thank, John... I'll take a look... I'm a dummy in this "realm" at this point and want to understand...
My whole objective is to ideally keep multiple signal chains from overloading and if it is a specious argument, it's worth knowing...gotta lot of reading to do on both sides of the fense within this thread...

It bothers me that I and others have been accused of giving out misinformation. This is something I have been very much fighting against ever since i have been on this forum. I have been wrong in the past. Not often, however.  When I am wrong I will broadcast that fact and try my best to correct the error. 
 
Goddard has accused me and just about all that have participated in this thread of being wrong. In fact it is he that is totally wrong. he has somehow confused 1 bit recording with 20/24 bit PCM recording. They are very different things and are interesting as a study in their differences but the technologies are very different. The one has no reason for being interjected in this thread. 
 
 

OK, Gurus...
I stepped away to pour over a number of cited references... and of all of them, this one seemed to ring true or at least was couched in a fashion that I can relate to... what I came away with is Sonar probably does this internally anyway, at least for effects (does it do it for busses and tracks?) and therefore the real world distinction between say, what I'm running (96/24, primarily for Latency with my Forte) and 96/32 is non-existant...except for much bigger files/disk&ram demands... 
 
http://www.sonicscoop.com...pth-is-probably-wrong/
excerpt:
"32 Bits and Beyond
Almost all native DAWs use what’s called “32-bit Floating Point” for audio processing. Some of them might even use 64 bits in certain places. But this has absolutely no effect on either the raw sound “quality” of the audio, or the dynamic range that you’re able to play back in the end.
What these super-high bit depths do, is allow for additional processing without the risk of clipping plugins and busses, and without adding super-low levels of noise that no one will ever hear.  This extra wiggle room lets you do insane amounts of processing and some truly ridiculous things with your levels and gain-staging without really thinking twice about it. (If that happens to be your kind of thing.)
To get the benefit of 32-bit processing, you don’t need to do anything.  Chances are that your DAW already does it, and that almost all of your plugins do too. (The same goes for “oversampling,” a similar technique in which an insanely high sample rate is used at the processing stage).
Some DAWs also allow the option of creating 32-bit float audio files. Once again, these give your files no added sound quality or dynamic range. All this does is take your 24-bit audio and rewrite it in a 32-bit language.
In theory, the benefit is that plugins and other processors don’t have to convert your audio back and forth between 24-bit and 32-bit, thereby eliminating any extremely low-level noise from extra dither or quantization errors that no one will ever hear.
To date, it’s not clear whether using 32-bit float audio files are of any real practical benefit when it comes to noise or processing power. The big tradeoff is that they do make all of your projects at least 50% larger. But if you have the space and bandwidth to spare, it probably can’t hurt things any.
Even if there were a slight noise advantage at the microscopic level, it would likely be smaller than the noise contribution of even one piece of super-quiet analog gear.
Still, if you have the disk space and do truly crazy amounts of processing, why not go for it? Maybe you can do some tests of your own. On the other hand, if you mix on an analog desk you stand to gain no advantage from these types of files. Not even a theoretical one..."
and...
"All Signal, No Noise
To give a proper explanation of the mechanics of just how the relationship between bit depth and noise floor works (and why the term “resolution” is both technically correct and so endlessly misleading for so many people) would be beyond the scope of this article. It requires equations, charts, and quite possibly, more intelligence than I can muster.
The short explanation is that when we sample a continuous real-world waveform with a non-infinite number of digital bits, we have to fudge that waveform slightly in one direction or another to have it land at the nearest possible bit-value. This waveform shifting is called a “quantization error,” and it happens every time we capture a signal. It may sound counter-intuitive, but this doesn’t actually distort the waveform. The difference is merely rendered as noise.
From there, we can “dither” the noise, reshaping it in a way that is even less noticeable. That gives us even more dynamic range. At 16 bits and above, this practically unnecessary. The noise floor is so low that you’d have to go far out of your way to try and hear it. Still, it’s wise to dither when working at 16 bits, just to be safe. There are no real major tradeoffs, and only a potential benefit to be had. And so, applying dither to a commercial 16-bit release remains the accepted wisdom.
Now You Know
If you’re anything like me, you didn’t know all of this stuff, even well into your professional career in audio. And that’s okay.
This is a relatively new and somewhat complex field, and there are a lot of people who can profit on misinforming you about basic digital audio concepts.
What I can tell you is that the 22-year olds coming out of my college courses in audio do know this stuff. And if you don’t, you’re at a disadvantage. So spread the word.
Thankfully, in a field as stimulating, competitive and ever-evolving as audio or music.
Keep on keeping up, and just as importantly, keep on making great records on whatever tools work for you – Science be damned."
 

I think I'll call it a night... and thanks to all for an interesting mental exercise... I like Justin's closing thought...
 "lifelong learning is half the point of getting involved..."
 

Vastman
OK, Gurus...
I stepped away to pour over a number of cited references... and of all of them, this one seemed to ring true or at least was couched in a fashion that I can relate to... what I came away with is Sonar probably does this internally anyway, at least for effects (does it do it for busses and tracks?)

Yes, it does apply to busses and tracks. This is why you can have signals within your mix that go over 0db without clipping.

John
I think Goddard is confused about 1 bit audio recording and the rest of the audio recorders. 
 

God only knows if that's it. What's evident, though, is that he's got absolutely no idea what he's talking about. It's sort of fascinating.

I know what you mean. It is fascinating but only in the way an accident is. LOL 

OK before I let myself get caught up in putting some one down let it end here and now. Often threads like this can and often do result in conflict. That is not the way we should let things happen any more. Goddard was only posting what he thought was true. There is no sin in that. 
 
I like it when some one posts a correction when they believe it is needed. We need to let members feel this place will not jump on them just because of a disagreement. 
 
We need to keep this place free of intimidation or make people feel uncomfortable no matter what is posted.   

It's very odd. Bits of knowledge being deployed as a kind of one-upmanship, but not in the service of actual understanding. It's inaccurate of me to say "has no idea what he's talking about". There are clearly remembered facts here. But they're being flung around in a hideous mish-mash of misleading irrelevance and deliberate obfuscation. I remain perplexed as to what the point of it is.
 
The article at the top of the thread is a good and correct layman's explanation of the topic. It'd be worth taking issue with it if it was making incorrect claims, but that's not what's happened here. It's seems more that Goddard simply doesn't like the idea of layman's explanations, even if they are accurate.

Nod as good as a wink...
 
I think I heard something more from my vocals using 48k a long time ago but it was very subjective and subtle.
 
I tried higher rates but could tell no more difference and lost half of my UAD resources.
 
48k runs smoother in my DAW because my korg synth is 48k native, so is my TC interface and my VL2.
 
Lot less stumbling and stalling in 48k for me.
 
 
 
 

Gee whilickers, it's been awhile since we've had a heated multi-page technical discussion! And one with a pretty decent signal-to-noise ratio, too. (Look up similar threads on Gearslutz to see just how uninformed and rude such conversations can get.)
 
If nothing else, this has prompted folks to seek additional self-education on the subject. Way to go, CW forum.