EE 395 Digital Signal Processing

NED University of Engineering & Technology
Department of Electrical Engineering
LAB MANUAL
For the course
Digital Signal Processing

(EE-395) For T.E.(EE)
Instructor name:__________________________________
Student name:____________________________________
Roll no: Batch:___________________
Semester: Year:________________
To be filled by lab technician
Attendance: Present out of _____ Lab sessions
Attendance Percentage ________
To be filled by Lab Instructor
Lab Score Sheet
Roll No. Rubric Rubric Rubric Rubric Rubric Rubric OEL/PBL Final LAB Attendance Final weighted Score for
based based based based based based Rubric Rubric Percentage MIS System
Lab I Lab II Lab III Lab IV Lab V Lab VI Score Score [10(A)+10(B)+5(C)]/25
A B C Round to next higher
multiple of 5
Rubric based labs for EE-395 DSP: 1, 2, 3, 6, 8, 10
Note: All Rubric Scores must be in the next higher multiple of 5 for correct entry in MIS system.
LAB MANUAL
For the course
Digital Signal Processing

(EE-395) For T.E.(EE)
Content Revision Team:

Mr. Muhammad Omar & Mr. Nabeel Fayyaz
Last Revision Date: 29-12-2020
Approved By
The Board of Studies of Department of Electrical Engineering
____________________ ____________________
____________________ ___________________
____________________ ____________________
CONTENTS
Psychomotor Level: 4
CLO: Experimentally verify the analytical and design techniques developed for discrete time systems and
signals. [PLO (5)]
S.No. Date Title of Experiment Total Marks Signature
To study the effects of Sampling in Discrete Time

1 Signals.
To study the effects of Quantization in Discrete Time

2 Discrete Valued Signals.
To study and verify Discrete-Time convolution and its

3 properties.
4 To study Discrete-Time correlation.
The Discrete Fourier Transform as a Linear

5 Transformation
Studying Discrete Fourier Transform using an audio

6 signal example
7 Relationship between Laplace and CTFT.
8 Relationship between Z transform and DTFT.
9 Designing FIR filters with windowing
10 Designing IIR filters using FDA tool
11 Open Ended Lab1 (due after lab 2)
12 Open Ended Lab2(due after lab 6)

LAB SESSION 01
OBJECTIVE:
To study the relationship between discrete-time and continuous time signals by examining
sampling and aliasing.
THEORY:
Signals are physical quantities that carry information in their patterns of variation. Continuous-
time signals are continuous functions of time, while discrete-time signals are sequences of
numbers. If the values of a sequence are chosen from a finite set of numbers, the sequence is known
as a digital signal. Continuous-time, continuous-amplitude signals are also known as analog
signals.
Analog phenomenon is continuous – like a human speech of speaker, or a continuously rotating

disc attached to the shaft of motor etc. With analog phenomena, there is no clear separation
between one point and the next; in fact, between any two points, an infinite number of other points
exist. Discrete phenomenon, on the other hand, is clearly separated. There's a point (in time or
space), and then there's a neighboring point, and there's nothing between the two.
Signal processing is concerned with the acquisition, representation, manipulation, transformation,

and extraction of information from signals. In analog signal processing these operations are
implemented using analog electronic circuits. Converting the continuous phenomena of images,
sound, and motion into a discrete representation that can be handled by a computer is called analog-
to-digital conversion. Digital signal processing involves the conversion of analog signals into
digital, processing the obtained sequence of finite precision numbers using a digital signal
processor or general purpose computer, and, if necessary, converting the resulting sequence back
into analog form. When stored in a digital computer, the numbers are held in memory locations,
so they would be indexed by memory address. Regardless of the medium (either sound or an
image), analog-to-digital conversion requires the same two steps: Sampling and Quantization.
Sampling: This operation chooses discrete (finite) points at which to measure a continuous
phenomenon (which we will also call a signal). In the case of sound, the sample points are evenly
separated in time. In the case of images, the sample points are evenly separated in space.
Sampling Rate: The number of samples taken per unit time or unit space is called the sampling
rate. The frequency of sampled/discrete phenomenon (signal) can be calculated as
fd = F /Fs (cycles/sec )/(samples/sec) = cycles/ samples
Where, F = Frequency of analog or continuous phenomenon (signal). [Unit: cycles/sec]
Fs = Sampling frequency or sampling rate [Unit: samples/sec]
fd = Frequency of Discrete phenomenon (signal). [Unit: cycles/sample]
Sampling Theorem: A continuous time phenomenon or signal like x(t) can be reconstructed
exactly from its samples x(n) = x(nTs), if the samples are taken at a rate Fs = 1/Ts
that is greater than twice the frequency of the signal being sampled i.e. Fs ≥ 2 ∗ F.
Mathematically,
Aliasing: A common problem that arises when sampling a continuous signal is aliasing, where a
sampled signal has replications of its sinusoidal components which can interfere with other
components. It is an effect that causes two discrete time signals to become indistinct due to
improper sampling (fd>1/2 cycles/sample).
PROCEDURE:
1. Simulate and plot two CT signals of 10 Hz and 110 Hz for 0 < t < 0.2 secs.
2. Sample at Fs = 100 Hz and plot them in discrete form.
3. Observe and note the aliasing effects.
4. Explore and learn.
STEPS:
1. Make a folder at desktop and name it as your current directory within MATLAB.
2. Open M-file editor and type the following code:
clear all;close all;clc;

F1 = 10;
F2 = 110;
Fs = 100;
Ts = 1/Fs;
t = [0 : 0.0005 : 0.2];
x1t = cos(2*pi*F1*t);
x2t = cos(2*pi*F2*t);
figure,
plot(t,x1t,t,x2t, 'LineWidth',2);
xlabel('cont time (sec)');
ylabel('Amp');
xlim([0 0.1]);
grid on;
legend('10Hz','110Hz');
title('Two CTCV sinusoids plotted');
3. Save the file as P011.m in your current directory and ‘run’ it, either using F5 key or writing
the file name at the command window.
(Check for the correctness of the time periods of both sinusoids.)
Now add the following bit of code at the bottom of your P011.m file and save.
nTs = [0 :Ts : 0.2];

n = [1 : length(nTs)-1 ];
x1n = cos(2*pi*F1*nTs);
x2n = cos(2*pi*F2*nTs);
figure,
subplot(2,1,1),
stem(nTs,x1n,'LineWidth',2);
grid on;
xlabel('discrete time (sec)');
ylabel('Amp');
xlim([0 0.1]);
subplot(2,1,2)
stem(nTs,x2n,'LineWidth',2);
grid on;
title('110Hz sampled')
xlabel('discrete time(sec)');
ylabel('Amp');
xlim([0 0.1]);
1. Before hitting the ‘run’, just try to understand what the code is doing and try to link it with
what we have studied in classes regarding concepts of frequency for DT signals.
2. Now ‘run’ the file and observe both plots.
To see what is really happening, type the following code at the bottom of your existing P011.m
file and run again.
figure,
plot(t,x1t,t,x2t);
hold;
stem(nTs,x1n,'r','LineWidth',2);
xlabel('time (sec)');
ylabel('Amp');
xlim([0 0.05]);
legend('10Hz','110Hz');
3. Observe the plots.
RESULT:
Explain (write) in your own words the cause and effects of what you just saw.
LAB TASKS:
1. Consider the following CT signal:

x(t) = sin (2 pi F0 t). The sampled version will be:
x(n) = sin (2 pi F0/Fs n),
where n is a set of integers and sampling interval Ts=1/Fs.
Plot the signal x(n) for n = 0 to 99
for Fs = 5 kHz and F1 = 0.5, 2, 3 and 4.5 kHz. Explain the similarities and differences among
various plots.
2. Generate a tone in MATLAB with varying frequency f = 1000,2000,3000,4000, 5000, 6000,

