Comm Manual 2020
Comm Manual 2020
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TABLE OF CONTENTS
Lab - 1: Introduction to Telecoms Trainer ......................................................................... 5
1.1 Apparatus ..........................................................................................................................................5
1.2 Introduction .......................................................................................................................................5
1.3 Modules .............................................................................................................................................5
1.4 Modeling Equations ...........................................................................................................................9
1.5 Modeling Equations .........................................................................................................................11
1.6 Lab Outcome ...................................................................................................................................11
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Lab - 6: SSB-SC Modulation Demodulation of Signals .....................................................51
6.1 Apparatus ........................................................................................................................................51
6.2 Introduction .....................................................................................................................................51
6.3 Generating SSBSC ............................................................................................................................52
6.4 The Experiment................................................................................................................................53
6.5 SSBSC Demodulation .......................................................................................................................56
6.6 Lab Outcomes ..................................................................................................................................58
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12.4 Result and Analysis ..........................................................................................................................99
12.5 4.2 Phase Lock Loop .....................................................................................................................101
12.6 Lab Outcomes ................................................................................................................................103
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Instructions for Students
1. If a student is absent from a lab due to any reason, he/she will have to get written permission from the
Dean to perform that lab. LAB GRADING POLICY CAN BE CHANGED BY LAB INSTRUCTOR
ANYTIME BEFORE THE FINAL LAB EXAM.
2. Students may bring their text books to the lab, so that they can refer to theory and diagrams whenever
required.
3. Labs will be graded in parallel entry fashion; one entry in the assessment sheet given at the end of every
lab and another entry in the instructor’s record which will be scaled according to the percentage of
respective labs. This system of keeping records will keep students aware of their performance throughout
the lab.
4. The general assessment sheet at the end of every lab looks as shown below
Name: _______________________
Reg #: ______________________
CLO-2 PLO-5 All required data is Data recorded and Data recorded Data recorded
recorded and presented presented is and presented is only with
accurately and completely complete but partially considerable
in the required format accuracy is not as complete. The assistance from
per requirement. required format instructor
The required format is followed but
is followed with few
deficiencies
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Introduction to Telecoms Trainer
1.1 Apparatus
EMONA 101 Trainer Kit
1.2 Introduction
As its name implies, the Emona Telecoms-Trainer 101 is used to help people learn about communications
and telecommunications principles. It lets you bring to life the block diagrams that fill communications
textbooks. A “block diagram" is a simplified representation of a more complex circuit. An example is shown
in Figure 1 below.
1.3 Modules
1.3.1 Master Signals, Speech and Buffer Modules
Analog Digital
A 2kHz sine wave A 2.083kHz square wave
A 100kHz sine wave An 8.33kHz square wave
A 100kHz cosine wave A 100kHz square wave
Each signal is available on a socket on the module's faceplate that's labeled accordingly.
TESTING:
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Connect the set-up as shown below:
Block diagram for the above 2kHz sine wave can be drawn as:
Check each of the Master Signals on oscilloscope. Draw their block diagrams below mention their
frequency and time period:
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The Speech Module:
Sine waves are important to communications. They are used extensively for the carrier signal in many
communications systems. Sine waves also make excellent test signals. However, the purpose of most
communications equipment is the transmission of speech (among other things) and so it is useful to examine
the operation of equipment using signals generated by speech instead of sine waves. The Emona Telecoms-
Trainer 101 allows you to do this using the Speech Module.
TESTING:
TASK:
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The Buffer Module:
Amplifiers are used extensively in communications and telecommunications equipment. They are often used
to make signals bigger.
Why?
They're also used as an interface between devices and circuits that can‘t normally be connected. The Buffer
module in the Emona Telecoms-Trainer 101 is an amplifier that can do both.
TESTING:
Make sure you connect the black leads to the ground of EMONA 101 trainer.
Set Scope’s mode control to Ext/Dual position.
