Analog Audio Signal to Decibels with an A weighting filter.

I have a microphone attached to a pre amp. The sensitivity is about 15mv/pa.
In labview I have written the A weighting filter myself. I want to know what the procedure is

for audio processing in labview. Should I convert the signal to Pascal via the sensitivity of the mic and then convert to decibels (20log(signal/20ref)) and

then take the FFT to apply my weighting filter? Do I need to convert the signal to a powerspectrum? Sort of confused on the process.