PCS Manual v4

COMSATS Institute of Information Technology Lahore
Department of Electrical Engineering
EEE351 –Principles of Communication Systems

Lab Manual
Prepared By
Engr. Assad Ali
Engr. Arslan Khalid
Engr. Tabassum Nawaz
Engr. Ayesha Naz
Supervised By
Dr. Sobia Baig
Name:
(Block Letters)
Registration No.:
Revision History
S.No. Update Date Performed by
1 Lab Manual Preparation Sep- Engr. Rizwan

2013 Asif
2 Lab Manual Review Dec- Dr. Sobia Baig
2014
3 Layout modifications Jan- Ayesha Naz
2015
4 Lab Manual Final Review Feb- Dr.Hafiz Asif
2015
5 Lab Manual modifications and Review Aug- Ayesha Naz
2015
6 Lab Manual Modification according to Aug- Ayesha Naz,
OBE Requirements. 2017 Assad Ali
7 Lab Manual Review Sep- Dr. Sobia Baig
2017
| EEE351| Principles Communication System Lab Manual i

Preface
The field of telecommunication has reconstructed the frontier upon which the world has come to
intricately interweave the lives of people across the globe. The industry has catapulted in a very short
span of time, and needless to say it would continue to do so for centuries to come. This manual has
been written as a laboratory reference for the course titled EEE 351 Principles of Communication
Systems. This is a basic course for students of BS.Engineering. Students for whom Principles of
Communication Systems is the only one course offered in the field of communication.
The course starts with a brief review of Fourier analysis and random processes. Basic analog
communication systems, including Amplitude Modulation and Frequency Modulation systems are
covered next. Then, digital communication systems using Pulse Code Modulation (PCM), Pulse
Amplitude Modulation is also discussed. The performance of communication systems in the
presence of noise is also analysed. We will consider the effects of inter symbol interference and
noise and ways to mitigate them using software and hardware. In the laboratory, the student will
perform experiments which demonstrate the basic principles of analog and digital communication
systems, also covered in the theoretical part of curriculum.
All experiments described in this manual are performed on TIMS (Telecommunications Instructional
Modelling System) as well as Discreet Components. TIMS is a modular system for modelling
telecommunications block diagrams whereas the simulations are performed within the environment
of MATLAB which is interactive software for scientific and engineering calculations. Simulations
can model the behaviour of real systems with remarkable degree of precision.
| EEE351| Principles Communication System Lab Manual ii

Books
Text Books
1. Modern Digital and Analog Communication Systems by B.P Lathi
Reference Books
1. Introduction to Digital and Analog Communication Systems by Simon Haykin and

Michael Moher, Wiley, 4th edition.
2. Fundamentals of Communication Systems by Hohn G. Proakis and Masoud Salehi,
Prentice
Hall, 2004.
Learning Outcomes
1. Display the performance of various analog and digital modulation techniques using
software and hardware tools (Level: P5)
2. Manipulate various parameters for evaluation of analog and digital communication
systems using software and hardware tools (Level: P5)
Grading Policy
Grades Letter Grade Credit Points Percentage Marks

A ( Excellent) 4.0 90and above
A- 3.7 85-89
B+ 3.3 80-84
B (Good) 3.0 75-79
B- 2.7 70-74
C+ 2.3 65-69
C (Average) 2.0 60-64
C- 1.7 55-59
D (Minimum passing) 1.3 50-54
F (Failing) 0.0 Less than 50
| EEE351| Principles Communication System Lab Manual iii

List of Equipment
 EMONA Telecommunications Instructional Modelling System(TIMS)
 Instek Digital Oscilloscope

 Multimeter
 Regulated Power Supply
 Function Generators
 Breadboard and other Discreet Components.
Software Resources
 MATLAB
 SIMULINK
| EEE351| Principles Communication System Lab Manual iv

Table of Contents
Revision History i
Preface ii
Books iii
Learning Outcomes iii
Grading Policy iii
List of Equipment iv
Software Resources iv
Table of Contents v
Experime nt Relevant to Learning Outcome vii
To sketch Frequency Domain Representation of Signals for Analysis of Communication

Systems Using MATLAB. 8
To Construct a Modulator for Amplitude Modulation in Analog Communication Systems

Using Discrete Components. 19
To Cconstruct an Envelope Detector Using Discrete Components for Demodulation in Analog
Communication Systems. 24
To Assemble A Modem For Single Side Band Amplitude Modulation Using TIMS Trainer For
Analog Communication Systems 31
To Assemble A Modem For Quadrature Amplitude Modulation Using TIMS Trainer For
Analog Communication Systems. 36
To Assemble Phase Locked Loop (PLL) using TIMS Trainer for Phase detection in Analog
To Construct Frequency Modulator Using Discrete Components For Analog Communication
Systems. 48
To Display Sampling of Analog Signals for Digitization Using Discrete Components And
MATLAB. 52
To Display Various Line Coding Schemes For Pulse Code Modulation Using TIMS Trainer 60
To an Assemble Encoder for Pulse Code Modulation (PCM) Using TIMS Trainer 66
To assemble Decoder for Pulse Code Modulation (PCM) Using TIMS trainer 71
To Construct a Modem Based on Amplitude Shift Keying Using Discrete Components for
Digital Communication Systems 75
To Construct Modem Based on Binary Phase Shift Keying Using Discrete Components for
Digital Communication Systems 81
To Construct Modem Based on Frequency Shift Keying Using Discrete Components for Digital
| EEE351| Principles Communication System Lab Manual v

| EEE351| Principles Communication System Lab Manual vi
Experiment Relevant to Learning Outcome
Learning Outcome Relevant Experiment
Display the performance of various analog and

digital modulation techniques using software and Lab # 01-02, 03, 04, 05, 07, 10, 11, 12, 13, 14
hardware tools.
Manipulate various parameters for evaluation of

analog and digital communication systems using Lab# 06, 08, 09
software and hardware tools
| EEE351| Principles Communication System Lab Manual vii

LAB # 1: To sketch Frequency Domain Representation of Signals
for Analysis of Communication Systems Using MATLAB
Objective
 To display magnitude and phase response of Fourier series coefficients using MATLAB
for visualization of signals in communication systems.
Lab Instructions
 This lab activity comprises of two parts: Lab Exercises and Post-Lab Viva session.
 The students should perform and demonstrate each lab task separately for step-wise
evaluation
 Only those tasks that completed during the allocated lab time will be credited to the
students. Students are however encouraged to practice on their own in spare time for
enhancing their skills.
 A student, who will not bring manual in lab, will be awarded with zero marks in
respective lab.
Lab Report Instructions

All questions should be answered precisely to get maximum credit. Lab report must ensure
following items:
 Lab objectives
 Answers of Exercise/viva Questions.
 Results (graphs/patterns) duly commented and discussed
 Conclusion
Part-I Fourier Series

Introduction
In mathematics, a Fourier series decomposes periodic functions or periodic signals into the sum
of a (possibly infinite) set of simple oscillating functions, namely sine’s and cosines (or complex
exponentials). The study of Fourier series is a branch of Fourier analysis.
Using Euler’s Equation, we can convert the standard Rectangular Fourier Series into an
exponential form. Even though complex numbers are a little more complicated to comprehend,
we use this form for many reasons:
 Only need to perform one integration

 A single exponential can be manipulated more easily than a sum of sinusoids
 It provides a logical transition into a further discussion of the Fourier Transform
In Lab
| EEE351| Principles Communication System Lab Manual 8

In this lab, you will perform the following tasks:
Lab Task 1
1. Evaluate the Fourier series coefficients using
Plot the magnitude | Dn | (in volts) and phase ∠Dn (in degrees) of the first twenty-one
coefficients
{n= -10,… 0,…, 10} versus frequency (in rad/sec).
2. Plot two periods of g(t) , directly i.e., by creating a vector of samples of g(t) and plotting
that vector.
3. Plot an approximation to g(t) using these first twenty-one terms of the exponential
Fourier series.
Make a function file sinc1.m using the following code listing.
Listing 1.1: Sinc function

functiony = sinc 1 ( x ) % sinc function implementation
k = length ( x ) ;
fori = 1 : k
ifx ( i ) == 0
y( i)= 1;
else
y ( i ) = sin ( x ( i ) ) /x ( i ) ;
end
end
end
Sketch the Plot for listing 1.1

Lab Task 2
Magnitude and Phase of Fourier Series Coefficients
Write the following code in MATLAB and run it.
Listing 1.2: Magnitude and Phase Response

n = [ -1 0 : 1 0 ] ;
z = n*( pi/4) ;
Dn = 0.25* exp(- I *z ) . *sinc 1 ( z ) ;
magDn = abs (Dn) ;
argDn = angle (Dn) *(180/ pi ) ;
w = 0.5*n ;
stem(w,magDn) , xlabel( ' Frequency in rad/ s e c ( units of pi ) ' )
grid
title( 'Magnitude o f the Exponential Fourier series coefficients' )
stem(w, argDn ) , xlabel( ' Frequency in rad/ s e c ( units of pi ) ' )
ylabel( ' degrees ' )
grid
title( ' Phase of the Exponential Fourier series coefficients')
Sketch the Plot for listing 1.2

Lab Task 3
Sum of Fourier Series Coefficients:
The signal g(t) can be reconstructed by adding the Fourier coefficients. To verify this we find the
sum given by
The following code in Listing 1.3 is used to find the sum in (1.2). The output is shown in
Figure 1.1.
Listing 1.3: Approximation of g(t)
n = [ -1 0 : 1 0 ] ;
z = n*( pi/4) ;
Dn = 0.25* exp(-i*z ) . * sinc 1 ( z ) ; % symbol . * means elementbyelement multiplication
nwo = n*( pi /2) ; % define eleven frequencies for sum
t = [ 0 : 0 . 0 1 : 8 ] ; % define sampling points
BIG= nwo ' * t ;
g = Dn* exp( i*BIG) ; % here ' s where the sum is done
plot( t , real ( g ) ) , grid ,
xlabel( ' Seconds ' )
title( ' Approximation to g ( t ) using the first ten components of the fourier series’)

