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.

Hello,

We're sorry that we don't know what you want.

Instead, we let you know useful sites.

We've been using ffmpeg please visit the ffmpeg official site: http://www.ffmpeg.org/download.html
or doom9(https://forum.doom9.org/).

Best regards,
The Bandicut Team.