If you think of the original analog music as a sound wave then converting it to digital means approximating that wave as digital bits. The sampling rate is how many points on that wave you are taking per unit time so the higher the sampling rate the closer you are to the original sound. Lower sampling rate means higher compression but lesser audio quality.
Similarly the bit rate is effectively 'how much' information you're encoding at each point so again, lower bit rate means higher compression but lower audio quality.
Compression algorithms generally use pyschoacoustics to try to determine what information can be lost with the least amount of audible difference. In some sections of a track this may be more or less than in others so using a variable bit rate enables you to achieve higher compression without a 'big' audible drop in quality.
It's well explained here: Link
I don't know the details of those codecs but generally what you should use and what parameters you should pass depends on what you're trying to achieve and for what purpose. For portable use where audio quality might not be paramount you might want to pass lower values to achieve smaller file sizes - for audiophile speakers you probably want to pass the maximum.