Mark's Homework 15: Difference between revisions

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==MATLAB Script==
==MATLAB Script==
The following is currently '''NON-WORKING'''. A working script will follow.
<pre>
<pre>
%Author: Mark Priddy
%Homework #15 MATLAB script
%Description: Demonstrates oversampling and cancelling of the effects of the D/A converter
clear all;
clear all;
close all;
close all;

N = 10; %Number of sample points for our original signal
tmax=2; %Time, in seconds, to extend our original signal. We then sample this signal N times
%Number of sample points for our original signal
N = 20;
oversampling = 4; %Number of oversampling points
%Time, in seconds, to extend our original signal. We then sample this signal N times
T=tmax/N; %Calculate T, the sample period
tmax=2;
%Number of oversampling points
oversampling = 4;

%Calculations
T=tmax/N;
t=0:T:(N-1)*T;
t=0:T:(N-1)*T;

x=sin(2*pi*t)+sin(2*pi*2*t)+sin(2*pi*5*t); %This is our signal to over sample
%This is our signal to over sample
x=sin(2*pi*t)+sin(2*pi*2*t)+sin(2*pi*5*t);

%The following three lines is our oversampling/interpolating filter
N1=(oversampling-1)*N;
N1=(oversampling-1)*N;
X=fft(x);
X=fft(x);
X1 = (N1+N)/N*[X(1:N/2), zeros(1, N1), X(N/2+1:N)]; %This is our interpolating filter
X1 = (N1+N)/N*[X(1:N/2), zeros(1, N1), X(N/2+1:N)];


%This preserves our interpolated signal for graphing later on
x2= ifft(X1);
x2= ifft(X1);


%
%This is to cancel out the effects of the D/A Converter
%This is to cancel out the effects of the D/A Converter
%
sT = T/100; %sT is hwo many times we are going to sample our pulse p(t)
stmax = oversampling*T; %This is how wide our pulse has to be when we oversample
%sT is how many times we are going to sample our pulse p(t)
sT = T*oversampling;
sN = stmax/sT; %This is the total number of sampling points for sampling p(t)
%This is how wide our pulse has to be when we oversample
stmax = oversampling*T;
%This is the total number of sampling points for sampling p(t)
sN = stmax/sT;
t = 0:sT:stmax-sT;
t = 0:sT:stmax-sT;
%This is p(t), the pulse of the D/A converter, sampled at our oversampling rate
p = u(t+1e-9) - u(t-T/(2*oversampling)) + u(t - (oversampling*T-T/(2*oversampling))); %This is p(t) sampled
p = u(t+1e-9) - u(t-T/(2*oversampling)) + u(t - (oversampling*T-T/(2*oversampling)));
%P(f) is calculated:
P = fft(p);
P = fft(p);
P(161) = 1e-19;
iP = 1./P; %Take 1/P(f) so we cancel the effects of the D/A converter (which is P(f))
numpoints = 2*T/sT;
%We can only go out to 1/(2*T) on the 1/P(f). The rest we need to zero out.
iP = [iP(1:numpoints) zeros(1, sN-numpoints*2-1) iP(sN-numpoints:sN)]
iP = iP(1:sN/(N1+N):sN); %Down sample 1/P(f) so we can element multiply with our interpolating filter


%Take 1/P(f) so we cancel the effects of the D/A converter (which is P(f))
%This plots P(f)
iP = 1./P;
f = -1/(2*sT):1/(sN*sT):1/(2*sT)-1/(sN*sT);
Ps = fftshift(P); %Shift P(f)
%Figure 4 plots P(f) shifted
figure(4)
plot(f, abs(Ps));
%Figure 5 plots P(f) shifted and zoomed in the f axis
figure(5)
plot(f(sN/2-numpoints+1:sN/2+numpoints), abs(Ps(sN/2-numpoints+1:sN/2+numpoints)));


X1 = X1 .* iP; %Element multiply our interpolating filter with our D/A converter effect canceller filter
%Element multiply our interpolating filter with our D/A converter effect canceller filter
X1 = X1 .* iP;
x1 = ifft(X1);
x1 = ifft(X1);


%
%Figure 1 plots the original signal and our interpolated signal (with D/A
%Graphs
%converter effects cancelled.)
%
%Figure 1 plots the original signal and our interpolated signal (with D/A converter effects cancelled.)
figure(1)
figure(1)
t1 = 0:N/(N+N1)*T:(N-1/(N1+N))*T;
t1 = 0:N/(N+N1)*T:(N-1/(N1+N))*T;
plot(t1, x1, 'bo');
hold on
t=0:T:(N-1)*T;
t=0:T:(N-1)*T;
plot(t, x, 'ro', t, x, 'r-');
plot(t1, x1, 'bo', t, x, 'r-o', t1, x2, 'g-');
legend('Interpolated signal with D/A effects cancelled', 'Original sampled signal', 'Interpolated oversampled signal')
hold off
xlabel('Time (s)');
ylabel('Amplitude');