8000, 9000, 25000,-1000,-2000,-3000 Hz with Fs = 8000 samples/sec.
Listen to the tones,and observe at Sounds like what frequency?
Also Specify whether Aliasing is happening or not.
3. Record a sentence in your voice.(you may use Simulink /audacity to record).Change Fs

=44100, 22050, 11025, 8192, 4096 , 2048 , 1024
and observe
a) Voice quality during playback [Excellent/Good/OK/Bad]
b) File size in kilobytes
c) Aliasing happening or not?
Course Code: EE-394 Course Title: Digital Signal Processing

Laboratory Session No.: ________________________ Date: ______________________________
Psychomotor Domain Assessment Rubric for Laboratory (Level P3)
Extent of Achievement
Skill(s) to be assessed
0 1 2 3 4
Software Menu Unable to Little ability and Moderate ability Reasonable Demonstrates
Identification and understand understanding of and understanding of command over
Usage: and use software menu understanding of software menu software menu
Ability to initialise, software operation, makes software menu operation, makes no usage with frequent
configure and operate menu many mistake operation, makes major mistakes use of advance
software environment lesser mistakes menu options
under supervision, using
menus, shortcuts,
instructions etc.
10% 0 10 20 30 40
Procedural Little to no Slight ability to Mostly correct Correctly recognises Correctly recognises
Programming of given understanding use procedural recognition and and uses procedural and uses procedural
Signal Processing of procedural programming application of programming programming
Scheme: programming techniques for procedural techniques with no techniques with no
Practice procedural techniques coding given programming errors but unable to errors and runs
programming algorithm techniques but run processing processing
techniques, in order to makes crucial scheme successfully successfully
code specific signal errors for the
processing schemes given processing
scheme
15% 0 15 30 45 60
Relating Theoretical Completely Able to recognise Able to recognise Able to recognise Able to recognise
Concepts, Equations unable to some relation relation between relation between relation between
and Transforms to relate between signal signal processing signal processing signal processing
Code: between processing concepts and concepts and concepts and
Recognise relation signal concepts and written code, written code, able to written code, able to
between signal processing written code, unable to do do some completely
processing concepts concepts and unable to do manipulations manipulations manipulate code in
and written code and written code, manipulations line with theoretical
manipulate the code in unable to do concepts
accordance of manipulations
requirements
15% 0 15 30 45 60
Detecting and Unable to Able to find error Able to find error Able to find error Able to find error
Removing Errors: check and messages and messages and messages in messages in
Detect detect error indications in indications in software as well as software along with
Errors/Exceptions and messages and software but no software as well understanding of the understanding
in simulation and indications in understanding of as understanding detecting all of to detect and rectify
manipulate code to software detecting those of detecting some those errors and them
rectify the simulation errors and their of those errors their types
types and their types
15% 0 15 30 45 60
Page 1 of 2
0 1 2 3 4
Graphical Visualisation Unable to Ability to Ability to Ability to Ability to
and Comparison of understand understand and understand and understand and understand and
Signal Processing and utilise utilise utilise utilise visualisation utilise visualisation
Scheme Parameters: visualisation or visualisation and visualisation and and plotting and plotting
Manipulate given plotting plotting features plotting features features features
simulation under features with frequent successfully but successfully, successfully, also
supervision, in order to errors unable to partially able to able to compare and
produce graphs/plots compare and compare and analyse them
for measuring and analyse them analyse them
comparing signal
processing parameters
15% 0 15 30 45 60
Following step-by-step Inability to Able to recognise Able to recognise Able to recognise Able to recognise
procedure to complete recognise and given lab given lab given lab procedures given lab procedures
lab work: perform given procedures and procedures and and perform them and perform them
Observe, imitate and lab procedures perform them perform them by by following by following
operate software to but could not following prescribed order of prescribed order of
complete the provided follow the prescribed order steps, with steps, with no
sequence of steps prescribed order of steps, with occasional mistakes mistakes
of steps frequent mistakes
10% 0 10 20 30 40
Recording Simulation Inability to Able to recognise Able to recognise Able to recognise
Observations: recognise prescribed or prescribed or prescribed or
Observe and copy prescribed or required required simulation required simulation
prescribed or required required simulation measurements but measurements and
simulation results in simulation measurements records them records them
__
accordance with lab measurements but does not incompletely completely, in
manual instructions record according tabular form
to given
instructions
10% 0 10 30 40
Discussion and Complete Slight ability to Moderate ability Reasonable ability Full ability to discuss
Conclusion: inability to discuss recorded to discuss to discuss recorded recorded
Demonstrate discussion discuss observations and recorded observations and observations and
capacity on the recorded draw conclusions observations and draw conclusions draw conclusions
recorded observations observations draw conclusions
and draw conclusions and draw
from it, relating them to conclusions
theoretical
principles/concepts
10% 0 10 20 30 40
Total Points (out of 400)
Weighted CLO (Psychomotor Score) (Points/4)
Remarks
Instructor’s Signature with Date
Page 2 of 2
LAB SESSION 02
OBJECTIVE:
To observe the quantization effects on sampled signals and to understand how quantization leads
to quantization error. In this lab, we will investigate the influence of the number of quantization
levels on the quality of digitized signal. Method of selection of ADC is also a part of this lab session.
THEORY:
Everything stored on a computer is discrete time discrete valued signal. Because computer has finite
number of registers and each register is a finite length register. We take too many samples to give
the ‘effect’ of continuous time signals. But actually they are discrete time. We also take very fine
resolution of amplitude axis to give the effect of continuous valued signal but due to finite word
length of the computer register, the stored variables are already quantized. This lab aims to explain
the quantization effects in a computer.
Regardless of the medium (audio or image), the digitization of real world analog signal usually
involves two stages: sampling, i.e. the measurement of signal at discretely spaced time
intervals, and quantization, i.e. the transformation of the measurements (amplitudes) into
finite-precision numbers (allowed discrete levels), such that they can be rep resented in
computer memory. Quantization is a matter of representing the amplitude of individual
samples as integers expressed in binary. The fact that integers are used forces the samples to
be measured in a finite number of bits (discrete levels). The range of the integers possible is
determined by the bit depth, the number of bits used per sample. The bit depth limits the
precision with which each sample can be represented.
Bit Depth:
Within digital hardware, numbers are represented by binary digits known as bits—in fact, the
term bit originated from the words Binary digit. A single bit can be in only one of two possible
states: either a one or a zero. When samples are taken, the amplitude at that moment in time must
be converted to integers in binary representation. The number of bits used to represent each
sample, called the bit depth (bits/sample) or sample size, determines the precision with which
the sample amplitudes can be represented. Each bit in a binary number holds either a 1 or a 0. In
digital sound, bit depth affects how much you have to round off the amplitude of the wave when
it is sampled at various points in time
The number of different values that can be represented with b-bit is 2b .The largest decimal
number that can be represented with an b-bit binary number is 2b - 1. For example, the decimal
values that can be represented with an 8-bit binary number range from 0 to 255, so there are
256 different values (levels of ADC). A bit depth of 8 allows 2 8=256 different discrete levels
at which samples can be approximated or recorded. Eight bits together constitute one byte. A
bit depth of 16 allows 2 16 = 65,536 discrete levels, which in turn provides much higher
precision than a bit depth of 8.
The number of bits in a data word is a key consideration. The more bits used in the word, the better
the resolution of the number, and the larger the maximum value that can be represented. Some
computers use 64-bit words. Now, 264 is approximately equal to 1.8 x 1019—that's a pretty large
number. So large, in fact, that if we started incrementing a 64-bit counter once per second at the
beginning of the universe (≈20 billion years ago), the most significant four bits of this counter
would still be all zeros today.
To simplify the explanation, take an example of ADC with a bit depth of 3, 2 3 = 8
quantization levels ranging from -4 to 3 are possible in signed magnitude representation. For
bipolar ADCs (or signed magnitude representation), by convention, half of the quantization
levels are below the horizontal axis (that is 21, of the quantization levels). One level is the
horizontal axis itself (level 0), and 2b-1 − 1levels are above the horizontal axis.Note that since
one bit is used for the signed bit (in 2-complementformat), the largest magnitude corresponds
to 2^(b -1 ). (not 2b). When a sound is sampled, each sample must be scaled to one of the 8
discrete levels. However, the samples in reality might not fall neatly onto these levels. They
have to be rounded up or down by some consistent convention.
QUANTIZATION ERROR:
The samples, which are taken at evenly-spaced points in time, can take on the values only
at the discrete quantization levels to store on our computer. Therefore quantization leads
to a loss in the signal quality, because it introduces a “Quantization error”. Quantization
error is sometimes referred to as '"Quantization noise". Noise can be broadly defined as
part of an audio signal that isn’t supposed to be there. However, some sources would argue
that a better term for quantization error is "distortion", defining distortion as an unwanted
part of an audio signal that is related to the true signal.
The difference between the quantized samples and the original samples constitutes
quantization error or rounding error (if round-off method is used). Xe(n) = Xq(n) − x(n).
The lower the bit depth, the more values potentially must be approximated (rounded),
resulting in greater quantization error
To calculate the required bit depth of ADC i.e. bits/sample, there are two important
points which we must have to consider:
a) How much noise is already present in the analog signal?
b) How much more noise can be tolerated in the digital
signal? Signal-to -noise-ratio- SNR (of analog signal)
Before looking at SNR specifically in the context of digital imaging and sound, let's
consider the general definition. Signal-to-noise ratio can generally be defined as the ratio
of the meaningful content of a signal versus the associated background noise.
SNR = 10log10 (Px /Pe )

