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can I do this (multiple update rates) with the apple audio environment?
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can I do this (multiple update rates) with the apple audio environment?


  • Subject: can I do this (multiple update rates) with the apple audio environment?
  • From: Cor Jansen <email@hidden>
  • Date: Thu, 18 Mar 2004 21:17:30 +0100

Hello all,

I'm new to this group. I subscribed because I want to try to find out if the Apple audio environment is
suitable for my project.
I want to write a additive sysnthesis module (audio unit?).
As far as I understand, audio is processed by doing calculations on an array of
samples (or more arrays if more inputs are used), producing arrays of output samples.
The problem is that for the additive synthesis project I want to build a need some other kind of processing.

First of all I need to generate the output signal by calculating the output of a (large) number of
sinewave oscillators. This can be done as mentioned above (producing an array of samples each time the unit
gets time to de the calculations). But now the problem. The sinewave oscillators have amplitude and frequency
enveloppes as input. The update rate of these envelops is much lower than the sample rate and in fact the update rate
depends on the controlled frequency. So the real sound processing of the module is not really related to the sample rate.
I do not want to process these envelops at intervals of a fixed number of samples.
And what is also important, I want the time used for the processing of the enveloppes to be flexible. It would be
wunderfull if I could make a system where I can do more processing only if time is available. In this way the sound quality
decreases gradually (and not abrupt) when there is a short on processing time.

I think what I would like most is that my module gets time each N samples to do the sinewave calculations plus the
processing of envelops that have highest priotiry (e.g. higher frequencies). Then my module would get each 2*N samples
the possibility to process other envelops. Each 4*N samples it should get time to process another set of enveloppes etc.
And I guess about 32*N samples it should process the enveloppes of the lowest frequencies.
It should be shuch that when processing for example the 32*N frame, this process may span the whole 32*N samples, thus the
processing ot the shorter frames should interupt the longer frames!!!

Is this possible with the apple audio core? I really hope so!
I hope I could make myself clear enough and someone could point me into the right direction how to do this on the mac.

Thanks, Cor Jansen.
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