On Wed, Jan 1, 2014 at 1:21 PM, Joel Roth <joelz@xxxxxxxxx> wrote:
I was curious, if doubling the sample rate is a
practical way to reduce latency for live effects
processing. I would think it would reduce latency by half.
It would: you mention "practical", i'm not sure I'd call it that.
If one wanted to avoid the tradeoff of handling twice the
usual amount of audio data,
CPU load will go up, since there is 2x more of data to process,
which also means every plugin / host has 2x more work to do.
Adds up quickly if you're doing things like convolution reverbs
or other CPU intense processing..
I was curious if ALSA sample
rate conversion, or some other clever hack could be used to
get low latency advantage of the high sample rate, while
actually dealing with 48k streams through JACK.
Theoretically possible I suppose, it seems like an awful lot of
effort to get a few less ms latency..
Latency below ~3ms isn't percievable at all IMO: most will agree.
Why not run jack at 64 frames, 2 buffers? That'll achieve approx
3ms (on 44.1kHz and 48kHz).. which is fine for the purpose?
Perhaps I'm missing something, are you doing mulitple passes
trough the sound-card that you're adding its latency two or more times?
trough the sound-card that you're adding its latency two or more times?
Cheers, -Harry
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