Upsample / Downsample a Signal from 1920.93Hz to 1920Hz 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?
Assuming the signal was reasonably band-limited before sampling, you can just use a high quality interpolation, such as a Sinc interpolation kernel, to calculate new equally spaced points at any lower density.

For small frequency deltas, even a long interpolation kernel or high degree polynomial interpolation can be a lot more efficient than naively upsampling and then downsampling.