Altera_Forum
Honored Contributor
18 years agoFIR Filter Explanation
I have a, hopefully straightforward, question that may aid in my understanding of filtering, and DSP in general.
Consider a system which instantiates an Alter FIR function. It is fed with an input stream from an Analog to Digital converter. The input is constrained to be a sine wave of 16 KHz. The A2D is clocked at 7.8 MHz and presents a new value on each clock cycle. The FIR is designed to pass frequencies from 10 KHz to 100 KHz and reject others. The FIR output is fed into a DAC and observed. Should I expect to see the sine wave? I do not, hence the question... My DSP knowledge is old and rusty, but do I need to transform this in any way, or do I feed this time domain signal directly into the FIR? Also, in creating the FIR, The MegaFunction Wizard askes for certain parameters. The sampling rate is one of them along with the cutoff frequencies. Just to be clear, am I right in using the 7.8 MHz value as the sampling frequency, even though it appears to be wildly oversampled? Thanks