the 2nd one sounded 'thinner' to me...and couldn't tell if it was volume, listening on good headphones.   they both were very close and if the difference is that big I should start laying asphalt as a living and sell my guitars, it aint gonna happen....  although asphalt layers do make some good money!
 
I was gonna pop them in SoundForge, but I figured someone else would do that (like John above  ;-)  )... and be better capable of making such an analysis.

Beepster
This does seem to be a rather contentious issue which strikes me as a little weird.

 
Yes, it's like religion to some people or something. I honestly don't care one way or the other but if I hear a difference, I want to know why and if it's an improvement, then I want to implement whatever causes an improvement.

John
OK but why would they be there anyway if (the higher frequencies) frequencies above half the sample rate are filtered out? And if this is happening than it must be considered a distortion. 

Distortion is exactly why it happens. The audio interface takes great pains to assure that no frequencies above Nyquist enter the system, and if you only mixed pure audio tracks with no intentional distortion you'd never have to worry about aliasing. But if distortion happens within a plugin, intentional or otherwise, harmonics are going to be generated. 
 
Plugins whose primary purpose is distortion, such as amp sims, are designed to deal with it because the designer knows up front that it's going to be a problem. That's why the better products offer 4x or higher oversampling, followed by filtering. Illegal frequencies are still generated internally within the plugin, but none of them make it back out to the DAW.
 

...wishing the problems weren't there in the first place doesn't make them go away; raising the sample rate does. Most of the foldover with 96kHz bounces back into a range that's above 20kHz so we don't hear it.

This implies that aliasing is an inherent and unavoidable byproduct of DSP. It isn't. Many types of processing never produce aliasing/distortion. For example, "boosting the treble" does not cause harmonic distortion. It will, however, exacerbate any existing distortion by boosting the harmonics.
 
Adding saturation does cause harmonic distortion, but again if your saturator causes aliasing it's the saturator's fault, not your choice of sample rates. Raising the sample rate would only be a band-aid to cover up perhaps half the fallout from uncontrolled harmonic distortion. That saturator will still sound bad at 96 KHz, just marginally less-bad.
 
Does 96 KHz push "most of the foldover...above 20 KHz"? Hmm, maybe. Depends on how you define "most". "Most" harmonics on most distorted sounds easily fall under the Nyquist frequency even at 44.1 KHz.
 
Let's look at a practical example, an electric guitar played through a high-gain amp sim. You play a very high note on your guitar, say with a fundamental frequency of 1.3 KHz (an octave above an open high-E string). The amp sim will generate harmonics at 3x, 5x, 7x, 9x, etc. The 15th harmonic is 19500 Hz, still legal at 44.1 KHz. You have to get up to the 17th harmonic before changing the sample rate would deliver any benefit. I didn't do the math, but the level of the 17th harmonic is going to be down more than 90 dB from the fundamental. IOW, inaudible. 
 
Synthesizers are a different can o' worms. A so-called "supersaw" contains both even and odd harmonics plus many sum and difference frequencies from the interplay between multiple pitch- and phase-modulated voices. Synthesizer designers have to anticipate such harmonic complexity. Improperly designed oscillators can indeed generate out-of-control harmonics. If Z3ta+ falls into that category, then I maintain that it's a design flaw. Zebra, for example, does not do this as far as I can tell. 
 

I agree upsampling and oversampling helps, but both rely on interpolation and in the case of oversampling, stuffing in zeroes and interpolating on playback because attempting to interpolate while recording creates its own issues. Running at 96kHz for sounds that are generated synthetically provides "real" data for each sample.

 
There is nothing inherently bad with how upsampling works, even the "stuffing in zeroes" part. In fact, this is what happens every time you record anything! Your audio interface isn't recording at 96 KHz, it's recording at some multiple of 96 KHz, at least 64x that rate. So you're using upsampling all the time. If it were messing up fidelity we'd still be recording directly at 96 KHz like they did in the early 60's when 96 KHz became the standard - precisely because of the lack of oversampling at the time.
 
Sample rate determines the highest frequency you can process. Period. It has nothing to do with "precision". Calculations aren't going to be more precise at 96 KHz. Precision is determined within the code itself, e.g. whether or not the programmer uses double-precision floating point values for all the internal math.
 

Saying I need better plug-ins is all well and good, but that's sort of like saying all my plug-ins should be 64-bit, which they should be. But some aren't, so I have to use bridging, which is a flawed technology but serves its purpose.

