Yes John I agree with you too. It has been an eye opener for me as well. I have always assumed that virtual instruments running at any sample rate would give a very similar result and so Craig thought and started out too and found out different. I am quite surprised myself in what it has shown up.
 

Anderton
One panelist said that to have a reference point for good sound, you should listen to vinyl. I disagreed - I said the reference point for good sound is live music in a room with good acoustics

Yes I tend to agree with you Craig in this eg an orchestra or a live totally un amplified Jazz ensemble for example. But I do have some fond memories of how good vinyl can sound and in some cases it can be an amazing reference point. For example in my Hi Fi heyday in the 70's the finest system I have every heard in my life was a premium turntable and arm fitted with an Ortofon SL15Q pickup (freq response flat to 50 kHz!) feeding a super high end preamp and driving Williamson class A valve amps feeding Quad Electrostatic speakers. (flat to over 23kHz) Also Sheffield Lab records where the studio is connected direct to the cutting lathe with nothing in between! Tape does not compare to this.
 
This has to be heard to be believed and it is what I still have in my mind as a reference point. Pretty hard to beat. There is vinyl and then there is vinyl!
 
PS I owned the above setup for a long time so I know what it sounds like. I also added sub woofer to the Quads. EVERY genre of music sounds unbelievable on this setup. Electronic music sounds ridiculous on this as well.
 
 
 

drewfx1
 
Basically with 2 tests and different results - and neither of them independently confirmed - things are at a dead end until there is some further credible evidence of some kind, which I am unaware of.
 

That was my first reaction, but I was told there have been additional experiments along these lines and the presenter said he'd be happy to provide me with links. I'll request them when I'm back home next week.

John
Putting aside the issues here I just want to say this has been one of the finest epic threads I can remember. All the participants deserve a pat on the back. 

What I'm hoping is that it will inspire others to conduct their own tests and report back on the results. I do keep hearing from people I trust that they prefer recording with 96kHz. They don't really want to because their system gets more squirrely and the file size gets bigger, but they do it regardless and not because of client requirements or fashion. 
 
I'd particularly like to hear from more people who don't use distortion or virtual instruments as to whether they hear a difference.

Anderton
A couple engineers have told me that certain converters perform better at 96kHz than 44.1kHz and some perform worse. If it sounds better or worse at 96 it could have nothing to do with the sample rate itself, but how the converter reacts to that sample rate.
 

Bit has been saying this for years. He points to the matching of certain hardware components. In other words, the engineers building the DAC choose components that will be better suited to one sample rate or another.
 
Bit?
 
I'd also be interested in directions for finding out for my particular unit, what that answer is.

.

.

I record at 96/24 not because individual tracks sound any better, but when mixing everything together, the end result is more open than when I mix at 44.1/24. The fact that some VST's render better at 96k may also have something to do with it, whether it is due to poor programming, bad drivers or whatever, I'll take it. Anything that makes getting a mix where I want it to be is a plus.
 
Regards,
Dan

Again, I point to the fact there are now coming up on 200 posts on this matter, which tells me that even if there is a difference, it just cannot be that significant.
 
I am meanwhile cheerfully tooling along at 48/24.  :)
 
Bob Bone
 

mike_mccue
How do we find out if a dsp programmer has incorporated CPU saving short cuts and or round offs to their products' calculations? How do we find out what oversampling options are specifically doing with those calculations. For example; Is "2x" oversampling in a dsp residing in a 44.1kHz project actually using all the increments available in 88.2kHz or does that just bump it up to twice the detail of the actual calculations happening in the dsp?
 
For example; How do we know that a synthesizer isn't just grabbing every other sample from a stream and mashing stuff together into the "synthesis"? I guess we'd have to ask.

I would judge whether they took short cuts by ear. If something sounds good in the way that I use it, curiosity aside I don't really care why. And if it doesn't sound good, or something else sounds better, I wouldn't use it. And the "something else sounds better" is a big reason why it makes sense for plugin authors not to take shortcuts. 
 
"Detail" isn't the right word. Higher sampling rates allow for higher frequencies and this drives the frequencies that will cause aliasing higher as well.
 
In an ideal world, CPU wouldn't be a limiting factor (and today it is much less than it was a few years back), and the plugins' authors would choose an appropriate sample rate to eliminate audible problems for the type of processing that they're doing.
 
Where it gets tricky is some things will only really require a higher rate with certain settings - i.e. a compressor may be absolutely fine unless you push it to really fast attack/release times and set the threshold so it's constantly crossing back and forth. So do you always oversample it and waste CPU for the 90% of uses where it will be fine, or never oversample and potentially get some unpleasant distortion with specific settings, or do you give the user a choice - knowing that most of them will have no idea when it needs to be turned on and when it will have no benefit?

This is for Mike...
 
Image with the Mic Cable plugged in but not including the microphone. In this case 48v Phantom is on.

 
Same situation in the next photo, but 48v Phantom is off