More accurate recording at higher sample rates at the expense of file Size and Bandwidth.

Resolution of sound. Like the difference between a 72dpi and 300dpi photo.

Higher is better, but you will not hear any difference between a 48khz and a 44.1khz on a CD since a CD is only limited to 44.1

Actually, they all allow you to choose a different sample rates.

Latency means the delay between the time when the input is received, processed then output.

DSP = digital signal processor So a DSP Mixer is an onscreen digital mixer. So there is zero latency (delay) if you use a software mixer vs. a hardware mixer.

The DSP mixer in the Firebox routes the incoming signal directly to the monitor/headphone outputs, without a trip through the computer.

Global probably means you can't turn Phantom Power off. It's always 48v.

Global means they have one switch for the phantom power on both mic preamps, instead of a switch on each mic input.

Only for extreme professionals with electrostatic headphones will tell any difference.

20hz to 20khz is pretty much universal. That's beyond most people's hearing range.

Analog to Digital Converter and Digital to Analog Converter. The range between the weakest audible signal and the strongest clean signal. A narrow dynamic range will result in clipping distortion (rasp or buzz) when you set the minimum signal up in clearly audible range. You have to compress the signal to pass it all cleanly. A wide dynamic range allows relatively uncompressed program material to pass without distortion.

No. They are so low these days.

Assuming they're using decent converters (and these are all decent, really), the dynamic range is determined by the bit depth. Since all of these can record and playback 24-bit audio, they'll all be about the same in dynamic range, and it exceeds the dynamic range of human hearing.

With S/N ratio -100, it's so far down the only hiss you hear is from your power amp.

In analog and digital communications, signal-to-noise ratio, often written S/N or SNR, is a measure of signal strength relative to background noise. The ratio is usually measured in decibels (dB).

When there is no sound present, in many amps and other audio components, there is usually a bias or hiss. signal to noise ratio tells you how much of this hiss there is.

S/N only really counted in the past with transistor, tube amps and tape players. Digital has almost made it a non-issue.

No, just means you need to use the provided adapter. It is a cost saver. One plug vs. 4 plugs. It is also a space saver on the box.

no, it's not a problem. It's a reasonable way for them to get around the limited amount of panel space on the back of the unit.

Yes, if you want to record to a cassette deck or some other stereo device.

non-issue. These all have several outputs, conversion of any pair to a stereo output would take a simple y-cable.

Balanced signal uses a third conductor to deliver the bias current at a controlled level, and keep it separate from the grounded conductor. It reduces RF noise.

...basically, if you are running a chord over 25 feet, use balanced. Vice versa.

The wiring three-pin balanced connectors is designed to reduce noise and interference. You can use much longer cable lenghts from a balanced connector.

Bottom line, a balanced pair has better noise resistance, but it's not really critical on high-level signals, like the line level used to drive an amp or powered monitors. On microphones, balanced cables are normal, and quite important.

Having a wide range of gain levels available allows use of a wide range of input devices. The ones you'll be most concerned with are mics and guitars.

How loud the preamp can output. It is nothing to worry about.

Pretty much.

It's similar to compression. You set a high threshold and a high ratio, so all it does is clip off transient peaks (pops and thumps in the microphone for instance, or pick attack on trebly guitar) without compressing the overall sound in an audible way... If the interface has the dynamic range to carry the peaks, you might compress the signal downstream to make it fit your mix. If not, you may have to limit it before it hits the interface, or reduce the gain. If the limiter is built in, it's a nice feature.

If you have someone scream in the mic, the limiter will drop the volume level automatically so that it will not distort the recording.

I tend to stay away from these and try to keep the volumes at reasonable levels. You do not want to limit your dynamics.

The Edirol blurb suggests the limiter is valuable as a field tool, where your first take is all you'll get. It's for use in live situations, where you can't control everything that happens. In the studio, set a level that'll cover your dynamic range and avoid limiting.

Human hearing spans from about 200hz to 20khz, when you're very young. You can feel things below 200, but not so much hear them. It's more important to get even reproduction across the audible spectrum than to reproduce above 20khz.

Not really. If you were to listen to a cymbol using both units, I doubt you would be able to identify which recorded what.