Measure the input voltage first by channel 1. Then by moving the gain knob of buffer to middle
position. Measure the output voltage at channel 2.
TASK:
Draw the block diagram of above observed phenomena, making sure both channels are labeled:
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The gain of electronic device is generally given as:
Gain =
Will the gain be same for all knob positions? Reason your answer:
1. The VCO module has only two outputs (1 analog und 1 digital) whereas the Master Signals module
has six (3 analog and 3 digital)
2. The VCO module‘s output frequency is variable whereas the frequency of the Master Signal module‘s
outputs is fixed (at 2kHz, 8kHz und l00kHz)
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THE BLOCK DIAGRAMS OF EACH OF ABOVE MODULES ARE
PROVIDED
ADDER MODULE
VCO MODULE
Through the above block diagrams, design/implement each of the 3 modules and get them checked by your
lab instructor. Make sure you can justify the purpose of each module.
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1.5 Modeling Equations
Mathematics is an important part of communications and telecommunications. As you will learn, the output
of all communications systems can be described mathematically with an equation. Although the mathematics
that you will need for today’s experiment is relatively light. The EMONA Telecoms-Trainer 101 can model
communications equations to bring them to life. This experiment will introduce you to modeling of equations
through different modules studied above.
As already performed write down the general equation for the adder module:
_____________________________________________________________________
EQUATION: OUTPUT =
_____________________________________________
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LAB 1 ASSESSMENT1.
Student Name: ______________________ Reg. #: _________________
Date: _________________
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Introduction to Communication Toolbox-MATLAB
2.1 Apparatus
MATLAB
2.2 Introduction
This document is designed to act as a tutorial for an individual who has had no prior experience with Simulink
and it also serve as the lab handout to work on Simulink. Simulink is a graphical extension to MATLAB for
the modeling and simulation of systems. In Simulink, systems are drawn on screen as block diagrams. Many
elements of block diagrams are available (such as transfer functions, summing junctions, etc.), as well as
virtual input devices (such as function generators) and output devices (such as oscilloscopes). Simulink is
integrated with MATLAB and data can be easily transferred between the programs. In this tutorial, we will
introduce the basics of using Simulink to model and simulate a system. Simulink is supported on Unix,
Macintosh, and Windows environments, and it is included in student version of MATLAB for personal
computers. For more information on Simulink, visit the MathWorks website.
The idea behind this lab is that you can view it in one window while running Simulink in another window
on the desktop. Do not confuse the windows, icons, and menus in this tutorials for your actual Simulink
windows. Most images in these tutorials are not live, they simply display what you should see in your own
Simulink windows. All Simulink operations should be done in your Simulink windows.
2.3 Starting Simulink
1. Start MATLAB.
2. Simulink is an extra toolbox that runs on top of MATLAB.
3. To start this, type “simulink” in the Command Window or click on the Simulink icon.
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Figure 2-3-2: Simulink Interface
5. The Simulink Library Browser window should now appear on the screen. Most of the blocks needed
for modeling basic systems can be found in the subfolders of the main "Simulink" folder (opened by
clicking on the "+" in front of "Simulink").
2.4 Basic Elements
There are two major classes of elements in Simulink: blocks and lines. Blocks are used to generate, modify,
combine, output, and display signals. Lines are used to transfer signals from one block to another.
2.4.1 Blocks
The subfolders underneath the "Simulink" folder indicate the general classes of blocks available for us to
use:
1. Continuous: Linear, continuous-time system elements (integrators, transfer functions, state-space
models, etc.)
2. Discrete: Linear, discrete-time system elements (integrators, transfer functions, state-space models,
etc.)
