Still waiting to hear if Prism and Wavestation have a HQ/oversampling mode and if there is a difference.

Time to bump this thread, it is getting lost and is seriously important!
 
Craig has brought up a very valid point. I think my second experiment confirms the fact that when you are recording from the outside into your DAW and even when that instrument is something very complex sounding such as a Kawai K5000 it can be said there is probably very little difference between sessions running at 44.1K or 96K. Like we used to think and that seems to hold true.
 
But NOT when a VST synth such as Native 'Prism' is being used to create the parts and being used digitally right from the get go. Then it seems 96K makes a big difference. So much so that I was thinking of doing a rebuild for a new setup but now I am re thinking going 96K all the way as it makes such a difference to some VST's running at that sample rate.
 
Something to think about for sure.
 
 

bitflipper

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.

I think you're still missing the point, it's not about high frequencies we hear, it's about high frequencies that cause distortion in such a manner that it creates artifacts at frequencies we can hear...and which speakers can reproduce, and microphones can pick up.
 
The thing that surprised me the most about the experiments I did wasn't that the non-oversampled processors and instruments sounded better at 96kHz, but that the audible improvements remained when downsampled back to 44.1kHz. In retrospect that probably shouldn't have been a surprise, because 44.1kHz is quite capable of reproducing sound within the audible range. So if I could hear an improvement in the audible range at 96kHz, it makes sense it would translate to 44.1kHz.

 I think you're still missing the point, it's not about high frequencies we hear, it's about high frequencies that cause distortion in such a manner that it creates artifacts at frequencies we can hear...and which speakers can reproduce, and microphones can pick up.

No, I'm not missing the point. This angle has been discussed many times before.
 
Supersonic frequencies do mix acoustically to produce sum and difference frequencies, and the latter are indeed audible. Cymbals, for example, can have twice as much energy above 20 KHz as below it. That beautiful mash-up of frequencies bounce around the room and come back to the ear as a very complex - and definitely audible - sound. Without those inaudible frequencies, the cymbal would sound thin and cheap.
 
However, it isn't necessary to record those supersonic frequencies to get a great cymbal sound, because even though they're actively involved in the final sound, that activity takes place in the air, before getting to the microphone. By the time we encode it as digital data, the magic's already happened.
 
Guitar amplifiers are different from acoustical instruments such as cymbals, though, because their acoustical output is generated by components that are physically incapable of reproducing supersonic frequencies. You may very well have harmonics generated by vacuum tubes as high as 100 KHz, but most don't make it past the output transformer and the ones that do certainly don't make it past the speakers. Nobody bothers putting Earthworks microphones on a Fender Twin, for good reason.
 
Your best candidate for justifying higher sample rates probably lies in virtual instruments and effects that can create frequencies beyond Nyquist: software synthesizers, distortion plugins and fast-attack dynamics processors. None of this applies to sampled instruments, nor to non-distorting processors such as reverbs and equalizers.
 
Even in those cases where a processor or an oscillator might generate supersonic content, a mere doubling of the sample rate isn't an effective way to deal with it. While there will be fewer frequencies to alias, there'll still be plenty of potentially harmful harmonics to make trouble. You'd really need to at least quadruple the sample rate to be effective, but nobody's making a case for 192 KHz sample rates.
 
I'm not saying that there necessarily isn't some kind of justification for 96 KHz. But over the years people have tried and tried and grasped at increasingly unlikely straws to make the case and so far nobody's come up with a rationale that's supported by science.
 
So Craig, who's going to be on that panel with you? Any real engineers?

bitflipper
Your best candidate for justifying higher sample rates probably lies in virtual instruments and effects that can create frequencies beyond Nyquist: software synthesizers, distortion plugins and fast-attack dynamics processors.

 
I thought it was clear at the very outset that's what this thread was all about, and why I started it. If you've been under the assumption that WASN'T what the thread was about, then your comments make more sense.
 
