[LAD] AoIP question

Fons Adriaensen fons at linuxaudio.org
Sun Oct 5 20:25:07 UTC 2014

On Sun, Oct 05, 2014 at 08:39:11PM +0200, tom at trellis.ch wrote:

> As a scenario, at point a) an analog signal is injected that will be
> played back (analog) at point b) with the lowest possible (and constant)
> latency.
> How do you intend to handle diverging clocks of the audio interfaces
> (ADC/DAC) at both (a/b) ends?


1. Sync the HW sample rates to an explicit or implicit reference
   provided by the network protocol. Requires special audio HW.
   A few normal audio interfaces (e.g. some RME cards) would allow
   to do this as well, but I know of no software that uses this

2. Resample at the receiver, as njbridge does.

3. Use some other trick. E.g. for VOIP a classical one is to
   modify the lenght of the pauses between words or phrases.



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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