3. Functions & Tables: User-defined functions and tables for interpolating function values
4. Math: Mathematical operators (sum, gain, dot product, etc.)
5. Nonlinear: Nonlinear operators (coulomb/viscous friction, switches, relays, etc.)
6. Signals & Systems: Blocks for controlling/monitoring signal(s) and for creating subsystems
7. Sinks: Used to output or display signals (displays, scopes, graphs, etc.)
8. Sources: Used to generate various signals (step, ramp, sinusoidal, etc.)
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Blocks have zero to several input terminals and zero to several output terminals. Unused input terminals are
indicated by a small open triangle. Unused output terminals are indicated by a small triangular point. The
block shown below has an unused input terminal on the left and an unused output terminal on the right.
2.4.2 Lines
Lines transmit signals in the direction indicated by the arrow. Lines must always transmit signals from the
output terminal of one block to the input terminal of another block. One exception to this is that a line can
tap off of another line. This sends the original signal to each of two (or more) destination blocks, as shown
below:
Lines can never inject a signal into another line; lines must be combined through the use of a block such as
a summing junction. A signal can be either a scalar signal or a vector signal. For Single-Input, Single-Output
systems, scalar signals are generally used. For Multi-Input, Multi-Output systems, vector signals are often
used, consisting of two or more scalar signals. The lines used to transmit scalar and vector signals are
identical. The type of signal carried by a line is determined by the blocks on either end of the line.
2.5 Building a System
To demonstrate how a system is represented using Simulink, we will build the block diagram for a simple
model consisting of a sinusoidal input multiplied by a constant gain, which is shown below:
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Simulink Library Browser (looks like a blank page). Building the system model is then accomplished through
a series of steps:
The necessary blocks are gathered from the Library Browser and placed in the model window.
The parameters of blocks are then modified to correspond with the system we are modeling.
Finally, the blocks are connected with lines to complete the model.
Each of these steps will be explained in detail using our example system. Once a system is built, simulations
are run to analyze its behavior.
1. Click on the "+" in front of "Sources" (this is a subfolder beneath the "Simulink" folder) to display
the various source blocks available for us to use.
2. Scroll down until you see the "Sine Wave" block. Clicking on this will display a short explanation of
what that block does in the space below the folder list:
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Figure 12-5-1-3 Gathering Blocks 2
2.5.2 Modifying the Blocks
Simulink allows us to modify the blocks in our model so that they accurately reflect the characteristics of the
system we are analyzing. For example, we can modify the Sine Wave block by double-clicking on it. Doing
so will cause the following window to appear:
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values can be entered into Simulink just as they have been shown. Next, we modify the Gain block by double-
clicking on it in the model window. The following window will then appear:
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route themselves). After drawing in the lines and repositioning the blocks, the example system model should
look like:
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Once the Scope window appears, click "Autoscale" button in its toolbar (looks like a pair of binoculars) to
scale the graph to better fit the window. Having done this, you should see the following:
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Note that the only difference between this output and the one from our original system is the amplitude of
the cosine curve. In the second case, the amplitude is equal to 1, or 1/10th of 10, which is a result of the gain
value being 1/10th as large as it originally was.
2.6 Implementation of Modulation
Now we introduce the spectrum analyzer as used in frequency domain analysis. To identify various types of
linear modulated waveforms in time and frequency domain representations the following procedure must be
followed and to implement theoretically functional circuits using the Communications Module Design
System (CMDS).
2.6.1 Equipment List
PC with MATLAB and Simulink
Spectrum Analyzer and Function Generator.
This section deals with looking at the spectrum of simple waves. We first look at the spectrum of a simple
sine wave. To Start Simulink: Start MATLAB then type Simulink on the command line. A Simulink Library
Window opens up as shown below:
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Figure 12-6-1-2: Design for Viewing the Spectrum of a Simple Sine Wave
Figure 12-6-1-3 shows the time-domain sine wave and the corresponding frequency domain is shown in
Figure 12-6-1-4. The frequency domain spectrum is obtained through a buffered-FFT scope, which
comprises of a Fast Fourier Transform of 128 samples which also has a buffering of 64 of them in one frame.