 

Even in those cases where a processor or an oscillator might generate supersonic content, a mere doubling of the sample rate isn't an effective way to deal with it.

 
An audible improvement is by definition effective. I've played comparison files for many people and they all hear the difference. Jeff Evans ran his own tests with different gear and heard an improvement. I can pick out which file is which 100% of the time. So can Steve Fortner from Keyboard magazine. None of this "well, better than chance." There are plenty of references on the web that justify the math behind what kind of audible artifacts you're likely to obtain at various sample rates, and their intensity; 96kHz is an improvement over 44.1kHz yet remains relatively cost-effective.
 

While there will be fewer frequencies to alias, there'll still be plenty of potentially harmful harmonics to make trouble.

 
But you said "Let's look at a practical example, an electric guitar played through a high-gain amp sim...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." So which is it? Are the harmonics from distortion going to be "inaudible," or have "plenty of potentially harmful harmonics to make trouble?" 
 
I think the answer lies in between, which is why simply doubling the sample rate produces an audible improvement.
 

You'd really need to at least quadruple the sample rate to be effective, but nobody's making a case for 192 KHz sample rates.

 
Actually, there are people within the AES, CEA, and the record industry debating whether 96k is enough so it is incorrect that "nobody's making a case for it." These involve some very heated discussions by people who have impeccable academic credentials in digital audio engineering. Unfortunately, 192kHz really cuts down on the ability of digital audio interfaces to stream audio using existing computer ports (although I haven't stress-tested Thunderbolt yet). Just as 96kHz was not practical when the CD was invented and so those wanting a higher sample rate than 44.1kHz were overruled, the same could happen with 192kHz, and for the same reasons. Someone who records only classical music could completely justify the argument that 44.1kHz is all that's needed. NIN might have a harder time...so how much should consumers have to pay if they only listen to NIN? Or only to classical music? Tradeoffs must be made in the real world. 
 

I'm not saying that there necessarily isn't some kind of justification for 96 KHz. But over the years people have tried and tried and grasped at increasingly unlikely straws to make the case and so far nobody's come up with a rationale that's supported by science.

 
The premise of this thread is supported by the science of aliasing, Nyquist, clock frequencies, foldover distortion, the harmonics generated by particular processes internal to the computer, and the effects of raising sampling rates through higher clock speeds or oversampling. I think those provide more than enough scientific rationale. There is both practical justification for what I claimed and theoretical justification that will remain in place as long as the concepts behind aliasing, Nyquist, clock frequencies, foldover distortion, the harmonics generated by particular processes internal to the computer, and the effects of raising sampling rates through higher clock speeds or oversampling are generally accepted by mainstream science.
 
There are recording/mastering engineers who make platinum records and hear the difference. There are audio engineers who are grounded in digital audio theory and can confirm what the recording/mastering engineers hear. The final piece of the puzzle is that the lay people I've played comparative files for can reliably detect a difference (as can a pro like Jeff Evans), and agree which recording protocol produces superior high-frequency reproduction with particular types of program material. That kinda nails it shut as far as I'm concerned.
 

Jeffs test analyzed with HarBal 3 
http://www.screencast.com/t/8v30AG0YIPlE

.

if your virtual thing is "emulating" something real, then the more accurate the maths the more accurate the emulation - it cracks me up that this stuff gets argued about like it's a religion or something... science, people, it's science!

gswitz
 
I think I see Bit's point-- If we can do it manually by up-sampling the project then down-sampling, why can't the plugin do it for us? This seems reasonable conjecture to me.

That's besides the point, which is that most of them don't. You can choose to say this is the way the world should be and complain that it isn't, or you can do something about it. I've chosen the latter.
 
His other point is that "science" doesn't support my contention that increasing the sample rate to 96kHz while recording with certain virtual processes has worthwhile benefits. ("But over the years people have tried and tried and grasped at increasingly unlikely straws to make the case and so far nobody's come up with a rationale that's supported by science.") Yet it clearly can improve the sound quality of certain processors, and science clearly explains why that is so. I don't think that can be disputed unless you don't accept theories regarding digital audio that have been developed and tested over decades.
 
