Re: [LAU] Use of 96 kHz sample rate to lower latency

Harry van Haaren wrote:
> 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?

For a live submix, the routing I want to use is exemplified by:

system:capture_5 --> Nama:sax_in_1 --( ecasound )--> Nama:sax_out_1
Nama:sax_out_1 --( ecasound )--> system:playback_11

If I understand correctly, the latency not due to Ecasound
in this graph is the soundcard roundtrip plus the number of
hops which contribute (frames*buffers/sample-rate) latency
per.

I'm adding one more hop, so that would be 3ms
at 64/2, 4ms at 64/3, and 5.3ms at 128/2.

Regards,

Joel

> Cheers, -Harry

-- 
Joel Roth
  
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