I play on a GK backline 1x12, and, although it can't compare with something like the AI, you should still be able to get a pretty decent sound in a lower-volume environment (I'm assuming you're not using your 1x10 to play with a mic-ed drummer without going through a PA system)
From my experience, I seem to get a best sound with higher gain (11 o'clock) and lower volume (9 o'clock), with all the EQ's (including the preamp's) at 12 o'clock. Of course, the gain cannot be so high that the input signal is clipped. For restaurant gigs, (although less so for big band gigs), I'm happy enough with the sound that I'm not yet convinced about spending $2k getting a AI Focus + Wizzy, which would be a definite improvement (I've tried Chris Fitzgerald's rig, and if $$ wasn't part of the equation, I'd def go for those).
Looking at the baggs specs on their site, you should be fine with the 600 ohm output impedence of the DI. As a side note, even though GK backline has a 1 M ohm input impedence, I can never get the sound that I can tolerate without going through a preamp (I use Realist->K&K).
Of course, there's also the actual playing technique too. My amp definitely seem to know when I practice and when I slack off.
Concerning why these $300 amps don't quite compare with $1000+ amps... Below is my (at least current) understanding of it.
The ideal frequency response of an amp is a constant magnitude for every frequency. This basically means that whether you're playing on a high register or a low register, the volume is exactly the same. Naturally, this is difficult to achieve: more circuitry is required to produce a more ideal frequency response. I suspect what they do is get the frequency response of the amp by itself, then make hard-wired IIR bandpass filters with some order (higher order -> more ideal, but more circuitry) in conjunction with op amps to get the ideal flat response. This still doesn't take into account the acoustics of the room.
Air, esp in conjunction with a wall, is a natural low pass filter: higher frequencies are attenuated at a much faster rate than the lower frequencies. Here's why: you probably know that each sound signal corresponds to a sine wave. With higher frequencies, you're more likely to have the sound reflect against a wall, at which point this sine wave is reflected back with enough phase delay to cause the waves to add destructively. When the phase is exactly 180 degrees, you basically have a sine wave and an inverted sine wave added together, which effectively cancels the signal. I believe this is how noise-cancellation works.
Anyways, to account for this, the EQ's have to be adjustable so that the amp can be used in the presence of air. These EQ controls are potentiometers that control the op amp that powers IIR band pass filters of a certain order, and, as mentioned above, higher order->more ideal=>more expensive. I'll mention that ideal band pass filters are ones where magnitude is attenuated evenly throughout the frequency range, which is actually impossible to achieve b/c this would require the impulse response of the filter to be a sinc function, which is defined for infinite time range... and no impulse response can be defined for the past, which is already half the signal.
lowEndRick, I know this is probably more than what you cared for... The reason I wrote this is prob more for me than for you, since I'm an Electrical Engineering student (and an ex-Music Performance student) and I'm always curious about how much of what I'm learning in school is actually applicable in real life. For some reason, there seem to be a lot of bass players that go into Electrical Engineering, so if one of you read this and see a flaw in my understanding somewhere, (which there may well be since everything here is based on my current understanding of the theories based solely on 2 undergrad signals course, and none on actual experience), I would be more than happy to be proven wrong. I find that to be the quickest way to learn.