The property block of the B-FFT is also displayed in figure 12-6-1-4.
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Figure 12-6-1-4 Sine Wave – Frequency Domain
From the property box of the B-FFT scope the axis properties can be changed and the Line properties can be
changed. The line properties are not shown in the above. The Frequency range can be changed by using the
frequency range pop down menu and so can be the y-axis the amplitude scaling be changed to either real
magnitude or the dB (log of magnitude) scale. The upper limit can be specified as shown by the Min and
Max Y-limits edit box. The sampling time in this case has been set to 1/5000.
Note: The sampling frequency of the B-FFT scope should match with the sampling time of the input time
signal. Also as indicated above the FFT is taken for 128 points and buffered with half of them for an overlap.
Calculating the Power: The power can be calculated by squaring the value of the voltage of the spectrum
analyzer.
Note: The signal analyzer if chosen with half the scale, the spectrum is the single-sided analyzer, so the
power in the spectrum is the total power.
Similar operations can be done for other waveforms – like the square wave, triangular. These signals can be
generated from the signal generator block.
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Figure12-6-1-4: Property box of the Spectrum Analyzer
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Figure 12-6-1-5 Modulation
The spectrum of the output when km=1 was shown below. The following figure demonstrates the waveform
multiplication. A sine wave of 1kHz is generated using a sine wave generator and multiplied with a replica
signal. The input signal and the output are shown in figures. The input signal as generated by the sine wave
is shown in figure.
The output of the multiplier is shown in figure and the spectral output is shown in figure. It can be seen that
the output of the multiplier in time domain is basically a sine wave but doesn’t have the negative sides. Since
they get cancelled out in the multiplication.
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Figure 12-6-1-7 Multiplier Spectrum
If a DC component was present in the input waveform, then
𝑦 = 𝑘𝑚 ∗ (𝑐𝑜𝑠(2𝜋(1,000)𝑡) + 𝑉𝑑𝑐 )2
2
= 𝑘𝑚 ∗ (𝑐𝑜𝑠 2 (2𝜋(1,000)𝑡) + 2𝑐𝑜𝑠(2𝜋(1000)𝑡) ∗ 𝑉𝑑𝑐 + 𝑉𝑑𝑐 )
The effect of adding a dc component to the input has the overall effect of raising the amplitude of the 2KHz
component and decreases the 2KHz component. However, for a value of Vdc = 0.1V, the 1KHz component
reduces and for any other increase in the Vdc value, the 1KHz component increases.
1
0.9
0.8
0.7
0.6
Magnitude
0.5
0.4
0.3
0.2
0.1
0
-2500 -2000 -1500 -1000 -500 0 500 1000 1500 2000 2500
Frequncy
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2.7 Lab Outcomes
At the end of this lab you should know following:
1. Simulink Handling for Communication Systems
2. Building communication systems in Simulink
3. Time domain analysis of a system in MATLAB
4. Frequency domain analysis of a system in MATLAB
5. Performing modulation through Communications toolbox
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LAB 2 ASSESSMENT
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Fourier Analysis and Spectrum Analysis of Signals
3.1 Apparatus
DC power Supply (726 86)
Frequency Synthesizer (736 031)
Spectrum Analyzer
Oscilloscope
3.2 Introduction
In mathematics, Fourier transform (FT) is an operation that transforms one complex-valued function of a
real variable into another and is used for the aperiodic signals. For periodic signals we use the fourier series.
Aperiodic signals can be converted to periodic signals by limiting them and repeating in a regular pattern
with same limits. In such applications as signal processing, the domain of original function is typically time
and is accordingly called the time domain. That of the new function is frequency, and so the Fourier domain
is often called frequency domain representation of the original function. It describes which frequencies are
present in the original function. This is in a similar spirit to the way that a chord of music can be described
by notes that are being played. In effect, the Fourier transform decomposes a function into series. The term
Fourier transform refers both to the frequency domain representation of a function and to the process or
formula that "transforms" one function into the other.
