[linux-audio-dev] Audio synchronization, MIDI API
John Check
j4strngs at bitless.net
Mon Aug 16 07:40:37 UTC 2004
On Sunday 15 August 2004 04:36 am, Steve Harris wrote:
> On Sat, Aug 14, 2004 at 11:39:48 -0400, John Check wrote:
> > > I don't think raw ethernet will buy us anything over using UDP. These
> > > few usecs less simply won't matter.
> > > (but with ethernet you would have the disadvantage that you loose
> > > routability)
> >
> > It probably wouldn't be the best idea to have an audio system on any kind
> > of accessible network anyway. Switches or even bridged ethernet would do
> > the trick for a cluster of audio processing hosts.
>
> But if youre going to do that, why use ethernet? You'd need dedicated NICs
Why climb a mountain? If you want to find reasons not to do something it's
easy.
> and switches, so you may as well use firewire, which has dedicated
> realtime channels, more bandwidth and doesnt require switching. 400meg
> Firewire cards are down to about 7 or 8 euros in the UK now.
>
Good idea, but ethernet is still an adequate low cost option. I have code to
do it over ethernet that I've yet to test and it doesn't require me spending
a penny (stop giggling).
> The only disadvantage is that you can't (right now) cheaply run firewire
> over long distances, but taht will change once firewire over CAT5 cards
> come down in price, and this is rarely an issue with clusters anyway.
>
I agree. It's either a problem or it isn't.
> > > In that case I guess sample skipping/duplication (from time to time) is
> > > the way to go.
> > > It's not hard to do but the hard thing is to keep buffers small and
> > > have the algorithm react fast to eventual drifts in sample rate.
> > > I measured the sample rate of some consumer (44100Hz) audio cards and
> > > usually it is quite stable but it varies
> > > 3-4 samples from card to card.
> > > eg card1 had 44096Hz , card2 44098Hz
> > > so assuming we use 128 frames buffers
> > > and the samplerate delta between cards is 2 samples we must insert/skip
> > > 2 samples per second in one of the cards.
> > > which means insert/skip a sample in one of each 172 buffers (of
> > > 128frames each) processed.
> > >
> > > Any good suggestion how to best implement it ?
>
> You can't just duplicate or drop samples, it will sound terrible. You need
> to do some resampling. For clusters this shouldn't be an issue, just
> make one device be the i/o machine and sync everything else off that.
>
> If you absolutly have to have multiple machines doing i/o then you will
> need some complicated resampling stuff. Fons has been working on it, to
> allow soft-sync between 2 jack systems, but I've not tried it yet.
>
> - Steve
More information about the Linux-audio-dev
mailing list