# Why do we need to normalize a WAV file before calculating the FFT?

I am trying to calculate the FFT of a signal stored in a WAV file using SciPy. I found a solution here, but it seems like we need to perform this step before the FFT:

b=[(ele/2**8.)*2-1 for ele in a] # this is 8-bit track, b is now normalized on [-1,1)


where the signal samples are stored in array a. Can someone explain the significance of this step?

Thanks.