The Fourier transform/series and its generalizations are the subject of Fourier analysis. In this specific case,
both the time and frequency domains are unbounded linear continua. It is possible to define the Fourier
transform/series of a function of several variables, which is important for instance in the physical study of
wave motion and optics.
3.3 Theory
Every periodic signal which satisfies Dirac conditions can be represented by simple sine waves of different
frequency and amplitude. The numeric value of the frequency and the amplitude can be obtained by taking
the Fourier transform of the signal.
For our case frequency is not important, because it has nothing to do with the shape of the signal. It is fixed
in the frequency synthesizer. (e.g. if the frequency of first harmonic is “f” than the frequency of 2nd harmonic
is “2f” and so on). We are interested in Phase and amplitude only. Frequency Synthesizer (736 031) has been
set with cos terms so
For cos terms set phase = 0
For sin terms set phase = 90 (so that cos becomes sin)
For –cos terms set phase = 180
For –sin terms set phase = 270
3.4 Experiment
You have been assigned a signal (time domain) as mentioned below, take its Fourier transform to obtain the
numeric value of magnitude and frequency. This gives us the fourier series. Set these parameters on the
“Frequency Synthesizer (736 031)”. Attach the output of the synthesizer to the Oscilloscope and draw the
signal below.
Now we shall also see the effect of using more number of Fourier terms (harmonics). Creating the
symmetrical square wave on Frequency Synthesizer shall be shown to you in the lab, and then you have to
create a wave as assigned by the lab instructor.
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Create your signal with all harmonics on, draw it and then starting from the highest frequency harmonic
switch off the harmonic component one by one and draw the result shown on the screen of the oscilloscope,
in the are given below.
TASK:
The Symmetrical Square wave signal: Fourier series of the wave is given as follows:
4A 1 1 1
[sin(2πft) + sin(2π3ft) + sin(2π5ft) + sin(2π7ft) … ]
π 3 5 7
_____________________________________________________
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n Phase Degree S(n) [Amplitude] V
1
2
3
4
5
6
7
8
With _____ harmonics off With all harmonics off except ____
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LAB 3 ASSESSMENT
Date: _________________
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Amplitude Modulation and Demodulation of Signals
4.1 Apparatus
EMONA 101 Trainer Kit
4.2 Introduction
In an amplitude modulation (AM) communications system, speech and music are converted into an electrical
signal using a device such as a microphone. This electrical signal is called the message or baseband signal.
The message signal is then used to electrically vary the amplitude of a pure sine wave called the carrier
usually has a frequency that is much higher than the message's frequency. Figure below shows a simple
message signal and an unmodulated carrier. It also shows the result of amplitude modulating the carrier with
the message. Notice that the modulated carrier's amplitude varies above and below its unmodulated
amplitude.
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Figure 5-2-1-1: Envelope Detection
4.3 Mathematical Modeling
In Communication Theory, the mathematical model that defines The AM signal is:
When the message is a simple sine wave (like in Figure 4-2-1) the equation's solution (which necessarily
involves some Trigonometry that is not shown here) tells us that the AM signal consists of three sine waves:
In other words, for every sine wave in the message, The AM signal includes a pair of sine waves - one above
and one below the carrier's frequency. Complex message signals such as speech and music are made up of
thousands sine waves and so the AM signal includes thousands of pairs of sine waves straddling carrier.
These two groups of sine waves are called the side bands and so AM is known as double-sideband full carrier
(DSBFC). Importantly, it's clear from this discussion that the AM signal doesn't consist of any signals at the
message frequency. This is despite the fact that the AM signal's envelopes are the same shape as the message.
4.4 The Experiment
In the experiment the main aim is to visualize how amplitude modulation is achieved practically satisfying
the mathematical model and communication theory.
1. Set the Trigger Source control to the CH1 (or INT) position.
2. the Mode control is set to the CH1 position.
