On Thursday 06 April 2006 19:26, Jack Carroll wrote:

>  Yup, I agree.  It takes non-zero time to convert the raw bit stream
> into Ethernet frames.  One feature the protocol would have to support is
> synchronization of the streams coming from different digitizer channels,
> including digitizers on different subnets.  I'd say 10 microsecond
> synchonization accuracy would probably be adequate; that's about 0.1" or 2.5
> mm sound source location accuracy.

Would maybe be a good idea to ask the Jack people about latencies. I've seen
people on LKML who've tried to do audio with latencies that I thought were
absolutely insane at first.
I'm not convinced ethernet is quick enough :)
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