Figure 1.1: Approximation to g(t) using the first ten components of the Fourier Series
We can approximate g(t) using the first ten components of the Fourier series. The following code is used
to generate two periods of function g(t). It uses a customized unit step function, u(t), a copy of
which is also provided below. The ability to use the unit step function to write piecewise
functions proves extremely effective. Below is code listing.
Listing 1.4: g(t) using unit step
gt= (u( t ) -u( t-1) ) + (u( t-4) -u( t-5) ) + u( t-8) ;

plot( t , gt)
grid ,xlabel( ' seconds ' )
title( 'The r e a l g ( t ) ' )
axis( [ 0 8 -0.2 1. 2 ] ) ;
Listing 1.5: u(t) Implementation
function[ y ] = u( x )
y=0.5+0.5*sign ( x ) ;
end
Sketch the Plot for listing 1.4 & 1.5

Part-II Discrete Fourier Transform (DFT)
Introduction
In mathematics, the DFT is a specific kind of discrete transform, used in Fourier analysis. It
transforms one function into another, which is called the frequency domain representation of the
original function (which is often a function in the time domain).
The DFT requires an input function that is discrete. Such inputs are often created by sampling a
continuous function, such as a person's voice. The discrete input function must also have a
limited (finite) duration, such as one period of a periodic sequence or a windowed segment of a
longer sequence.
Discrete Fourier Transform (DFT) decomposes a sequence of values into components of
different frequencies. This operation is useful in many fields but computing it directly from the
definition is often too slow to be practical. An FFT is a way to compute the same result more
quickly, computing a DFT of N points using the definition, takes O(N2) arithmetical operations,
while an FFT can compute the same result in only O(N log N) operations.
The difference in speed can be substantial, especially for long data sets where N may be in the
thousands, the computation time can be reduced by several orders of magnitude in such cases.
This huge improvement made many DFT-based algorithms practical. FFTs are of great
importance to a wide variety of applications, from digital signal processing and solving partial
differential equations to algorithms for quick multiplication of large integers.

In particular, the DFT is widely employed in signal processing and related fields to analyze the
frequencies contained in a sampled signal, to solve partial differential equations, and to perform
other operations such as convolutions or multiplying large integers. A key enabling factor for
these applications is the fact that the DFT can be computed efficiently in practice using a fast
Fourier transform (FFT) algorithm.
In Lab
Lab Task 1
Plotting the Signal
This is the code segment for plotting the signal g(t). Vector g contains samples of function g(t),
which is formed by concatenating three individual vectors g1, g2 and g3. The result in Figure 1.2
shows the plot of g(t) signal.
Listing 2.1: Plotting signal g(t)
g1 = [0:1/32:1-(1/32) ] ;
g2 = [ -1:1/32:1-(1/32) ] ;
g3 = [ -1:1/32:0-(1/32) ] ;
g = [ g1 , g2 , g3 ] ;
t = [ -1:1/64:1-(1/64) ] ;
plot( t , g ) ;
axis( [ -1.5 1.5 -1.5 1.5 ] )
grid
Lab Task 2
Discrete Fourier Series using FFT
Using the following code we get the Fourier series coefficients.
Listing 2.2: FFT of g(t)
z = fft( g ) ;
stem( t , z ) ;
If we plot this calculated fft what we get is an arrangement of those total 128 coefficients one by
one i.e. they are not arranged as a normal Fourier series spectrum. FFTSHIFT command helps us
to reach there. We will get a plot using FFTSHIFT command such that DC component is at the
centre, and plot gets the shape of a normal Fourier series plot.
Listing 2.3: Shifted version of FFT of g(t)

z1 = fftshift( z ) ;
n=[1:1:128] ;
a = n-65;
f = 0.5* a ;
stem( f , abs ( z1 ) )

Figure 1.2: Plot using the MATLAB function 'FFTSHIFT'.
The above approach uses the MATLAB commands FFT and FFTSHIFT.
Comments
 FFT function plot the Fourier series but with the DC component.
 FFTSHIFT shifts the DC component to the center of spectrum.
 Magnitude of the Fourier series is plotted against frequency to remove the complex part
from the Fourier series.
Lab Task 2
1. Write MATLAB code to get the spectrum of g(t) and also sketch the output.
2. Write MATLAB code to get the magnitude and phase of g(t).

Lab Task 3
Given the signal in Figure 1.3:
1. Plot the signal in MATLAB for two periods of time.
2. Find its Fourier series and plot it. The plot should have DC component at the center.

Figure 1.3: Example plot for the assignment.

Performance Viva
Total/15
(10 Marks) (5 Marks )
Performance /4
Results /3
Lab Report /3
Comments

Lab # 2: To Construct a Modulator for Amplitude Modulation in Analog
Communication Systems Using Discrete Components
Objectives
 To construct amplitude modulator and observed output signals using various discreet
components.
Equipment Required
Resistors, Capacitors, Oscilloscope, Multimeter, Regulated power supply, Function generators,
Breadboard and Connecting Wires.
Lab Instructions
evaluation
 A student who will not bring manual in lab, will be awarded with zero in respective lab.

following items:
 Lab objectives
 Conclusion
Introduction
An amplitude modulated signal is defined as:
AM = (A + m(t) ) cosωt ........ 1
= m(t)cosωt+Acosωt ........ 2
= [low frequency term m(t)] x [high frequency term c(t)] ........ 3

Here:
‘A’ is the DC value. For modelling convenience eqn. (1) has been written into two parts in eqn.
(2).
Block Diagram
Equation (1) can be represented by the block diagram of Figure 1 .
Figure 1: Block diagram of AM
To make 100% amplitude modulated signal adjust the ADDER output voltages independently to
+1 volt DC and 1 volt peak of the sinusoidal message. Figure 2 illustrates what the
oscilloscope will show
Figure 2: AM, with A=1, as seen on the oscilloscope
The depth of modulation ‘m’ can be measured either by taking the ratio of the amplitude of
the AC and DC terms at the ADDER output, or applying the formula:
m =P-Q/P+Q

Figure 3: the oscilloscope display for the case A = 1.5
where P and Q are the peak-to-peak and trough-to-trough amplitudes respectively of the AM
waveform of Figure 3. Note that Q = 0 for the case m = 1. To vary the depth of modulation use
the G gain control of the ADDER. Notice that the ‘envelope’, or outline shape, of the AM signal
of Figure 3 is the same as that of the message provided that m ≤1.
The envelope of the AM signal is defined as |a(t)|. When m ≤1 the envelope shape and the
message shape are the same. When m > 1 the envelope is still defined as |a(t)|, but it is no longer
the same shape as the message. Note that eqn.(4) is still applicable - the trough is interpreted as
being negative.
Significance of A
First note that the shape of the outline, or envelope, of the AM waveform (lower trace), is exactly
that of the message waveform (upper trace). As mentioned earlier, the message includes a DC
component, although this is often ignored or forgotten when making these comparisons. You can
shift the upper trace down so that it matches the envelope of the AM signal on the other trace.
Now examine the effect of varying the magnitude of the parameter 'm'. This is done by varying
the message amplitude with the ADDER gain control G.

Figure 4: Amplitude Modulation model graph
Lab Task 1: Generation of AM using discrete components

MATLAB Exercises
Q1. Write the given code on MATLAB and verify the results of the performed experiment.
Draw its Simulink diagram as well.
Q2. Use MATLAB to generate and display an AM wave for 100% modulation, under
modulation and over modulation.
Carrier frequency, fc=5kHz
Amplitude of carrier frequency, Ac=9
Performance Viva
Total/15
Performance /4
Results /3
Lab Report /3

LAB # 3:To Cconstruct an Envelope Detector Using Discrete Components for
Demodulation in Analog Communication Systems
Objectives
 To assemble an envelope detector using TIMS trainer for demodulation of amplitude
modulated signals.
 To construct envelop detector using discreet components for demodulation of
amplitude modulated signals.
 To sketch the recovered message signal with envelop detector using MATLAB for
demodulation of amplitude modulated signals.
Lab Instructions
evaluation
 A student, who will not bring manual in lab, will be awarded with zero in respective lab.

following items:
 Lab objectives
 Conclusion
Introduction
Envelopes:
When we talk of the envelopes of signals we are concerned with the appearance of signals in the
time domain. Text books are full of drawings of modulated signals, and you already have an idea
of what the term ‘envelope’ means. It will now be given a more formal definition.
Qualitatively, the envelope of a signal y(t) is that boundary within which the signal is contained,
when viewed in the time domain. It is an imaginary line.

This boundary has an upper and lower part. You will see these are mirror images of each other.
In practice, when speaking of the envelope, it is customary to consider only one of them as ‘the
envelope’ (typically the upper boundary).
Although the envelope is imaginary in the sense described above, it is possible to generate, from
y(t), a signal e(t), having the same shape as this imaginary line. The circuit which does this is
commonly called an envelope detector.
The Diode Detector:
The ubiquitous diode detector is the prime example of an envelope generator. It is well
documented in most textbooks on analog modulation. It is synonymous with the term ‘envelope
demodulator’ in this context.
But remember: the diode detector is an approximation to the ideal. We will first examine the
ideal circuit.
The Ideal Envelope Detector:

The ideal envelope detector is a circuit which takes the absolute value of its input, and then
passes the result through a lowpass filter. The output from this lowpass filter is the required
envelope signal. See Figure 3.1.
Figure 3.1: the ideal envelope recovery arrangement
The truth of the above statement will be tested for some extreme cases in the work to follow; you
can then make your own conclusions as to its veracity.
The absolute value operation, being non-linear, must generate some new frequency components.
Among them are those of the wanted envelope. Presumably, since the arrangement actually
works, the unwanted components lie above those wanted components of the envelope.
It is the purpose of the lowpass filter to separate the wanted from the unwanted components
generated by the absolute value operation.
The analysis of the ideal envelope recovery circuit, for the case of a general input signal, is not a
trivial mathematical exercise, the operation being non-linear. So it is not easy to define
beforehand where the unwanted components lie
Lab Tasks
Lab Task 1: Generation of AM using TIMS
Implement this hardware model on TIMS and observe the output.

