[LAD] AoIP question

Fons Adriaensen fons at linuxaudio.org
Sun Oct 5 22:29:18 UTC 2014

On Sun, Oct 05, 2014 at 11:35:21PM +0200, tom at trellis.ch wrote:

> I know this, a real pearl! It's working amazingly well. The data on the
> receiver is not 1:1 equal to what is being sent. This works for most cases
> though.

The digital data is not the same, but it shouldn't be if your
sample rate is not the same as the sender's. The analog signal
will be the same, as if it was sampled by your HW.

> Many middle-class audio interfaces have an S/PDIF input that can drive the
> internal clock (interface is slaved to S/PDIF). S/PDIF is rate-less, any
> (?) rate would work. Now, if we'd have an external device that does
> nothing else than providing a variable clock to the audio interface via
> S/PDIF, software-controlled, it would be possible to match / align
> decoupled, network-connected hosts. Scenario: endpoint audio interface is
> slaved to that variable "speed" S/PDIF generator, that is software
> controlled. Some clever tool could then adjust the speed to match the
> sender's rate. This would allow non-resampled ~isochronous audio on remote
> hosts for cheap.

The feedback loop in njbridge that now controls the resampling
ratio could be used to generate the SPDIF signal, or a word clock,
or to control the sample rate directly as would be possible with
some RME cards. But there's no standard to do either of those.



A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)

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