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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.

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Hi ykhalil,


Here's a pretty detailed explanation of signal processing in LabVIEW:


Also, are you using the NI Sound and Vibration Toolkit?




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