As I understand it (and that ain’t saying much), the typical sampling rate used to create an MP3 file from a continuous or analogue signal, is 44,100 Hz.
Assuming that you had the necessary computational power, would there be any gain to the apparent signal quality by interpolating between adjacent samples? I suppose the easiest way to do this would be to calculate the interpolated “sample” simply by using the mean value, at all ‘relevant’ frequencies, of adjacent samples (thus winding up with a signal that’s effectively 88, 200Hz) .
Alternatively, but more complex, would be to interpolate after taking into account, not just the values, but the trend of the values of samples over some range before and after the adjacent samples. In addition to taking into account the trend, I can imagine a technique to calculate interpolated “samples” by also using some sort of weighting method (wherein the closer in time a sample is to the adjacent samples of interest, the more its values would contribute to the interpolation).
Although computationally demanding, interpolating in some way would seem save bandwidth. Is it being done?