When capturing microphone audio, I noticed that my static buffer eventually overflows.
So I timed the samples, and low and behold I'm getting an actual sample rate of 44102-ish Hz (my timer is very accurate).
Is this the actual sample rate recorded by my microphone, or is windows sending me samples twice?
Doesn't this cause all sorts of synchronization problems. Say if you're listening to a live recording, wont a noticeable lag start to accumulate if sample rates aren't exact? How is this problem solved traditionally? Is dropping a sequence of samples, or padding zeros normal?