Where, Pxand Pe are average power of the analog signal and noise signal respectively.
A signal is any type of communication – something a person says to you, a digital
signal sending an image file across a network, a message posted on an electronic
bulletin board, a piece of audio being played from a cassette, etc. The noise is the
part of the message that is not meaningful; in fact, it gets in the way of the message
intended in the communication. You could use the term signal-to-noise ratio to
describe communications in your everyday life. If you know someone who talks a
lot but doesn't really convey a lot of meaning, you could say that he or she has a
low signal-to-noise ratio. Web-based bulletin board and chat groups are sometimes
described as having a low SNR – there may be quite a few postings, but very much
meaningful content. In these first two examples, the noise consists of all the empty
"filler" words. In the case of a digital signal sent across a network, the noise is the
electronic degradation of the signal. On a piece of audio played from cassette, the
noise could be caused by damage to the tape or mechanical imperfections in the
cassette player.
In analog data communication (analog signals), the signal-to-noise ratio is defined
as the ratio of the average power in the signal versus the power in the noise level.
In this context, think of a signal being sent over a network connection compared
to the extent to which the signal is corrupted. This is related to the genera l usage
of the term described above. This usage of the term SNR applies to analog signals.
SIGNAL-TO-QUANTIZATION-NOISE-RATIO- SQNR (OF ADC):
Using finite word lengths prevents us from representing values with infinite precision, increases
the background noise in our spectral estimation techniques etc. The amount of error implicit in a
chosen bit depth can be measured in terms of the signal-to-noise ratio (SNR).
For a digitized image or sound, the signal-to-noise ratio is defined as the ratio of the maximum
sample value versus the maximum quantization error. In this usage of the term , the ratio
depends on the bit depth chosen for the signal. Any signal encoded with a given bit depth will
have the same ratio. This can also be called signal-toquantization-noise ratio (SQNR), but you
should be aware that in many sources the term signal-to-noise ratio is used with this meaning
as well. (Henceforth, we'll use the term SQNR to distinguish this measurement from SNR.)
Practical A/D converters are constrained to have binary output words of finite length.
Commercial A/D converters are categorized by their output word lengths, which are normally
in the range from 8 to 16 bits. There is no infinite bit ADC. So whenever we will digitize our
signal, we will always have a quantization error. Quantization error represents the quality of
quantization process but the total error may also turn out to be zero, so signal-toquantization-
noise-ratio (SQNR) is used to describe the quality of quantization process and it can be defined
as
SQNRA/D = 10 log 10(Px /Pe)

Where, Px~and Pe are average power of the DTCV (sampled) signal and quantization error
signal respectively.
PROCEDURE:
1. Simulate a DTCV sinusoid of 1/50 cycles/sample with length of the signal be 500.
2. Choose the no. of significant digits for round-off and apply to the signal generated above.
3. Compute the error signals and SQNR
4. Explore and observe.
STEPS:
2. Open M-file editor and write the following code:
clear all;
close all;
clc;
fd1 = 1/50;
n = [0 : 499 ];
q=input('No. of Digits after decimal points to be retained (0-9): ');

x1 = cos(2*pi*fd1*n);
Px1 = sum(abs(x1).^2)/length(x1);
x1q = round(x1*10^q)/10^q;
x1e = x1 -x1q;
Pe1 = sum(abs(x1e).^2)/length(x1e);
SQNR = 10*log10(Px1/Pe1);
disp(['The Signal to Quantization Noise Ratio is: ' num2str(SQNR) '
dB.' ]);
figure,
subplot(2,1,1);
plot(n,x1,n,x1q);
xlabel('indices');
ylabel('Amp');
xlim([0 49]);
ylim([-1.1 1.1]);
legend('DTCV','DTDV');
subplot(2,1,2);
plot(n,x1e);
xlabel('indices');
ylabel('Error');
xlim([0 49]);
3. Save the file as P021.m in your current directory and run it.
Explore and take notes.

Now modify the above code as follows and save as another file P022.m.
clear all;
close all;
clc;
fd1 = 1/50;
n = [0 : 499 ];
q = [0 : 10];
% No. of Digits after decimal points to be retained for num = 1 :

length(q)
x1 = cos(2*pi*fd1*n);
Px1 = sum(abs(x1).^2)/length(x1);
x1q = round(x1*10^q(num))/10^q(num);
x1e = x1 -x1q;
Pe1 = sum(abs(x1e).^2)/length(x1e);
SQNR(num) = 10*log10(Px1/Pe1);
end
figure,
plot(q,SQNR);
xlabel('Significant Digits');
ylabel('SQNR (dB)');
xlim([q(1) q(end)]);
1. Before hitting the ‘run’, just try to understand what the code is doing and try to link it with
the previous code.
2. Now ‘run’ the file and observe the results.
RESULT:
Explain (write) in your own words the cause and effects of what you just saw.
LAB TASKS:
1. Effects of Quantization with variable precision levels

Simulate a DTCV sampled composite signal of 𝑓𝑑1=125 samples/sec and
𝑓𝑑2=150 samples/sec with length of the signal be 250 samples.
Take the desired number of significant digits from user as an input. Then choose the method of
Quantization (round-off, floor & ceil) and apply to the signal generated above.
Compute the quantization error signals and SQNR.
2. Simple sinusoid quantized to various bits per sample

Generate a 100 Hz sinusoid sampled at 10000 samples/sec and quantized at 1_bit/sample.
Now increase the bit depth for various numbers of bits per sample (2, 3, 4, 5, 6, 7, 8) and attach
plots. You can use two column format for plotting (but the diagrams should be visible).
3. Audio signal quantization to various bits per sample

Use your recorded voice in last session and quantize it at 1 bit /sample.
Change bit depth to 2,3,4 and then listen and take notes of your observations.
Decide no. of bits for audio until quality stops improving.

