Professional Documents
Culture Documents
Supervised By
Dr. SOBIA BAIG
Semester ___________________________
Revision History
Date
S. No. Activity Performed by
(DD/MM/YYYY)
1 Lab Manual Preparation 02/09/2013 Engr. Rizwan Asif
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 behavior of real systems with remarkable degree of
precision.
Learning Outcomes
Theory CLOs
After successfully completing this course, the students will be able to:
1. Analyze the behavior of amplitude and angle modulated signals both in time and frequency
domain to comprehend the analog communication systems (Level: C4).
2. Design and analysis of digital data communication system using constituent basic
building blocks (Level: C5).
3. Evaluate performance of digital communication system using basic communication
parameters (Level C6).
Lab CLOs
After successfully completing this course, the students will be able to:
4. Display the performance of various analog and digital modulation techniques and manipulates
the associated parameters for their evaluation u s i n g software and hardware tools (PLO5/P5)
5. Students will be able to explain the communication system’s design effectively with the help of
documentation and technical reports. (PLO10/A2)
6. Analyze the performance of various concepts of analog and digital communication systems using
hardware and software tools. (PLO9/A2)
PLO3
PLO4
PLO5
PLO9
CLO
CLO1 x C4
CLO2 x C5
CLO3 x C6
CLO4 x P5
CLO5 x A2
CLO6 x A2
Lab 10
Lab 11
Lab 12
Lab 13
Lab 14
Lab 1
Lab 2
Lab 3
Lab 4
Lab 5
Lab 6
Lab 7
Lab 8
Lab 9
CLO
CLO4 x x x x x x x x x X x x x X
CLO5 x x x x x x x x x X x x x X
CLO6 x x x x x x x x x X x x x X
Grading Policy
The final marks for lab would comprise of Lab Assessment (25%), Lab S1 (10%), Lab S2 (15 %) and
Lab Terminal (50%).
S-I 0.5*(S-I Exam result) + 0.5* (average of lab evaluation of Lab 1-4)
S-II 0.5*(S-II Exam result) + 0.5*[ (average of lab evaluation of Lab 5-8) * 1.5]
Terminal 0.5*(Terminal Exam result) +0.25*[(average of lab evaluation of Lab 9-12) *5] +
0.10*[(average of lab evaluation of Lab 5-8) *5] + 0.15*[(average of lab evaluation of Lab 1-4) *5]
A/Q Marks: For CEP designated courses: Add CEP marks out of 25.
For Non-CEP designated courses: [(Average of lab evaluation of Lab (1-12)) * 2.5]
The minimum pass marks for both lab and theory shall be 50%. Students obtaining less than 50%
marks (in either theory or lab, or both) shall be deemed to have failed in the course. The final marks
would be computed with 75% weight to theory and 25% to lab final marks.
List of Equipment
EMONA Telecommunications Instructional Modelling Systems (TIMS)
Instek Digital Oscilloscope
Multimeter
Regulated Power Supply
Functional Generators
Breadboard and other Discrete Components
Software Resources
MATLAB
SIMULINK
Safety Instructions
Preface ..........................................................................................................................................................................ii
LAB # 1 .......................................................................................................................................................................10
To Sketch Frequency Domain Representation of Signals for Analysis of Communication Systems Using MATLAB
....................................................................................................................................................................................10
Objectives ....................................................................................................................................................................... 10
Pre-Lab ........................................................................................................................................................................... 10
In-Lab Tasks .................................................................................................................................................................... 11
Lab Assessment .............................................................................................................................................................. 19
LAB # 2 .......................................................................................................................................................................20
To Construct a Modulator for Amplitude Modeling in Analog Communication Systems Using Discrete
Components ................................................................................................................................................................20
Objectives ....................................................................................................................................................................... 20
Pre-Lab ........................................................................................................................................................................... 20
In-Lab Tasks .................................................................................................................................................................... 23
Assessment ..................................................................................................................................................................... 24
LAB # 3 .......................................................................................................................................................................25
To Construct an Envelope Detector Using Discrete Components for Demodulation in Analog Communication
Systems .......................................................................................................................................................................25
Objectives ....................................................................................................................................................................... 25
Pre-Lab ............................................................................................................................................................................ 25
In-Lab Tasks .................................................................................................................................................................... 27
Lab Assessment .............................................................................................................................................................. 31
To Assemble a Modem for Single Side Band Amplitude Modulation Using TIMS Trainer for Analog
Communication Systems ............................................................................................................................................32
Objectives ....................................................................................................................................................................... 32
Pre-Lab ............................................................................................................................................................................ 32
In-Lab Tasks .................................................................................................................................................................... 33
Lab Assessment .............................................................................................................................................................. 36
LAB # 5 .......................................................................................................................................................................37
To Assemble a Modem for Quadrature Amplitude Modulation using TIMS Trainer for Analog Communication
Systems .......................................................................................................................................................................37
Objectives ....................................................................................................................................................................... 37
Pre-Lab ............................................................................................................................................................................ 37
In-Lab Tasks .................................................................................................................................................................... 38
Lab Assessment .............................................................................................................................................................. 41
LAB # 6 .......................................................................................................................................................................42
To Assemble Phase Locked Loop (PLL) using TIMS Trainer for Phase Detection in Analog Communication
