[linux-audio-dev] Beginner question about ring buffer and block processing
cournape at gmail.com
Tue May 24 04:07:39 UTC 2005
I try to get my head around a basic problem, and I don't manage to
get around it: how to adapt to variable sized blocks of samples given by
a ladspa or jack host/server to an existing, fixed length block
processing algorithm, for example a N points fft. AFAIK, the ring buffer
is the standart tool to solve this problem, but I don't understand how
this helps ?
I tried to read the source of jamin, which implements this kind of
scheme in io.c, but I couldn't manage to get the whole thing.
I would be glad to hear any hint about the right way to use
ringbuffer for this task,
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