0 1 2 3 4
menus, shortcuts,
instructions etc.
10% 0 10 20 30 40
scheme
15% 0 15 30 45 60
requirements
15% 0 15 30 45 60
15% 0 15 30 45 60
Page 1 of 2
0 1 2 3 4
comparing signal
15% 0 15 30 45 60
10% 0 10 20 30 40
__
to given
instructions
10% 0 10 30 40
theoretical
principles/concepts
10% 0 10 20 30 40
Remarks
Page 2 of 2
LAB SESSION 03
OBJECTIVE:
To study impulse response, observe convolution technique in signal processing, and verify
different properties like causality, commutative, distributive and associative properties.
THEORY:
1. Convolution is given as :y(n) = x(n)*h(n) =
i.e.one can compute the output y(n) to a certain input x(n) when impulse response h(n) of
that system is known. Convolution holds commutative property.
2. The length of the resulting convolution sequence is N+M-1,where N and M are the
lengths of two convolved signals respectively.
3. In causal system, the outputs only depend on the past and/or present values of inputs and
NOT on future values. This means that the impulse response h(n) of a causal system will
always exist only for n≥ 0.
PROCEDURE:
1. We have the impulse response of a system as h(n) = { 3,2,1,-2,1,0,-4,0,3}

↑
2. For x(n)={1,-2,3,-4,3,2,1}
↑
STEPS:
clear all;
close all;
clc;
h = [3 2 1 -2 1 0 -4 0 3]; % impulse response
org_h = 1; % Sample number where origin exists
nh = [0 : length(h)-1]- org_h + 1;
x = [1 -2 3 -4 3 2 1]; % input sequence
org_x = 1; % Sample number where origin exists
nx = [0 : length(x)-1]- org_x + 1;
y = conv(h,x);
ny = [nh(1)+ nx(1) : nh(end)+nx(end)];

figure,
subplot(3,1,1),
stem(nh,h);
xlabel('Time index n');
ylabel('Amplitude');
xlim([nh(1)-1 nh(end)+1]);
title('Impulse Response h(n)');
grid;
subplot(3,1,2),
stem(nx,x);
xlim([nx(1)-1 nx(end)+1]);
title('Input Signal x(n)');
grid;
subplot(3,1,3)
stem(ny,y);
xlim([ny(1)-1 ny(end)+1]);
title('Output Obtained by Convolution');
grid;
1. Save the file as P031.m in your current directory and ‘run’ it.
2. Calculate the length of input signal (N) and impulse response (M) used in above task?
3. Calculate the length of the output sequence and verify the result with N+M-1
4. Try to learn, explore the code and make notes.
5. Now modify the above code such that h(n)= {3,2, 1, -2,1,0,-4,0,3}(origin is shifted) and
check for causality.
↑
RESULT:
EXERCISE:
1. What will happen if we input x(n)={0,0,1,0,0} into the above system.

↑
2. Can you prove the commutative property of the convolution?
3. Modify the code to prove Associative and Distributed properties of the convolution.
4. Convolve your recorded sound with drumloop.wav. Note your observation
a) Plot the output.

b) Listen the output

0 1 2 3 4
menus, shortcuts,
instructions etc.
10% 0 10 20 30 40
scheme
15% 0 15 30 45 60
requirements
15% 0 15 30 45 60
15% 0 15 30 45 60
Page 1 of 2
0 1 2 3 4
comparing signal
15% 0 15 30 45 60
10% 0 10 20 30 40
__
to given
instructions
10% 0 10 30 40
theoretical
principles/concepts
10% 0 10 20 30 40
Remarks
Page 2 of 2
LAB SESSION 04
OBJECTIVE:
To study discrete time correlation and apply it to real data to observe the correlation
between two signals.
THEORY:
1. Correlation is given as where ‘l’ is the lag. This is called cross-correlation and it
gives the magniyude and location of similarity between two signals. The correlation
between x(n) and y(n) . It is given as:
2. Generally rxy(l) = ryx(l). These two are the same when x(n) and y(n) are the same signals
or when x(n) and y(n) are even symmetric signals .
3. The length of the resulting correlation sequence is N+M-1, where N and M are the
lengths of the two signals.
4. Correlation may also be computed using convolution algorithm with a modification that
we need to fold one of the signals before applying convolution.
Mathematically, rxy(n)= x(n) * y(-n)
STEPS:
1. Generate two sinusoids of length 10 and fd = 0.1 with variable phase.

2. Apply correlation and check for certain properties such as magnitude and location of
maximum correlation with varying phases.
PROCEDURE:
1.Make a folder at desktop and name it as your current directory within MATLAB. -
2.Open M-file editor and write the following code: )
clear all;
close all;
clc;
n = [0:9];
ph1 = 0;
ph2 = 0;
x = sin(2*pi*0.1*n + ph1);
org_x = 1;
nx = [0 : length(x)-1]- org_x + 1;
y = sin(2*pi*0.1*n + ph2);
org_y = 1;
ny = [0 : length(y)-1]- org_y + 1;
rxy = xcorr(x,y);
nr = [nx(1)-ny(end) : nx(end)-ny(1)];
[maxR indR] = max(rxy);
disp(['The correlation at lag zero is: ' num2str(rxy(find(nr==0)))

'.']);
disp(['The maximum correlation is at lag ' num2str(nr(indR)) '.']);
|
figure,
subplot(3,1,1),
stem(nx,x);
xlim([nx(1)-1 nx(end)+1]);
title('Signal x(n)');
grid;
subplot(3,1,2),
stem(ny,y);
xlim([ny(1)-1 ny(end)+1]);
title('Signal y(n)');
grid;
subplot(3,1,3)
stem(nr,rxy);
xlim([nr(1)-1 nr(end)+1]);
title('Cross Correlation');
grid;
Save the file as P041.m in your current directory and ‘run’ it.
Learn the specific logical bits of the code and make notes
Now modify the phase of the second signal to pi/2 (it will make it cosine) and observe the
correlation at lag zero. Modify the phase again to ‘pi’ and observe.
1. Check for auto-correlation (ph1 = ph2) that the lag zero value gives the energy of the
Signal.
2. Observe that the commutative property does not hold.
RESULT:
Please write in exercise book.
EXERCISE:
1. Now modify the phase of the second signal to pi/2 (it will make it cosine)and observe the
correlation at lag zero.
2. Modify the phase again to ‘pi’ and observe.
3. Check for auto-correlation (ph1 = ph2) that the lag zero value gives the m energy of the
signal.
4. Observe that the commutative property does not hold.
5. Modify the code, such that the correlation is obtained using convolution command.
6. Calculate correlation between voltages of any two phases of a 10HP motor Using the data
given below. First use Ms. Excel to copy data and then calculate correlation.
Voltage A Min Voltage B Min Voltage C Min

189.358 153.917 195.735
189.175 159.719 201.877
188.783 161.575 186.718
188.757 172.186 187.659
176.995 173.206 205.876
180.472 176.865 204.831
180.524 176.917 192.494
180.262 189.28 199.839
181.778 189.828 211.887
179.975 189.462 211.94
178.642 189.253 212.462
180.315 188.94 193.749
180.707 190.377 200.492
180.262 190.194 201.433
180.628 190.064 202.635
180.315 189.907 200.701
179.635 189.541 203.289
179.243 189.567 202.635
179.4 189.619 200.989
180.576 189.044 197.591
180.837 189.123 199.865
180.184 189.332 201.093
180.08 189.097 201.041
177.675 189.044 199.656
175.297 189.018 198.558
173.99 189.123 204.595
LAB SESSION 05
OBJECTIVE:
To study the computer implementation of Discrete FourierT transform and Inverse Fourier
Transform using Twiddle factor.
THEORY:
The formulas for the DFT and IDFT are given as