Systems .......................................................................................................................................................................42
Objectives ....................................................................................................................................................................... 42
Pre-Lab ............................................................................................................................................................................ 42
In-Lab Tasks .................................................................................................................................................................... 47
Lab Assessment .............................................................................................................................................................. 49
LAB # 7 .......................................................................................................................................................................50
To Construct Frequency Modulator using Discrete Components for Analog Communication Systems ..................50
Objectives ....................................................................................................................................................................... 50
In-Lab Tasks .................................................................................................................................................................... 50
Lab Assessment .............................................................................................................................................................. 53
LAB # 8 .......................................................................................................................................................................54
To Display Sampling of Analog Signals for Digitization using Discrete Components and MATLAB ......................54
Objectives ....................................................................................................................................................................... 54
Pre-Lab ............................................................................................................................................................................ 54
In-Lab Tasks .................................................................................................................................................................... 55
Lab Assessment .............................................................................................................................................................. 59
LAB # 9 .......................................................................................................................................................................60
To Display Various Line Coding Schemes for Pulse Code Modulation using TIMS Trainer ...................................60
Objectives ....................................................................................................................................................................... 60
Pre-Lab ............................................................................................................................................................................ 60
In-Lab Tasks .................................................................................................................................................................... 63
Lab Assessment .............................................................................................................................................................. 65
LAB # 10 .....................................................................................................................................................................66
LAB # 11 .....................................................................................................................................................................72
To Assemble Decoder for Pulse Code Modulation (PCM) using TIMS Trainer .......................................................72
Objectives ....................................................................................................................................................................... 72
Pre-Lab ............................................................................................................................................................................ 72
Lab Tasks ......................................................................................................................................................................... 73
Lab Assessment .............................................................................................................................................................. 75
LAB # 12 .....................................................................................................................................................................76
To Construct a Modem Based on Amplitude Shift Keying using Discrete Components for Digital Communication
Systems .......................................................................................................................................................................76
Objectives ....................................................................................................................................................................... 76
Pre-Lab ............................................................................................................................................................................ 76
Lab Tasks ......................................................................................................................................................................... 78
Lab Assessment .............................................................................................................................................................. 81
LAB # 13 .....................................................................................................................................................................82
To Construct Modem Based on Frequency Shift Keying using Discrete Components for Digital Communication
Systems .......................................................................................................................................................................82
Objectives ....................................................................................................................................................................... 82
Pre-Lab ............................................................................................................................................................................ 82
Lab Tasks ......................................................................................................................................................................... 84
Lab Assessment .............................................................................................................................................................. 87
LAB # 14 .....................................................................................................................................................................88
To Construct Modem Based on Binary Phase Shift Keying using Discrete Components for Digital Communication
Systems .......................................................................................................................................................................88
Objectives ....................................................................................................................................................................... 88
Pre-Lab ............................................................................................................................................................................ 88
Lab Tasks ......................................................................................................................................................................... 90
Lab Assessment .............................................................................................................................................................. 93
LAB # 1
To Sketch Frequency Domain Representation of Signals for Analysis of
Communication Systems Using MATLAB
Objectives
To display magnitude and phase response of Fourier series coefficients using MATLAB for
visualization of signals in communication systems
Pre-Lab
Read this experiment in its entirety to become familiar with objectives of this lab. Study in detail and
become familiar with the fundamentals of Fourier series and discrete time Fourier transform. The
instructor may provide the class some time to reflect upon these before proceeding with the lab.
