Old 08-12-2011, 06:29 PM   #41
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So does anyone have a project file to post in which this is exhibited in an obvious way?
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Old 08-12-2011, 06:30 PM   #42
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Originally Posted by timlloyd View Post
Then with all due respect, because of what I know about psychophysics I cannot logically interpret your preferences as anything more than potentially inaccurate and probably biased anecdote.

I really am sorry if this offends you. I have said the same sort of thing to other people and will continue to do so.

This isn't about baking noodles, it's about adherence to the scientific method in light of certain realities of sensory perception.
Very nicely stated. Should be on a plaque.
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Old 08-12-2011, 07:24 PM   #43
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Actually, if you take the trouble to really learn about 32 bit floating point you'll find that it's capable of representing a dynamic range in excess of 1500 dB.
This is technically true, however in many situations you do need more precision to have a 24 bit result that is accurate and not marred by a loss of precision.

32 bit floating point can represent all 24 bit samples exactly, but 32 bit floating point can't represent all sums of N (N>1) 24 bit samples without a loss of precision.

Example:

Take 32 24-bit integer samples, sum these, then apply approximately -30dB gain, then convert back to 24 bit integer. If you do this using all integer math or 64 bit floating point, you will get a result. If you do this using all 32 bit floating point, depending on the content you will likely get a slightly different result. The differences are very small, but measurable within a 24 bit result.

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Old 08-12-2011, 09:13 PM   #44
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I prefer integer.
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Old 08-12-2011, 11:10 PM   #45
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I'm so lost after reading this thread
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Old 08-13-2011, 12:01 AM   #46
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I'm so lost after reading this thread
math is hard, i'm not trying

https://www.youtube.com/watch?v=X8CrHrx2k5M
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Old 08-13-2011, 12:13 AM   #47
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Just did a null test of 64bit float v 39bit integer using a project with 7 tracks, no plugins. The original mix peaks at about -9db.

The difference file peaks at under -200db.

Heres a pic of Bitters display while its playing...



As you can see they null perfectly above the 37th bit so i cant see how there could be any audible difference in this case.

As the converters on any high end sound card are only 24bit, they couldn't even play it back and if you dither to 24bit, you'd wipe it out anyway.
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Old 08-13-2011, 12:22 AM   #48
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I have been using an ancient HDD MultiTrack recorder from Scope DSP platform, now in x64.
I have always noticed it's 32bit Integer sound was better than the quality I got from Cubase or Reaper
I also loved VDAT years ago when I was using scope cards. It really sounded better than any software sequencer back then.
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Old 08-13-2011, 02:27 AM   #49
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I'll try to expain my thoughts about 64bits floating point:
this could be a great solution at a scientific point but there's something that seems wrong to me when "practicing" (vs theory):
when using 64bFP the audio doesn't clip internally anymore, which seems to be fine at first sight but I've found that it mostly leads to bad working habits for beginners (and others...), because the red isn't really red anymore.
You can go over 0dBfs on your individual tracks without mostly hearing it but what really occurs is you're clipping on the master track unless you lower the master fader.
Beginners (and others) mostly don't lower the master fader, because it's hard to do it: the mix doesn't sound so good anymore at lower level unless you compensate with the monitoring level.
So they don't lower the master fader and they're sending a clipped mix to the mastering...
Other common practice these days is to mix with a limiter on the buss-master:
then 64bFP can be a dynamic killer in practice because the master never goes red and you hear no distortion even when your tracks are hitting +6 dBFS! But what happens is these 6dBs are eaten by the limiter, without really being aware of it.

So to me, the commercial thing "with 64bits Floating point you can't clip anymore" looks more like "with 64bFP every clipping will goes in your master unless you take care!".
And finally I would say that 64FP could have its responsability when it comes to ask why there are so many clipped mixes even before the mastering stage.

Does it make sense to anyone?
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Old 08-13-2011, 03:24 AM   #50
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I prefer integer.
What is it that you prefer?

Quote:
Originally Posted by Billoon View Post
Just did a null test of 64bit float v 39bit integer using a project with 7 tracks, no plugins. The original mix peaks at about -9db.

The difference file peaks at under -200db.

As you can see they null perfectly above the 37th bit so i cant see how there could be any audible difference in this case.

As the converters on any high end sound card are only 24bit, they couldn't even play it back and if you dither to 24bit, you'd wipe it out anyway.
Yes, this is my experience and opinion as well. Let's imagine someone using a Lavry DA10 for monitoring such signals. It has a dynamic range of about -110dB (as stated in the spec). So, when monitoring these signals, the differences between them are about 90dB below the noise floor of the unit ... and the Lavry is within the higher end of the DA market. D/A conversion usually is not "even" 24bit.

