Digital Signal Processing LAB

Lab Report
By

Nimra Noor (F18604006)

National University of Technology

Computer Engineering Department

Digital Signal Processing LAB

Submitted
To
Lec. Rafi

Semester: 6th
Session: Spring 2021
Credit Hours: 1
Time: 3 Hours

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Experiment List

Sr.No. Experiments

1 Plotting sinusoids using MATLAB

2 Introduction to Complex Exponentials

3 Spectrum Analysis
4 Sampling and Aliasing
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Experiment 4
Sampling and Aliasing using MATLAB

Objective
To investigate the sampling of continuous-time signals and the effects of aliasing. This lab covers
the implementation of sampling of a continuous time signal and to analyze the aliasing effect that
occurs due to improper sampling.
Theoretical Explanation

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Explanation of Procedure
A. Tools/Commands
1. MATLAB
B. Codes
1. Lab task 1

%TASK # 4.1
t=-0.5:0.0005:0.5;
fo1=3;fo2=7;fo3=13;
x1t=cos(2*pi*fo1*t);
x2t=cos(2*pi*fo2*t);
x3t=cos(2*pi*fo3*t);

fsa=10;fsb=13;fsc=26;fsd=52;fse=80;fsf=130;
%for x1
figure(1);
% For fsa
n=0:(0.5*fsa)-1;
xs1=cos(2*pi*fo1*n/fsa);
subplot(231)
stem(n,xs1)
% For fsb
n=0:(0.5*fsb)-1;
xs1=cos(2*pi*fo1*n/fsb);
subplot(232)
stem(n,xs1)
% For fsc
n=0:(0.5*fsc)-1;
xs1=cos(2*pi*fo1*n/fsc);
subplot(233)
stem(n,xs1)
% For fsd
n=0:(0.5*fsd)-1;
xs1=cos(2*pi*fo1*n/fsd);
subplot(234)
stem(n,xs1)
% For fse

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n=0:(0.5*fse)-1;
xs1=cos(2*pi*fo1*n/fse);
subplot(235)
stem(n,xs1)
% For fsf
n=0:(0.5*fsf)-1;
xs1=cos(2*pi*fo1*n/fsf);
subplot(236)
stem(n,xs1)
%for x2
figure(2);
% For fsa
n=0:(0.5*fsa)-1;
xs2=cos(2*pi*fo2*n/fsa);
subplot(231)
stem(n,xs2)
% For fsb
n=0:(0.5*fsb)-1;
xs2=cos(2*pi*fo2*n/fsb);
subplot(232)
stem(n,xs2)
% For fsc
n=0:(0.5*fsc)-1;
xs2=cos(2*pi*fo2*n/fsc);
subplot(233)
stem(n,xs2)
% For fsd
n=0:(0.5*fsd)-1;
xs2=cos(2*pi*fo2*n/fsd);
subplot(234)
stem(n,xs2)
% For fse
n=0:(0.5*fse)-1;
xs2=cos(2*pi*fo2*n/fse);
subplot(235)
stem(n,xs2)
% For fsf
n=0:(0.5*fsf)-1;
xs2=cos(2*pi*fo2*n/fsf);
subplot(236)
stem(n,xs2)

%for x3
figure(3);
% For fsa
n=0:(0.5*fsa)-1;
xs3=cos(2*pi*fo3*n/fsa);
subplot(231)
stem(n,xs3)
% For fsb
n=0:(0.5*fsb)-1;
xs3=cos(2*pi*fo3*n/fsb);

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subplot(232)
stem(n,xs3)
% For fsc
n=0:(0.5*fsc)-1;
xs3=cos(2*pi*fo3*n/fsc);
subplot(233)
stem(n,xs3)
% For fsd
n=0:(0.5*fsd)-1;
xs3=cos(2*pi*fo3*n/fsd);
subplot(234)
stem(n,xs3)
% For fse
n=0:(0.5*fse)-1;
xs3=cos(2*pi*fo3*n/fse);
subplot(235)
stem(n,xs3)
% For fsf
n=0:(0.5*fsf)-1;
xs3=cos(2*pi*fo3*n/fsf);
subplot(236)
stem(n,xs3)

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2. Lab task 2

For x1: f1= 3Hz but and for Fs just for 10,13,26Hz (it has limit 30 we can’t exceed it ),so I also did for 30Hz.

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For x1: f1= 7Hz

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For x1: f1= 13Hz

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3. Lab task 3

 p and q both have same values and my audio was at normal speed

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 When my p=3 and q= 1 my audio becomes fast with respect to originals audio

 When q=3 and p=1 my audio becomes slow that a 6 sec audio were playing in 15 sec

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Conclusion
The sample rate is how many samples, or measurements, of the sound are taken each second. The
more samples that are taken, the more detail about where the waves rise and fall is recorded and the
higher the quality of the audio. Also, the shape of the sound wave is captured more accurately.
When the sampling rate (Fs) is increased, the distance between the maximum frequency content
Fm and Fs/2 will increase. This increase in the gap between the maximum frequency content of the
signal and Fs/2 will ease requirements on the transition bands of the anti-aliasing analog filter.

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