Fourier Series
In mathematics, a Fourier series decomposed 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
Tasks
1. The periodic signal 𝑥(𝑡) is defined in one period as 𝑥(𝑡) = 𝑡𝑒 −𝑡 , 0 ≤ 𝑡 ≤ 6 . Plot the
approximate signals in 4 periods in time using 81 terms of the complex exponential and
trigonometric forms of Fourier series. For comparison reasons, plot the original signal 𝑥(𝑡)
over the same time interval.
2. The periodic signal in one period is given by:
1, 0 ≤ 𝑡 ≤ 1
𝑥(𝑡) = {
0 1≤𝑡≤2
Plot in one period the approximate signals using 41 and 201 terms of the complex exponential
Fourier series. Furthermore, each time plot the complex exponential coefficients.
In-Lab Tasks
Task 1
1. Evaluate the Fourier series coefficients using:
Plot the magnitude |Dn| (in volts) and phase ∠Dn (in degrees0 of the first twenty-one
coefficients, Let {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.
Task 2
Magnitude and Phase of Fourier Series Coefficients
Write the following code in MATLAB and run it.
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.
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
function[y] = u(x)
y = 0.5+0.5*sign(x)
end
Task 4
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 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
Task 5
Discrete Fourier Series using FFT
Using the following code, we get the Fourier series coefficients.
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 o reach
there. We will get a plot using FFTSHIFT command such as DC component is at the center, and plot
gets the shape of a normal Fourier Series plot.
COMMENTS
FFT function plot the Fourier series but with 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.
Task 6
1. Write MATLAB code to get the spectrum of g(t) and also sketch the output.
2. Write MATAB code to get the magnitude and phase of g(t).
Task 7
Consider 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.
Lab Assessment
Pre-Lab /1
Performance /3
Results /2
/10
Viva /2
Lab Report /2
LAB # 2
To Construct a Modulator for Amplitude Modeling in Analog Communication
Systems Using Discrete Components
Objectives
To construct amplitude modulator and observed output signals using various discrete
components
Equipment Required
Resistors, Capacitors, Oscilloscope, Multimeter, Regulated Power Supply, Function Generators,
Breadboard and Connecting Wires.
Pre-Lab
Read this experiment in its entirety to become familiar with objectives of this lab. Study in detail and
become familiar with the fundamentals of Amplitude Modulation and Double Side Band Suppressed
Carrier modulation. The instructor may provide the class some time to reflect upon these before
proceeding with the lab.
Introduction
An amplitude modulated signal is defined as:
𝐴𝑀 = (𝐴 + 𝑚(𝑡)) cos 𝜔𝑡 … … … … 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 equation 1 has been written into two parts in
equation 2.
Block Diagram
Equation 1 can be represented by the block diagram of Figure 2.1.
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 2.2 illustrates what the oscilloscope will
show.
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)
Where, P and Q are the peak-to-peak and trough-to-trough amplitudes respectively of the AM
waveform of Figure 2.3. 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 routine shape, of the AM signal of
Figure 2.3 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 equation 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.
In-Lab Tasks
1. Write the MATLAB code and verify the results of the performed experiment. Draw its
Simulink diagram as well.
2. 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
Assessment
Pre-Lab /1
Performance /3
Results /2
/10
Viva /2
Lab Report /2
LAB # 3
To Construct an Envelope Detector Using Discrete Components for
Demodulation in Analog Communication Systems
Objectives
Pre-Lab
Read this experiment in its entirety to become familiar with objectives of this lab. Study in detail and
become familiar with the fundamentals of Amplitude Modulation and demodulation. Further, also
become aware of the envelope detection techniques. The instructor may provide the class some time
to reflect upon these before proceeding with the lab.
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 views 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 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.
In-Lab Tasks
The ‘ideal rectifier’ is easy to build, does in fact approach the ideal for our purpose, 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.
T1 plug in the TUNEABLE LPF module. Set it to widest bandwidth, which is about 12 kHz (front
panel toggle witch 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 ADUIO 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.
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.