Look, people can do whatever they want. I don't care if their music is filled with distortion, but I do care whether mine is or not. So I will take steps to mitigate it and have an audible improvement in the quality of what I produce and presumably, Jeff Evans will do so too. I've put this information out there, it's not disputable, and people can either take advantage of it or not.
 
Remember, the whole reason I got started on this tangent was to show that running at 96kHz DIDN'T make a difference. I was wrong. As I said earlier, I'm not interested in being right or wrong; I'm interested in the truth. I guess that's not the way scientists are supposed to think

I'm not sure that we don't just have a misunderstanding about whether some would just consider 96kHz a "workaround" rather than "intrinsically better" in the situation described.
 
IOW it's about how one characterizes the situation, not what the facts are.

The "tone" of this thread took a wild turn for the worse IMO, which actually concerns me more than the topic itself (which Drew summarized well).

Thanks jps very much for the Harbal analysis.  I appreciate it.  Interesting the high end is obviously what it is but your curves also point out some serious differences in the mids and even the low end too.  No wonder the rendered version sounds so different.
 
Mccue's statement is a bit silly.  Nothing crappy about 'Prism' or 'Wavestation'  (or Z3ta in fact)  And what if as a composer I really want to use a certain VST for its sound but it did not happen to have an upsampling option built in.  But at 96K it sounds way different.  'Prism' is a very different animal and many don't even understand it completely.  Myself included in that.
 
Part of the problem is the type of music and how it is made.  A lot of people would never make music such as I have done in 'Prism' for example.  They think a DAW's job is confined to producing music recording everything live in through a sound card.  A situation which I agree at both sample rates seems to produce very similar results but not so when I am doing something a bit different with a virtual instrument such as 'Prism'.  And it being totally digital too.  Think outside the box.
 
Drew I cannot seem to be able to put 'Prism' into a high sample rate while the session remains at 44.1K. I may have to dig deeper but when I click on the arrow all the options are greyed out.  The one that is ticked is the match host rate option. (I am running the Reaktor player though with the full version of Prism. It may have something to do with that too.)
 
The facts are very  clear. 'Prism' sounds much better at 96K.  Thanks to Craig again for pointing that out.  As I am someone who relies a lot on virtual instruments at times I feel the need to re approach a newer setup at least having the option of running at 96K if I want it to. Before this I may not have considered that so much but now I am.
 
Yesterday I did another experiement using Prism (only) and choosing a combination of a delicate percussion Gamelan patch with some very ambient detailed intricate high end harmonic patches too and the differences were even more pronounced than the experiemnts I have posted. 
 
I got a slightly different response this time. The 96K version had some slightly clearer and taller transients on the Gamelan sound compared to a slightly duller and not so transient hits in the 44.1K version. (Gamelan only though) Some of the ambient sounds in the background (on the 96K version) were a still little smoother and less strident. Just goes to show results are not always the same but what was the same though is the 96K version still sounded different and overall better.
 
Interesting.  I think 'Prism' must be one of those synths that is just pushing the 44.1K boundaries very hard.  I am sure there are VST's that don't but if I want to use 'Prism' then I should be able to and I want and need to get the best possible sound from it.
 

Jeff Evans
Drew I cannot seem to be able to put 'Prism' into a high sample rate while the session remains at 44.1K. I may have to dig deeper but when I click on the arrow all the options are greyed out. The one that is ticked is the match host rate option. (I am running the Reaktor player though with the full version of Prism. It may have something to do with that too.)

It's funny. I have the full Reaktor (not the Player) and it allows me to change it but I seem to recall having this problem with things being grayed out in the past (I have an ensemble I created that can alias like crazy with some settings), but there was a way to make it work.
 