∑𝑁−1 𝑘𝑛
X(K) = 𝑛=0 𝑥(𝑛) 𝑊𝑁 ; k=0,1,……N-1
1 −𝑘𝑛
X(n) = ∑𝑁−1
𝑛=0 𝑋(𝑘) 𝑊𝑁 ; k=0,1,……N-1
𝑁
−𝑗2𝛱
Where by definition WN = 𝑒 𝑁
Which is an Nth root of unity. Where WN is called Twiddle Factor , also
[WN] = [𝑒 −𝑗2𝛱/𝑁 ]kn
𝑘𝑛
W = 𝑒 −𝑗2𝛱𝑘𝑛/𝑁
𝑁
DFT analysis equation in matrix form is
𝑘𝑛
XN=[ W ]xN
𝑁
DFT synthesis equation in matrix form is
𝑘𝑛 -1
xN= [W ] XN
𝑁
PROCEDURE:
TASK
Compute 4 point DFT of x(n)= ( 1,2,3,0).
STEPS
1.Generate given sequence in Matlab .
2.Take N-=4 to calculate 4-point DFT.
3.Define 0: N-1 point vector for time and frequency samples.
4.Define W matrix and then use DFT analysis equation to compute DFT.
close all,
clear all;
clc;
x=[1 ,2 ,3 ,0];
N=4;
n=[0:1:N-1];
k=[0:1:N-1];
WN=exp(-j*2*pi/N);
nk=n'*k;
WNnk=WN.^nk;
Xk=x*WNnk
LAB TASK
Prove DFT synthesis equation using DFT output generated from lab task.
LAB SESSION 06
OBJECTIVE:
To observe/find different frequency components in an audio signal and plot it with different x_
axes .
THEORY:
PROCEDURE:
1. Load an audio file ‘noisy.wav’ into Matlab.

2. There is a tone added to the speech in this file. The objective is to find the frequency of
this tone.
3. Computing the DFT of this signal;
4. Generating frequency vector in Hz.
5. Displaying the DFT and observing the frequency of the added tone.
STEPS
2. Copy the audio file ‘noisy.wav’ into your current directory.
3. Open M file editor and write the following code:
clear all; clc; close all;

[y,Fs,bits] = wavread('noisy.wav');
Ts = 1/Fs;
n = [0:length(y)-1];
t = n.*Ts; k = n;
Df = Fs./length(y);
F = k.*Df;
Y = fft(y);
magY = abs(Y);
sound(y,Fs);
figure,
subplot(2,1,1);
plot(F,magY);
grid on;
xlim([0 Fs/2]);
xlabel('Frequency (Hz)');
ylabel('DFT Magnitude');
title('Discrete Fourier Transform');
subplot(2,1,2);
plot(F,magY);
grid on;
xlim([0 2000]);
xlabel('Frequency (Hz)');
ylabel('DFT Magnitude');
title('Discrete Fourier
Transform');
4. Save the file as P081.m in your current directory and run it.
RESULT:
Explore and take notes.
EXERCISE:
Use recorded data,

1. Plot different frequencies present in it with
a) x-axis as time
b )x-axis as frequency. (Take FFT and plot).
2. Calculate the amount of energy present in fundamental frequency.
3. Calculate the amount of energy present in different harmonics.

0 1 2 3 4
menus, shortcuts,
instructions etc.
10% 0 10 20 30 40
scheme
15% 0 15 30 45 60
requirements
15% 0 15 30 45 60
15% 0 15 30 45 60
Page 1 of 2
0 1 2 3 4
comparing signal
15% 0 15 30 45 60
10% 0 10 20 30 40
__
to given
instructions
10% 0 10 30 40
theoretical
principles/concepts
10% 0 10 20 30 40
Remarks
Page 2 of 2
LAB SESSION 07
OBJECTIVE:
To study s-plane and plot impulse and frequency response for different pole zero location in s-
plane. Also to determine weather system is FIR or IIR.
THEORY:
The Laplace Transform of a general continuous time signal x (t) is defined as;
-st
X(S) = ∫ x(t) e dt.
Where the complex variable s=δ+ j w, with δ and w the real and imaginary parts. CTFT is a
subset of Laplace when δ =0. Since ‘δ’ information is not present in CTFT, therefore information
about stability can only be obtained from Laplace. If pole lies on L.H.S of s-plane, system is
stable. If pole lies on R.H.S of s-plane, system is unstable. If pole lies on y(jw)-axis, system is
marginally stable or oscillatory. If system has FIR, it is stable. If system is IIR, it can be stable or
unstable.
PROCEDURE:
Generate pole zero constellation in s plane.
1. Plot corresponding Frequency (Bode magnitude) response.
2. Plot impulse response and determine that the system is FIR or IIR.
3. Modify location of poles in s plane to observe the corresponding change in frequency and
impulse response.
STEPS.
clear all;
close all;
clc;
Num = poly([(0-(i*(pi/2))),(0+(i*(pi/2)))]);
Zeros=roots(Num)
Den = poly([-1,-1]);
poles=roots(Den) sys=tf(Num,Den)
figure;
subplot(3,1,1);
pzmap(sys);
xlim([-2 2]);
ylim([-4 4]);
subplot(3,1,2);
[mag phase w]=bode(sys);
mag=squeeze(mag);
plot(w,mag);
subplot(3,1,3);
impulse(sys);
H=dfilt.df1(Num,Den);A=isfir(H)
RESULT:
1. Learn the specific logical bits of the code and make notes.
2. Observe the plots.
3. Now, explain (write) in your own words the cause and effects of what you just saw.
EXERCISE:
Change the location of poles from L.H.S of s-plane to y axis first, and then to R.H.S of s-
plane and observe the effects.
LAB SESSION 08
OBJECTIVE:
To study z-plane and plot impulse and frequency response for different pole zero location in z-
plane.Also to determine weather system is FIR or IIR.
THEORY:
The z - Transform of a general discrete time signal x(n) is defined as;
X (z) = ∑∞
𝑛=𝑜 𝑥 (𝑛) z
-n
Where the complex variable z=r ∠w , with r the radius and w the angle. DTFT is a subset of z
transform when r =1. Since ‘r’ information is not present in DTFT, therefore information about
stability in discrete time can only be obtained from z transform. If pole lies inside the unit circle,
system is stable. If pole lies outside the unit circle, system is unstable. If pole lies at the unit circle,
system is marginally stable or oscillatory. If system has FIR, it is stable. If system is IIR, it can be
stable or unstable.
PROCEDURE:
1. Generate pole zero constellation in z plane.

2. Plot corresponding Frequency (Bode magnitude) response.
3. Plot impulse response and determine that the system is FIR or IIR.
4. Modify location of poles in z plane to observe the corresponding change in frequency and
impulse response.
STEPS:
clear all;
close all;
clc;
Num = poly([(0-(i*(pi/2))),(0+(i*(pi/2)))]);
Den = poly([-1,-1]);
Num1 = poly([j,-j]);
Den1 = poly([exp(-1),exp(-1)]);
sys1=tf(Num1,Den1,1)
figure;
subplot(3,1,1);
pzmap(sys1);
xlim([-2 2]);
ylim([-4 4]);
subplot(3,1,2);
[mag phase w]=bode(sys1);
mag=squeeze(mag);
plot(w,mag);
xlim([0 100])
subplot(3,1,3);
impulse(sys1);
H=dfilt.df1(Num,Den);
A=isfir(H)
figure;
pzmap(sys1)
grid on;
RESULT:
1 Learn the specific logical bits of the code and make notes.
2 Observe the plots.
3 Now, explain (write) in your own words the cause and effects of what you just saw.
EXERCISE:
Change the location of poles from inside the unit circle to outside and at the unit circle and
observe and note the changes.

