When I open a new file it askes me what "sample rate" I want. I can choose from 2,000Hz all the way up to 192,000Hz. I've been spinning fro a pretty long time but im really new to recording, plus im shit when it comes to computers. What is Sample Rate? Does it effect the quality of the sound?

It also asks me what "Bit Depth" I want, again what does that mean and does it effect the wuality of the sound?

thanks

It definately affects the sound quality.

I'm not going to give the full explanation about samplerate and bit depth, cuz that will take very long, but I'll try to simplify as much as possible.

The signal that comes out of your mixer (well most actual mixers anyway) is analog (a certain voltage). Computer data, thus also audio, is digital data (1 and 0's). So when you are recording to your computer, the analog signal will be converted to digital data (your soundcard contains elements called analog to digital convertors at the inputs, and digital to analog convertors at the outputs).
But what is the main difference between analog and digital. Well, an analog signal is a continuous signal (you have a certain value for each moment), while digital data is discrete (you only have a certain value for each time segment). So to convert analog to digital, we're gonna do what's called sampling. We are going to take a value each X time. And that X time is defined by the sampling frequency. Hertz is the opposite of second (1 hertz = 1/1 second). So for example if you say the sampling frequency is 44100 Hz, a digital sample will be taken every 1/44100 second. Or in other words, in one second 44100 samples will be taken.
Why are there different sampling frequencies, and why are they always funny numbers. That's because there's a certain law, from Shannon-Nyquist. That law says that the sampling frequency must be at least double the value of the highest frequency that's present in the analog signal you are sampling. If this requirement is not met, you'll get aliasing. I'm not really going to explain that further, just know, aliasing isn't good. You'll hear artifacts.

The most used frequency nowadays (cd for example) is 44100. It is said that the normal bandwith of an analog audio signal is 20 Hz to 20 KHz. So the highest possible frequency should be 20 KHz. When we apply Nyquist, the sampling frequency should be at least 40 KHz, but they left a margin (because at the end there's a low pass filter against aliasing, and it hasn't an infinitely steep cut), so they set it to 44.1 KHz.

Ok, but why are there lower or higher sampling frequencies also? Well, lower that's if you need more space, or if a high frequency isn't required.
But why higher? It takes up more space (more samples per second). Again this is because of that aliasing problem. I told you something about a low pass filter didn't I? Well there's a problem with analog filters. If they have a steep cutting curve, there will be problems concerning phase. Phase problems are bad (you lose some frequencies and such). When using higher samplerates the filter has to be less steep. Less steep, less phase problems. Easy as that. On some occasions (cdplayers for examples) they even use oversampling. Sampling at very high rates, so they can add digital filters (using delays off all sorts), which gives an even better result).

Now, you must know most consumer cards (Sounblaster and such) can record up to 48000 Hz. 48 kHz was (and still is in most studio's) the standard sampling frequency (it's shifting to 96 kHz now). But for home recordings, 44.1 kHz is the best way to go, as most of the time it's for burning on cd. And cd only accepts 44.1 KHz 16 bits. Most people aren't familiar with downsampling and dithering on professional level, so stick with the standard rates. Besides, higher frequency means more hard disk space...

Phew, that was for sampling frequency, next chapter : bit depth.

Now I hope you understand the frequency part.
What's the bit depth. Well let's simplify by saying it's the amount of values you get to define a sample.

Let's even more simplify (purists won't like this, but what the heck)and say it's the volume. Each sample as a certain value according to it's volume. The more bits you use, the more precise you can pinpoint that value.
Each sample will get a binary code as its value. If you used 16 bits, it means the code will consist of 16 ones and zeroes. There are 65536 (2^16) possible combinations with that code.
If you used 24 bits, the number of combinations would raise to 16777216. Needless to say each sample will be pinpointed way more precisely.
Ok all those numbers look nice, but what does it mean in practical use. Well the bit depth affects the dynamic range. That's the relation between the softest and the loudest part of a signal.
Let's start with 8 bits. This was the bit depth used on the first consumer soundcards. A big simplification says each bit is 6 dB of dynamic range. So 8 bits = 48 dB dynamic range. The human ear has a dynamic range of about 120 dB. So you see, 48 isn't really that great.
16 bits = 96 dB. That's better. That's the actual consumer standard. It's enough for most people. Why? Well take a very loud signal (a starting plane for example). That's about 120-130 dB SPL loud. That's also our pain threshold. You don't want to hear a plane at real volume on your stereo do you?
At 24 bit you get 144 dB dynamic range. That's very good. It's used in studio's, because having a massive dynamic range, makes it easier to tweak the sound. Once it's ready to produce on cd, they'll decrease to 16 bit anyway. (There are even 32 bit cards). Of course these ranges are purely theoretical. Usually a cd doesn't have 96 dB, at best I'd say 90 dB. But it's enough.

But there is another difference. As I said earlier, converting from analog to digital, is going from a continuous signal to a discrete signal. This always implies errors (quantisation errors). And we will hear those errors as noise. The lower the bit depth, the more noise (try recording on 8 bit, and compare it to a 16 bit sample).

So, a high bit depth will make the sound punchier, but again, you need more space, and most consumer cards work in 16 bit (even the Audigy. It's acclaimed 24/96 are pure bollocks. It works on these setting internally, but it can't record that way).
Again I'd say, if you just want to make a mix on cd, stick to the cd standard 16/44.1 . If you are planning to do a big commercial release (or are an audiophile) 24 bit recordings are really a must. I can't really tell you go for 96 kHz, becuase nobody agrees on that. It's very easy to tell a 16 bit recording from a 24 bit one, but pretty difficult to tell the difference between 48k and 96k unless you have trained ears. So my bet is stay 48 for now if you want to make pro recordings. The only way I see a big advantage in 96k right now is when you use music making programs. On the same buffer rate, a higher sampling frequency has a lower latency...

http://www.hammersound.net/audiobasics/audiobasics.html
A very good page if you want more details.

Hope this helped.

Good answer!

My fingers still hurt

Ya thanks a lot for the help. That was the perfect answer to my question