On Wed, Jan 1, 2014 at 1:21 PM, Joel Roth <joelz@email-addr-hidden> 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?
Cheers, -Harry
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Received on Wed Jan 1 16:15:13 2014
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