0 1 2 3 4
menus, shortcuts,
instructions etc.
10% 0 10 20 30 40
scheme
15% 0 15 30 45 60
requirements
15% 0 15 30 45 60
15% 0 15 30 45 60
Page 1 of 2
0 1 2 3 4
comparing signal
15% 0 15 30 45 60
10% 0 10 20 30 40
__
to given
instructions
10% 0 10 30 40
theoretical
principles/concepts
10% 0 10 20 30 40
Remarks
Page 2 of 2
LAB SESSION 09
OBJECTIVE:
Object of this lab is introduction to digital filters and its types, design FIR filter and study how it
performs filtering on a signal. Further truncate different types of FIR filter like Low Pass, High
Pass, Band Pass using different windows like rectangular, Kaiser Etc. and compare the results
obtained from different windows.
THEORY:
The process of deriving a realizable transfer function of a digital filter by considering given
frequency response specifications is known as digital filter design. The digital filter can be
classified as:
1. Finite –duration impulse response (FIR) filter

2. Infinite –duration impulse response (IIR) filter
In MATLAB, there are built in functions which can be used to design digital filter like IIR and
FIR.
The different types of FIR filters are listed as follows:
• Window techniques based FIR filter.

1. Rectangular windows.
2. Hamming window.
3. Hanning window.
4. Blackman window.
5. Barlett window.
6. Kaiser window.
• Equiripple linear phase FIR filter.
• Least square error FIR filters.
The different types of IIR filters are listed as follows:
• Butterworth filter
• Chebyshev Type I filter
• Chebyshev Type II filter
• Elliptic filter
FIR digital filter operates on digital sample values. It uses current and past input samples to
produce a current output sample. It does not use previous output samples. There are various
types of FIR filter based on need viz. low pass, high pass, band pass and band stop, Low pass
filter.
Following points are usually considered to design FIR filter other the window type.
INPUT:
• Window Type
• Passband and stopband ripples
• passband and stopband edge frequencies
• sampling frequency
• order of the filter
• window coefficients
OUTPUT:
• magnitude and phase responses
COMPARISON OF FIR AND IIR FILTERS

1. FIR filters are Finite Impulse Response filters with no feedback, whereas IIR contains
feedback.
2. Transfer function of FIR filter does not contain any non-trivial poles. Their frequency
response is solely dependent on zero locations. IIR filters contain poles as well as zeros.
3. As there are no poles, FIR filters cannot become unstable; however, IIR filters can
become unstable if any pole lies outside the unit circle in z-plane.
4. More number of coefficients is needed to design the desired filter in FIR than IIR.
PROCEDURE:
TASK-1
1. Create a signal vector containing two frequencies as:

i) 100 Hz. and ii) 150 Hz. with Fs = 1000 Hz.
2. Design two band pass FIR filters with 64 coefficients and with pass bands as i) 125 to
175 Hz. and ii) 75 to 125 Hz.
3. Use both filters on the created signal and observe their outputs.
4. Plot frequency responses and pole-zero constellations of both filters and note
observations.
close all; clear all; clc; % Frequencies in Hz.
F1 = 100; F2 = 150;
% Sampling Frequency in samples/sec.

Fs = 1000;
t = [0 : 1/Fs : 1]; % Time Vector

F = Fs*[0:length(t)-1]/length(t); % Frequency Vector
x = exp(j*2*pi*F1*t)+2*exp(j*2*pi*F2*t); % Signal Vector
bh = fir1( 64 , [125 175]/500); % filter coeffs.

bl = fir1( 64 , [75 125]/500); % filter coeffs.
[hh,wh]=freqz(bh,1,length(t),'whole'); % Frequency
response for filter 1
[hl,wl]=freqz(bl,1,length(t),'whole'); % Frequency
response for filter 2
% Filter operation - see filtfilt in help to learn what it
does
yh = filtfilt(bh,1,x);
yl = filtfilt(bl,1,x);
% Plotting
figure, subplot(5,1,1),
plot(F,abs(fft(x)));
xlim([0 Fs/2]);
title('FFT of original signal');
subplot(5,1,2),
plot(F,abs(hh));
xlim([0 Fs/2]);
title('Frequency response of Filter One');
subplot(5,1,3),
plot(F,abs(fft(yh)));
xlim([0 Fs/2]);
title('FFT of filtered signal from filter one');
subplot(5,1,4),
plot(F,abs(hl));
xlim([0 Fs/2]);
title('Frequency response of Filter Two');
subplot(5,1,5),
plot(F,abs(fft(yl)));
xlim([0 Fs/2]);
title('FFT of filtered signal from filter two');
xlabel('Hz.')
% Pole Zero Constellations
[bh,ah] = eqtflength(bh,1);
[zh,ph,kh] = tf2zp(bh,ah);
[bl,al] = eqtflength(bl,1);
[zl,pl,kl] = tf2zp(bl,al);
figure,
subplot(1,2,1),
pzplane(bh,ah);
xlim([-1.5 1.5]);
ylim([-1.5 1.5]);
title('Filter_One');
subplot(1,2,2),
pzplane(bl,al);
xlim([-1.5 1.5]);
ylim([-1.5 1.5]);
title('Filter Two');
TASK -2
Write a program to design a FIR filter using Hanning windows,take inputs from user for design
values of filter.
close all;
clear all;
clc;
fp=input('Enter the pass band frequency');

fs=input('Enter the stop band frequency');
rp=input(' Enter the pass band attenuation');
rs=input('Enter the stop band attenuation');
f=input(' Enter the sampling frequency');
% Calculating filter order
num=-20*log10(sqrt(rp*rs))-13;
dem=14.6*(fs-fp)/f;
n=ceil(num/dem);
n=abs(n);
% Normalizing the frequencies
wp=2*fp/f;
ws=2*fs/f;
wn=(ws+wp)/2;
%Adjusting the filter order. The order of window must be an odd
number
%and the order of filter must be one less than that of the
window
if (rem(n,2)==0)
m=n+1;
else
m=n;
n=n-1;
end
%Window sequence calculation
w=hann(m);
%Calculation of filter coefficients
b=fir1(n,wn,'low',w);
%Plotting the filter response
freqz(b,1,500,3000);
TITLE('Magnitude and Phase response');
TASK-3
Write a program for FIR(Finite Impulse Response) filter like Low pass FIR filter, High pass FIR
filter, Band pass FIR filter and Band stop FIR filter using Rectangular window using MATLAB .
ALGORITHM:
LOW PASS FILTER:
Step 1: Read the input sequence
Step 2: Perform low pass filter calculations
Step 3: Plot the output sequences
HIGH PASS FILTER:

Step 2: Perform high pass filter calculations
BAND PASS FILTER:

Step 2: Perform band pass filter calculations
BAND STOP FILTER:

Step 2: Perform band stop filter calculations
PROGRAM:
clc;
clear all;
close all;
rp=input('Enter the passband ripple(rp):');
rs=input('Enter the stopband ripple(rs):');
fp=input('Enter the passband frequency(fp):');
fs=input('Enter the stopband frequency(fs):');
f=input('Enter the sampling frequency(f):');
wp=2*fp/f;
ws=2*fs/f;
num=-20*log10(sqrt(rp*rs))-13;
dem=14.6*(fs-fp)/f;
n=ceil(num/dem);
n1=n+1;
if(rem(n,2)~=0)
n1=n;
n=n-1;
end
y=boxcar(n1);
%Low pass filter
b=fir1(n,wp,y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,1);
plot(m);
ylabel('Gain(db)->');
xlabel('(a)Normalised frequency->');
%High pass filter
b=fir1(n,wp,'high',y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,2);
plot(m);
ylabel('Gain(db)');
xlabel('(b)Normalised frequency');
%Band pass filter
wn=[wp*ws];
b=fir1(n,wn,y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,3);
plot(m);
ylabel('Gain(db)');
xlabel('(c)Normalised frequency');
%Band stop filter==============
wn=[wp*ws];
b=fir1(n,wn,'stop',y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,4);
plot(m);
ylabel('Gain(db)');
xlabel('(d)Normalised frequency-');
EXERCISE:
Q1. Perform Q3.using Hamming and Kaiser Window.
Compare results of designed filters using three different windows on a single plot.
LAB SESSION 10
OBJECTIVE:
Object of this lab is to design different IIR filter using FDA tool.
THEORY:
Filter Design and Analysis Tool (FDA Tool) is Graphic User Interface for designing and analyzing
filters. It is used to design FIR and IIR filters by entering the desired filter specifications, or by
importing filter from MATLAB workspace or by adding, moving or deleting poles and zeros. After
designing a filter, the response can be viewed and analyses in other Graphic User Interface tool
named Filter Visualization Tool (FV Tool) linked with FDA Tool. The different types of responses
that can be viewed are listed below:
• Magnitude response
• Phase response
• Group delay
• Phase delay
• Impulse response
• Step response
• Pole-zero plot
• Zero-phase plot
OPENING FDA TOOL WINDOW:
FDA Tool can be opened using command:
fdatool
Figure A
The different steps involved in designing a filter using FDA Tool can be listed as:
1. Selection of type of response required

