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How would i down-sample a .wav file then reconstruct it using nyquist? - in matlab

This is all done in MatLab 2010

My objective is to show the results of: undersampling, nyquist rate/ oversampling

First i need to downsample the .wav file to get an incomplete/ or impartial data stream that i can then reconstuct.

Heres the flow chart of what im going to be doing So the flow is analog signal -> sampling analog filter -> ADC -> resample down -> resample up -> DAC -> reconstruction analog filter

what needs to be achieved:

F= Frequency

F(Hz=1/s) E.x. 100Hz = 1000 (Cyc/sec) F(s)= 1/(2f)

Example problem: 1000 hz = Highest frequency 1/2(1000hz) = 1/2000 = 5x10(-3) sec/cyc or a sampling rate of 5ms

This is my first signal processing project using matlab.

what i have so far.

% Fs = frequency sampled (44100hz or the sampling frequency of a cd)

[test,fs]=wavread('test.wav'); % loads the .wav file
left=test(:,1);

% Plot of the .wav signal time vs. strength

time=(1/44100)*length(left);
t=linspace(0,time,length(left));
plot(t,left)
xlabel('time (sec)');
ylabel('relative signal strength')

**%this is were i would need to sample it at the different frequecys (both above and below and at) nyquist frequency.*I think.***

soundsc(left,fs) % shows the resaultant audio file , which is the same as original ( only at or above nyquist frequency however) 

Can anyone tell me how to make it better, and how to do the various sampling at different frequencies?

This is all done in MatLab 2010

My objective is to show the results of: undersampling, nyquist rate/ oversampling

First i need to downsample the .wav file to get an incomplete/ or impartial data stream that i can then reconstuct.

Heres the flow chart of what im going to be doing So the flow is analog signal -> sampling analog filter -> ADC -> resample down -> resample up -> DAC -> reconstruction analog filter

what needs to be achieved:

F= Frequency

F(Hz=1/s) E.x. 100Hz = 1000 (Cyc/sec) F(s)= 1/(2f)

Example problem: 1000 hz = Highest frequency 1/2(1000hz) = 1/2000 = 5x10(-3) sec/cyc or a sampling rate of 5ms

This is my first signal processing project using matlab.

what i have so far.

% Fs = frequency sampled (44100hz or the sampling frequency of a cd)

[test,fs]=wavread('test.wav'); % loads the .wav file
left=test(:,1);

% Plot of the .wav signal time vs. strength

time=(1/44100)*length(left);
t=linspace(0,time,length(left));
plot(t,left)
xlabel('time (sec)');
ylabel('relative signal strength')

**%this is were i would need to sample it at the different frequecys (both above and below and at) nyquist frequency.*I think.***

soundsc(left,fs) % shows the resaultant audio file , which is the same as original ( only at or above nyquist frequency however) 

Can anyone tell me how to make it better, and how to do the various sampling at different frequencies?