oldhead wrote:

Also remember that your final product's going to be 16 bits. You record
at 24 precisely because that *does* give you room to play.


Excuse me for being green, but . . .

Does this mean that if I use an interface capable of a 192kHz sample
rate, the quality of my final track will be determioned by the sample
rates of any plug-ins or DSP effects I use?



The things that you are doing willingly *with* the
plugins will have a much greater impact than the
arithmetic overhead of the plugins themselves.

Unfortunately, I do not know that mixed sample
rates are a fact in any plugins. I know that the
VST SDK publishes the sample rate via a method in
the SDK, and I figure the coders have to make their
own choices. I do not use a DAW that allows mixed
sample rates; if I had one, I would not trust it
to do that correctly without considerable evidence.

There is only amplitude and time in plugins. Any phase
or frequency information is inferred by integration of
the current sample with the context of the samples that came
before. All those transforms pretty much have sample rate
as a "furniture" variable - a manifest constant, if you
will. A "#define".

I see no reason that a plugin with a lower sample rate should
work any less well than one with a higher sample rate - indeed,
it should just require smaller buffers. That really
depends on how much data the transforms need to be well behaved.
SFAIK, that ain't muc for most. Delay/amplitude transforms,
which include compression, echo and reverb require no context
beyond things like the time constants. EQ? That's a filters
thing, and I'm not qualified. I don't think an FFT filter
costs that much preload-data. And the sample rate would cancel out
with the time constants fo the filter, anyway...

I wish there was a published, well vetted metric of operational
quality for plugins, but there ain't. That's too bad...

--
Les Cargill