Figure 3.2: TIMS model for AM generation
Lab Task 2: Envelop detection using TIMS

The Ideal Model:
The TIMS model of the ideal envelope detector is shown in block diagram form in Figure 3.2.
Figure 3.3: Modeling the ideal envelope detector with TIMS
The ‘ideal rectifier’ is easy to build, does in fact approach the ideal for our purposes, and one is
available as the RECTIFIER in the TIMS UTILITIES module. For purposes of comparison, a
diode detector, in the form of ‘DIODE + LPF’, is also available in the same module; this will be
examined later.
The desirable characteristics of the lowpass filter will depend upon the frequency components in
the envelope of the signal as already discussed.
We can easily check the performance of the ideal envelope detector in the laboratory, by testing
it on a variety of signals.

The actual envelope shape of each signal can be displayed by observing the modulated signal
itself with the oscilloscope, suitably triggered .
AM Envelope:
For this part of the experiment we will use the generator of Figure 3.4, and connect its output to
the envelope detector of Figure 2.
Figure 3.4: Generator for AM and DSBSC
T1 plug in the TUNEABLE LPF module. Set it to its widest bandwidth, which is about 12 kHz
(front panel toggle switch to WIDE, and TUNE control fully clockwise). Adjust its passband
gain to about unity. To do this you can use a test signal from the AUDIO OSCILLATOR, or
perhaps the 2 kHz message from the MASTER SIGNALS module.
T2 model the generator of Figure above and connect its output to an ideal envelope detector. For
the lowpass filter use the TUNEABLE LPF module. Your whole system might look like that
shown modeled in Figure 3.5 below.
Figure 3.5: modulated signal generator and envelope recovery

T3 set the frequency of the AUDIO OSCILLATOR to about 1 kHz. This is your message. T4
adjust the triggering and sweep speed of the oscilloscope to display two periods of the message
(CH2-A). T5 adjust the generator to produce an AM signal, with a depth of modulation less than
100%. Don`t forget to so adjust the ADDER gains that its output (DC + AC) will not overload
the MULTIPLIER; that is, keep the MULTIPLIER input within the bounds of the TIMS
ANALOG REFERENCE LEVEL (4 volt peak-to-peak). This signal is not symmetrical about
zero volts; neither excursion should exceed the 2 volt peak level. T6 for the case m < 1 observe
that the output from the filter (the ideal envelope detector output) is the same shape as the
envelope of the AM signal a sine wave.
Lab Task 3: Envelop detection using envelop detector

 Generate AM signal (D1, R1, R2, C1, L1) with modulation index less than 1 i.e. µ < 1.
 Build the envelope detector (D2, C2, and R3) as shown in figure 1 below.
 Display the demodulated output of the envelope detector on the oscilloscope.
 Compare the message signal to the demodulated signal on the oscilloscope.
 Draw the message, modulated and the demodulated signal on the next page.
 Investigate the effect of variation of varying the message frequency and modulation
index.
(a)

(b)
Figure 3.6 (a): Envelop detector circuit diagram (b) Circuit diagram for amplitude modulation with envelop
detector
MATLAB Exercise
Q1. Write the given code on MATLAB and verify the results of the performed experiment.
Draw its Simulink diagram as well.
% Define the time interval
ts=0.00001;
t= -0.1:ts:0.1;
% Define the functions m(t) and c(t)
m=cos(2*pi*10*t);
c=cos(2*pi*1000*t);
A=input('Enter the DC value');
% Perform the multiplication
g=(A+m).*c;
% for asynchronous demodulation
%rectification
y=abs(g);
% Create the filter
cutoff=500;
[a b]=butter(5,2*cutoff*ts);
% Get the output after the filter;
z=filter(a,b,y);
% Plot the input and output on the same graph
figure (1)
subplot(2,1,1)
plot(t,m)
title('Message Signal')
subplot(2,1,2)
plot(t,c)
title(' Carrier Signal')
figure (2)
subplot(3,1,1)
plot(t,g)

title('Modulated Signal')
subplot(3,1,2)
plot(t,y)
title('For demodulation')
subplot(3,1,3)
plot(t,(z-A/2))
title('Recovered Signal')
Q2. Use MATLAB to generate and display an AM wave for 100% modulation, under modulation
and over modulation.
Carrier frequency, fc =5kHz
Amplitude carrier frequency, Ac = 9
Performance Viva
Total/15
Performance /4
Results /3
Lab Report /3
Comments

LAB # 4: To Assemble A Modem For Single Side Band Amplitude
Modulation Using TIMS Trainer For Analog Communication Systems
Objectives
 To assemble a Modulator and demodulator using TIMS trainer for single side band
amplitude modulation.
 To sketch the SSB amplitude modulated signal using MATLAB for amplitude
modulation.
Equipment Required
Adder, Audio Oscillator, Multiplier, Phase Shifter, Quardrature Phase Splitter, Voltage
Controlled Oscillator
Lab Instructions
evaluation
 A student who will not bring manual in lab , will be awarded with zero in respective lab.

following items:
 Lab objectives
 Conclusion
 Take home assignment solved

Introduction
The method of sideband removal is to make two DSBSC signals, identical in all respects
except for their relative phasing. If this is suitably arranged the two DSBSC can be added,
whereupon the two upper sidebands (say) cancel, whilst the two lower add. An
arrangement for achieving this is illustrated in Figure 4.1
Figure 4.1: SSB Generation using the phasing method
The block labelled ‘QPS’ is a quadrature phase splitter. This produces two output signals, I and
Q, from a single input. These two are in phase quadrature. In the position shown in the diagram it
will be clear that this phase relationship must be maintained over the bandwidth of the message.
So it is a wideband phase splitter.
There is another phase shifter in the diagram, but this works at one frequency only - that of the
carrier. Wideband phase shifters (Hilbert transformers) are difficult to design. The phase splitter
is a compromise. Although it maintains a (relatively) constant phase difference of 90 between its
two outputs, there is a variable (with frequency) phase shift between both output and the
common input. This is acceptable for speech signals (speech quality and recognition are not
affected by phase errors) but not good for phase-sensitive data transmission.
Lab Tasks
Lab Task 1: SSB Modulation

Figure 4.2: The SSB phasing generator model
To align this generator it is a simple matter to observe first the ‘upper’ DSBSC (upper in the
sense of the ADDER inputs), and then the lower. Adjust each one separately (by removing the
appropriate patch lead from the ADDER input) to have the same output amplitudes (say 4 volt
peak-to-peak) Then replace both ADDER inputs, and watch the ADDER output as the PHASE
SHIFTER is adjusted. The desired output is a single sine wave, so adjust for a flat envelope. A
fine trim of one or other of the ADDER gain controls will probably be necessary.
The gain and phase adjustments are non-interactive. The magnitude of the remaining
envelope will indicate, and can be used analytically, to determine the ratio of wanted to
unwanted sideband in the output. This will not be infinite. The QPS, which cannot be adjusted,
will set the ultimate performance of the system. Which sideband has been produced? This can be
predicted analytically by measuring the relative phases of all signals. Alternatively, measure it!
Demonstrate your knowledge of the system by re-adjusting it to produce the opposite
sideband.
Lab Task 2: SSB Demodulation

An SSB signal can be demodulated with a product demodulator. But a product demodulator is
not an SSB demodulator in the strict sense. A true SSB demodulator can distinguish
between a lower sideband and an upper sideband. This experiment investigates the phasing
type demodulator, block diagrams of which are shown in Figure 4.1. It would be helpful,
though not essential, that the Lab Sheet entitled SSB generation has been completed.
A model of demodulation is given below:
Figure 4.3: Model of phasing-type SSB demodulator
An SSB received signal is required. If such a signal were derived from a single tone message,
and based on a 100 kHz (suppressed) carrier, it can be simulated by a single sine wave either
just above or just below 100 kHz. This can be obtained from a VCO.
After patching up the model it is necessary to align it. With an input signal (VCO) at, say,
102 kHz (simulating an upper sideband):
 Examine the waveforms throughout the model. Most will be un-familiar.

 Use the oscilloscope to set the phase shift through the PHASE SHIFTER to about 90.
 With only one input at a time into the ADDER, set its output to say 2 volt peak-to peak.
 Connect both inputs to the ADDER. Minimize the output from the LPF by alternately
Adjusting the PHASE SHIFTER and one ADDER gain control (why not maximize the ADDER
output in the above procedure?). The above procedure used an upper sideband for alignment. It is
now set to receive the lower sideband of a 100 kHz carrier. Verify this by tuning the VCO to the
region of the lower sideband. Alternatively, institute whatever change you think is necessary
to swap from one sideband reception to the other. Conversion of the summer from an

ADDER to a SUBTRACTOR would do it (insert a BUFFER AMPLIFIER, which acts as an
inverter, into one path to the ADDER); what other methods are there? Notice that by removing
one input from the ADDER you have a DSBSC receiver. Observe that it will still demodulate the
simulated SSB.
MATLAB Exercise
Write the given code on MATLAB and verify the results of the performed experiment.
Draw its simulink diagram as well.
ts=0.00001;
t= -0.1:ts:0.1;
% Define the functions m(t) and c(t)
m= 2*sin(2*pi*5*t);
c=2*sin(2*pi*300*t);
mh= 2*sin((2*pi*5*t)+(pi/2));
ch=2*sin((2*pi*300*t)+(pi/2));
md=m.*c;
modh=mh.*ch;
mod=md+modh;
subplot(5,1,1)
plot(t,m)
title('message signal');
subplot(5,1,2)
plot(t,mh)
title('hilbert message signal');
subplot(5,1,3)
plot(t,md)
title('modulated signal');
subplot(5,1,4)
plot(t,modh),title('modulated hilbert signal');
subplot(5,1,5)
plot(t,mod)
title('SSB modulated signal');
Performance Viva
Total/15
Performance /4
Results /3
Lab Report /3

LAB # 5: To Assemble A Modem For Quadrature Amplitude Modulation
Using TIMS Trainer For Analog Communication Systems
Objectives
 To assemble a Modulator and demodulator using TIMS trainer for Quadrature

