On Sun, Oct 05, 2014 at 08:39:11PM +0200, [email protected] 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?
Either 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 capability. 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. Ciao, -- FA 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) _______________________________________________ Linux-audio-dev mailing list [email protected] http://lists.linuxaudio.org/listinfo/linux-audio-dev