Careful, let's not bring bit-depth into the discussion. It's irrelevant to the topic. In fact I would not be the one to say that all your plugins should be 64-bit, because I know that would have no discernible impact on fidelity.
 
Now, I do understand and agree with your underlying argument, which is that you're stuck with the tools at hand, and a higher sample rate may improve the performance of some of them. I certainly couldn't afford to replace all of my plugins! So yeah, if I found out that most of my plugins were deficient I'd be seriously bummed and looking for any way to work around their limitations.
 
To put your premise another way, you could rephrase it thus: "if your plugins are substandard, a higher sample rate might or might not help". I'm good with that.
 

Speaking of recording at higher sample rates somebody once told me 88.2 is the best to record at simply because it's 44.1 doubled. They said the math that is computed is handled better at this sample rate? I don't know a thing about it. I'm not that technical when it comes to this sort of thing, but it makes sense to me. I record at 24 bit/44.1 but I think I'm going to try recording at the higher sample rates. Why not?

Geo524
Speaking of recording at higher sample rates somebody once told me 88.2 is the best to record at simply because it's 44.1 doubled. They said the math that is computed is handled better at this sample rate? I don't know a thing about it. I'm not that technical when it comes to this sort of thing, but it makes sense to me. I record at 24 bit/44.1 but I think I'm going to try recording at the higher sample rates. Why not?

Urban myth, unfortunately. One of those concepts that sounds reasonable but just ain't so.
 
As for the "why not", the only real downside is the extra disk space you'll use and the extra CPU load. If you have gobs of disk, and most do nowadays, that's not much of a barrier. What you're more likely to encounter is your CPU running out of steam sooner, meaning fewer soft synths and effects and more freezing. The upside is lower latency, if that's important to you - assuming your CPU can handle the same buffer sizes at the higher rate.

Craig to me the 27 is the better sounding clip.
 

I am sidestepping the whole discussion - I am quite happy with the results I get with using 48 k, and I never have to worry about dropouts or otherwise taxing my system, in any way.
 
I will instead continue to concentrate on the quality and strength of a song's writing and playing, and I happily drive a Nissan Cube.  :)
 
Bob Bone

Grem
Craig to me the 27 is the better sounding clip.

I understand that, a few of the non-oversampled Guitar Rig amps sound "better" to my ears than the oversampled ones. I guess this means that if the world goes to 96. someone will need to create an undersampling button to get "that vintage digital sound so popular in the 2010s"
 
But the point of the comparison is about the accuracy with which the sound represents what the synth was generating. (If I had any synth sound in a track with that many highs, I'd run for the QuadCurve's lowpass filter.) 28 has high frequencies that simply don't exist in 27, and they're not distortion byproducts.
 

bitflipper

Geo524
Speaking of recording at higher sample rates somebody once told me 88.2 is the best to record at simply because it's 44.1 doubled. They said the math that is computed is handled better at this sample rate? I don't know a thing about it. I'm not that technical when it comes to this sort of thing, but it makes sense to me. I record at 24 bit/44.1 but I think I'm going to try recording at the higher sample rates. Why not?

Urban myth, unfortunately. One of those concepts that sounds reasonable but just ain't so.

Actually I doubt you're old enough to know this, but that was true at one point. It's easy to forget that digital audio has been around for a long time, and that audio engines used to be 16-bit or [gasp] even less. When DAT was invented, it was deemed that the math involved in sample converting from 44.1 to 48 was so daunting (1.0884353741496599) that it would discourage digital copying. 
 
Quality sample rate conversion is not trivial, and the accuracy has improved dramatically over the past 30 years. This site is really interesting: 
 
http://src.infinitewave.ca/
 
Check out the difference in sample rate converters between Ableton Live 7 and Ableton Live 9, and that was only a few years' difference...then consider the days when we had 16-bit engines. Back then, conversions from 88.2 to 44.1 did sound better than 96 to 44.1. Fortunately that period didn't last long, but it did exist. 
 
Oh, and if you want to feel good about Sonar, while you're on that site compare it to a bunch of other DAWs. They used 8.5, but I assume the sample rate conversion didn't get any worse in the X-series.

bitflipper
The biggest difference between Craig's files is that one drops off sharply above 18 KHz and the other does not. There was also a small level difference. After level-matching them and applying a LPF so that their spectra were more closely matched, the 96->44.1 file sounded more pleasing to my ear and I could not reliably distinguish between the two in an A/B test.

