alias FIRLS filter order
I have some audio files recorded in 48kHz sampling frequency. I will have to examine some audio characteristics and need to do that for lower sampling frequencies to see when they start to fail. I am going to test the downsampled audio files at 24kHz, 16kHz, 12kHz, and 8kHz.
I found MATLAB function resample(x,P,Q,N) . It's fairly easy to understand but there's one thing I'd like to ask. In description it says it applies an anti-alias FIRLS filter during the re-sampling process. This is understandable. I don't know what should I apply for N because the accuracy depends on the N parameter. What values should I use to obtain decent results in downsampling. Thanks for help.
Also, It says if I downsample from high to low sampling frequency I should do that in intermediate stages. Could someone also suggest something on this. Cheers! :)
You can even use the easier command y = decimate(x,r)
The documentation mentions "For better results when r is greater than 13, divide r into smaller factors and call decimate several times.", so with your highest factor of 6 times decimating you don't have to worry.
The default 8th order IIR and 30th order FIR seem sufficient to me, but if you doubt them you can plot your decimated signal over the original and increase the order to see the effects.
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