From a programming standpoint, there are some important considerations that need to be observed when changing the sampling rate of a discrete-time digital audio signal. I examine them here, and begin discussing the design of an anti-aliasing interpolating filter that is used in the process of resampling.
Aliasing occurs when there are frequencies present in a signal that are greater than the Nyquist limit (half of the sampling rate). What happens in such a case is that the sampling rate is not high enough to properly capture (sample) the high frequency of the signal, and so the frequency “folds over” and creates aliases that are mirror images of the original frequency.
