Equalising an audio signal to represent true audio

Hi,

I am working on a project with the aim of recording audio (acoustic instruments) and then re-creating them synthetically using code. At the moment I am trying to analyse the sound files as best I can, and I would like to compensate for the frequency response of the microphone to improve the files.

I have attached necessary m-files and will now explain exactly how it works

1. Choose whether to record, save and analyse a new sound file; analyse an existing file; or record a swept-sine signal

(r) - Recording and saving is pretty self explanatory.

(s) - The swept sine signal is a generated signal that goes from 20Hz to 20000Hz in t seconds. This is recorded back into the computer and an analysis of the frequency amplitude shows how responsive the microphone & speakers are at each frequency.

(a) - The analysis section uses fft and then displays 1. The waveform, 2. The frequency spectrum over time, 3. The frequency response curve and 4. The probability distribution.

2. Choose any required parameters.

Now what I'm attempting is to create a frequency response using a swept-sine signal (giving me the best possible frequency response of the microphone). Then I want to inversely apply this reading, like an equaliser, to any new recorded signals so that the data is as true to real life as possible. I'm having a lot of difficulty doing this, and could do with some pointers.

Thanks for your time

EDIT: Wrong sound_analysis.m file uploaded this has now been fixed

[Michael]

I know this is an old thread, but I'm also trying to apply a frequency response curve to audio data to recreate the original sound. I'm not sure how to go about this, and am just wondering if you ever worked it out. Thanks for your help.