FFT stands for Fast Fourier Transform. So, it's a fast way of doing Fourier Transform. Fourier Transform is used to transform a (periodic) signal between the time base (which you can see on the normal oscilloscope screen) and the frequency base (a plot where you can see all the containing frequences).
If you put on a 1kHz signal on your channel and let it run a FFT it will show 1 line at 1kHz. It will also show the amplitude, which you can determine by it's height. Note that this is in the decibel scale. I think it's 0dB if your signal originally went top-top in your normal oscilloscope screen. The big advantage of this type of measuring is you can see how much noise a signal has, how your mixers and other signal transforming circuits work.
For example. If you make a FM stereo coder you have 2 audio channels (the audio for the left and the right speaker) and an output (MPX channel). In the past FM has been used to be received by mono radio's. It was just connecting the received audio signal up to a speaker. If you want to send out stereo, you need 2 different channels; L and R. Because you can't just leave away the R (or the L) channel, they made a dual side band at 38kHz with L-R, the mono sound or L+R on the normal band and added a pilote tone of 19kHz (which shouldn't be of that high amplitude). It would look something like this, except the SCA band:
If you would make a mixer that computes the L+R sound, the L-R sound, a 19kHz pilot tone and a DSB modulator (to make the L-R a 38kHz band where the signal folds out from the 38kHz point), you will need such a tool as FFT to verify it's functionality. We had to make this device for a college project, where it was very important to see even the smallest signals which are basically invisible on the normal scope screen. If you see a signal of -50 dB or even less in the FFT screen, that's something of 10000x as small as 0dB. Might not seem important, but it will when you are going to mix signals or amplify them.
edit: I believe it was also a rule that the 38kHz was supposed to be damped, otherwise you would have to put like 25% or even more of the transmitting power into that signal. So , we did find an IC (MC1496) that could suppress the carrier, but it performed disappointing. In these terms I am speaking about a carrier signal of a few millivolts against 2Vpp of the original audio signal. We could see it on FFT screen very clearly, no way we see that on the time base. Still though, we expected the carrier signal to be in the regions of -65dB, because that's what the specified for a much higher frequency.
There are a lot of books about Fourier Analysis but I won't recommended buying those. The reason is because they are all very theoretic and I think you should look for the practical usage of this first. I think you will find it a lot used in telecommunications and signal processing. Most of these books will describe a bit of it and also how it works.
edit:: Oh yeah, I know that DSP book and it's indeed worth a read. It's more focussed on digital signals, but those require the same theory.