Claiming to be able to hear differences between these types of files when the differences between them exist below the thermal noise of your monitoring chain verges on the ridiculous, and really need to be backed up with abx listening ...

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Does it make sense to anyone?
I understand what you're saying, but it's not the fault of the medium (in this case 64-bit). It's the fault of people not understanding that gain staging is still important in the floating-point digital domain. People really need to understand that ... it's not remotely hard to grasp so I have no sympathy tbh
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Old 08-13-2011, 03:58 AM   #51
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I'll try to expain my thoughts about 64bits floating point:
this could be a great solution at a scientific point but there's something that seems wrong to me when "practicing" (vs theory):
when using 64bFP the audio doesn't clip internally anymore, which seems to be fine at first sight but I've found that it mostly leads to bad working habits for beginners (and others...), because the red isn't really red anymore.
You can go over 0dBfs on your individual tracks without mostly hearing it but what really occurs is you're clipping on the master track unless you lower the master fader.
Beginners (and others) mostly don't lower the master fader, because it's hard to do it: the mix doesn't sound so good anymore at lower level unless you compensate with the monitoring level.
So they don't lower the master fader and they're sending a clipped mix to the mastering...
Other common practice these days is to mix with a limiter on the buss-master:
then 64bFP can be a dynamic killer in practice because the master never goes red and you hear no distortion even when your tracks are hitting +6 dBFS! But what happens is these 6dBs are eaten by the limiter, without really being aware of it.

So to me, the commercial thing "with 64bits Floating point you can't clip anymore" looks more like "with 64bFP every clipping will goes in your master unless you take care!".
And finally I would say that 64FP could have its responsability when it comes to ask why there are so many clipped mixes even before the mastering stage.

Does it make sense to anyone?
Then what are the best setting to hear clipping.
I remember my old days with Cakewalk 9 Audio and it distorted the signal very "easy" if you are not careful and it was really easy to reach the limits of the mater bus and hear the clipped signal. I'm not sure but it was the same with Cubase 32 5v
Also I used to mix through my Cream Ware Luna Mixer and the quality was different and great...
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Old 08-13-2011, 04:10 AM   #52
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This is technically true, however in many situations you do need more precision to have a 24 bit result that is accurate and not marred by a loss of precision.

32 bit floating point can represent all 24 bit samples exactly, but 32 bit floating point can't represent all sums of N (N>1) 24 bit samples without a loss of precision.

Example:

Take 32 24-bit integer samples, sum these, then apply approximately -30dB gain, then convert back to 24 bit integer. If you do this using all integer math or 64 bit floating point, you will get a result. If you do this using all 32 bit floating point, depending on the content you will likely get a slightly different result. The differences are very small, but measurable within a 24 bit result.
Agreed. (As I said, I've been dealing with these formats for a very long time. My little dig at 64 bit was actually a bit tongue-in-cheek.)

The purpose of my original post was to point out the very significant mathematical difference between integer and floating point. I find it very difficult to see how integer mixing could ever be preferred to floating point. I'd love to see some examples posted for ABX testing.
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Old 08-13-2011, 04:14 AM   #53
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but it's not the fault of the medium (in this case 64-bit). It's the fault of people not understanding that gain staging is still important in the floating-point digital domain. People really need to understand that ... it's not remotely hard to grasp so I have no sympathy tbh
you sir are pompous!

some of us have women to please

that's why we ask you lot these questions

PS my ears hear no difference between 64bfp & 39i
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Old 08-13-2011, 04:19 AM   #54
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You're not required to call me "Sir" except on special occasions.
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Old 08-13-2011, 05:10 AM   #55
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Then what are the best setting to hear clipping.
I remember my old days with Cakewalk 9 Audio and it distorted the signal very "easy" if you are not careful and it was really easy to reach the limits of the mater bus and hear the clipped signal. I'm not sure but it was the same with Cubase 32 5v
Also I used to mix through my Cream Ware Luna Mixer and the quality was different and great...
Yes, that's what I mean, before Floating Point (integer) a clipping could be heard as soon as it occured.
The problem with Floating Point is that sometimes red is not red and it tends to some bad mixing habits: "let's go red to get this song loud" and so on...
And the resulting mixes tends to be squashed/clipped on the master.
This technology (FP) has something of a "non-traditional audio levelling" in it that is easily fooling people...
When you choose a technology for your software you have to take care what people will tend to do with it, no?
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Old 08-13-2011, 05:48 AM   #56
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A Saw wave coming from an Oberheim OBX sounds different than the SE-1X Saw wave, always has, yet on an Oscilliscope they look exactly the same.
Greed causes me to use and like both..
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Old 08-13-2011, 06:08 AM   #57
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yet on an Oscilliscope they look exactly the same.
Greed causes me to use and like both..
Are you sure about that? What if you compared them in the frequency domain instead of the time domain ... you would be able to more easily see the slight differences in harmonic structure due to their band-limiting ...
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Old 08-13-2011, 06:14 AM   #58
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A Saw wave coming from an Oberheim OBX sounds different than the SE-1X Saw wave, always has, yet on an Oscilliscope they look exactly the same.
Greed causes me to use and like both..
You are of course wrong about them 'looking exactly the same' if they do not sound identical, but this has got absolutely nothing to do with this thread either way.