Generate AM signal (D1, D2, 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 3.6 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) Envelope Detector Circuit Diagram (b) Circuit Diagram for Amplitude Modulation with Envelope Detector
Lab Assessment
Pre-Lab /1
Performance /3
Results /2
/10
Viva /2
Lab Report /2
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 single side band amplitude modulated signal using MATLAB for amplitude modulation
Equipment Required
Adder, Audio Oscillator, Multiplier, Phase Shifter, Quadrature Phase Splitter, Voltage Controlled
Oscillator
Pre-Lab
Read this experiment in its entirety to become familiar with objectives of this lab. Study in detail and
become familiar with the fundamentals Double Side Band and Single Side Band Amplitude
Modulation. The instructor may provide the class some time to reflect upon these before proceeding
with the lab.
Introduction
The method of sideband removal is to make two DSDSC 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.
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.
In-Lab Tasks
Lab Task 1: SSB Modulation
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 volts peak-to-peak).
Then replace both ADDER inputs and watch the ADDER outputs 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 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
|EEE 351 | Principles of Communication Systems Lab Manual 33
Lab Experiment | 4
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.
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 mode. Most will be unfamiliar.
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.
|EEE 351 | Principles of Communication Systems Lab Manual 34
Lab Experiment | 4
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.
Lab Assessment
Pre-Lab /1
Performance /3
Results /2
/10
Viva /2
Lab Report /2
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 amplitude
modulation
To sketch the quadrature amplitude modulated signal using MATLAB for amplitude
modulation
Equipment Required
TIMS modules of Multiplier, Phase Shifter, Adder and Audio Oscillator
Pre-Lab
Read this experiment in its entirety to become familiar with objectives of this lab. Study in detail and
become familiar with the fundamentals of Quadrature Amplitude Modulation and Demodulation.
The instructor may provide the class some time to reflect upon these before proceeding with the lab.
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 later it is
often called a Quadrature Phase Shift Keyed (QPSK) modulator.
This lab sheet investigated 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.
|EEE 351 | Principles of Communication Systems Lab Manual 37
Lab Experiment | 5
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.
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 to be constructed.
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.
In-Lab Tasks
Lab Task 1: QAM Modulation
Figure 5.3 shows a model of the block diagram of a QAM modulator, show in Figure 5.1.
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?
The 100 kHz carrier (sin(ωt) or cos(ωt)) 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 the 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. 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.
Lab Assessment
Pre-Lab /1
Performance /3
Results /2
/10
Viva /2
Lab Report /2
LAB # 6
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 discrete components and TIMS trainer for
phase detection in analog communication systems
Pre-Lab
Read this experiment in its entirety to become familiar with objectives of this lab. Study in detail and
become familiar with the fundamentals of Phase Locked Loop. The instructor may provide the class
some time to reflect upon these before proceeding with the lab.
Introduction
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 6.1 represents the block diagram of PLL showing its basic function connected
together in a feedback loop.
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 fosc is 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 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 signal 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
PLL. The lock range is wider than the capture range. So, if the VCO output frequencies fosc is
plotted against the incoming frequency fi, we obtain the PLL steady state characteristics shown in
Figure 6.2. the characteristics simply shows that fosc=fi in the locked condition, and that
fosc=fo=constant when PLL is unlocked. A hysteresis can be observed in the fosc(fi) characteristic
because the capture range is smaller than the lock range.
Phase Detector
The phase detector on the 4046 is simply an XOR logic gate as shown in Figure 6.3, 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 ration Dφ that depends on
the phase difference between Vi and Vosc.
VDD = φ π
The periodic signal Vφ(t) at the output of the XOR phase detector can be written as the Fourier
series:
𝑉𝜔(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 = 𝑉𝐷𝐷𝜑𝜋
Loop Filter
The output Vφ(t) of the phase detector is filtered by an external low-pass filter. In Figure 6.3, 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 = 12𝜋𝑅𝐶
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
is called the gain or the sensitivity 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 Figure 6.5. Note that Vo id 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 Figure 6.4 it is easy to see that 𝜑𝜋, 𝑉𝑜 decreases. Of
course, the characteristics is periodic in 𝜑 with period 2𝜋. The range 0 ≤ 𝜑 ≤ 𝜋 is the range where
the PLL can operate in the locked condition.