Do you have the latest and greatest version of Player? It would be a shame if the Reaktor Player didn't let you adjust the sample rate when running as a VST. Maybe someone else here can confirm this behavior with either Reaktor or Reaktor Player?

gswitz
 
On a different note, I'm not sure that bringing up qualifications helps the conversation... saying the Jeff is Pro (suggesting that Bit isn't as qualified at least in part because he's challenging your assertions).
 

 
I'm never suggested Bit isn't as qualified. Take what I say at face value, which was that Jeff is a pro - it's not just me saying that there's a difference. He hears what I hear too, and he's not some crazy person selling cables for $3,000 a foot (and I'm NOT implying Bit sells cables at $3,000 a foot, either)...
 

I like the topic and I like how everyone has conversed on it. I want to speak up and say I also like Bit and how he's presented his knowledge. I don't think anything you or Jeff has contributed invalidates Bits challenges. They are reasonable challenges. Same with DrewFX. That dude's freakin smart too.

 
I will admit it is frustrating to have to repeat once more that I've said oversampling is a solution. Bit's saying that too, so there's no argument there. His challenges are reasonable, but he is shooting the messenger. Those challenges need to be addressed to the coders who for whatever reason have created an issue, not to the musicians who have found a practical solution that can be implemented today.
 
I've stated that oversampling-capable processors running at 44.1kHz produce results that, to me, sound equivalent to running at 96kHz. I've also said that sample rate converting something recorded at 96kHz to 44.1kHz preserves any benefit from recording at 96kHz, so I'm not saying it has anything to do with hearing anything above the audible range.
 
I honestly don't know how I can state it any more clearly than that.
 
My disagreement with Bit is his saying that when it comes to recording at 96kHz, "nobody's come up with a rationale that's supported by science" as if I'm just pulling some cork-sniffing crackpot theory out of the air. That, and what appeared to be a condescending comment about whether there would be any "real" engineers on the panel, did not sit well. I have not belittled anyone's knowledge, accused anyone of propagating misinformation, painted broad strokes about "crappy" products, or claimed to have superior knowledge. I heard what I heard, was very much surprised at what I heard, wanted to know why, and found a reason for what I heard that's supported by science. All that means is a) I have ears, and b) I know how to do research. Nothing more, nothing less.
 
The science definitely supports the concept that signals above the Nyquist frequency can interfere with a clock and cause undesirable artifacts. There really is no dispute about that. Anyone is welcome to challenge that, but they would have to come up with extremely compelling arguments to show that current scientific thinking about the relationship among sampling rates, clock frequencies, aliasing, and undesirable/audible artifacts is wrong.
 

I wouldn't want Bit to feel quelled b/c some one is being more successful in the industry than he is (or not, idk).

 
I don't want him to feel quelled. I am always willing to stand corrected when appropriate, but I do not think it's appropriate to dismiss hours of work, testing, and research as having no scientific basis. I don't make stuff up; I have yet to see anything presented here that negates what I heard, or the reasons given for why I heard it. However...
 

The news about the research in ultra-high frequency related brain activity is interesting. It makes sense to me. I imagine the brain activity coming up like a slowly growing pad. :-)
 
Also, thank you vey much for the super post (147 and 148) about where you were coming from. It helped me understand better why I was reading you as upset with Bit.
 
When Bit was talking about Amp Sims and stuff he said they don't reproduce sounds above certain frequencies (excuse me if I'm too roughly paraphrasing, Bit). Curious about my Mics and Gear, I plugged in my Studio Projects B3 Microphone (nothing super fancy) and took a screen shot of RMEs spectral analyzer as I made Shhhh sounds into the Mic.
 
From this picture, you can clearly see that the Mic picks up frequencies well above Audible range. In fact, there were frequencies all the way up to Nyquist for 192. If you zoom in on the image, you can see the freqs defined on the X axis.

 
This is a picture where the room was quiet (computer on -- AC on quiet in the background). I touched the screen to take the screen shot. I couldn't take it with keyboard or mouse without capturing the sound of the clicks.