You claimed to hear something that you can not possibly hear. This is a fact and nothing you might ever write could change something about it.
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Old 08-13-2011, 06:49 AM   #59
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I beg to differ. And don't worry I am not insulted, as I assume you never had the pleasure of using such vintage gear.

Why do Piano tuners use the Oscillascope + their ears....??

Because the tools were meant to get relative pitch as referencing, then the tuner could apply his trusted ears, and place the false beats where he chooses, according to the imperfect mechanics...

Why did Horowitz and most performing Concert Pianists fly a certain tuner around the globe to tune the Piano they were suppose to use....?
Same reason why a certain engineer or producer gets called.
Their ears and gear combined are awesome and very signature.

Worshippers of science and naysay-ers remind me of the VST versus Analog, Mac Vrs. PC arguments.
You use what "you think" sounds the best. Get the science and nature working together....

Is that beyond grasping...?
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Old 08-13-2011, 06:57 AM   #60
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Is that beyond grasping...?
Not in the slightest, but you're missing the point.
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Old 08-13-2011, 07:11 AM   #61
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Xite it seems your trying to set up science as a "straw man" by comparing it to completely subjective concepts.

Science deals in things like ABx tests and massive statistical analysis to create results with the least bias possible and even then the bias is analysed in and of it's self and stated.

respectively though I'm not angry about your suggestions just that there are ways to actually breakthrough subjectiveness and find the hidden ways are minds are so easily fooled
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Old 08-13-2011, 07:29 AM   #62
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Totally agree. hearing the difference from a Curtis Chipset to another CEM version just makes me wonder what other imperfections we might not discover until years after the fact just like the Oscilliscope. Just like every Hammond or Rhodes sounded different, but back then to those who swore by the Scope, many musicians and Piano Tuners were deaf and dumb, well acording to them.

Just so maybe I might be enlightened, how would I nullify or ABX test my old VDAT app. I am curious if there's a mathematical explanation or scientific reason it sounded so much better than DAWs did for years.
It even bothered engineers with PT Racks that this nifty app had such a sparkly sound.

Much appreciated...
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Old 08-13-2011, 07:46 AM   #63
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It's a complete failure of methodology to try to use an oscilloscope to look for subtle differences in sound. Pretty much any DAW with a half-decent graphical rendering of the waveforms would be better. And as mentioned, a spectrum analyzer would be even more revealing.
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Old 08-13-2011, 08:12 AM   #64
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Agree, but my 30 Band Spectrum Analyser won't tell me what the differences I hear from SX2 and VDAT.
Does nullifying reveal this, ..? And if so I would appreciate a link as today I am loading some new instruments from LASS/VSL and have time to do some experimenting...
I'd love to see why I find certain details sound better than others.
It might also explain why I hear my new 96k hardware synth sounds better than my old Nord.....And no offense to any Nord users, these areas are so touchy, explanations and opinions seem to insult people, and I assure you, those are not my intentions.

Thanks
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Old 08-13-2011, 08:28 AM   #65
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Personally I find it exciting if some brings hard evidence (objective evidence) that challenges the growing evidence for the amount of tricks our brains play on us

I appreciate that you hear a difference but these conversations would be much more useful to all involved if we can somehow test these things for ourselves and get to the bottom of it

As far as I'm concerned sound is the really easy part of it, emulation is not that difficult compared to the power of the mind

Gearslutz has had some great listening tests that really show up some factors.

Let's find something like this here. ABx tests do help a lot and nulling is a massive help too
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Old 08-13-2011, 08:32 AM   #66
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Uploading my A/B project...
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Old 08-13-2011, 08:39 AM   #67
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I'd love to.
I have several vintage synths, 6 x various 4U DAW's for live work.
I am fascinated by this stuff, and simply would like an explantion of why I hear differences in mixes, DSP vrs. Native, all of it.

