According to IK Multimedia's chief engineer, physical guitar amps generate harmonics well above the audible range and part of their emulation process is to reproduce those frequencies. He also said that high-gain amp sims often deliver 60dB of gain.

Amplifiers, yes. Guitar speakers, no. And most microphones couldn't pick them up anyway.
 
Stick a microphone in front of a guitar speaker cabinet and play a fat distorted chord, record it at 192 KHz and analyze the spectrum. The amplitudes of supersonic harmonics will be very, very small - if detectable at all - and they'll be the product of unpleasant intermodulation distortion. Your typical guitar amp and speaker will roll off steeply over about 12 KHz, and even if it didn't your microphone won't pick up much beyond 20 KHz. Certainly not the ubiquitous SM-58 that's so commonly used for this purpose.
 
Have a good time at the show, Craig, and remember: if you don't come back with snapshots it didn't happen.

.

Anderton

Sanderxpander
I would think at timescales like this simply moving your head into a different angle would have a more significant effect (and thus negate any difference between the output of the two speakers). I haven't read Moorer's article yet but generally speaking this seems a pretty wild conclusion to draw from a carefully done experiment in very controlled conditions. That's not really how science works (although the media would like it to).

It's not just Moorer, check out the article I linked to from JARO. It has references going back to the late 50s. It almost seems this is an "everyone knows that" kind of thing in the field.
 
I'm not saying it's right or wrong. I haven't done the experiments myself. But I'm not arrogant enough to say that I flat out don't accept it because it doesn't seem right, or obsequious enough to flat out accept it because a bunch of researchers with doctorate degrees tell me it's so. 
 
The one thing I DO flat out accept is that instruments without oversampling sound better when recorded at 96kHz and I have files that demonstrate that to more than my satisfaction. And as I side note, I looked for instruments and processors that had switchable oversamping capabilities...there aren't that many. Either it's done internally and is transparent to the user, but then I don't understand why they sound better if run at a higher sample rate, or it's simply not built into the design.

I'm not disputing the validity of methodology of the experiments (even though "from the 50's" means "ancient and outdated" from a biological and psychological perspective) because I didn't read them in full, nor do I plan to. I'm disputing your extrapolations from them. You seem to take the conclusions from these very controlled experiments and apply them to a wide range of real world scenarios. That's not how science works, sadly. There are too many variables. You can't use an experiment that seems "kind of similar" or "touching the edges of your point" and use it to make your case. You would have to set up a separate experiment with the exact variable you want to measure in mind, taking care to avoid all others influencing it. As it is, they did not test for what you're using it for.

As a result of Craig's experiments I decided to do one myself. I did all this inside Studio One. In my case I have used two virtual instruments that are complex in nature. The first is Natve Instruments Prism. This is an amazing synth that can generate very complex upper patials. Harmonics that not only have complex amplitude opeartions on them but the frequencies can move smoothly sideways creating even more 'movement' in an ambient patch for example. The way the harmonics move in Prism patches has to be seen to be believed. Here the Prism sound is called 'The Witcher' I played an ambient midi part to control this.
 
The second VST is Korg Wavestation with 'Deep Atmosphere' selected in RAM 11. Wavestation adds fatness and also beautiful and complex movement within a patch. There is always a lot going on in any Wavestation patch! This is blended quite lower under the Prism sound. So now we have a quick ambient thing lasting 30 seconds. A blend of both parts.
 
There is no processing or eq on any of these VST's. Levels are set to unity as well.
 
I created a session at 96 K and rendered the result out at 96K. All bit depths are at 24 bit. I also exported the same 96K session down to 44.1 Khz and that is the wave in my download. Next I created a session all at 44.1 KHz. Still 24bit. Set up the same two synths and got both of the midi tracks to play them and exactly the same music resulted. I exported that session at 44.1 Khz. I think I level matched these two waves OK as well. Timing is spot on as if you line them up you won't hear a shift in timing.
 