 To sketch the Quadrature amplitude modulated signal using MATLAB for
Equipment Required
TIMS modules of Multiplier, Phase Shifter, Adder and Audio Oscillator
Lab Instructions
evaluation

following items:
 Lab objectives
 Conclusion
Introduction
There are two messages, A and B. whilst these are typically independent when they are
analog; it is common practice for them to be intimately related for the case of digital
messages. In the former case the modulator is often called a quadrature amplitude modulator
(QAM), whereas in the latter it is often called a quadrature phase shift keyed (QPSK)
modulator.
This Lab Sheet investigates an analog application of the modulator. The system is then
described as a pair of identical double sideband suppressed carrier (DSBSC) generators, with
their outputs added. Their common carriers come from the same source, but are in phase

quadrature. The two DSBSC are overlaid in frequency, but can be separated (by a suitable
receiver) because of this phase difference.
Note that the two paths into the ADDER are labelled ‘I’ and ‘Q’. This refers to the phasing of
the DSBSC in phase and quadrature.
Channel 2
. input
Figure 5.1: QAM modulator
Please complete the Lab Sheet entitled QAM - generation, which describes the generation of a
quadrature amplitude modulated signal with two, independent, analog messages. That generator
is required for this experiment, as it provides an input to a QAM demodulator. A QAM
demodulator is depicted in block diagram form in Figure 5.1.In this experiment only the
principle of separately recovering either message A or message B from the QAM is
demonstrated. Only one half of the demodulator need be constructed.
Figure 5.2: QAM demodulator
Such a simplified demodulator is shown in the block diagram of Figure 5.2. This is the
Structure you will be modelling. By appropriate adjustment of the phase either message A or
Message B can be recovered.
Lab Tasks
Lab Task 1: QAM Modulation
Figure 5.3 shows a model of the block diagram of a QAM modulator, shown in Figure 5.1.

Sin µt
Figure 5.3: QAM Generation Block Diagram
The 100 kHz quadrature carriers come from the MASTER SIGNALS module. Note that these do
not need to be in precise quadrature relationship; errors of a few degrees make negligible
difference to the performance of the system as a whole - transmitter, channel, and receiver. It is
at the demodulator that precision is required - here it is necessary that the local carriers
match exactly the phase difference at the transmitter.
The two independent analog messages come from an AUDIO OSCILLATOR and the
MASTER SIGNALS module (2 kHz).
Setting up is simple. Choose a frequency in the range say 300 to 3000 Hz for the AUDIO
OSCILLATOR (message ‘A’).Confirm there are DSBSC at the output of each MULTIPLIER.
Adjust their amplitudes to be
Equal at the output of the ADDER, by using the ADDER gain controls (remove the ‘A’ input
when adjusting ‘g’, and the ‘B’ input when adjusting ‘G’). Since the QAM signal will (in
later experiments) be the input to an analog channel, its amplitude should be at about the
TIMS ANALOG REFERENCE LEVEL of 4 volt peak-to peak.
What is the relationship between the peak amplitude of each DSBSC at the ADDER output, and
their sum?
To what should the oscilloscope be triggered when examining the QAM? Is the QAM of a
‘recognizable’ shape? For the case when each message could lie anywhere in the range 300 to
3000 Hz, what bandwidth would be required for the transmission of the QAM?
Lab Task 2: QAM Demodulation

Set up the transmitter according to the plan adopted in the Lab Sheet entitled QAM -
generation. Synchronize the oscilloscope to, and observe, say, the ‘A’ message, on CH1 -A. A
model of the block diagram of Figure 4.2, which is a demodulator, or receiver, is shown in
Figure 4.4.

Figure 4.4: Channel A or B demodul ator
The 100 kHz carrier (sinωt or cosωt) comes from MASTER SIGNALS. This is a ‘stolen’
carrier. In commercial practice the carrier information must be derived directly from the
received signal. Remember to set the on-board switch SW1 of the PHASE SHIFTER to the HI
range.
The 3 kHz LPF in the HEADPHONE AMPLIFIER can be used if the messages are restricted to
this bandwidth. Observe the output from this filter with the oscilloscope on CH2-A. Since
message A is already displayed on CH1-A, an immediate comparison can be made. Probably
both messages will be appearing at the filter output, although of different amplitudes. Being on
different frequencies the display will not be stationary.
Now slowly rotate the coarse control of the PHASE SHIFTER. The output waveform should
slowly approach the shape of message A (if not, flip the 180 front panel toggle switch). Note that
the phase adjustment is not used to maximize the amplitude of the wanted message but to
minimize the amplitude of the unwanted message. When this minimum is achieved then what
remains, by default, is the wanted message. Provided the phasing at the transmitter is anywhere
near quadrature there should always be a useful level of the wanted message. The magnitude of
the wanted waveform will be the maximum possible only when true quadrature phasing is
achieved at the transmitter. An error of 450 at the transmitter, after accurate adjustment at the
receiver, results in a degradation of 3 dB over what might have been achieved. This is a
signal-to -noise ratio degradation; the noise level is not affected by the carrier phasing.
Phase Vision Multiplex

The arrangement just examined has been called phase division multiplex - there are two
channels sharing the same frequency space. Separation - demultiplexing - is by virtue of their
special phase relationships.

To enable carrier acquisition from the received signal there needs to be a small ‘pilot’ carrier,
typically about 20 dB below the signal itself. A filter is used to separate this from the message
sidebands. TIMS can easily demonstrate such a system by using a phase locked loop (PLL) as
the filtering element.
MATLAB Exercise
Write the MATLAB code and verify the results of the performed experiment. Draw its Simulink
diagram as well.
Performance Viva
Total/15
Performance /4
Results /3
Lab Report /3
Comments

Lab #06: To Assemble Phase Locked Loop (PLL) using TIMS
Trainer for Phase detection in Analog Communication Systems .
Objectives
 To construct and assemble phase lock loop using discreet components and TIMS trainer
for phase detection in analog communication systems.
Lab Instructions
evaluation
 A student who will not bring manual in lab, will be awarded with zero in respective lab.

following items:
 Lab objectives
 Conclusion
Background Theory
Phase-Locked Loop Phase-Locked Loop is a device which is used to track the phase and
frequency of an incoming signal. It uses a voltage-controlled oscillator (VCO),the output of
which can be automatically synchronized (”locked”) to a periodic input signal. The locking
property of the PLL has numerous applications in communication systems (such as frequency,
amplitude, or phase modulation/demodulation, analog or digital),clock and data recovery ,self-
tunable filters, frequency synthesis etc.
Following figure represents the block diagram of PLL showing its basic function connected
together in a feedback loop.

Figure 6.1: Block diagram of a Phase-Locked(PLL)
 Voltage-Controlled oscillator(VCO)
 Phase detector(PD)
 Low-pass loop filter(LPF)
VCO is an oscillator of the frequency of which fosc is proportional to input voltage Vo .The
input voltage to VCO determines the frequency fosc of the periodic signal Vosc at the output of
the VCO. //Phase comparator is device that compares the phase of the output signal of VCO and
the incoming signal and produces a signal proportional to the phase difference between the
incoming signal and the VCO output signal. The output of the phase detector is filtered by a low-
pass loop filter. The loop is closed by connecting the filter output to the input of the VCO. When
the loop is locked on the incoming signal Vi ,the frequency of the VCO output foscis exactly
equal to the frequency fi of the periodic signal Vi
fosc = fi
The basic function of PLL is to maintain the frequency lock(fosc=fi) between the input and the
output signals even if the frequency fi of the incoming signal varies with time. Assuming that the
PLL is in the locked condition and then if the frequency fi of the incoming signal increases
slightly, the phase difference between the VCO signal and the incoming signal will begin to
increase in time. As a result, the filter output voltage Vo increases, and the VCO output
frequency fosc increases until it matches fi ,thus keeping the PLL in the locked condition. //The
range of frequencies from fi=fmin to fi=fmax where the locked PLL remains in the locked
condition is called the lock range the PLL.If the PLL is initially locked, and fi becomes smaller
than the fmin, or if fi exceeds fmax, the PLL fails to keep fosc equal to fi ,and the PLL becomes
unlocked ,i.e. fosc !=fi .When the PLL is unlocked ,the VCO oscillates at the frequency fo called
the subtitle center frequency ,or the free-running frequency of the VCO .The lock can be
established again if the incoming singal frequency fi gets closed enough to fo. The range of
frequencies fi=fo-fc to fi=fo+fc such that the initially unlocked PLL becomes locked is called the
capture range of the PLL. The lock range is wider than the capture range. So, if the VCO output
frequency fosc is plotted against the incoming frequency fi , we obtain the PLL steady state
characteristics shown in Fig 6.2. The characteristics simply shows that fosc=fi in the locked
condition, and that fosc=fo=constant when the PLL is unlocked. A hysteresis can be observed in
the fosc(fi) characteristic because the capture range is smaller than the lock range.
Figure 6.2: steady-state characteristics of the basic PLL
The 4046 Phase-Locked Loop IC
A diagram of the 4046 PLL is shown in Fig 6.3. A single positive supply voltage is needed for
the chip .The positive supply voltage VDD is connected to pin 16 and the ground is connected to
pin 8.In the lab we will use +VDD=+15V.The incoming signal Vi goes to the input of an internal
amplifier at the pin 14 of the chip. The internal amplifier has the input biased at about
+VDD/2.Therefore ,the incoming signal can be capacitive coupled to the input as shown in Fig
6.3.The incoming ac signal Vi of about one volt peak-to-peak is sufficient for proper operation.
The output Vi2 of the amplifier is internally connected to one of the inputs of the phase detector
on the chip.
Phase Detector
The phase detector on the 4046 is simply an XOR logic gate is shown in Fig, 12.4, where, with
logic low output (Vφ=0V) when the two inputs are both high and low and the logic high output
Vφ=VDD) otherwise. Following figure shows the operation of the XOR phase detector when the
PLL is in the locked condition. Vi2 and Vosc are two phase-shifted periodic square-wave signals
at the same frequency fosc = fi and with 50 percent duty cycle .The output of the phase detector
is a periodic square wave signal Vφ(t) at the frequency 2fi ,and with the duty ratio Dφ that
depends on the phase difference between Vi and Vosc.
VDD = φ π (6.1)

