How to sample audio at Nyquist frequency with MSP430F5438?
You can do your entire project in a desktop PC.  In fact, if I had to do it, I would start with the desktop:


a .wav file is already sampled at a high frequency, often 44100 or 48000 Hz.
determining the highest frequency can be done with an FFT.  For prototyping, I would link FFTW. 
downsampling to an arbitrary frequency is a bit hard, because downsampling involves low-pass filtering.  You need to set up a filter for each frequency.  Look at libsamplerate and see how it sets up a SINC function to convolve against.
Converting back to the original sample rate will involve another low-pass filter.  See again libsamplerate.


I believe I would implement this in several passes, for ease of debugging:


Get everything working in Matlab or Octave first.  Octave has libraries to do all the filtering and Fourier analysis.
Get everything working in C on PC, linking FFTW and libsamplerate for the downsampling / upsampling.
Rewrite the C code with explicit-width variable types (e.g., int16_t instead of "short") and replace FFTW and libsamplerate with own code so that it compiles standalone.
In C for the MSP430 or whatever DSP you've got, write interrupt routines to sample data on the ADC and output it on the DAC.  Test that this works, just going from input to output.
Take the working code from step 3 and compile it for the MSP430 or whatever.  Then wedge it in to the working code from step 4 to operate on the sampled data between ADC and DAC.


This may seem like a lot of steps, but it is much more likely to make a working result than heroically coding everything up in one huge MSP430 application, then trying to debug it on the dev board.