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UIU - DSP - EEE - 312 - Lab Report 03

United International University Course No. EEE 312 Digital Signal Processing Lab Section A Lab Report 03 was submitted by Thajid Ibna Rouf Uday and MD. Ashif Jardary to their professor Haider Adnan Khan. The report contains answers to three questions: 1) definitions of band limited signals and the sampling theorem, 2) a MATLAB code demonstrating sampling of a signal at different sampling rates, and 3) another MATLAB code showing the effect of different sampling frequencies on a signal.
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0% found this document useful (0 votes)
100 views3 pages

UIU - DSP - EEE - 312 - Lab Report 03

United International University Course No. EEE 312 Digital Signal Processing Lab Section A Lab Report 03 was submitted by Thajid Ibna Rouf Uday and MD. Ashif Jardary to their professor Haider Adnan Khan. The report contains answers to three questions: 1) definitions of band limited signals and the sampling theorem, 2) a MATLAB code demonstrating sampling of a signal at different sampling rates, and 3) another MATLAB code showing the effect of different sampling frequencies on a signal.
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© © All Rights Reserved
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United International University

QUEST FOR EXCELLENCE


Course No. EEE 312
Course Title: Digital Signal Processing Lab

Section: A

Lab Report 03
Submitted To:

Haider Adnan Khan


Dept of EEE
United International University

Submitted By

Thajid Ibna Rouf Uday


ID: 021 121 102
MD. Ashif Jardary
ID: 021 122 039

Date of Submission: March 08, 2015

Answer no: 01
a) Band Limited Signal: A signal is said to be band limited if its spectrum is zero outside the
frequency range F> B, where B is the maximum frequency of a signal.
For example, a continuous-time finite-energy signal x(t) is band limited if its Fourier transform
X(F) = 0 for F > B.
b) Sampling Theorem: If the highest frequency contained in an analog signal xa(t) is Fmax = B
and the signal is sampled at a rate Fs> 2Fmax= 2B then xa(t) can be exactly recovered from its
sample value using the interpolation function.
c) Reconstruction using sinc interpolation function:
The frequency response of an L.P.F.(low pass filter) is rect function. Thus the output of the
L.P.F. is a sinc function. i.e. when a signal is passed through L.P.F. its output is a sum of sinc
function. Now if the frequency domain signal is rect function then its time domain signal will be
sinc function. Thus when we pass a sampled signal through a L.P.F. the output is a combination
of the sinc function which in terms generate the original signal.
Sinc(Z); Which

Figure : Band limited reconstruction of the signal The dots show the signal samples, the dashed
lines show the component sinc functions, and the solid line shows the unique bandlimited
reconstruction from the samples obtained by summing the component sinc functions.

Answer no: 02
%% 2 a
clear all
clc
t=0:0.0001:0.5;
xa=sin(1000*pi*t);
Ts=0.0001;
N1=round(1/Ts);
n1=0:N1/2;
x1=sin(1000*pi*n1*Ts);
subplot(3,1,1)
plot(t,xa,n1*Ts,x1,'bo');
axis([0,.05,-1,1]);
ylabel('x1(n)');
title('sampling of x_{a}(t)using Ts=0.1ms')
%%2 b
Ts=0.001;
N2=round(1/Ts);
n2=0:N2/2;
x2=sin(1000*pi*n2*Ts);
subplot(3,1,2)
plot(t,xa,n2*Ts,x2,'ro');
axis([0,.05,-1,1]);
ylabel('x2(n)');

title('sampling of x_{a}(t)using Ts=1ms')


%% 2 c
Ts=0.00001;
N3=round(1/Ts);
n3=0:N3/2;
x3=sin(1000*pi*n3*Ts);
subplot(3,1,3)
plot(t,xa,n3*Ts,x3,'ko');
axis([0,.05,-1,1]);
ylabel('x3(n)');
title('sampling of x_{a}(t)using
Ts=0.01ms')

Answer no: 03
clear clc
clc
t=-1:0.0001:1;
x=exp(-1000*abs(t));
Fs = 500;
Ts=1/Fs;
N1=round(1/Ts);
n1=-N1:N1;
x1=exp(-1000*abs(n1*Ts));
subplot(2,1,1)
plot(t,x,n1*Ts,x1,'ro');
axis([-.009,.009,-1,1]);
title('sampling frequency Fs=5000
sample/sec')
ylabel('x(n)');
%%
Fs = 1000;
Ts=1/Fs;
N2=round(1/Ts);
n2=-N2:N2;
x2=exp(-1000*abs(n2*Ts));

subplot(2,1,2);
plot(t,x,n2*Ts,x2,'ro');
axis([-.009,.009,-1,1]);
title('sampling frequency Fs=1000
sample/sec')
ylabel('x(n)');

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