10/28/87)
This is a question for DSP jocks on the net...... I am designing a really sharp FIR filter for digital audio. The purpose of the filter is to prevent aliasing for a sample rate of 44.1 KHz. For 16 bits I need to be -96 dB at 22.05 KHz. I would like the passband to be 20.0 KHz. My question is this.... Which FIR design technique should work best??? a) Kaiser Window and Remez Exchange Algorithm b) Equi-ripple iterative approach. Dropping 96 dB in only 10% of the bandwidth is a REALLY sharp filter! It would also be nice to have less than .01 dB of passband ripple. I have a hunch as to which one works the best but I'd like to hear some opinions from someone else. Thanks. Charlie Thompson Austin,TX
hildum@iris.UUCP (10/28/87)
Hello. You might want to check out the Signetics SAA7030, which is designed for filtering the output of a CD system. This chip uses a 4x resampling system to allow a much cheaper (lower order) filter. Eric Hildum