I want to:
Input an analog audio track
Using the microcontroller ADC, convert it to a digital output
Then have the microcontollers/boards timer sample the data at selected intervals
Resample the "Sampled audio track" at twice the highest frequency content
Convert it back to analog as a perfect reconstruction of the initial audio track
Using Fourier Analysis I will determine the highest frequency at which I will sample the track at.
It sounds easy enough and straight forward, but what I need is to program this in C and utilize my MSP430 chip/experimenters board to sample the track. I'm going to be using Texas Instruments CCS and Octave for my programming and debugging. I am using the MSP430F5438A Experimenter Board:http://www.kynix.com/Detail/99534/MSP430F5438A.html
Is C the right language for this? Can I get any examples of how to sample the track at Nyquist frequency using C? What code in C will tell the board to utilize the ADC component? And any recommended information that is similar or that will help me on this project.