Figure 6.3: CMOS 4046 PLL
Figure 6.4: Operation of the phase detector with XOR gate
The periodic signal Vφ(t) at the output of the XOR phase detector can be written
as the Fourier series:
Vφ(t) = Vo + X k=1 Vk sin((4kπfi)t − θk)
where Vo is the dc component of Vφ(t),and Vk is the amplitude of the kth harmonic at the
frequency 2kfi .The dc component of the phase detector output can be found easily as the
average of Vφ(t)over a period TI=2
Vo = VDDΦ π

Loop filter
The output Vφ(t) of the phase detector is filtered by an external low-pass filter. In Fig12.3,the
loop filter is a simple passive RC filter. The purpose of the low-pass filter is to pass the dc and
low-frequency portions of Vφ(t) and to attenuate high-frequency ac component at frequencies
2kfi .The simple RC filter has the cut-off frequency:
fp = 1 2πRC (6.4)
The cut-off frequency should be smaller than the input frequency for the output of the filter to be
approximately equal to Vo. Vo is proportional to the phase difference between the incoming
signal Vi and the signal Vosc from the VCO and the constant of proportionality,
KD = VDD pi (6.5)
is called the gain or the sensivity of the phase detector .This expression is valid for 0≤ φ ≤ π .The
filter output VO as a function of the phase difference φ is shown in Fig 6.5.Note that Vo if Vi
and Vosc are in phase (φ=0),and that it reaches the maximum value Vo=VDD when the two
signals are exactly out of phase(φ=π).From Fig 6.4 it is easy to see that for φ π,V0 decreases. Of
course, the characteristic is periodic in φ with period 2π.The range 0 ≤ φ ≤ π is the range where
the PLL can operate in the locked condition.
Figure 12.5: Characteristics of the phase detector
Voltage Controlled Oscillator
The voltage VO controls the charging and discharging currents through an external capacitor C1,
and therefore determines the time needed to charge (discharge) the capacitor to a predetermined
threshold level. As a result, the frequency fosc of the VCO output is determined by VO. The
VCO output Vosc is a square wave with 50percent duty cycle and frequency fosc. As shown in
Fig 6.3, the VCO characteristics are user-adjustable by three external components:R1 ,R2 and
C1.When Vo is zero ,VCO operates at the minimum frequency fmin and when V0=VDD, the
VCO operates at the maximum frequency fmax. The actual operating frequencies can differ
significantly from the values predicted by the above equations. So, one may need to tune the
component values by experiment. For fosc between the minimum fmin and the maximum fmax,
the VCO output frequency fosc is ideally a linear function of the control voltage Vo. The slope
Ko = 4fosc 4VO
is called the gain or the frequency sensitivity of the VCO,in HZ V.
Lock Range and Capture Range
Once the PLL is in the locked condition ,it remains locked as long as the VCO output frequency
fosc can be adjusted to match the incoming signal frequency fi ,i.e., as long as fmin ≤ fi ≤ fmax.
When the lock is lost, the VCO operates at the free-running frequency fo, which is between the
fmin and fmax.To establish the lock gain , i.e. to capture the incoming signal again, the incoming
signal frequency fi must be close enough to fo. Here ‘close enough’ means that fi must be in the
range from fo-fc to fo+fc, where 2fc is called the capture range. The capture range 2fc is smaller
than the lock range fmax-fmin as shown in Fig.6.2 .The capture range 2f − c depends on the
characteristics of the loop filter. For the simple RC filter, a very crude, approximate implicit
expression for the capture range can be found as:
fc ≈ VDD 2 Ko q 1 + ( fc fp ) 2 (6.7)
where, fp is the cut-off frequency of the filter, VDD is the supply voltage and KO is the VCO
gain. Given ko and fp this relation can be solved for fc numerically which yields an approximate
theoretical prediction for the capture range 2fc. If the capture range is much larger than the cut-
off frequency of the filter, fc fp >>1,then the expression for the capture range is simplified.
2fc ≈ q 2KOfpVDD (6.8)
Note that the capture range 2fc is smaller if the cut-off frequency fp of the filter is lower. It is
usually desirable to have a wider capture range, which can be accomplished by increasing the
cut-off frequency of the filter. On the other hand a lower cut-off fp is desirable in order to better
attenuate high frequency components of vφ at the phase detector output and improve noise
rejection in general.
In Lab
 Set the values of C1=0.03uF,R1=R2=18K Ω.
 Find out the fmin and fmax of the VCO. To find fmin, simply connect the VCO input
(pin 9) to ground and to find fmax connect pin 9 to VDD.
 Find out the free-running frequency fo of the VCO. This is the frequency of the output
signal when input is not applied to phase detector.

 Apply an incoming signal Vi from the signal generator. Adjust its frequency to
approximately match the free-running frequency f0 of the VCO. When Vi is applied, the
PLL should operate in the locked condition, with fo exactly equal to fi .The locked
condition can be easily verified by observing Vi and Vosc simultaneously on a dual-trace
oscilloscope. If fi=fosc, stable waveforms of both Vi and Vosc can be observed.
Otherwise, one of the waveforms on the scope screen is blurred or is moving with respect
to the other.
 By changing the frequency of the incoming signal, determine the actual lock range of the
PLL i.e determine the maximum and the minimum frequency fi such that starting from
the locked condition the PLL remains in the locked condition. The lock range should be
equal to fmax-fmin.
 Record the readings for the lock range and the capture range below.
fmin = ............
fmax = ............
fcap1=fo-fc = ............
fcap2=f0+fc = ............
Lock range=fmax- fmin = ..............
Capture range=fcap2-fcap1 = ............
Performance Viva
Total/15
Performance /4
Results /3
Lab Report /3
Comments

Lab # 07: To Construct Frequency Modulator Using Discrete
Components for Analog Communication Systems
Objectives
 To construct PLL based frequency modulator using discreet components for analog
communication systems.
Lab Instructions
evaluation

following items:
 Lab objectives
 Conclusion
In Lab
Build the circuit shown in Fig. 7.1. This uses the VCO portion of the 4046 PLL.

Figure 7.1: PLL (CD 4046 BC)
 First, investigate it using the “test input” circuit that is shown in Figure 7.2. Find the
frequency and sketch the waveform for the three VCO input voltages shown in the table
below. From that information, determine the FM constant , Kf, for your modulator. See
the data analysis section below for guidance in this calculation.
 Second, instead of the ”test input” circuit, use, as the input, the function generator with
the sinusoidal output listed as follows:
o Frequency = 5 kHz (fm = modulating frequency)
o Amplitude = 2 volts (p-p)
o D.C.Offset = 5V
Figure 7.2: Test Input Circuit
 Examine the time-domain signal at the VCO output. It should look similar to the plot of
Figure 7.3. Essentially, this is a rectangular waveform with a varying frequency , i.e., a
frequency that is modulated. The maximum and minimum frequencies, fmax and fmin,
can be determined using the following formulas:
fmin = 1 T1
fmax = 1 T2
 Write an expression for the time domain output, assuming that the output waveform is
sinusoidal like. What is the β for your signal? Examine the spectra using the spectrum
analyzer. (Make the Connection to the spectrum analyzer using a high impedance scope
probe). Sketch the spectra and measure the power in significant sidebands (powers
greater than one percent of the total transmitted power). Record this data in the table
shown in the ‘report’ section.
Figure 7.3: Time-domain signal at the VCO output
Data Analysis
The FM constant, Kf , can be determined by plotting the VCO’s output frequency v/s the VCO’s
input voltage. This should give (approximately) a straight line, its slope is Kf in hertz per-volt.
You will want to convert it to radians/second-per-volt in order to write the expression for the FM
signal you generate. To find β , use
β = [peak modulating tone amplitude/modulating tone frequency (in Hz)] kf .
An alternative method is given by
β = fmax − fmin fm

Performance Viva
Total/15
Performance /4
Results /3
Lab Report /3
Comments

LAB # 8: To Display Sampling of Analog Signals for Digitization
Using Discrete Components And MATLAB
Objectives
After completing this lab, you should be able to:
 Trace the Nyquist sampled signal for understanding of sampling theorem.
 Display the recovered message signal using sampled values in MATLAB.
 Construct a sampler circuit using discreet components for digitization of analog signals.
Pre-Lab Exercise
Read this experiment in its entirety to become familiar with objectives of this lab. Study in detail
and become familiar with the basics of sampling theorem provided with this laboratory
experiment and in chapter 6 of the reference book. The instructor may provide the class some
time to reflect upon these before proceeding with the lab. Furthermore, you need to complete the
following tasks on MATLAB and also attach codes with detailed annotations on them:
 Generate a band-limited random signal in MATLAB

 Plot the signal in time and frequency domain
Background
Analog signals, which are the most familiar type of signal, are continuous functions of time in
the sense that their amplitudes are defined explicitly for every instant of time. However, there is
another important class of signals, usually referred to as sampled signals, for which the
amplitude is defined (non-zero) only for a certain discrete instant of time. Figure 2.1 displays an
example of both the analog and a sampled signal. Sampled signals are used in pulse-modulation
communication systems, in sampled data control systems, and when digital computers are used
as part of an analog system.