%Figure 2 plots just our interpolated signal, but this time with lines
%Figure 2 plots just our interpolated signal, but this time with lines connecting each point.
%connecting each point.
figure(2)
figure(2)
plot(t1, x1, 'b-');
plot(t1, x1, 'b-');
title('Interpolated oversampled signal');
xlabel('Time (s)');
ylabel('Amplitude');


%Figure 3 plots the frequency response of our original sampled signal.
%Figure 3 plots the frequency response of our original sampled signal.
figure(3)
figure(3)
Xs((N+1)/2:N) = X(1:(N+1)/2); %The fftshift didn't work for me
Xs(1:(N+1)/2) = X((N+1)/2:N); %So this manually shifts the FFT
f = -1/(2*T):1/(N*T):1/(2*T)-1/(N*T);
f = -1/(2*T):1/(N*T):1/(2*T)-1/(N*T);
plot(f,abs(Xs))
plot(f,abs(fftshift(X)))
title('Frequency response of the origianl signal');
%figure(4)
xlabel('Frequency (Hz)');
%plot(f, abs(fftshift(X1)))
ylabel('Amplitude');
</pre>


To run this script, you'll also need another script, called "u.m" with the following contained inside. This must be located in the same folder as the main script above.
<pre>
function [out] = u(t)
out=(1+sign(t))./2;
end
</pre>
</pre>

I couldn't find MATLAB's step function, so I created my own.


==Graphs==
==Graphs==

[[Image:OVR20071203Fig1.png|thumb|left|694px| Figure 1. The sampled signal with only 7 sample points.]]

[[Image:OVR20071203Fig2.png|thumb|left|694px| Figure 2. The DFT of Figure 1.]]

[[Image:OVR20071203Fig3.png|thumb|left|894px| Figure 3. The same as Figure 1 but with 600 sample points, and zoomed in.]]


<br clear="all"/>


==Explanations==
==Explanations==

Revision as of 16:59, 3 December 2007

Problem

Make a MATLAB script to do four times oversampling and a filter so as to eliminate as much as possible the effect of the D/A converter that follows the interpolation filter.

MATLAB Script

%Author: Mark Priddy
%Homework #15 MATLAB script
%Description: Demonstrates oversampling and cancelling of the effects of the D/A converter
clear all;
close all;

%Number of sample points for our original signal
N = 20; 
%Time, in seconds, to extend our original signal. We then sample this signal N times
tmax=2; 
%Number of oversampling points
oversampling = 4; 

%Calculations
T=tmax/N; 
t=0:T:(N-1)*T;

%This is our signal to over sample
x=sin(2*pi*t)+sin(2*pi*2*t)+sin(2*pi*5*t);

%The following three lines is our oversampling/interpolating filter
N1=(oversampling-1)*N;
X=fft(x);
X1 = (N1+N)/N*[X(1:N/2), zeros(1, N1), X(N/2+1:N)];

%This preserves our interpolated signal for graphing later on
x2= ifft(X1);

%
%This is to cancel out the effects of the D/A Converter
%
%sT is how many times we are going to sample our pulse p(t)
sT = T*oversampling; 
%This is how wide our pulse has to be when we oversample
stmax = oversampling*T; 
%This is the total number of sampling points for sampling p(t)
sN = stmax/sT; 
t = 0:sT:stmax-sT;
%This is p(t), the pulse of the D/A converter, sampled at our oversampling rate
p = u(t+1e-9) - u(t-T/(2*oversampling)) + u(t - (oversampling*T-T/(2*oversampling)));
%P(f) is calculated:
P = fft(p);

%Take 1/P(f) so we cancel the effects of the D/A converter (which is P(f))
iP = 1./P;

%Element multiply our interpolating filter with our D/A converter effect canceller filter
X1 = X1 .* iP; 
x1 = ifft(X1);

%
%Graphs
%
%Figure 1 plots the original signal and our interpolated signal (with D/A converter effects cancelled.)
figure(1)
t1 = 0:N/(N+N1)*T:(N-1/(N1+N))*T;
t=0:T:(N-1)*T;
plot(t1, x1, 'bo', t, x, 'r-o', t1, x2, 'g-');
legend('Interpolated signal with D/A effects cancelled', 'Original sampled signal', 'Interpolated oversampled signal')
xlabel('Time (s)');
ylabel('Amplitude');

%Figure 2 plots just our interpolated signal, but this time with lines connecting each point.
figure(2)
plot(t1, x1, 'b-');
title('Interpolated oversampled signal');
xlabel('Time (s)');
ylabel('Amplitude');

%Figure 3 plots the frequency response of our original sampled signal.
figure(3)
f = -1/(2*T):1/(N*T):1/(2*T)-1/(N*T);
plot(f,abs(fftshift(X)))
title('Frequency response of the origianl signal');
xlabel('Frequency (Hz)');
ylabel('Amplitude');

To run this script, you'll also need another script, called "u.m" with the following contained inside. This must be located in the same folder as the main script above.

function [out] = u(t)
out=(1+sign(t))./2;
end

I couldn't find MATLAB's step function, so I created my own.

Graphs

Figure 1. The sampled signal with only 7 sample points.
Figure 2. The DFT of Figure 1.
Figure 3. The same as Figure 1 but with 600 sample points, and zoomed in.



Explanations