6 Apr
2009
6 Apr
'09
3:19 p.m.
On Mon, Apr 6, 2009 at 9:20 AM, Luis Garrido
<luisgarrido(a)users.sourceforge.net> wrote:
The output buffer is not your only audio buffer, what we are dealing
with here is the size of the LADSPA processing buffer. This is a
simple matter of sending data from one buffer to another, as suggested
above, using a ring buffer is you most likely solution.
passed by
SDL_mixer to my sfx processing callback. 1880 samples at
sample rate 44100 (halved for 22050). I'm trying to understand how this
figure is arrived at, and if I can rely on it (before any audio
The buffer size you get from an audio backend is normally difficult to
predict, and I wouldn't advice you to do so. Sometimes it may even
depend on the hardware you are using, so it is not guaranteed that it
will be the same on another machine. I don't know whether SDL
encapsulates that for you, that would be more a question for SDL
support.
Most audio processing apps use ring buffers to account for that variability.
Just use any value you think provides you with a reasonable latency.
If you get more from the backend, just make several calls to the
LADSPA, it shouldn't mean much overhead. If you get less, just process
that bit or save it for the next iteration.
L
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