Edited for clarity.
I have a set of speech signals that were peak normalized. I want to adjust the gain of each signal so they have an equal loudness (same idea as replay gain).
I am calling the function :
The signal I'm passing to the function is the original recording.
First I wanted to use the ISO 532-2 method, since the sound presentation will be done with headphones, but my signal is time-varying (speech signal), so I must use the ISO 532-1 method instead. How would I compensate for the non-linearity of the headphones and the fact that the sound will not be presented in a free field? I thought of just building a filter to pre-process the signal before feeding it to acousticLoudness() but I'm not sure where to start.
The headphones used are Sennheiser HDA300 (spec sheet).
One I have a "replay gain" value for each signal based on an arbitrary reference, I will calibrate the amplifier by measuring the dBSPL output of a 1kHz signal.