bitflipper
 
Craig, I did not realize that you'd taken offense to my query about whether there'd be any "real" engineers on the panel. You have misunderstood me, and that is my fault. I should have explained what I meant. I'll attempt to do that now, at the risk of diverting the thread along yet another point of contention: the definition of "engineer".

 
Got it, and your explanation is much more in character with the "real" Bitflipper  But I've also been a little more sensitive about things lately...there's been a lot in my personal life involving illness and death in those close to me...which shouldn't spill over into my public life, but that's hard to avoid.
 
Anyway, I don't have much more to add to the thread either, although given how many directions it's taken, it's sort of turning into some kind of reference. I do plan to do some serious experimenting, and see what turns up. If I find out anything other than what I've found so far with amp sims and virtual instruments, I'll post the results. Meanwhile, it seems plenty of other people have useful contributions. I'm learning a lot.

mike_mccue
 
My interpretation of what it says is that really nice sounding stereo systems sound really nice.
 

My interpretation is that spending more care on mixing and mastering in order to please more discriminating ears is why the hi-res ones sound better: 
 
Partly because these recordings have not captured a large portion of the consumer market for music, engineers and producers are being given the freedom to produce recordings that sound as good as they can make them, without having to compress or equalize the signal to suit lesser systems and casual listening conditions. These recordings seem to have been made with great care and manifest affection, by engineers trying to please themselves and their peers. They sound like it, label after label. High-resolution audio discs do not have the overwhelming majority of the program material crammed into the top 20 (or even 10) dB of the available dynamic range, as so many CDs today do.

I don't mean to speak for Jeff, but I think his statement about 96/24 was more situational - i.e., there are situations where it's demonstrably better and the resolution makes an audible difference.
 

DeeringAmps
And, was it Z3ta2, when NOT oversampling, sounds better at 96k; BFD!

 
Well, I'm trying to be a gentleman regarding products from other manufacturers. I think it's more "diplomatic" to pick a Cakewalk product that can emulate the performance of other virtual instruments that can't oversample.
 

For 99% of the work done is Sonar I'm pretty sure we're "safe" at 44.1.

 
I think it depends on what kind of music you do. Given how many virtual instruments and saturation algorithms I use, I'm not in the "safe" camp but I would guess most people are. 
 

ESPECIALLY if the user is going to "master" (small m) his/her own material and convert to mp3.

 
I'm very glad you brought that up, because it's something I forgot to mention. If something does sound better at 96kHz, as noted previously any "goodness" remains if converted down to 44.1. But, it also sounds better if converted to a quality MP3 (e.g., 256 kbps).
 

And I'm pretty sure that 48/24 is WAY better than the best tape machine, especially after about 50 passes.

 
There are a lot of people who would argue that, but I'm not one of them. FWIW, if you think about it, tape with a 100kHz bias oscillator has a sort of "sample rate" of 100kHz because it's lining up all those little magnetic particles at that rate. 
 

But I don't have to admit nutin'.
Nobody saw me do it...

 
What happens in Sonar, stays in Sonar. Well, at least until you release it...
 

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DeeringAmps
"You just have to admit that 96K 24 bit is a higher resolution and mostly better sounding"
NO! YOU have made that statement, this was not the premise that Craig is/was pursuing.
Stay on topic.
Craig has stated that Amplitube in Hi res does not benefit working at 96k, and IIRC IK states its not as efficient.
And, was it Z3ta2, when NOT oversampling, sounds better at 96k; BFD!
If YOU hear a difference, then by all means run at 96k.
For 99% of the work done is Sonar I'm pretty sure we're "safe" at 44.1.
ESPECIALLY if the user is going to "master" (small m) his/her own material and convert to mp3.
And I'm pretty sure that 48/24 is WAY better than the best tape machine, especially after about 50 passes.
But I don't have to admit nutin'.
Nobody saw me do it...
 
T

That is EXACTLY what I took away from it. I wrote a long post and then deleted it and just added the post above, that I get it (what Craig did) and that it does sound better. I have nothing else to add other than questions but my understanding of the OP was the same. Some VSTs, not Amplitude, not oversampled, recorded at 96, mixed down to 44.1 sound better because aliases get folded back into things we can here.... NOW, don't ask me about the mechanics of it but I understand the simplicity of doing the task and I certainly hear the result.. and I am glad to know about it. I don't guess I have to understand what is going on under the hood to hit the gas and turn the wheel.
 
J

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Sifted through this epic thread and my conclusion is as follows:
 
If it is so that some or most VI's/amp sims/FX sound better operating at 96k and retain that sound even after being downsampled, I really really want SONAR to have the option of "super-freezing" or "over-bouncing" tracks within a 44.1 project.
 
The process would take a track with, say, an amp-sim effect, upsample to 96 to make the magic happen and downsample again to give me a 96k magic dust-infused 44.1 wav without me having to run the whole project @ 96k from beginning to end!
 
Did that make any sense at all?
 
Sven

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SvenArne
Sifted through this epic thread and my conclusion is as follows:
 
If it is so that some or most VI's/amp sims/FX sound better operating at 96k and retain that sound even after being downsampled, I really really want SONAR to have the option of "super-freezing" or "over-bouncing" tracks within a 44.1 project.
 
The process would take a track with, say, an amp-sim effect, upsample to 96 to make the magic happen and downsample again to give me a 96k magic dust-infused 44.1 wav without me having to run the whole project @ 96k from beginning to end!
 
Did that make any sense at all?
 
Sven

That makes a lot of sense. I don't know the mechanics/difficulty of how/if it would work, but it sure seems like it would. I even like the term "over-bouncing."
 
I think you may have come up with my favorite kind of idea - something so obvious no one ever thought of it before  I guess it would likely extend the time required to bounce, but so what?
 
Must talk to the Bakers about this...

Thinking about it some more...I think you're right with it being more like a super-freeze; I think it might be closer to super-rendering, operationally speaking.
 
Seems the best way to handle this would be that upon invoking "super-render" or whatever it would be called, it would do the 44>96>44 render and then archive what it is you rendered (amp sim track, virtual instrument, or whatever). That way you could return to the track if needed, but it wouldn't be influencing the CPU.
 
Another advantage: You wouldn't need to take a CPU hit while recording from enabling oversampling on your plug-ins. It's like you could always record in "draft" mode to keep the latency down, then when you're ready to mix, do the super-render trick to end up with a higher-fidelity audio file, which wouldn't stress your computer out much anyway,
 
Now I REALLY must talk to the Bakers about this.........!!!