2. Selection of type of filter design
3. Specifying the filter order
4. Entering the filter specifications
5. Entering the magnitude specifications
After providing the information listed above, filter can be designed and its response can be
viewed and analysed.
The complete description of the FDA Tool window and different steps required to design a filter
are elaborated below:
1. Selecting response type: The desired response type is selected from the list of available
options, i.e., lowpass, highpass, bandpass, bandstop, differentiation, multiband, peaking etc.
2. Type of design method: The design can be of FIR or IIR filter. Depending upon whether
FIR Or IIR filter design is selected, further options are available in the dropdown menu.
In IIR filter design, the different options available in dropdown menu are as given below:
• Butterworth
• Chebyshev type I
• Chebyshev type II
• Elliptic
• Maximally flat
• Least Pth-norm
• Const least Pth-norm
In FIR filter design the options available are listed as follows:
• Equirriple
• Least square
• Window
• Const least squares
• Complex equiripple
• Least Pth norm
• Constrained equiripple
• Generalized equiripple
• Constrained band equirriple
• Interpolated FIR
The options available depend upon the selection of response type.
3. Filter order: Under this, two options available are as

• User-defined order: Under this option user has to enter the order of the filter.
• Minimum order: The other option available for selecting the filter order is
minimum order. It is calculated by system itself.
4. Filter specifications: Depending upon the response type and design method selected,
the graphical representation of generalized filter specifications appear in the display
region of FDA Tool. These specifications are ‘Frequency Specifications’ and
‘Magnitude Specification’.
These specifications are provided by the user, as per filter design requirement, in the
appropriate blocks.
5. Designing filter: After all the requisite information is entered, a filter can be designed by
clicking the ‘Design Filter’ button available at the bottom of the window. Filter |
coefficients are calculated and magnitude response appears in the display region.
(Note: ‘Design Filter’ button will be disabled once the filter coefficients are computed. This
button will be enabled again in case any changes are made in the filter specifications.)
6. Displaying filter responses: Once the filter coefficients are calcu
lated as per the specifications provided by the user, the display region will show
magnitude response of the designed filter. The other filter response characteristics
can be viewed in the display region or FV Tool. The response to be viewed can be
selected from the different icons displayed on the toolbar shown in Figure below.
Figure : Different Response Icons on the Toolbar
(NOTE: The different responses for display can also be selected from the
‘Analysis’ menu on menu bar.)
7. Current filter information: The information about the designed filter is given in
the ‘Current Filter Information’ region of FDA Tool window as shown
in Figure A The information provided is about the ‘structure’, ‘order’, ‘stability’ and
‘source’
• Storing a filter
The designed filter is stored by clicking ‘Store Filter’ button in the
‘Current Filter Information’ region.
• Filter manager
The ‘Filter Manager’ button opens up a new Filter Manager window
(Figure B) showing the list of filters stored. This window also has
options as: Edit current filter, Cascade, Rename, Remove and FV Tool.
Figure B Filter Manager Window

To cascade two or more filters, highlight the designed filters and press ‘Cascade’ button.
A new cascaded filter is added to the ‘Filter Manager’.
8. Saving and opening filter design session:

The filter design session is saved as MAT-file and can be used later on. It can be saved by
clicking save icon or selecting save session
option in File menu and giving desired session name. Similarly, the saved session
can be opened by clicking open icon or by selecting open option in file menu and
selecting the previously saved filter design session.
FILTER VISUALIZATION TOOL:
The response characteristics can be viewed in a separate window by selecting the ‘Filter
Visualization Tool’ (FV Tool) from ‘view’ menu or clicking the ‘Full View Analysis’ button
on the toolbar. The FV Tool window is shown in Figure C
FV Tool has most of the menus on the menu bar and icons on the toolbar similar to that FDA
Tool with some additional icons which are mainly used to work with representation of the
responses.
Figure C Filter Visualization Tool(FV Tool) window

IIR FILTER DESIGN USING FDA TOOL
TASK-1
Design an IIR Butterworth band pass filter with the following
specifications:
Normalized pass band edges at 0.40 and 0.65
Normalized stop band edges at 0.3 and 0.75
Pass band ripple 1 dB
Minimum stop band attenuation 40 dB
Show (i) Magnitude response (ii) Phase response (iii) Group delay (iv) Phase delay response.
Solution:
As per the given specifications, the requisite data is entered in new FDA Tool window
as shown in Figure
Figure. FDA Tool Window Showing Specification Entered and Magnitude Response for Task-1.
The filter is designed for minimum order so as to reduce the complexity of the design.
In case, it has to be designed for user defined order, then the order of the filter has to be
calculated first by user using appropriate formulas or MATLAB function.
The other responses can be viewed by clicking on the appropriate icon on the toolbar and
responses obtained are shown in Figures below
Figure Magnitude Response in dB
Figure Phase Response

Figure Group Delay Response
Figure. Phase Delay
These responses can also be viewed in FV Tool.

TASK-2
Design a Type II Chebyshev IIR lowpass filter with the following specifications:
Passband frequency 1,200 Hz
Stopband frequency 1,700 Hz
Sampling frequency 6,000 Hz
Passband ripple 0.50 dB
1. Show magnitude response.

2. Show pole/zero plot.
Solution: FDA Tool Window showing given specifications duly entered and magnitude
response in response display region is shown in Figure.
Figure. FDA Tool Window for Example

By using the FV Tool, the magnitude response and pole/zero plot is obtained as separate figures and is
shown in Figures.
Figure. Magnitude Response for Task-2.
Figure Pole/zero Plot for Q2

TASK-3:
Design an elliptic IIR low pass filter with following specifications:
Pass band ripple 0.5 dB
Stop band attenuation 40 dB
Pass band frequency 800 Hz
Stop band frequency 1,000 Hz
Sampling frequency 4,000 Hz
1. Show magnitude and phase response in the same window.

2. Obtain filter information.
Solution:
As per given specifications, the requisite data is entered in FDA Tool window. By clicking
appropriate icon on toolbar, the magnitude and phase responses are obtained in the same
window.
1. These magnitude and phase responses obtained are viewed in FV Tool window also and
are shown in Figure .
Figure . Magnitude and Phase Response for Task-3.
2. To obtain the information about the filter ‘Filter Information’ icon on Toolbar of FDA
Tool Window is clicked or ‘Filter Information’ option is selected from ‘Analysis’ menu. The
detailed filter information appears in the display region as shown in Figure a, b and c.
Figure ‘Filter information’ for Task-2
The filter information is obtained by scrolling down the text in the window shown
in Figure 15.41b.
EXERCISE:
Record Your Voice at home while turn any motor of your house ON.
Design a filter using FDA Tool.
Remove sound of motor from recorded signal.
Listen the output signal.


