Hello kind people of the audio computing world,
I have an array of samples that respresent a recording. Let us say that it is 5 seconds at 44100Hz. How would I play
Take a look at the "Elephant" paper in Nosredna's answer to this (very similar) SO question: How do you do bicubic (or other non-linear) interpolation of re-sampled audio data?
Sample implementations are provided starting on page 37, and for reference, AShelly's answer corresponds to linear interpolation (on that same page). With a little tweaking, any of the other formulas in the paper could be plugged into that framework.
For evaluating the quality of a given interpolation method (and understanding the potential problems with using "cheaper" schemes), take a look at this page:
http://www.discodsp.com/highlife/aliasing/
For more theory than you probably want to deal with (with source code), this is a good reference as well:
https://ccrma.stanford.edu/~jos/resample/