(a)
(b)
Figure 8.1: Examples of (a) an analog signal; (b) a sampled signal
The process of generating sampled signals, sometimes called pulse-amplitude modulationis

illustrated in Fig. 4.2. The analog input signal, , is multiplied by the uniform pulse
train, , and the resulting output signal, , is non-zero only when and are both
non-zero. The analog signal, , is said to have been sampled by the sampling signal, .
(b)
(a)
(c)
(d)
Figure 8.2: Generation of sampled signals
Sampled signals such as in Figure 2.2(d) are useful only if they contain the same
information as the original signal, , as shown in Figure 2.2(b). That is to say, must be
recoverable from . The conditions under which such a recovery of the original signal
occurs, constitute a statement of the sampling theorem. Briefly these conditions are:
 The original signal must be a band-width limited function (i.e., have no frequency
components outside the frequency interval , and

 The frequency of the sampling signal, , must be greater than .
In-Lab Exercise
Task 1 (Verification of Sampling Theorem using TIMS Trainer)
The sampling process is illustrated by a switch that can be operated with the help of pulse train
as shown in Figure 2.3. The input to this switch is a single audio tone. To model the arrangement
of Figure 2.3 with TIMS the modules required are a TWIN PULSE GENERATOR(only one
pulse is used), and a DUAL ANALOG SWITCH (only one of the switches is used). Note that,
for the sampling method being examined, the shape of the top of each sample is the same as that
ofthe message. This is often called natural sampling.
Figure 8.3: Sampling a Sine wave
With the help of above given arrangement and TIMS modules draw a circuit diagram (with
proper labeling) that satisfied the sampling theorem. Also implement the designed circuit on
TIMS trainer.
Circuit diagram:

Having generated a train of samples, now observe that it is possible to recover, or reconstruct (or
interpolate) the message from these samples. Now with the help of Fourier series analysis, and
consideration of the nature of the sampled signal, you can already conclude that the spectrum of
the sampled signal will contain components at and around harmonics of the switching signal, and
hopefully the message itself. If this is so, then a low pass filter from the HEADPHONE
AMPLIFIER would seem the obvious choice to extract the message. The reconstruction circuitry
is illustrated in Figure 2.4.
Figure 8.4: Reconstruction circuit
You can confirm that it recovers the message from the samples by connecting the output of the
DUALANALOG SWITCH to the input of the LPF in the HEADPHONE AMPLIFIER
module, and displaying the output on the oscilloscope.
Task 2 (Design and Implementation of Flat top sampling circuit using discrete
components)
During transmission, noise is introduced at top of the transmission pulse which can be easily
removed if the pulse is in the form of flat top. Here, the top of the samples are flat i.e. they have
constant amplitude. Hence, it is called as flat top sampling or practical sampling. Flat top
sampling makes use of sample and hold circuit.
The sample-and-hold operation is simple to implement, and is a very commonly used method of
sampling in communications systems. In its simplest form the sample is held until the next
sample is taken. So it is of maximum width. This is illustrated in Figure 2.5 below.

Figure 8.5: Sample and hold operation
In the above example the sampling instant is coincident with the rising edge of the clock
signal.In practice there may be a ‘processing delay’ before the stepped waveform is presented at
the output.
In this very task you need to design a circuit diagram of flat top sampling using discrete
components. Also design a low pass filter to reconstruct the message signal from its sampled
output. After designing the circuit, implement it on the bread board to test its validity. The circuit
must be designed with following parameters:
(sine wave)
(duty cycle i.e. pulse wave)
Circuit Diagram:
Task 3 (MATLAB)
Use MATLAB to see the effects of sampling. Generate a baseband signal. This signal is sampled
every sec. Simulate sampling with an impulse train by sampling every sec.

Post Lab
Use MATLAB to see the effects of sampling. Generate a baseband signal. This signal is sampled
every sec. Simulate sampling with an impulse train by sampling every sec. Let represent
the baseband signal, the sampled signal and represent the length of .
Hint: ;
After generating the sampled signal , attempt reconstruction of with a lowpass filter. Let
the reconstructed signal be .
Hint:
Plot , the sampled signal, and the reconstructed signal in time and frequency domain. Repeat
by sampling different rates including a case where aliasing occurs. Discuss the results.

Results
Attach results of the Lab Tasks.

Record the terms and sections that require further elaboration
Performance Viva
Total/15
Performance /4
Results /3
Lab Report /3
Comments

LAB # 9: To Display Various Line Coding Schemes For Pulse Code
Modulation Using TIMS Trainer.
Objectives
 To assemble the circuit for implementation of various line codes e.g. NRZ, RZ,
Manchester etc using TIMS trainer.
 To display line coded waveform for PCM using trainer.
Pre-Lab Exercise
and become familiar with the basics of line coding provided with this laboratory experiment and
in chapter 7 of the reference book. The instructor may provide the class some time to reflect
upon these before proceeding with the lab.
Plot the following bit sequence in MATLAB as a sequence of perfect square waves.
Background
The process for converting digital data into digital signal is said to be Line Coding. Digital data
is found in binary format. It is represented (stored) internally as series of 1s and 0s. Digital signal
is denoted by discreet signal, which represents digital data.There are three types of line coding
schemes available as shown in Figure 9.1:
Figure 9.6: Types of Line Coding Schemes
Uni-polar Encoding
Unipolar encoding schemes use single voltage level to represent data. In this case, to represent
binary 1, high voltage is transmitted and to represent 0, no voltage is transmitted. It is also
called Unipolar-Non-return-to-zero, because there is no rest condition i.e. it either represents 1
or 0.
Figure 9.7: Uni-polar Signal
Polar Encoding
Polar encoding scheme uses multiple voltage levels to represent binary values. Polar encoding
techniques are of four types:
 Polar Non-Return to Zero (Polar NRZ)
It uses two different voltage levels to represent binary values. Generally, positive
voltage represents 1 and negative value represents 0. It is also NRZ because there is no
rest condition.NRZ scheme has two variants: NRZ-L and NRZ-I.
Figure 9.8: Polar NRZ Signal

NRZ-L changes voltage level at when a different bit is encountered whereas NRZ-I
changes voltage when a 1 is encountered.
 Return to Zero (RZ)
Problem with NRZ is that the receiver cannot conclude when a bit ended and when the
next bit is started, in case when sender and receiver’s clock are not synchronized.
Figure 9.9: Polar Return zero Signal
RZ uses three voltage levels, positive voltage to represent 1, negative voltage to

represent 0 and zero voltage for none. Signals change during bits not between bits.
 Manchester
This encoding scheme is a combination of RZ and NRZ-L. Bit time is divided into two
halves. It transits in the middle of the bit and changes phase when a different bit is
encountered.
Bipolar Encoding
Bipolar encoding uses three voltage levels, positive, negative and zero. Zero voltage represents
binary 0 and bit 1 is represented by altering positive and negative voltages.
Figure 9.10: Bipolar Signal

Advantages of Line Coding
There are many reasons for using line coding. Each of the line codes you will be examining
offers one or more of the following advantages:
 Spectrum shaping and relocation without modulation or filtering. This is important in

telephone line applications, for example, where the transfer characteristic has heavy
attenuation below .
 Bit clock recovery can be simplified.

 DC component can be eliminated; this allows AC (capacitor or transformer) coupling
between stages (as in telephone lines). Can control baseline wander (baseline wander
shifts the position of the signal waveform relative to the detector threshold and leads to
severe erosion of noise margin).
 Error detection capabilities.
 Bandwidth usage; the possibility of transmitting at a higher rate than other schemes over
the same bandwidth.
In-Lab Exercise
Task 1 (TIMS Trainer)
Design a TIMS circuit diagram using a Line encoder module that will encode a binary data into
different line coding schemes. Also, design its receiver that will decode the line coded signal to
provide original binary data. Moreover, implement the designed circuit on TIMS trainer.
Circuit diagram:

In TIMS, the LINE-CODE ENCODER accepts a TTL input, and the output is suitable for
transmission via an analog channel. As signal passes through the communication channel it could
be corrupted due to channel’s impact or noise. Here, it is re-generated by a detector. The TIMS
detector is the DECISION MAKER module. Finally, the TIMS LINE-CODE DECODER
module accepts the output from the DECISION MAKER and decodes it back to the binary TTL
format.
The LINE-CODE ENCODER serves as a source of the system bit clock. It is driven by a master
clock at (from the TIMS MASTER SIGNALS module). It divides this by a factor of
four, in order to derive some necessary internal timing signals at a rate of . This then
becomes a convenient source of a TTL signal for use as the system bit clock.
Because the LINE-CODE DECODER has some processing to do, it introduces a time delay. To
allow for this, it provides a re-timed clock if required by any further digital processing circuits
(e.g., for decoding, or error counting modules).
Post-Lab
Write a MATLAB program to line-encode your complete registration number using uni-polar
and polar NRZ Coding.
Results
Attach results of Lab tasks.

Performance Viva
Total/15
Performance /4
Results /3
Lab Report /3
Comments

LAB # 10: To Assemble Encoder for Pulse Code Modulation
(PCM) Using TIMS Trainer
Objectives
 To assemble various block i.e. Sampler, Quantizer, Encoder for Pulse Code Modulation
signal using TIMS trainer.
Pre-Lab Exercise
and become familiar with the basics of pulse code modulation provided with this laboratory
time to reflect upon these before proceeding with the lab. Furthermore, explore the following
MATLAB functions and write a short summary about their usage:
 ;
Background
Information in an analog form cannot be processed by digital computers so it's necessary to
convert them into digital form. After converted to digital signal, it is easy for us to process the
signal such as encoding, filtering the unwanted signal and so on. PCM is a term which was
formed during the development of digital audio transmission standards. Digital data can be
transported robustly over long distances unlike the analog data and can be interleaved with other
digital data so various combinations of transmission channels can be used.
PCM doesn`t mean any specific kind of compression, it only implies PAM (pulse amplitude
modulation) - quantization by amplitude and quantization by time which means digitalization of
the analog signal. The range of values which the signal can achieve (quantization) is divided into
segments, each segment has a segment representative of the quantization level which lies in the
middle of the segment.
The value that a signal has in certain time is called a sample; the process of taking samples is
called quantization by time. After quantization by time, it is necessary to conduct quantization by

amplitude. Quantization by amplitude means that according to the amplitude of sample one
quantization segment is chosen (every quantization segment contains an interval of amplitudes).
PCM modulation is commonly used in audio and telephone transmission. The main advantage is
the PCM modulation only needs sampling frequency to maintain the original quality of
audio. Figure 4.1is the block diagram of PCM modulation. First of all a low pass filter is used
that removes the noise in the audio signal. After that the audio signal will be sampled to obtain a
series of sampling values. Next, the signal will pass through a quantizer that defines the levels.
Then the signal will pass through an encoder to encode the quantization values and then convert
to digital signal. In fact, the process of quantization can be achieved at one time by A/D
converter. However, we should pay attention on the quantization levels. For example, if the bits
for PCM modulation is 3, then the quantization levels is , which is 8 steps. If the bits for
PCM are 4, then the quantization levels is , which is 16 steps. The increasing of bits of
PCM modulation will prevent the signal from distortion, but the bandwidth will also increase due
to the increasing of the capacity of data. An encoder utilizes ‘ ’ output terminals; therefore, we
need to convert the parallel data to serial data, which is the way that satisfy the data format of
PCM modulation.
Figure 10.11:Block diagram of PCM Encoder
Modulation process is executed in three steps, 1) Sampling, 2) Quantization, and 3) Encoding.