3. Locate the Adder module and turn its G and g controls fully anti-clockwise.
4. Locate the Variable DCV module and turn its DC Voltage control almost fully anti-clockwise.
5. Set the connections as shown in Figure 5-4-1:
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Figure 5-4-1: Message for Amplitude Modulation
The set-up in Figure 5-4-1 can be represented by the block diagram in Figure 5-4-2. It implements the
highlighted part of the equation:
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Now perform following steps for Amplitude modulation of the message:
1. Set the scope’s Channel 1 Vertical Attenuation control to the 0.5V/div position.
2. Use the scope’s Channel 1 Vertical Position control to move the trace so that it lines up with the
horizontal line in the middle of the scope’s screen.
3. While watching the Adder module’s output on the scope, turn its g control clockwise until the DC
level is 1V.
4. While watching the Adder module’s output on the scope, turn its G control clockwise to obtain a
1Vp-p sine wave.
5. Now modify the above set-up as shown below. Do not disconnect the previous connections.
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TASK:
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Figure 5-5-1: Amplitude Demodulation Block Diagram
As you can see, the rectifier stage chops the AM signal in half letting only one of its envelopes through (the
upper envelope in this case but the lower envelope is just as good). This signal is fed to an RC LPF which
tracks the peaks of its input. When the input to the RC LPF is a rectified AM signal, it tracks the signal’s
envelope. Importantly, as the envelope is the same shape as the message, the RC LPF’s output voltage is also
the same shape as the message and so the AM signal is demodulated.
A limitation of envelope detector shown in Figure 5-2-1-1 is that it cannot accurately recover the message
from over-modulated AM signals. To explain, recall that when an AM carrier is over-modulated the signal’s
envelope is no-longer the same shape as the original message. Instead, the envelope is distorted and so, by
definition, this means that the envelope detector must produce a distorted version of the message.
4.6 The Experiment
Modify Figure 5-4-3 as shown below. Make sure not to disassemble the previous connections.
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4.7 Lab Outcomes
At the end of this lab you should know following:
1. Differentiation between the message signal and carrier signal.
2. Modulation of signal amplitude.
3. Demodulation of signal amplitude.
4.8 Simulation Task
1. Implement AM modulation demodulation in MATLAB. Message signal is 2KHz Sine wave
whereas carrier is 8 KHz Sine wave.
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LAB 5 ASSESSMENT
Date: _________________
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Double Side-Band Suppressed Carrier Modulation
Demodulation of Signals
5.1 Apparatus
EMONA 101 Trainer Kit
5.2 Introduction
DSBSC is a modulation system similar but not exactly AM (which was explored in last experiment).
Like AM, DSBSC uses a microphone or some other transducer to convert speech and music to an electrical
signal called the message or baseband signal. The message signal is then used to electrically vary the
amplitude of a pure sinewave called the carrier. And like AM, the carrier usually has a frequency that is
much higher than the message’s frequency.
Figure 6-2-1 below shows a simple message signal and an unmodulated carrier. It also shows the result of
modulating the carrier with the message using DSBSC.
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Figure 6-2-2: DSBSC Envelope Detection
Instead, alternating halves of the envelopes form the same shape as the message as shown in Figure 6-2-3.
One with a frequency equal to the sum of the carrier and message frequencies
One with a frequency equal to the difference between the carrier and message frequencies
Importantly, the DSBSC signal doesn’t contain a sinewave at the carrier frequency. This is an important
difference between DSBSC and AM.
That said, as the solution to the equation shows, DSBSC is the same as AM in that a pair of sinewaves is
generated for every sinewave in the message. And, like AM, one is higher than the unmodulated carrier’s
frequency and the other is lower. As message signals such as speech and music are made up of thousands of
sinewaves, thousands of pairs of sinewaves are generated in the DSBSC signal that sit on either side of the
carrier frequency. These two groups are called the sidebands.
