On Thu, Nov 20, 2014 at 11:12:09PM +0100, Ede Wolf wrote: > Thanks very much. I do know about zita-a2j any may end up using it > for recording smtpe through onboard audio, at least as an > experiment, but for standard audio I would like to avoid resampling > whenever possible. Also, maybe pure superstition, that a2j attached > soundcard always feels as just being second best, not treated equal. You say 'feels' and that's what it is. There is no objective reason why resampling would reduce sound quality provided it's done as it should be done (which is the case for ajbridge). Basically the process is the same as the anti-aliasing filter of an AD or DA converter. And almost all such converters include digital resampling as part of the process. > So is there a realistic chance that I would be presented with 52 > inputs instead of 26 as of now? Without using a2j? > For synchronizing both cards I'd use external wordclock. The next release of ajbridge will have the option to bypass the resampling in that case. The only remaining effect would be a somewhat higher latency for the second card (because the periods are not synchronised). 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-user mailing list Linux-audio-user@xxxxxxxxxxxxxxxxxxxx http://lists.linuxaudio.org/listinfo/linux-audio-user
