22 Jun
2009
22 Jun
'09
11:49 p.m.
On Tue, 2009-06-23 at 03:54 +0200, Esben Stien wrote:
Dennis Schulmeister
<linux-audio-dev(a)windows3.de> writes:
There's the buffer argument again. Even a highly compressed
low-bandwidth VoIP stream needs to be buffered by the application
because of the annoying factor not being latency but jitter. If you do a
"regular" phone call from one mobile phone to another you easily get 1
second latency. The point is you don't notice it unless you're standing
next to the other person.
But then what's the reason behind ultra-small hardware buffers which
need to be refreshed very quickly in contrast to the comparatively large
buffers needed for the input stream in the first place?
I see the reasoning for games, though.
Yours sincerely,
Dennis Schulmeister
--
Dennis Schulmeister - Schifferstr. 1 - 76189 Karlsruhe - Germany
Tel: +49 721/5978883 - Mob: +49 152/01994400 - eMail: dennis(a)windows3.de
Now moved to the corridor: Hermes! (http://ncc-1701a.homelinux.net)
Besides that: http://www.denchris.de - http://www.motagator.net/bands/65
<GnuPG KeyIDs: B8382C97, 01AD62DE>
audio performance has always been fine for
typical desktop usage
without rt privileges me thinks. Because the issue of latency is
relative in that case. But maybe I'm missing an important point.
Sure, like VoIP.