I've been told this is how I should be preprocessing audio samples, but what information does this method actually give me? What are the alternatives and why shouldn't I use them.
Fourier transform is used to transform audio data to get more information (features).
For example, raw audio data usually represented by a one-dimensional array,
x[n], which has a length
n (number of samples).
x[i] is an amplitude value of the i-th sample point.
Using the Fourier transform, your audio data will be represented as a two-dimensional array. Now,
x[i] is a not a single value of amplitude, but a list of frequencies which compose original value at the i-th frame (a frame consists of a few samples).
See the image below (from wikipedia), the red graph is an original value of n samples before transformed, and the blue graph is a transformed value of one frame.