jaddie - I didn't know that the DAC did filtering. Thanks for explaining that. However, in your example of making a 44.1KHz file into an 88.2KHz file, if the interpolation is skewed from a straight line interpolation, wouldn't the distortion component be at half the new sampling frequency, since only every other sample is skewed? Also, if a 44.1KHz AAC were decoded at 88.2KHz, not every interpolated sample would be skewed the same amount, or in the same direction. For example, in one cycle of an 11.025KHz cosine wave, the first sample would be an original*, the second would be slightly higher than a straight line interpolation, the third: an original, the fourth: slightly lower than a SLI, the fifth: an original, the sixth: slightly lower than a SLI, the seventh: an original, the eighth: slightly higher than a SLI. The interpolated samples would fit right on the curve of the cosine wave and so no distortion component would be added. With a straight line interpolation, however, the interpolated samples would not fit on the curve of the cosine wave. In fact they would add a low amplitude, odd shaped distortion component. This distortion component would be repeated at a rate of 11.025KHz.
Well, at least you're getting the point: resampling doesn't improve anything other than changing the sampling frequency that must be filtered by the anti-aliasing filter. Your comment about not realizing the DAC has a filter is interesting. I wonder how many others out there don't realize that the jagged waveforms they see when they zoom in on a waveform in a DAW never actually get out of the DAC. Could explain a lot of interest in high rates!