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 2fc is smaller than the lock range
fmax-fmin as shown in Figure 6.2. The capture range 2fc-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 2𝐾𝑜 𝑞 1 + (𝑓𝑐 𝑓𝑝)2
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 2𝐾𝑜 𝑓𝑝 𝑉𝐷𝐷
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 Tasks
Lab Task 1
Set the values of C1 = 0.03µF, R1=R2=18KΩ.
Find out the fmin and fmax of the VCO. To find fmin simply connect the VCO input (pin9) 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 fo 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 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 = fo+fc =…………...
lock range = fmax-fmin = ………………….
Capture range = fcap2-fcap1 = ……………..
Lab Assessment
Pre-Lab /1
Performance /3
Results /2
/10
Viva /2
Lab Report /2
LAB # 7
To Construct Frequency Modulator using Discrete Components for Analog
Communication Systems
Objectives
To construct PLL based frequency modulator using discrete components for analog
communication systems.
In-Lab Tasks
Lab Task 1:
Build the circuit shown in Figure 7.1. This uses the VCO portion of the 4046 PLL.
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 DC offset = 5V
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 of 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.
Data Analysis
The FM constant, Kf, can be determines 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/seconds-per-volt in order to write the expression for the FM signal
l you generate. To find β, use
Lab Assessment
Pre-Lab /1
Performance /3
Results /2
/10
Viva /2
Lab Report /2
LAB # 8
To Display Sampling of Analog Signals for Digitization using Discrete
Components and MATLAB
Objectives
Pre-Lab
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 proceedings with the lab. Furthermore, you need to complete the following tasks on
MATLAB and also attach the 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 8.1 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.
zero. The signal, , is said to have been sampled by the sampling signal .
Sampled signals such as in Figure 8.2 (d) are useful if they contain the same information as the
original signal , as shown in Figure 8.2 (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 bandwidth limited function (i.e. have no frequency
components outside the frequency interval (-fb, fb) and
The frequency of the sampling signal must be greater than .
In-Lab Tasks
With the help of above given arrangement and TIMS modules, draw a circuit diagram (with proper
labelling) 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 reconstructed (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 8.4.
You can confirm that it recovers the message from the samples by connecting the output of the
DUAL ANALOG SWITCH to the input of the 3 kHz LPF in the HEADPHONE AMPLIFIER
module and displaying the output on the oscilloscope.
Lab 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 is flat i.e. they have
constant amplitudes. 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 8.5 below:
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)
Circuit Diagram
Lab Assessment
Pre-Lab /1
Performance /3
Results /2
/10
Viva /2
Lab Report /2
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.
suing TIMS trainer
To display line coded waveform for PCM using trainer
Pre-Lab
Read this experiment in its entirety to become familiar with objectives of this lab. study in detail 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 wave.
𝑋 = [0 1 1 0 0 1 0]
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 discrete signal, which represents digital data. There are three types of line coding
schemes available as shown in Figure 9.1.
Uni-Polar Encoding
Unipolar encoding schemes use signal voltage level to represent data. In this case, to represent binary
1, high voltage is transmitted and to represent 0, no voltage transmitted. It is also called Unipolar-
Non-return-to-zero, because there is no rest condition i.e. it either represents 1 or 0.
Polar Encoding
Polar encoding scheme uses multiple voltage levels to represent binary values. Polar encoding
techniques are of four types:
Manchester
This encoding scheme is a combination of RZ and NRZ-L. bit time is divided into two
halves. It transmits 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.
In-Lab Tasks
Lab 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 DECISISON 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 8.333 kHz (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 2.083 kHz. This, then becomes a
convenient source of a 2.083 kHz TTL signal for use as the system bit clock.
Because the LINE-CODE DECODER has some processing to do it, 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).
Write code
Lab Assessment
Pre-Lab /1
Performance /3
Results /2
/10
Viva /2
Lab Report /2
LAB # 10
To Assemble an 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
Read this experiment in its entirety to become familiar with objectives of this lab. study in detail and
become familiar with the basics of pulse code modulation 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 proceedings with the lab. Furthermore, explore the following MATLAB functions
and write a short summary about their usage:
𝑌 = 𝑞𝑢𝑎𝑛𝑡(𝑋, 𝑄);
𝑂𝑢𝑡 = 𝑐𝑜𝑚𝑝𝑎𝑛𝑑(𝑖𝑛, 𝑝𝑎𝑟𝑎𝑚, 𝑣);
Background
Information in an analog from cannot be processed by digital computers so its 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 Pulse Amplitude Modulation
(PAM)- quantization by amplitude and quantization by time which means digitization 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 samples one
quantization segment is chosen (every quantization segment contains an interval of amplitudes).
