This question already has an answer here:
I've just learned that the A/D converter I am using is not sampling at my desired sampling rate. I request 20 seconds of data at 1920Hz and I receive 38400 points of data that were actually sampled at 1920.93Hz.
So for example, if I sample a 30Hz sine wave and chop the signal up into 300 sections, the phase of the sine wave in section 1 differs from the sine wave in section 300. If my sampling rate was 1920Hz, as expected, the phase of each section would line up perfectly.
Now, to correct for this, is there any way of processing the signal such that it appears as if it was sampled at the correct sampling rate through interpolation or other means?