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

You're right in that you need apply lowpass filtering on your signal. Any signal over 5500 Hz will be present in your downsampled signal but 'aliased' as another frequency so you'll have to remove those before downsampling.

It's a good idea to do the filtering with floats. There are fixed point filter algorithms too but those generally have quality tradeoffs to work. If you've got floats then use them!

Using DFT's for filtering is generally overkill and it makes things more complicated because dft's are not a contiuous process but work on buffers.

Digital filters generally come in two tastes. FIR and IIR. The're generally the same idea but IIF filters use feedback loops to achieve a steeper response with far less coefficients. This might be a good idea for downsampling because you need a very steep filter slope there.

Downsampling is sort of a special case. Because you're going to throw away 3 out of 4 samples there's no need to calculate them. There is a special class of filters for this called polyphase filters.

Try googling for polyphase IIR or polyphase FIR for more information.