RE: [AudioBox] FW: another wild Richard idea.... just to drive you crazy!
> -----Original Message-----
> From:
AudioBox@yahoogroups.com
> [mailto:
AudioBox@yahoogroups.com] On Behalf Of jdow
> Sent: Sunday, October 15, 2006 1:42 AM
> To:
AudioBox@yahoogroups.com
> Subject: Re: [AudioBox] FW: another wild Richard idea....
> just to drive you crazy!
>
> Increasing the latency does not reduce CPU load materially.
> Arguably it increases the CPU load if it is done wrong.
> Reducing the latency to less than three or four packets does
> not allow for times the machine decides to do its own thing.
> But more than 4 packets starts to get silly and increases the
> computing challenge "slightly". The real problem is pushing
> the samples in from the source, through filters, delays,
> level shifts, matrix connection calculations, and on out the output.
>
The this is counter to all of my experiences with DAWs then. I don't
know how all this buffer stuff works well enough to really give
examples, but all I know is if my CPU is running out of gas with too
many plug-ins or software synths, etc.... on any of my DAW applications,
(i.e. I'm getting clicks, pops, drop outs, etc...) I can go into the
ASIO control panel, and turn up the buffers (ie. which adds latency) and
then get more out of the same CPU. I can get a lot more work out of the
same CPU with 1024 buffers than with 64, etc, etc....
> (And live is less work than clip because there is no file
> system overhead involved.) {^_^}
But live audio does have to pass through any buffers twice correct?
Once on the way in, and again on the way out?
Richard B. Ingraham
RBI Computers and Audio
http://www.rbicompaudio.20m.com/
rbI@...
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