I'd love for someone here to explain how I can ABx or Nullify. As I said I have all day to test things out, it's too cloudy to catch any rays, and I have 6 more hours of DVD loads ahead of me.

What do you personally do when you nullify,,,,
And ABx testing seems to be blind testing, but I am not sure of the terminology.

But I ask these questions from developers of DSP apps, and one explanation of VDATs' quality was because it had no drivers to pass through, comes straight from the DSP chip unadulterated.......
I have no idea what he's trying to explain to me....
Plus he's German and there could be a language barrier.

Thanks for any links to the actual methods used.
Searching GS was insane.
So many angry defenders of sciences versus the " Pro Tools " experts, etc.
Rather frustrating.....

Cheers
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Old 08-13-2011, 08:40 AM   #68
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Wouldn't it be funny if the dev's added the code for the option between float/integer but forgot to actually wire it up. LMAO
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Old 08-13-2011, 08:42 AM   #69
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They didn't, at least not for the < 24int modes ... they quite obviously involve truncation.
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Old 08-13-2011, 08:54 AM   #70
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*** File removed ***

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Old 08-13-2011, 09:15 AM   #71
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I'm just on laptop speakers atm, but you've got:

- some kind of amp hum and very quiet rhythmic noise bursts (very beginning)
- distorted guitar that sounds like an amp sim
- drum reverb .......
- more guitars that sound like sims
- keys that sound like a VSTi
- some clipping in various places

The above exist in the difference signal. The existence of reverb is a definite sign of bad methodology. Bypass or render that 'verb or it's an invalid test.

The only thing that I would like to know is, are these amp sims and a keys VSTi? If they are, then unless you know with 100% certainty that their processing doesn't involve any kind of randomness/noise whatsoever, they invalidate the comparison for the same reasons as the reverb.

Render the reverb return, the amp sims and VSTi (if they are indeed those) to audio and bounce again without those plug-ins to remove them as variables.

If these things get cleared up, then we might be able to make a comparison but until then, definitely not.
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Old 08-13-2011, 09:17 AM   #72
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Facepalm...........
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Old 08-13-2011, 09:45 AM   #73
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The existence of reverb is a definite sign of bad methodology. Bypass or render that 'verb or it's an invalid test.
You want perfect science methodology, you do it.
I don't care so much for the science side of art.

This A/B wasn't designed to please the 'numbers' people.
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Old 08-13-2011, 09:49 AM   #74
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This A/B wasn't designed to please the 'numbers' people.
No but the original reason for A/B was to expose those who imagine things instead of actually hearing them. If that imagination leads to better art, I support it fully but if the numbers don't crunch, then the artist should accept it for what it is, not call it something else.
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Old 08-13-2011, 09:51 AM   #75
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You want perfect science methodology, you do it.
I did do it, and as stated in the second post of this thread, my files cancelled down to dither.
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Old 08-13-2011, 09:54 AM   #76
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Is that beyond grasping...?
It all still has got nothing to do with this thread.

Just because we have the ability to see all kinds of different colours doesn't mean old fotographs from the beginning of last century are more than just black&white.

What you are doing in this thread, is the same as if you were claiming you can see all kinds of colours in such fotographs (because you were imagining them). Someone comes along and says 'sorry, bud, but that's not possible' and proves it to you beyond the shadow of a doubt but you ignore it and then start ranting about all those beautyful sundowns and England's green and pleasant hills and the blue eyes of your girlfriend as if that would give any credibility to your claim that you see these fotographs in colour.
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Old 08-13-2011, 09:57 AM   #77
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I did do it, and as stated in the second post of this thread, my files cancelled down to dither.
You don't post it tho.
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Old 08-13-2011, 10:00 AM   #78
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I can't post the individual bounces (or I would have), but I can do it again and post the difference signal if you like? It sounds like dither noise ...

You can do the same test as me if you eliminate the variables that I suggested. You don't need me to show you this, you can do it yourself. Or you can render your sims and VSTis and reverb to audio, send them to me as stems and I'll do it.

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Old 08-13-2011, 10:16 AM   #79
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I see nothing good coming out of this thread unless, of course, people change their "tone"
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Old 08-13-2011, 10:17 AM   #80
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Render the reverb return, the amp sims and VSTi (if they are indeed those) to audio and bounce again without those plug-ins to remove them as variables.

there's a simpler way - all he needs to do is to render twice with each summing method and then check with a nulling test whether the difference between the two summing methods is significantly greater than the difference between each of the Pairs of bounces which use the same summing method.
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