PCM modulation is commonly used in radio and telephone transmission. The main advantage is that
the PCM modulation only needs 8 kHz sampling frequency to maintain the original quality of audio.
Figure 10.1 is 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 to the quantization levels. For example, if the bits for PCM modulation is 3,
then the quantization levels are 23 = 8, which is 8 steps. If the bits for PCM are 4, then the
quantization levels are 24 = 16, 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 utilized ‘n’ 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. 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.
Lab Tasks
Lab Task 1: TIMS TRAINER
Design a TIMS circuit diagram using a PCM encoder module that will generate a PCM encoded
signal for a sinusoidal input signal with 2 kHz 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 ± 2.0 volts range of the TIMS ANALOG REFERENCE LEVEL. This is in
keeping with the amplitude limits set for all analog modules.
Circuit Diagram
The codeword is assembled into time frame together with other bits as may be required
(described below). In the TIMS PCM ENCODER (and many commercial system) 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.
Lab Assessment
Pre-Lab /1
Performance /3
Results /2
/10
Viva /2
Lab Report /2
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
Read this experiment in its entirety to become familiar with objectives of this lab. Study in detail and
become familiar with the basics of pulse code demodulation provided with this laboratory
experiment and in chapter 6 of the reference book. The instructor may provide the class more time to
reflect upon these before proceeding with the lab.
Generate a sinusoidal signal with amplitude 1 and ω = 1. 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 identified 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 a ‘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 reconstructing filter is provided in the module.
Lab Tasks
Circuit Diagram
Lab Assessment
Pre-Lab /1
Performance /3
Results /2
/10
Viva /2
Lab Report /2
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 discrete
components.
Pre-Lab
Read this experiment in its entirety to become familiar with objectives if this lab. study in detail and
become familiar with the fundamentals of ASK modulation provided with this laboratory experiment
and in chapter 7 of the reference book. The instructor may provide the class more time to reflect
upon these before proceeding with the lab.
Background
Amplitude Shift Keying (ASK) in the context of digital communications is a modulation process
which imparts to a sinusoidal 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.
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 12.2, where switch is opened and closed by unipolar binary sequence.
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 increases
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.
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:
Lab Tasks
Circuit Diagram
Circuit Diagram
Lab Assessment
Pre-Lab /1
Performance /3
Results /2
/10
Viva /2
Lab Report /2
LAB # 13
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 discrete
components.
Pre-Lab
Read this experiment in its entirety to become familiar with objectives of this lab. study in detail and
become familiar with the fundamentals of FSK modulation 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.
Background
In Frequency Shift Keying (FSK), the carrier frequency is shifted (i.e. from on 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
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:
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 modulation 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 bandlimited 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.
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 generated 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.
Lab Tasks
Circuit Diagram
Circuit Diagram
Lab Assessment
Pre-Lab /1
Performance /3
Results /2
/10
Viva /2
Lab Report /2
LAB # 14
To Construct Modem Based on Binary Phase Shift Keying using Discrete
Components for Digital Communication Systems
Objectives
To construct a modulator and demodulator for Binary Phase Shift Keying using discrete
components.
Pre-Lab
Read this experiment in its entirety to become familiar with objectives of this lab. study in detail and
become familiar with the fundamentals of BPSK modulation 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.
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.
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.
There is something special about the waveform of Figure 14.2. The wave shape is ‘symmetrical’ at
𝜔
each phase transition. This is because the bit rate is a sub-multiple of the carrier frequency 2𝜋. In
addition, the message transitions have been timed to occur at a zero-crossing of the carrier. Whilst
this is referred to a ‘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. 2 below. This requires a
local synchronous carrier. In this experiment a stolen carrier will be used.
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.
|EEE 351 | Principles of Communication Systems Lab Manual 89
Lab Experiment | 14
Lab Tasks
Circuit Diagram
Circuit Diagram
Lab Assessment
Pre-Lab /1
Performance /3
Results /2
/10
Viva /2
Lab Report /2