I am doing some Fourier transforms of audio (.wav) data using the FFT command in MATLAB. The input values are numbers between -1.0 and 1.0
My understanding is that after taking the absolute value (modulus) of the output of the FFT, I should get values that have units of amplitude, but the actual values are on the order of thousands. This doesn't make sense as theoretically I should be able to sum the Fourier components to get the original signal back. I feel like the output should then also be between 0 and 1, so what's up here? My guess is that the FFT algorithm blows it out of proportion but I'm not sure what value to use to scale it back.