Asynchronous SRC's do not "reduce input jitter" but rather they very efficiently convert it. Time domain jitter to the input side of the ASRC is converted to amplitude jitter on the output side. Reclocking with a very low jitter local clock is the correct solution. If ASRC is needed, reclock first to reduce input side jitter and then use the ASRC.
I believe the discussion on 'interpolation' is talking about such techniques as used in the old Audio Alchemy DTIPro and DTIPro32, and the new Perpetual Technologies P-1A, and copycats. The act of taking a signal and changing to a higher data rate (typically higher but could also be lower) is handled in the traditional way and gains nothing in precision. The output may indeed have more bits of resolution than the input, but those bits are merely necessary to provide for the accurate positioning of the amplitude for the new samples, located at points inbetween the incoming samples.
The 'interpolation', which is called Resolution Enhancement in the P-1A, can operate on the data even when there is no rate conversion. So it is not a result of upsampling (or downsampling) and filtering. So what is it? Without giving away the store, I can tell you that it runs a real time analysis on the incoming audio signal and estimates what is missing. For example, a low level analog sine wave when sampled at 16 bits will tend to be "squarer". This is a direct consequence of the low resolution of the quantized sine wave. Dither and noise shaping will tend to mask the squareness but those are long term statistical remedies and looking over a short sequence will appear to be a noisy square wave. If you knew, a priori, that the signal was a sine wave, it is easy to reconstruct to as many bits of precision as you need. That's where the heuristics come in and the algorithm has to 'best guess' what the missing resolution is, or give up and let pass as is.
Fortunately, music is full of patterns and those are in use everyday in ways that we are all familiar with. Reducing to mp3 or Dolby Digital format is, in part, finding those patterns and using the information to decide what is important and what is not when encoding the signal. Resolution Enhancement works on the decode side. It takes the music, finds the patterns, and reconstructs what was left out by the quantization process. It's not magic, but it is in a sense, creating something from nothing.
In the video arena, line doublers and quadruplers and scalers and other enhancement techniques are used to create something from nothing, and the results are very satisfying to the vast majority of users. So it goes with Resolution Enhancement. Most listeners perceive improvements in low level detail and imaging. It's something from nothing and it's based on decades of research and techniques commonly used and accepted in many other audio domains.
Bottom line - if you don't perceive a difference and prefer the certainty of the source 16 bits, turn off the processing and enjoy the music.