So, the presence of both sidebands but the absence of the carrier gives us the name of this modulation method
- double-sideband, suppressed carrier (DSBSC).
The carrier in AM makes up at least 66% of the signal’s power but it doesn’t contain any part of the original
message and is only needed for tuning. So by not sending the carrier, DSBSC offers a substantial power
saving over AM and is its main advantage.
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5.4 The Experiment
Use the same equipment as in experiment for Amplitude Modulation. Set-up the assembly as shown in Figure
6-4-1 to generate a DSBSC:
1. Adjust the scope’s Timebase control to view two or so cycles of the Master
Signals module’s 2kHz SINE output.
2. Set the scope’s Mode control to the DUAL position to view the DSBSC
signal out of the Multiplier module as well as the message signal.
3. Set the scope’s Channel 1 Vertical Attenuation control to the 1V/div
position and the Channel 2 Vertical Attenuation control to the 2V/div
position.
4. Draw the waveform of DSBSC below:
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5.5 DSBSC Demodulation
Unfortunately, the envelope detector cannot be used to demodulate a DSBSC signal.
To understand why, recall that the envelope detector outputs a signal that is a copy of its input’s envelope.
This works well for demodulating AM because the signal’s envelopes are the same shape as the message that
produced it in the first place. However, recall that a DSBSC signal’s envelopes are not the same shape as the
message.
Instead, DSBSC signals are demodulated using a circuit called a product detector (though product
demodulator is a more appropriate name) and its basic block diagram is shown in Figure 6-5-1. Other names
for this type of demodulation include a synchronous detector and switching detector.
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Figure 6-5-1: DSBSC Demodulation Block Diagram
As its name implies, the product detector uses multiplication and so mathematics is necessary to explain its
operation. The incoming DSBSC signal is multiplied by a pure sine wave that must be the same frequency
as the DSBSC signal’s suppressed carrier. This sine wave is generated by the receiver and is known as the
local carrier.
Importantly, recall that DSBSC generation involves the multiplication of the message with the carrier which
produces sum and difference frequencies. That being the case, this information can be substituted for the
DSBSC signal and the equation rewritten as:
When the equation is solved, we get four sine waves with the following frequencies:
(If you’re not sure why these sine waves are produced, it’s important to remember that for every pair of pure
sine waves that are multiplied together, two completely new sine waves are generated. One has a frequency
equal to the sum of the original sine waves’ frequencies and the other has a frequency equal to their
difference.)
Importantly, notice that two of the products are sine waves at the message frequency. In other words, the
message has been recovered. As the two message signals are in phase, they simply add together to make one
larger message.
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5.7 The Experiment
Keeping the settings as mentioned by the instructor (if not you may ask) setup the following assembly:
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Draw the observed waveform:
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Figure 6-7-2: DSBSC Message Recovery Connections
The additions to the set-up in Figure 6-7-1 can be represented by the block diagram in Figure 6-7-3. The
multiplier module and the RC Low-pass Filter on the Utilities module are used to implement a product
detector which demodulates the original message from the DSBSC signal.
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Figure 6-7-4: DSBSC Complete Demodulation Block Diagram
5.8 Lab Outcomes
At the end of this lab you should know following:
1. Differentiation between the DSBSC and AM.
2. Double Side Band Suppressed Carrier.
3. Recovering the message from DSBSC Demodulation.
5.9 Simulation Task
2. Implement DSB-SC modulation demodulation in MATLAB. Message signal is 2KHz Sine wave
whereas carrier is 8 KHz Sine wave.
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LAB 5 ASSESSMENT
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Single Side-Band Suppressed Carrier Modulation
Demodulation of Signals
6.1 Apparatus
EMONA 101 Trainer Kit
6.2 Introduction
Comparing the two communications systems considered earlier in this manual, DSBSC offers considerable
power savings over AM (at least 66%) because a carrier is not transmitted. However, both systems generate
and transmit sum and difference frequencies (the upper and lower sidebands) and so they have the same
bandwidth as each other.
