Okay, I think I have a lot of my issues resolved Sunrat! I found an article that I considered to be pretty informative, with explanations that either made sense or weren't too far above my head:
https://www.soundons...audio-interface
I think I already had a very general and broad-based understanding of this, but this helped provide some of the "infill" and "nuance". As I previously thought, this seems to confirm that the majority of us can't detect anything less than a 12ms latency, which suggests that my latency issue above was probably the result of importing a "click track" at 48000 sample rate, but recording at 44100 sample rate. I have a tendency to start tracks around the 20sec mark, rather than zero, so I can tinker with intros at a later stage if I'm inspired to do that. I think the different sample rates caused the start points to be off by a little. I was able to "nudge" the track to sound in sync, but it was a PITA and shouldn't have even been necessary. I imported a "click track" again, at 44100 and recorded a sample track next to it at the same rate and everything sounded fine. Shut it down, restarted, and checked again, and everything was fine. So I think I just did something stupid in the beginning; but apparently it was necessary so I could get a further understanding of bit depth, same rates, buffers, periods/buffer, and latency.
Without providing a play-by-play, I spent a couple hours tinkering with Cadence, Ardour, hardware, and software settings. Long story short, I've got my settings dialed in at 44100 sample rate and 128 buffer, with 3 periods/buffer and a low-latency kernel, which produce a latency of 2.3ms...for all intents & purposes, that is zero latency! However, on rare and seemingly random occasions, sometimes i get 1-2 xruns (according to Cadence), so I'm thinking of dialing the buffer up to 256, or perhaps adjusting the periods/buffer. Your thoughts?
And I also thought the concept of recording with the lowest physically possible latency, but then changing the buffer sizes for mix-down to create more headroom for plugins was an interesting concept. I would've never thought of that on my own. Conversely, I've never really dealt with a lot of plugins. I usually record the signal "straight in" from a direct injection (DI) box/pedal, only adding Lexicon/Pantheon reverb on the master bus. But if I'm trying to develop a "best practice" routine, perhaps I should allow for that increased skillset in the future? I'm also thinking of changing my default sample rate to 48000, as a matter of "best practices" with my system, as I've read that 48000 provides more "headroom" without a whole lot larger file sizes. And I'm assuming that 48000 sample rates will get converted to 44100 on exporting to CD/Redbook format?
So I think I have a lot of my questions answered...at least enough to generate some forward momentum. But referencing my prior post, I'm still wondering why you chose liquorix over low-latency? I had never heard of liquorix, but google seems to suggest it's a matter of personal preference, with everyone believing their preference is the superior one. Shades of old gnome/kde and nano/vi discussions, LOL! I installed low-latency because it's already there in the Ubuntu repos...should I be looking at liquorix? What advantage(s) does liquorix provide over low-latency?
Other than the intellectual enlightening you can provide to Hedon, I think I have a working setup now?! Ardour is a bit off for me, compared to Cubase, but it's similar enough that greater familiarity will come with use. Next up...trying to tackle latency issues with the Omega and Cubase, running on WinXP in a VirtualBox VM on the same computer as Linux/Ardour. If I can solve that, I've got 2 more computers to be refurbished and donated to like-minded musicians who may even want to collaborate?!
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