sure. usually audio APIs (like ASIO on Windows) operate in a double-buffered mode, where you submit a buffer A and receive a buffer B back "at the same time". The A buffer contains data you wish for the driver to output and the B buffer is what was recorded last.
let's assume you have an analog audio device that takes its input from soundcard outputs and sends its output back to a soundcard input
when you submit buffer A, you will get the samples processed by the analog device with the next submitted buffer
so not in the current B but next B
additionally because of how A/D and D/A converters work, there is a filtering and conversion latency that happens between the first sample being output and the first sample being recorded
so if you sent 512 samples of white noise in a buffer A, you would still get buffer Bnext of 512 samples length, but the actual first processed sample would appear after some silence. Let's say additional 30-or-so samples for filters and conversion
so now if you want to sample accurately mix dry and wet signals (where wet is what is processed by the analog device)
you need to align those two
hence my question, is this something that is fairly easy to do in SC?