Downsampling and applying a lowpass filter to digital audio

Question

I\'ve got a 44Khz audio stream from a CD, represented as an array of 16 bit PCM samples. I\'d like to cut it down to an 11KHz stream. How do I do that? From my days of engine

Answer 1

Notice (in additions to the other comments) that the simple-easy-intuitive approach "downsample by a factor of 4 by replacing each group of 4 consecutive samples by the average value", is not optimal but is nevertheless not wrong, nor practically nor conceptually. Because the averaging amounts precisely to a low pass filter (a rectangular window, which corresponds to a sinc in frequency). What would be conceptually wrong is to just downsample by taking one of each 4 samples: that would definitely introduce aliasing.

By the way: practically any software that does some resampling (audio, image or whatever; example for the audio case: sox) takes this into account, and frequently lets you choose the underlying low-pass filter.