An audio server need not be designed to add latency (beyond
that of the network itself, of course). With current networks,
this is very small, down to a few samples.
Existence proof is the AF audio server we did 10 years ago,
in which the server design itself did not enforce any latency: if
data arrived at the AF server before the sample had been
played (and the hardware permitted), it performed cut-through
and updated the samples immediately.
- Jim
On Wed, 2003-09-10 at 17:54, Ross Vandegrift wrote:
> On Wed, Sep 10, 2003 at 11:09:54AM -0400, Jim Gettys wrote:
> > The other promising work besides MAS is an audio server project
> > called "Jack".
> >
> > It is not clear it currently provides network transparency, but it
> > does boast low latency (required for telephony, teleconferencing and
> > gaming applications).
>
> No, jack is intended for apps with much stricter performance
> requirements - low latency, sample synchronization, and realtime
> transport. These are pretty critical for pro audio work - recording,
> production, soundtracking, overdubs, etc.
>
> It's very doubtful it will ever work over conventional networks - timing
> is just too critical to jack.
>
> Now, a specially designed network with ADAT synchronization could work,
> but I doubt anyone would want to port X11 to such a transport... ::-)
--
Jim Gettys <[EMAIL PROTECTED]>
HP Labs, Cambridge Research Laboratory
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