Here is the link for the Zip file. Good for about 7 days I suppose.
 
https://www.hightail.com/...ZUczRkJRdWNIcWNVV01UQw
 
On listening you will notice the original 44.1 K version is very much brighter. On Span there are a lot more harmonics jammed in and the response is flatter up high. Notice how smooth the 96K rendered version is in comparison. I think because of the complex nature of the sounds involved the differences are even more apparent. I believe the extra high end generated in the 44.1K version is a result of the session not being at 96K. The 96K version sounds better to me.
 
If you are wondering how did they sound at the time. When the session was in 96K it was the smooth sound the rendered wave has live. The Prism patch is not swamped in high end harnmonics at all and is quite smooth and analog sounding almost. Span shows this with less extra harmonic crowding in and the top end has that 6db/Oct roll off type slope instead.
 
The next part of the experiment for me is to do a test with an incoming hardware synth into two sessions and render one down as before. The hardware synth needs to be special for this test and I have the perfect one on loan right now for the job. A Kawai K5000 additive synth with patches that are very similar and very complex and moving etc.. Sounds most unlike anything. I can layer this with some dreamy JD800 patches for some depth. It will be interesting to see how they compare going through the audio interface instead of being generated internally. (the results may be similar too because if the session is opened up to 96K does not that mean the frequency response through the whole system will be wider now.)
 
There is a very compelling reason to work at this rate just to get this smooth more natural top end sound in instruments such as Prism producing these type of patches alone. It is obvious the good things are translated down to 44.1K
 
 

Neither of those have an oversampling or HQ version? I can very clearly hear the difference between "normal" Z3TA+ and the highest quality setting.

In the two versions of the experiment, one uses Sonar (or another software) to downsample (sample rate conversion) the final wav from 96K to 44.1K. The other method relies on all the various sound generators to generate "clean" high end at 44.1. From reading all of the above, it sounds like this alone could produce "differences" that may or may not be pleasant to the individual listener, even if all the software is "well behaved". It could happen.
 
I wonder if there could be a different result by upsampling the 44.1 wav to 96 and downsampling it again, to impose whatever high frequency filtering is carried out in your chosen downsampling s/w to compare it with the downsampled wav that was originally created at 96K.

Jeff, I wonder how much one may really conclude from tests in which the signal source is digital to begin with. You're essentially re-sampling and thereby introducing new variables.
 

I would think that multiple years of these issues, with hundreds of posts per thread, would tend to indicate that there may not be much of a difference at the end of the day, in practical terms.
 
If it takes this much discussion and back and forth with still no clear cut answer, isn't that telling in and of itself, in terms of it not being significant when all is said and done?
 
Bob Bone
 

Not really, I agree with Craig's general point that some plugins may not have an oversampling mode while they should and sometimes you may just forget to turn it on. You would avoid this by running at 96KHz. I can very clearly hear the difference between the Z3TA+ modes so as long as your computer is not complaining, why take the risk at all?

The whole point of the OP and original experiment was in fact to use only digital generating sources and see how they behave under different conditions.
 
The fact is the file of mine that was created at the start at 44.1 sounds quite different to the file that was created at 96K and rendered down to 44.1 The down sampling process has maintained the beautiful smooth sound from the 96 KHz session.
 
I see these differences varying according to the type of sounds you are making too. For highly additive complex harmonially rich sounds it might be good to work at 44.1 mostly but create a session up at 96k to just render out all the complex additive sounding stuff. Down sample it to 44.1 and drop it back into the 44.1 session. It is going to sound different no matter how you look at it. (I am not suggesting you do this for everything either it might be just one of those things where VST synths/additive synths really show up these differences with some patches too)
 
Span shows less harmonics (peaks are spaced further apart) in the 96 K version and a smooth roll off compared to the extra stuff (more harmonics squashed in) showing up in the 44.1 Khz version and the flatter roll off. Putting a LPF over the 44.1 KHz version will not turn it into the 96KHz rendered down to 44.1K version either.