[LAD] AoIP question
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)
> 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)
More information about the Linux-audio-dev