Audio equalisation filter based on recorded microphone sinesweep

Hi MATLABers,

I am trying to normalise the response of a set of microphones. I am trying to develop individual microphone audio filters based on a recorded sinesweeps from the microphones. I don't even know if this will work. I am not and expert in this area but I have tried implementing a frequency based normalisation filter with not much luck.

Summary of what i have done

X = fft(sinesweep)

N = mean(abs(X) % This is used as the mean for all microphones

Norm = 1./(X./N)

%This is then multiplied but the windowed recorded signal before reconstruction

Any help would be greatly appreciated, Matt