The details of each process can be seen in chapter 6 of the reference book.
In-Lab Exercise
Design a TIMS circuit diagram using a PCM encoder module that will generate a PCM encoded
signal for a sinusoidal input signal with frequency. Also, implement the designed circuit
on TIMS trainer.
The input to the PCM ENCODER module is an analog message. This must be constrained to a
defined bandwidth and amplitude range. The maximum allowable message bandwidth will

depend upon the sampling rate to be used. The Nyquist criterion must be observed. The
amplitude range must be held within the volts range of the TIMS ANALOG REFERENCE
LEVEL. This is in keeping with the input amplitude limits set for all analog modules.
Circuit diagram:
A brief description of the operation of the module follows:
 The module is driven by an external TTL clock.

 The input analog message is sampled periodically. The sample rate is determined by the
external clock.
 The sampling is a sample-and-hold operation. It is internal to the module, and cannot be
viewed by the user. What is held is the amplitude of the analog message at the sampling
instant. The amplitude of each sample is compared with a finite set of amplitude levels.
These are distributed (uniformly, for linear sampling) within the range ± 2.0 volts (the
TIMS ANALOG REFERENCE LEVEL). These are the system quantizing levels.
 Each quantizing level is assigned a number, starting from zero for the lowest (most
negative) level, with the highest number being , where is the available number
of levels.
 Each sample is assigned a digital (binary) code word representing the number associated
with the quantizing level which is closest to the sample amplitude. The number of bits ‘ ’
in the digital codeword will depend upon the number of quantizing levels. In fact,
.

 The code word is assembled into a time frame together with other bits as may be required
(described below). In the TIMS PCM ENCODER (and many commercial systems) a
single extra bit is added, in the least significant bit position. This is alternately a one or a
zero. These bits are used by subsequent decoders for frame synchronization.
 The frames are transmitted serially. They are transmitted at the same rate as the samples
are taken. The serial bit stream appears at the output of the module.
 Also available from the module is a synchronizing signal FS (‘frame synch’). This signals
the end of each data frame.
Task 2 (MATLAB)
Use MATLAB to observe the output of Pulse code modulation. This will constitute of two parts:
 Sampling an analog signal.
 Uniformly quantizing the samples.
 Encoding the quantizer output to digital pulses.
Plot the sampled, quantized and the encoded signal. Repeat by changing the quantization levels.
Also discuss the results.
Post Lab
Determine the mean squared error for a uniform quantizer with 12 quantization levels, each of
length 1, designed for a zero-mean Gaussian source with variance 4. It is assumed that the
quantization regions are symmetric with respect to the mean of the distribution.
Results
Attach results of the Lab Tasks.

Performance Viva
Total/15
Performance /4
Results /3
Lab Report /3
Comments

LAB # 11: To assemble Decoder for Pulse Code Modulation (PCM)
Using TIMS trainer
Objectives
 To assemble a decoder using various blocks of TIMS trainer to reconstruct message

signal in Pulse Code Modulation.
Pre-Lab Exercise
and become familiar with the basics of pulse code demodulation provided with this laboratory
time to reflect upon these before proceeding with the lab.
Generate a sinusoidal signal with amplitude 1 and . Using a uniform PCM scheme,
quantize it once to 8 levels and once to 16 levels. Plot the original signal and the quantized signal
on the same axes. Compare the resulting SQNRs in the two cases.
Background
The signal to be decoded in this experiment will be provided by you, using the PCM ENCODER
module as set up in previous experiment. A clock synchronization signal will be stolen from the
encoder. In the PCM DECODER module there is circuitry which automatically identifies the
location of each frame in the serial data stream. To do this it collects groups of eight data bits and
looks for the repeating pattern of alternate ones and zeros placed there (embedded) by the PCM
ENCODER in the LSB position.
It can be shown that such a pattern cannot occur elsewhere in the data stream provided that the
original bandlimited analog signal is sampled at or below the Nyquist rate. When the embedded
pattern is found an ‘end of frame’ synchronization signal FS is generated, and made available at
the front panel. The search for the frame is continuously updated.
The PCM DECODER module is driven by an external clock. This clock signal is synchronized
to that of the transmitter. For this experiment a ‘stolen’ clock will be used.
Upon reception, the PCM DECODER:

 Extracts a frame synchronization signal FS from the data itself (from the embedded
alternate ones and zeros in the LSB position), or uses an FS signal stolen from the
transmitter.
 Extracts the binary number, which is the coded (and quantized) amplitude of the sample
from which it was derived, from the frame.
 Identifies the quantization level which this number represents.
 Generates a voltage proportional to this amplitude level.
 Presents this voltage to the output VOUT . The voltage appears at VOUT for the duration of
the frame under examination.
 Message reconstruction can be achieved, albeit with some distortion, by low pass
filtering. A built-in reconstruction filter is provided in the module.
In-Lab Exercise
Design a TIMS circuit diagram using a PCM decoder module that will decode a PCM encoded
signal for a sinusoidal input signal with frequency. Also, implement the designed circuit
on TIMS trainer.
Circuit diagram:

Post-Lab
Sample and encode a sinusoidal message signal of frequency using a PCM encoder and
reconstruct the original message using PCM decoder in MATLAB.
Results
Attach results of Lab tasks.

Performance Viva
Total/15
Performance /4
Results /3
Lab Report /3
Comments

Lab # 12: To Construct a Modem Based on Amplitude Shift
Keying Using Discrete Components for Digital Communication
Systems
Objectives
 To construct a modulator and demodulator for Amplitude Shift Keying using discreet
components.
Pre-Lab Exercise
and become familiar with the fundamentals of ASK modulation provided with this laboratory
Background
Amplitude shift keying (ASK) in the context of digital communications is a modulation process
which imparts to a sinusoid two or more discrete amplitude levels. These are related to the
number of levels adopted by the digital message. For a binary message sequence there are two
levels, one of which is typically zero. Thus the modulated waveform consists of bursts of a
sinusoid. Figure 12.1 illustrates a binary ASK signal, together with the binary sequence which
initiated it. Neither signal has been band limited.
Figure 12.12: An ASK signal and the message
There are sharp discontinuities shown at the transition points. These result in the signal having an
unnecessarily wide bandwidth. Band limiting is generally introduced before transmission, in
which case these discontinuities would be ‘rounded off’. The band limiting may be applied to the

digital message, or the modulated signal itself. The data rate is often made a sub-multiple of the
carrier frequency.
One of the disadvantages of ASK is that it has not got a constant envelope. This makes its
processing (e.g., power amplification) more difficult, since linearity becomes an important
factor. However, it does make for ease of demodulation with an envelope detector. A block
diagram of a basic ASK generator is shown in Figure 8.2, where switch is opened and closed by
unipolar binary sequence.
Figure 12.13: Block Diagram of ASK
Bandwidth Modification: As already indicated, the sharp discontinuities in the ASK

waveform of Figure 12.1 imply a wide bandwidth. A significant reduction can be accepted
before errors at the receiver increase unacceptably. This can be brought about by band limiting
(pulse shaping) the message before modulation, or band limiting the ASK signal itself after
generation. Both these options are illustrated in Figure 12.3, which shows one of the generators
you will be modelling in this experiment.
Figure 8.14: ASK band-limiting

Demodulation methods: It is apparent from Figure 12.1 that the ASK signal has a well-
defined envelope. Thus it is amenable to demodulation by an envelope detector. A synchronous
demodulator would also be appropriate. Note that:
 Envelope detection circuitry is simple.

 Synchronous demodulation requires a phase-locked local carrier and therefore carrier
acquisition circuitry.
With band limiting of the transmitted ASK neither of these demodulation methods would recover
the original binary sequence; instead, their outputs would be a band limited version. Thus further
processing -by some sort of decision-making circuitry for example -would be necessary. Thus
demodulation is a two-stage process:
 Recovery of the band limited bit stream

 Regeneration of the binary bit stream
In-Lab Exercise
Design a TIMS circuit diagram of ASK modulator that employs an appropriate sinusoidal carrier
to modulate the baseband signal. Also, design its receiver with the help of envelope detector to
recover the baseband signal. Moreover, implement the designed circuit on TIMS trainer.
Circuit diagram:

Task 2 (Circuit design using discrete components)
Design a circuit diagram for ASK modulator using discrete components available in lab. Use
baseband signal and sinusoidal carrier of frequency and from the signal
generator respectively. Also employ envelope detector at the receiver to recover the baseband
message.
Circuit diagram:
Task 3 (MATLAB)
Use MATLAB to apply the ASK modulation to a bit sequence of 1 1 1 0 0 0.
Post-Lab
Use MATLAB to generate a ASK signal with appropriate sinusoidal carrier.

Results
Attach results of Lab tasks and comments on them.