0 1 2 3 4
menus, shortcuts,
instructions etc.
10% 0 10 20 30 40
scheme
15% 0 15 30 45 60
requirements
15% 0 15 30 45 60
15% 0 15 30 45 60
Page 1 of 2
0 1 2 3 4
comparing signal
15% 0 15 30 45 60
10% 0 10 20 30 40
__
to given
instructions
10% 0 10 30 40
theoretical
principles/concepts
10% 0 10 20 30 40
Remarks
Page 2 of 2
(Open Ended Lab 01)
Objective:
To convert an analog (voltage & current) signal into digital signal using ADC
(audio card) and display it on MATLAB Simulink environment.
Required Components:
1. Audio Card
2. Transformer (220V/12V)
3. Resistors (for VDR)
4. Veroboard
5. Audio jack
6. PC with MATLAB environment
Procedure:
• Using Transformer convert 220VAC from mains into 12VAC.

• Using VDR convert 12VAC to a voltage compatible to audio card (show all the
calculations of resistances with their power ratings).
• Set the sampling frequency of the audio card ADC in MATLAB Simulink
environment with proper justification
• Plot the acquired voltage waveform to Simulink scope.
• Mention the safe operating range of your equipment.
Project Summary: (Not more than one sheet)

Project Specification:
Calculations:
• VDR and Resistance power calculation

• Resolution of ADC
• Sampling Frequency
• Gain Factor
Attachments:
• Project Block Diagram
• Real Project Image
• Image of current and voltage plot (with proper labelling)

0 1 2 3 4
menus, shortcuts,
instructions etc.
10% 0 10 20 30 40
scheme
15% 0 15 30 45 60
requirements
15% 0 15 30 45 60
15% 0 15 30 45 60
Page 1 of 2
0 1 2 3 4
comparing signal
15% 0 15 30 45 60
10% 0 10 20 30 40
__
to given
instructions
10% 0 10 30 40
theoretical
principles/concepts
10% 0 10 20 30 40
Remarks
Page 2 of 2
(Open Ended Lab 02)
Objective:
To convert analog (Voltage and current) signal into digital signal using ADC
(audio card). Display it on MATLAB Simulink environment and perform
Spectral Analysis of the resulting current signal.
Required Components:
1. Audio Card
2. Current Sensor (current sensing resistor / hall effect sensor / CT)
3. Vero board
4. Audio jack
5. Harmonic producing Load (Electronic devices)
6. PC with MATLAB environment
Procedure:
➢ Using current sensor, convert the current flowing through load into an equivalent voltage.
➢ If required, using VDR to convert the voltage as obtained from current sensor to a voltage
compatible to audio card (show all the calculations of resistances with their power
ratings).
➢ Set the sampling frequency of the audio card ADC in MATLAB Simulink environment
with proper justification
➢ Plot the acquired current waveform to Simulink scope.
➢ Mention the safe operating range of your equipment.
➢ Plot the frequency spectrum of the obtained current and Voltage waveform. Use
windowing function to reduce DFT leakage if required.
➢ Also, plot the frequency spectrum of the line voltage as obtained in open ended lab 01.
Project Summary:
Project Specification:
Attachments:
• Project Block Diagram
• Real Project Image
• Image of current and voltage plot and Spectrum(with proper labelling)
Results:

0 1 2 3 4
menus, shortcuts,
instructions etc.
10% 0 10 20 30 40
scheme
15% 0 15 30 45 60
requirements
15% 0 15 30 45 60
15% 0 15 30 45 60
Page 1 of 2
0 1 2 3 4
comparing signal
15% 0 15 30 45 60
10% 0 10 20 30 40
__
to given
instructions
10% 0 10 30 40
theoretical
principles/concepts
10% 0 10 20 30 40
Remarks
Page 2 of 2

0 1 2 3 4
menus, shortcuts,
instructions etc.
10% 0 10 20 30 40
scheme
15% 0 15 30 45 60
requirements
15% 0 15 30 45 60
15% 0 15 30 45 60
Page 1 of 2
0 1 2 3 4
comparing signal
15% 0 15 30 45 60
10% 0 10 20 30 40
__
to given
instructions
10% 0 10 30 40
theoretical
principles/concepts
10% 0 10 20 30 40
Remarks
Page 2 of 2

EE 395 Digital Signal Processing

Uploaded by

Copyright:

Available Formats

EE 395 Digital Signal Processing

Uploaded by

Document Information

Original Description:

Copyright

Available Formats

Share this document

Share or Embed Document

Sharing Options

Did you find this document useful?

Is this content inappropriate?

Copyright:

Available Formats

EE 395 Digital Signal Processing

Uploaded by

Copyright:

Available Formats

NED University of Engineering & Technology

Department of Electrical Engineering

Digital Signal Processing

Attendance: Present out of _____ Lab sessions

Attendance Percentage ________

To be filled by Lab Instructor

Lab Score Sheet

Rubric based labs for EE-395 DSP: 1, 2, 3, 6, 8, 10

Digital Signal Processing

Content Revision Team:

The Board of Studies of Department of Electrical Engineering

S.No. Date Title of Experiment Total Marks Signature

To study the effects of Sampling in Discrete Time

To study the effects of Quantization in Discrete Time

To study and verify Discrete-Time convolution and its

4 To study Discrete-Time correlation.

The Discrete Fourier Transform as a Linear

Studying Discrete Fourier Transform using an audio

7 Relationship between Laplace and CTFT.

8 Relationship between Z transform and DTFT.

9 Designing FIR filters with windowing

10 Designing IIR filters using FDA tool

11 Open Ended Lab1 (due after lab 2)

12 Open Ended Lab2(due after lab 6)

Analog phenomenon is continuous – like a human speech of speaker, or a continuously rotating

Signal processing is concerned with the acquisition, representation, manipulation, transformation,

clear all;close all;clc;

nTs = [0 :Ts : 0.2];

3. Observe the plots.

1. Consider the following CT signal:

2. Generate a tone in MATLAB with varying frequency f = 1000,2000,3000,4000, 5000, 6000,

3. Record a sentence in your voice.(you may use Simulink /audacity to record).Change Fs

Course Code: EE-394 Course Title: Digital Signal Processing

SNR = 10log10 (Px /Pe )

SQNRA/D = 10 log 10(Px /Pe)

q=input('No. of Digits after decimal points to be retained (0-9): ');

Explore and take notes.

% No. of Digits after decimal points to be retained for num = 1 :

1. Effects of Quantization with variable precision levels

2. Simple sinusoid quantized to various bits per sample

3. Audio signal quantization to various bits per sample

Course Code: EE-394 Course Title: Digital Signal Processing

1. Convolution is given as :y(n) = x(n)*h(n) =

1. We have the impulse response of a system as h(n) = { 3,2,1,-2,1,0,-4,0,3}

ny = [nh(1)+ nx(1) : nh(end)+nx(end)];

1. What will happen if we input x(n)={0,0,1,0,0} into the above system.

4. Convolve your recorded sound with drumloop.wav. Note your observation

a) Plot the output.

Course Code: EE-394 Course Title: Digital Signal Processing

1. Generate two sinusoids of length 10 and fd = 0.1 with variable phase.

[maxR indR] = max(rxy);

disp(['The correlation at lag zero is: ' num2str(rxy(find(nr==0)))

Please write in exercise book.

Voltage A Min Voltage B Min Voltage C Min

The formulas for the DFT and IDFT are given as

Which is an Nth root of unity. Where WN is called Twiddle Factor , also

[WN] = [𝑒 −𝑗2𝛱/𝑁 ]kn

DFT analysis equation in matrix form is

DFT synthesis equation in matrix form is

x = exp(j2piF1t)+2exp(j2*piF2t); % Signal Vector