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

Previous message: [thread] [date] [author]
Next message: [thread] [date] [author]
To: Joel Roth <joelz@...>
Cc: Linux Audio User <linux-audio-user@...>
Date: Wednesday, January 1, 2014 - 1:56 pm

--001a113490fa12fbdf04eee90c83
Content-Type: text/plain; charset=ISO-8859-1

On Wed, Jan 1, 2014 at 1:21 PM, Joel Roth wrote:

> I was curious, if doubling the sample rate is a

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

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

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

--001a113490fa12fbdf04eee90c83
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

On W=
ed, Jan 1, 2014 at 1:21 PM, Joel Roth <joelz@pobox.com> 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 no=
t sure I'd call it that.=A0

If one wanted to avoid the tradeoff of handling twice the
usual amount of audio data, CPU load will go up, sin=
ce there is 2x more=A0 of data to process,which also means e=
very plugin / host has=A0 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.=A0The=
oretically 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 ap=
prox3ms (on 44.1kHz and 48kHz).. which is fine for the purp=
ose?Perhaps I'm missing something, are you doing mul=
itple passes
trough the sound-card that you're adding its latency two or more times?=
Cheers, -Harry

--001a113490fa12fbdf04eee90c83--

Previous message: [thread] [date] [author]
Next message: [thread] [date] [author]

Messages in current thread:
[LAU] Use of 96 kHz sample rate to lower latency, Joel Roth, (Wed Jan 1, 1:21 pm)
Re: [LAU] Use of 96 kHz sample rate to lower latency, Fons Adriaensen, (Wed Jan 1, 7:48 pm)
Re: [LAU] Use of 96 kHz sample rate to lower latency, Harry van Haaren, (Wed Jan 1, 9:33 pm)
Re: [LAU] Use of 96 kHz sample rate to lower latency, Harry van Haaren, (Wed Jan 1, 1:56 pm)
Re: [LAU] Use of 96 kHz sample rate to lower latency, Bill Gribble, (Thu Jan 2, 11:50 am)
Re: [LAU] Use of 96 kHz sample rate to lower latency, Harry van Haaren, (Thu Jan 2, 12:27 pm)
Re: [LAU] Use of 96 kHz sample rate to lower latency, Bill Gribble, (Thu Jan 2, 1:31 pm)
Re: [LAU] Use of 96 kHz sample rate to lower latency, Ralf Mardorf, (Wed Jan 1, 5:18 pm)