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.
"Amplitude jitter" is actually amplitude modulation. But ASRC does not do that... What it does do is grab data that has jitter, and since there is an independent clock working on the signal, it passes the jitter through, even if the output clock is more precise. This is because the jitter signal is resampled with the new clock. A fallacy of ASRC is that it is a means of eliminating input jitter.
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.
Although it would help, it would only do so inasmuch as it would with synchronous conversion...
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.
If the data rate is lower, interpolation is not performed, but rather "decimation"... Resolution is lost.
The fundamental method of interpolation via digital filter has not changed since it was first done in the late 1980's...
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.
There's where a lot of the confusion lies. The extra bits are solely for the positioning of new intermediate samples. It's not added resolution. (It's only "added resolution" if the media itself has that resolution in the first place. Where no resolution was lost between sampling the original signal and data storage.)
The 'interpolation', which is called Resolution Enhancement in the P-1A, can operate on the data even when there is no rate conversion.
This implies that if you took a 16 bit signal, and converted it to 20 or 24 bits, at the same sample rate, resolution could somehow be enhanced... The problem is there is no viable algorithm that could truly determine how the amplitude value should change relative to the 16-bit representation. (I think this is what the Genesis Digital Lens did. The product is now a boat anchor.)
Interpolation, no matter how you slice it, does *not* enhance resolution. (If there is no rate conversion, it's not even interpolation. Higher sample rates are required for interpolation to take place, because interpolation involves plotting new calculated values between existing values.)
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.
This is how every interpolation algorithm works. Be it by Perpetual or someone else... Except the "estimation" is actually calculation, per the D/A's filter algorithm. (Most-likely convolving the CD data with a truncated "sinc" function.)
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.
Only if the playback is non-filtered... Dither, in conjunction with the DAC's digital filter, produces a signal closer in semblance to the waveform, be it sine or other shape. (Provided the frequency is well below half the media sample rate.)
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.
But what about cases where the signal is not close to being a sine wave? In such cases, the result will be *less* precise.
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.
It sounds like the technology presumes the low-level signal is more like a sine wave. But if the original pre-digitized signal was really less like a sine wave, then such technology makes the playback more "wrong." But even worse, since different recordings and performances have subtle differences in low-level behavior, this technology would mask those differences, thinking it's always "sine waves." So in essence, you hear the DAC, as opposed to the recording.
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.
So for continuous wave patterns possibly an enhancement... For non-continuous patterns, likely a degradation...
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.
In digital processing, the available resolution is often optimized, but you cannot recover lost resolution, no matter how hard you try... What you call "resolution enhancement" is likely no different from optimized digital filters that exist elsewhere.
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.
Well, at least it's defeatable. The problem with a lot of ASRC players and DACs is the ASRC is *not* defeatable.
The bottom line- There is so much hype in digital audio products. The best way to choose your product is to toss the sales pitch out the window, and let your ears decide. (And if one ends up liking the Perpetual sound, that's fine too.)