Performance Viva
Total/15
Performance /4
Results /3
Lab Report /3
Comments

Lab # 13: To Construct Modem Based on Binary Phase Shift
Keying Using Discrete Components for Digital Communication
Systems
Objectives
 To construct a modulator and demodulator for Phase Shift Keying using discreet
components.
Pre-Lab Exercise
and become familiar with the fundamentals of FSK modulation provided with this laboratory
Use MATLAB to generate a FSK signal with appropriate sinusoidal carrier.
Background
In Frequency shift keying (FSK), the carrier frequency is shifted (i.e. from one frequency to
another) corresponding to the digital modulating signal. If the higher frequency is used to
represent a data ‘1’ & lower frequency a data ‘0’, the resulting FSK waveform is shown in
Figure 13.1.Thus,
Data =1 High Frequency
Data =0 Low Frequency
Figure 13.15: FSK waveform, derived from a binary message

Conceptually, and in fact, the transmitter could consist of two oscillators (on frequencies f1 and
f2), with only one being connected to the output at any one time. This is shown in block diagram
form in Figure 13.2 below.
Figure 13.16: An FSK Transmitter
Unless there are special relationships between the two oscillator frequencies and the bit clock
there will be abrupt phase discontinuities of the output waveform during transitions of the
message.
Bandwidth: Practice is for the tones f1 and f2 to bear special inter-relationships, and to be
integer multiples of the bit rate. This leads to the possibility of continuous phase, which offers
advantages, especially with respect to bandwidth control. Alternatively, the frequency of a single
oscillator (VCO) can be switched between two values, thus guaranteeing continuous phase -
CPFSK. The continuous phase advantage of the VCO is not accompanied by an ability to ensure
that f1 and f2 are integer multiples of the bit rate. This would be difficult to implement with a
VCO. Being an example of non-linear modulation, calculation of the bandwidth of an FSK signal
is a non-trivial exercise. It will not be attempted here.
FSK signals can be generated at baseband, and transmitted over telephone lines (for example). In
this case, both f1 and f2 (of Figure 13.2) would be audio frequencies. Alternatively, this signal
could be translated to a higher frequency. Yet again, it may be generated directly at ‘carrier’
frequencies.
It is also represented as a sum of two ASK signals. The two carriers have different frequencies
and the digital data is inverted. The demodulation of FSK is done by separated the modulated
signal into two parts by band pass filter tuned to mark and space frequencies. The demodulation
by this method is shown in Figure 13.3. The output from each BPF looks like an amplitude shift
keyed (ASK) signal. These can be demodulated asynchronously, using the envelope. The

decision circuit, to which the outputs of the envelope detectors are presented, selects the output
which is the most likely one of the two inputs. It also reshapes the waveform from a band-limited
to a rectangular form. This is, in effect, a two channel receiver. The bandwidth of each is
dependent on the message bit rate. There will be a minimum frequency separation required of the
two tones.
Figure 13.17: Demodulation by conversion to ASK
Another method of demodulation of FSK can be carried out by a PLL. As known, the PLL tries
to ‘lock’ the input frequency. It achieves this by generating corresponding O/P voltage to be fed
to the VCO, if any frequency deviation at its I/P is encountered. Thus the PLL detector follows
the frequency changes and generates proportional O/P voltage. The O/P voltage from PLL
contains the carrier components. Therefore, to remove this, the signal is passed through Low
Pass Filter. The resulting wave is too rounded to be used for digital data processing. Also,the
amplitude level maybe very low due to channel attenuation.
In-Lab Exercise
Design a TIMS circuit diagram of FSK modulator that employs an appropriate sinusoidal carrier
to modulate the baseband signal. Also, design its receiver with the help of PLL to recover the
baseband signal. Moreover, implement the designed circuit on TIMS trainer.
Circuit diagram:

Design a circuit diagram for FSK modulator using discrete components available in lab. Use
baseband signal of frequency from the signal generator. Also employ phase locked loop
at the receiver to recover the baseband message.
Circuit diagram:

Post-Lab
Write a MATLAB code for modulating the input signal using FSK modulation and then
demodulate using both synchronous and asynchronous method.
Results


Performance Viva
Total/15
Performance /4
Results /3
Lab Report /3
Comments

Lab # 14: To Construct Modem Based on Frequency Shift Keying
Using Discrete Components for Digital Communication Systems.
Objectives
 To construct a modulator and demodulator for Frequency Shift Keying using discreet
components.
Pre-Lab Exercise
and become familiar with the fundamentals of BPSK modulation provided with this laboratory
Use MATLAB to generate a BPSK signal with appropriate sinusoidal carrier.
Background
Consider a sinusoidal carrier. If it is modulated by a bi-polar bit stream according to the scheme
illustrated in Figure 14.1 below, its polarity will be reversed every time the bit stream changes
polarity. This, for a sine wave, is equivalent to a phase reversal (shift). The multiplier output is a
BPSK signal.
Figure 14.18: Generation of BPSK
The information about the bit stream is contained in the changes of phase of the transmitted
signal. Asynchronous demodulator would be sensitive to these phase reversals. The appearance

of a BPSK signal in the time domain is shown in Figure 14.2 (lower trace). The upper trace is the
binary message sequence.
Figure 14.19: A BPSK signal in time domain
There is something special about the waveform of Figure 14.2. The wave shape is ‘symmetrical’
at each phase transition. This is because the bit rate is a sub-multiple of the carrier frequency .
In addition, the message transitions have been timed to occur at a zero-crossing of the carrier.
Whilst this is referred to as ‘special’, it is not uncommon in practice. It offers the advantage of
simplifying the bit clock recovery from a received signal. Once the carrier has been acquired
then the bit clock can be derived by division.
Band limiting: The basic BPSK generated by the simplified arrangement illustrated in Figure
14.1 will have a bandwidth in excess of that considered acceptable for efficient communications.
If you can calculate the spectrum of the binary sequence, then you know the bandwidth of the
BPSK itself. The BPSK signal is a linearly modulated DSB, and so it has a bandwidth twice that
of the baseband data signal from which it is derived. In practice there would need to be some
form of bandwidth control. Band limiting can be performed either at baseband or at carrier
frequency. It will be performed at baseband in this experiment.
BPSK Demodulation: Demodulation of a BPSK signal can be considered a two-stage process.
 Translation back to baseband, with recovery of the band limited message waveform
which is achieved with a synchronous demodulator, as shown in Figure 14.3 below. This
requires a local synchronous carrier. In this experiment a stolen carrier will be used.

Figure 14.20: Synchronous demodulation of BPSK
 Regeneration from the band limited waveform back to the binary message bit stream.
Translation back to baseband requires a local, synchronized carrier. The translation
process does not reproduce the original binary sequence, but a band limited version of it.
The original binary sequence can be regenerated with a detector. This requires
information regarding the bit clock rate. If the bit rate is a sub-multiple of the carrier
frequency, then bit clock regeneration is simplified. In TIMS the DECISION MAKER
module can be used for the regenerator, and in this experiment the bit clock will be a sub-
multiple of the carrier.
In-Lab Exercise
Design a TIMS circuit diagram of BPSK modulator that employs an appropriate sinusoidal
carrier to modulate the baseband signal. Also, design its receiver that uses synchronous
demodulator to recover the baseband signal. Moreover, implement the designed circuit on TIMS
trainer.
Circuit diagram:

Design a circuit diagram for BPSK modulator using discrete components available in lab. Use
baseband signal and sinusoidal carrier of frequency and from the signal
generator respectively. Moreover, implement the circuit on trainer board.
Circuit diagram:
Post-Lab
Write a MATLAB code that modulates the following binary sequence

using BPSK modulation.

Results

Performance Viva
Total/15
Performance /4
Results /3
Lab Report /3
Comments

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COMSATS Institute of Information Technology Lahore

Department of Electrical Engineering

EEE351 –Principles of Communication Systems

S.No. Update Date Performed by

1 Lab Manual Preparation Sep- Engr. Rizwan

| EEE351| Principles Communication System Lab Manual i

| EEE351| Principles Communication System Lab Manual ii

1. Modern Digital and Analog Communication Systems by B.P Lathi

1. Introduction to Digital and Analog Communication Systems by Simon Haykin and

Grades Letter Grade Credit Points Percentage Marks

| EEE351| Principles Communication System Lab Manual iii

 Instek Digital Oscilloscope

| EEE351| Principles Communication System Lab Manual iv

To sketch Frequency Domain Representation of Signals for Analysis of Communication

To Construct a Modulator for Amplitude Modulation in Analog Communication Systems

| EEE351| Principles Communication System Lab Manual v

Learning Outcome Relevant Experiment

Display the performance of various analog and

Manipulate various parameters for evaluation of

| EEE351| Principles Communication System Lab Manual vii

Lab Report Instructions

Part-I Fourier Series

 Only need to perform one integration

| EEE351| Principles Communication System Lab Manual 8

Make a function file sinc1.m using the following code listing.

Listing 1.1: Sinc function

Sketch the Plot for listing 1.1

| EEE351| Principles Communication System Lab Manual 9

Listing 1.2: Magnitude and Phase Response

Sketch the Plot for listing 1.2

| EEE351| Principles Communication System Lab Manual 10

| EEE351| Principles Communication System Lab Manual 11

Listing 1.4: g(t) using unit step

gt= (u( t ) -u( t-1) ) + (u( t-4) -u( t-5) ) + u( t-8) ;

Sketch the Plot for listing 1.4 & 1.5

| EEE351| Principles Communication System Lab Manual 12

| EEE351| Principles Communication System Lab Manual 13

Listing 2.3: Shifted version of FFT of g(t)

| EEE351| Principles Communication System Lab Manual 14

| EEE351| Principles Communication System Lab Manual 15

| EEE351| Principles Communication System Lab Manual 16

| EEE351| Principles Communication System Lab Manual 17

| EEE351| Principles Communication System Lab Manual 18

Lab Report Instructions

= [low frequency term m(t)] x [high frequency term c(t)] ........ 3

Figure 1: Block diagram of AM

Figure 2: AM, with A=1, as seen on the oscilloscope

| EEE351| Principles Communication System Lab Manual 20

| EEE351| Principles Communication System Lab Manual 21

Lab Task 1: Generation of AM using discrete components

| EEE351| Principles Communication System Lab Manual 22

| EEE351| Principles Communication System Lab Manual 23

Lab Report Instructions

| EEE351| Principles Communication System Lab Manual 24

The Ideal Envelope Detector:

Figure 3.1: the ideal envelope recovery arrangement

Lab Task 1: Generation of AM using TIMS

Implement this hardware model on TIMS and observe the output.

| EEE351| Principles Communication System Lab Manual 25

Lab Task 2: Envelop detection using TIMS

Figure 3.3: Modeling the ideal envelope detector with TIMS

| EEE351| Principles Communication System Lab Manual 26

Figure 3.4: Generator for AM and DSBSC

Figure 3.5: modulated signal generator and envelope recovery