It's interesting that you found the 96k file more pleasing to the ear. However the point was not to apply an LPF to match their spectra, but that the one recorded at 96k represented the high frequencies more accurately, even when played back at 44.1kHz. Hey, I just report, it's up to other people to come up with the theories 
 
If you listen closely to the two files on headphones you'll also hear non-harmonic noise in the upper ranges of the 44.1 file but not in the one that was recorded at 96. At first I thought it was because the additional high-frequency content was masking the noise, but I don't think so. The high frequencies are in a different range.

Anderton
The plot thickens...
 
Actually, both Bitflpper and I might be wrong about 96 making a difference only with signals inside the computer. No less an authority than James A. Moorer wrote a paper that proposed, among other things, that hearing involves not just frequency and amplitude, but time and how it relates to localization when listening with both ears. He claims that most people can distinguish a time delay of 15 microseconds or more when a pulse is put into each ear, and that some people can differentiate delays as low as 3 to 5 microseconds. Given that a sample at 48kHz is about 21 microseconds and 10.5 microseconds at 96kHz, that means the minimum time delay most people can differentiate is actually less than one sample at 48kHz, but more than one sample at 96kHz. 

 
The sample time does not remotely equal the time resolution. This is a common misunderstanding of how sampling works. 
 
It depends on bit depth as well as sample rate, but the short answer is 48kHz has a timing resolution of FAR greater than 1/48,000.

Anderton
 
But the point of the comparison is about the accuracy with which the sound represents what the synth was generating. (If I had any synth sound in a track with that many highs, I'd run for the QuadCurve's lowpass filter.) 28 has high frequencies that simply don't exist in 27, and they're not distortion byproducts.

I am having some difficulty discerning how you are determining what the synth is actually producing and how you are deciding that one recording is more accurate to that production than another. More pleasant, maybe, but how accurate?  Many people find the distortions inherent in analog recording/processing to be "warmer" or whatever cork sniffing metaphor you prefer. But it is pretty easy to demonstrate that analogue recordings of a reference source are less accurate than a properly done digital version using instrumentation. With a signal produced in the box, what is the proper A/B comparison to determine if it is accurately recorded?

Anderton
 

If Z3ta+ falls into that category, then I maintain that it's a design flaw.

 
So do you consider including oversampling compensation for a design flaw, or simply good practice? If the answer is simply good practice and therefore it must always be done, then that restricts how many instances you can use and balance out CPU consumption. In some cases, there will be no audible difference whether the synth is loafing along at normal speed or oversampling, and many might choose not to oversample as a tradeoff for something like being able to overdub a guitar part with lower latency.
 
So the bottom line is this: if oversampling and filtering are good practices to compensate for deficiencies caused by generating undesirable harmonics that can create problems with lower sampling rates, why is recording at 96kHz inherently negative if it too can compensate for deficiencies caused by generating undesirable harmonics that can create problems with lower sampling rates? Aren't they the same thing, except one is applied as a global preference as opposed to a localized one?

I would say that if a modern plugin is aliasing, then it should at least give the option of oversampling itself. It's sort of a no brainer.
 
For older plugins that were developed when CPU power was much more limited it's more understandable why they might lack this functionality (particularly if they were already pushing the CPU envelope at that time), and that's where your solution makes sense.
 
But the point is that you're doing something that the plugin's developers should have already done in the first place. And for recent plugins they probably already did do it. And if it's a recent plugin and they didn't oversample - even though it was clearly necessary - then I would say it is absolutely a design flaw.

gswitzDrewFX, what would the time resolution of 24 bit 48kHz be? Can you figure it out? I'm curious.
 

Something on the order of 0.0000000000002 seconds.
 
Note that 1/48,000 = 0.0000208333333 seconds (Craig's 21 microseconds).

I was quite surprised at the difference between the two files. The 96kHz clip definitely has way more high frequency content, in fact the 44.1kHz sounded to me like it has some gentle low pass filtering in comparison. 
 
I have long noticed quite a large difference between having 2x oversampling turned on or off in Z3TA+2. It's not an open and shut case as to which sounds the best, however - it depends on the patch. For example when it's a bright plucky patch then the 2x oversampling sounds better to my ears. But for a warm sounding pad, I think it sounds better with the oversampling turned off - you get a much creamier "analog" sounding pad. 
 
As for Craig's two clips, I definitely prefer the sound of the 44.1kHz clip, it sure sounds a lot fatter and warmer, and the high frequencies in the 96kHz clip are way too harsh sounding. 
 
But don't people often say similar things about lower bit rates where some kinds of sounds are concerned? For example the much sought after "phat" sounding drums from an old 12-bit Akai sampler.