I thought folding back was addressed with a filter.

bitflipper

...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".

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. 

bitflipper
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.

 
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.
 

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.

 
The samples sound fine at 44.1 until you start adding synthetic processes like boosting the treble, adding saturation, etc. It doesn't matter to me why the results are an improvement, because I need to make the best-sounding music I can with the tools I have today. 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.
 

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.

 
As can harmonic-rich waveforms from synthesizers like Z3TA et al; it's not just distortion.
 

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.

 
But 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.
 
But I also don't think it's just about aliasing. One result that made so little sense I re-did the experiment to make sure was that the imaging of TH2 with its included reverb at 44.1kHz "wandered" compared to the same preset at 96kHz, where the image was rock-steady. The sound quality was the same. I'm theorizing that reverb is sufficiently complex that doubling the processing rate somehow tightened up the calculations and reduced variations between the left and right channels.
 
I checked with designers at Native Instruments and IK Multimedia when I first started running amp sims at 96kHz and I thought they sounded better, but didn't trust my ears. I asked if i was hearing things or whether there was an actual reason why they sounded better. I don't have their responses on this computer, but I can look it up. Independently, both of them mentioned improved computational precision as the main reason why, not distortion. Perhaps the experience with the TH2 reverb supports that.
 
The bottom line is I got incontrovertibly better sounds out of virtual instruments and plug-ins by running the project at 96kHz, even when sample rate converted to 44.1kHz and played back through a 44.1 audio engine.
 
To hear what I mean, here's a link (expires in six days) to download two files produced by the Z3TA+ 2. One was recorded with the project running at 44.1kHz, the other with the project running at 96kHz and converted back down to 44.1kHz. I'm not even going to say which is which, because it's audibly obvious which file reproduces high frequencies more cleanly and accurately. I recommend that anyone who wonders whether running a project at 96kHz can improve the sound listen to these two files.
 
P.S. I also read on the web that digital filters aren't perfect, which may be part of the story as well.
 
 

I remember asking about this stuff when I first started with Sonar and was told that in general 48 was fine for general audio stuff but some synths and effects could perform better at higher samplerates. They cited the effect higher SRs had on Zeta+ specifically so I guess you confirmed that.
 
I was also under the impression that this is the exact reason why some effects and instruments have those oversampling features (like GR which you mentioned in the original thread). That way you can work at lower rates but still get the full benefit of the effect/instrument.
 
Either way I had considered dropping to 48k but considering TH2 does not have oversampling and how often I use it AND the potential for MAYBE getting some improved sound quality I'm just going to switch back. I was only really doing it to save disk space and lower resource consumption but once I fix up my file locations and start being a little smarter about storage/cleanup I should be fine anyway.
 
Cheers.

Craig,
The Z3TA-2 has a render resolution option. Could this be affecting the results?
 
Michael

One more question: Wouldn't a higher sample rate also spread out quantization noise over a wider bandwidth? I also wonder about jitter. Wouldn't a higher sample rate distribute any jitter over a larger number of samples, which when interpolated and filtered, would give better results?
 
I'm not trying to come up with reasons to justify running at 96kHz, I'm trying to come up with reasons that explain why running at 96kHz sounds better for signals generated inside the computer. I don't think it's solely a question of poor plug-in design, because if it is, I sure have a lot of poorly-designed plug-ins  And if it is that common, well, all the more reason to compensate for that deficiency wherever possible.

microapp
Craig,
The Z3TA-2 has a render resolution option. Could this be affecting the results?
 
Michael

Synths and amp sims with oversampling do sound better with oversamplin engaged. Although theoretically running something at 96k should sound better than running at 48k with 2X oversampling, I don't hear a difference. But the bigger point is not everything has an oversampling option (conversely, Live's EQ8 has oversampling always turned on and you can't turn it off), so running at 96kHz is a one-size-fits all solution compared to hunting down the oversampling options and enabling them.
 
This is a new world for me so I'm still looking for answers. But I don't "have a dog in this fight." A lot of people come up with theories, then look for evidence that supports the theory. I was looking for evidence that 96kHz didn't make a difference, but found out I was wrong.

This does seem to be a rather contentious issue which strikes me as a little weird but I'm in the camp of "If ain't hurtin' anything and it might maybe possibly perhapsly make things sound better then I might as well do it?" All the sciency stuff just becomes something interesting to learn about.
 
I do actually notice a bit more crispness at 96k in general anyway. Always have. Might be my imagination but hey... I'm crazy anyway. Might as well roll with it. lol

Craig,
Quantization noise and jitter would only apply when going from analog to digital I would think. I would expect the synth output to be sync'ed with sonar's clock which in a freeze or bounce is not in real time. I suspect something else is going on here. The sample Z3ta-2 files you posted sound to me like what happens when Z3ta is switched from low to high resolution. (Similar to a fast bounce vs a fully rendered off-line bounce).
Perhaps there is an interaction between Z3ta output rate and sonar sample rate. Check the output resolution of the Z3ta. I am not in my studio right now but I will attempt to repro your results later tonight.
Michael

I agree with Beepster that this is a contentious issue. I wish plugin and synth vendors would provide a little more detail regarding their products. I don't mean for the free ones but if I pay top dollar for a pro bundle, it would be nice to know a little more about the internals. These are supposed to be pro products.
It is expected to get some detailed specs  with outboard gear, why not for synths/plugins?
Even with Sonar I have a list of unknowns regarding things like plugin headroom. Yes I know there is like 1000 db of dynamic range for 64 bit float but where exactly is 0 db within this ?
Just sayin'
Michael