Also 88.2 isn't as well supported because it's less used. For example the Roland Octa-Capture and Quad-Capture don't support 88.2 but do support 96.
 

Nice rationale plus confirmation for higher rates.  Here I come.
 
@

I have been using 96 for a while now. At least two yrs. When I started doing it, I just noticed just a better sound. I could/can not describe it at the time, nor can I put it into words today. But I hear something.
 
About 2 months ago, I started a project and decided to do everything in 48k. So some of the songs I had at 96k, I redid them in 48k. About 2 wks ago I had to open an older version of one of the songs to grab some info out of it. It was in 96. So I reset my VS100 to 96, opened the project and listened. I could hear the difference. And I thought it was just my mind playing tricks on me. But my wife was sitting there with me and without me saying/asking anything, she said,"Those guitars sound really good."
 
I made the decision to go back to 96 then.
 
Then Craig started posting about 96k.

Sorry, Craig, but  you're inadvertently propagating well-intentioned misinformation.
 

... amp sims, virtual instruments, etc. can easily generate signals that go above the clock, and fold back into the audio range.

This is a true statement, but anti-aliasing will be handled internally within a well-designed synth or distortion processor. If you have to increase your sample rate to make some plugins work better, then you need better plugins.
 
If AD sounds harsh at 44.1KHz it's a design flaw in AD. What is the sample rate for AD's samples? 44.1 KHz. Playing them back at the same rate they were recorded should yield the best fidelity. If it doesn't, and the reason is aliasing, then there is distortion happening within AD that either shouldn't be there or that should have been handled internally with upsampling and filtering.
 
Consider the most common scenario for generating "illegal" frequencies within your project: harmonic distortion. You might call it an exciter, an amp sim, a tape sim, a revitalizer, a tube emulator, or a saturator - they're all adding harmonics that can potentially include frequencies above Nyquist.
 
Such processors most often add odd-order harmonics. For example, a distorted 10 KHz signal's third harmonic of 30 KHz would exceed Nyquist at 44.1 KHz but not at 96 KHz. However, the fifth, seventh and ninth harmonics still exceed Nyquist, even at 96 KHz. IOW, raising your sample rate is only a partially-effective band-aid for mitigating problems in your plugins that shouldn't be there in the first place.
 
A simple test using a sinewave and a spectrum analyzer will tell you whether a plugin is causing aliasing. Many do, especially older plugins and freebies. The better ones absolutely do not, and will perform equally well at 44.1, 48, 88.2 or 96 KHz.
 
 

I'm not sure what "fold back" means. 

I'm not going to try and recreate this, have never had any problems at 44.1, and this sounds like a 'clock' problem...to this limited, technical person.  Can frequencies really "bounce" off the upper limit of the clock setting and be captured...like sound waves bouncing off a wall?

mixmkr
I'm not going to try and recreate this, have never had any problems at 44.1, and this sounds like a 'clock' problem...to this limited, technical person.  Can frequencies really "bounce" off the upper limit of the clock setting and be captured...like sound waves bouncing off a wall?

It seems to me that they would be filtered out. As stated it would imply that the higher frequencies are somehow morphing into lower frequencies. I have heard variations on this as an explanation why it sounds better to the person. I just wonder if a double blind test would give the same results.    

John
it would imply that the higher frequencies are somehow morphing into lower frequencies.

and so digital data can do this?  That sounds like a new plugin format to do "This"...if it can.  I'm sure someone would buy it.  Digital saturation... that's what I'd call it.

...it would imply that the higher frequencies are somehow morphing into lower frequencies.

That's exactly what's happening.
 
Frequencies above half the sample rate cannot be accurately reconstructed and are misinterpreted as lower, legal frequencies. The result will be a frequency that's the difference between the real frequency and the Nyquist frequency. 
 
At 44.1 KHz, the highest legal frequency is just under 22.05 KHz. If you try to encode a 23 KHz signal, what you'll get is 23 minus 22.05, or a 950 Hz tone. This is what people are referring to when they say it "folds back".

bitflipper
[If you try to encode a 23 KHz signal, what you'll get is 23 minus 22.05, or a 950 Hz tone. This is what people are referring to when they say it "folds back".

And so we're not hearing this somehow?  (when using 44.1).  That would be very audible as compared to 21KHz, or that upper region.  I doubt if I'm hearing over 15K anyway.  I'm sure people are cramming 23 K frequencies all the time.  They just don't hear it.