[music-dsp] Interpolation

[Paul Maddox]

Paul,

> Now I'm getting confused too (I'll have to go back and read
> what David said he thinks you mean...) but I also was talking
> about a 24dB moog filter or whatever.

hehe..
I hate email, its way to confusing..

> If you have an interpolator, then it's reading a series of samples and 
> outputting a series of samples.

yep.

> Any series of samples can be filtered by any sort of filter,
> but the normal situation is: you interpolate between 2 (or more) waveform 
> samples to produce your oscillator output samples, which you then filter. 
> But you could also filter the waveform samples just before the 
> interpolator needs them, and interpolate between the current and previous 
> filter output.
> But then see what I wrote above.  Lowpass filtering before
> interpolation is a way to reduce aliasing, but if you want
> to use the filter for synthesis too it's probably not worth
> the added complexity.

Ahhh, ok, I don't mean using a filter to reduce aliasing, i'm talking about 
moving the point at which the interpolation is done to save DSP clock 
cycles, which means I can do more with the same processor.
if I can, I'd like to do the interpolation last, this way, instead of the 
filter processing a buffer of 16 samples per cycle of 'voice', it would only 
do two samples, then perform the interpolation at the end..

i'll have a play with putting the interpolation in various positions over 
the next week or so.
I'll keep you guys posted.

Paul