Hi, I\'m having trouble understanding this question from my Signals and Systems

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Hi, I'm having trouble understanding this question from my Signals and Systems II class. The question is from a lesson regarding Sampling and Reconstruction. Here is the question:

Here is section 8 from the lab:

I understand you don't have access to the tada.wmv file. I ran the code using the sugested coordinates of [1 100 100 1] mentioned in the lab. This is the plot I got back:

Please explain this problem to me and provide suggested coordinates to use. How do I explain my answer using the resulting plots?

In the MATLAB code provided in Section 8 of the lab manual, a frequency equalizer is implemented. Copy the MATLAB code and run it in a script. a) Try three different sets of values for the equalization coefficients and include the frequency response plots generated. b) Does the frequency response correspond to what you entered as coefficients in all three cases? Explain by using the plots from part a)

Explanation / Answer

clear all; % Clears all existing memory and starts allocating memory from first
close all; % If any other MATLAB window such as graphs, plots, images are open, the following command terminates the windows
clc; %Clears the command window
[y,f]=wavread('signal.wav'); % Audio read is an in built function which takes audio signal as input. The following comman fetches the discrete values of signal in y and the sampled frequency in f. y has two columns if it is stereo type and has only one coloumn is it is mono.
y=y(:,1);%In either case of mono or stereo waves one column is enough for processing. We take the first column for processing.
ly=length(y);% To determine the length of the wave signal
ty=([1:ly]'-1)/f;% Creates a vector of dimension 1xly and divides all value with f. [1:ly] to create a row vector values from 1 to ly and ('-1) is to transpose to a column vector. This vector is the time domain.
w=([1:ly]'-1)/f;w=w(1:ly/2);% Frequency domain
fy=abs(fft(y));fy=fy(1:ly/2)/f;%fft generates the fourier tranform of the signal given in input

% Once you get familiar with the above block of code, you can try to
% analyze the second set of codes. They has just created each signal using
% the time domain and frequecy domain array produced in top code.

% hs1 hs2 hs3 and HS1, HS2 and HS3 are three different signals which can be viewed using the following commands
%figure
%plot(hs1)
%figure
%plot(hs2)
%figure
%plot(hs3)
%figure
%plot(HS1)
%figure
%plot(HS2)
%figure
%plot(HS3)

%conv function convolutes the given two input signals
%View the convoluted signals use the following
%figure
%plot(z1)
%figure
%plot(z2)
%figure
%plot(z3)

%coeff=[1 1 1 1] This is were you give inputs for your filter. [low,
%mid-low, mid-high, high]

%loglog(w,fy,w,fx,'--'); % The function used to plot signal. Similar to plot() function. The one which you see as output
%soundsc(y,f); plays original sound
%soundsc(ys,f); play equalized sound

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