As its name implies, the Single Sideband Suppressed Carrier (SSBSC or just SSB) system transmits only
one sideband. In other words, SSB transmits either the sum or the difference frequencies but not both.
Importantly, it doesn’t matter which sideband is used because they both contain all of the information in the
original message.
In transmitting only one sideband, SSB requires only half the bandwidth of DSBSC and AM which is a
significant advantage.
Figure 7-2-1 shows a simple message signal and an unmodulated carrier. It also shows the result of
modulating the carrier with the message using SSBSC. If you look closely, you’ll notice that the modulated
carrier is the same shape as the message but is not the same frequency as either the message or the carrier.
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6.3 Generating SSBSC
6.3.1 Filter Method:
A common method of generating SSB simply involves generating a DSBSC signal then using a filter to pick
out and transmit only one of the sidebands. This is known as the filter method. However, the two sidebands
in a DSBSC signal are close together in frequency and so specialised filters must be used. This means that
the filters for non-mainstream applications can be expensive.
SSB = (message × carrier) + (message with 90° of phase shift × carrier with 90° of phase shift)
If you look closely at the equation you’ll notice that it’s the sum of two multiplications. When the message
is a simple sine wave the solution of the two multiplications tells us that four sine waves are generated.
Depending on whether the message’s phase shift is +90° or -90° their frequencies and phase differences are:
These… Or these…
Carrier + message Carrier + message
Carrier - message Carrier - message
Carrier + message Carrier + message (180° phase shifted)
Carrier - message (180° phase shifted) Carrier – message
Regardless of whether the message’s phase shift is +90° or -90°, when the four sine waves are added together,
two of them are in phase and add together to produce one sine wave (either carrier + message or carrier –
message) and two of the sine waves are phase inverted and completely cancel. In other words, the process
produces only a sum or difference signal (that is, just one sideband).
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6.4 The Experiment
In this experiment you’ll use the Emona Telecoms-Trainer 101 to generate an SSB signal by implementing
the mathematical model for the phasing method. You’ll then use a product detector (with a stolen carrier) to
reproduce the message.
First we have to generate the SSBSC. For this follow the following steps:
1. Set up the scope per the instructions in Experiment 1. Ensure that its Trigger Source control is set to
the CH1 (or INT) position.
2. Set the scope’s Mode control to the DUAL or EXT position.
3. Setup the following assembly:
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1. Adjust the scope’s Timebase control to view two or so cycles of the two message signals.
2. Set the Phase Shifter module’s Phase Change control to the 180° position.
3. Set the Phase Shifter module’s Phase Adjust control to about the middle of its travel.
4. Check that the two message signals are out of phase with each other. (Note: At this stage, it doesn’t
matter what the phase difference is.)
5. Modify the set-up as shown in Figure 7-4-3. (Remember: Dotted lines show leads already in place.)
1. Use the scope to check that the lower Multiplier module’s output is a DSBSC signal.
2. Disconnect the scope’s Channel 2 input from the lower Multiplier module’s output and connect it to
the upper Multiplier module’s output.
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3. Check that the upper Multiplier module’s output is a DSBSC signal as well.
4. Locate the Adder module and set its G and g controls to about the middle of their travel.
5. Modify the assembly according to Figure 7-4-5.
The signal out of the SSB modulator is highly unlikely to be an SSB signal at this stage. The reasons for this
are: the two messages are probably not exactly 90 degrees out of phase or the gain of adder module for both
of its inputs is not same.
The next part of the experiment gets you to make the fine adjustments necessary to turn the set-up into a true
SSB modulator.
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For this follow the following steps:
1. Turn the Adder module’s G control fully anti-clockwise. Note: This removes the signal on the Adder
module’s A input from the set-up’s output.
2. Adjust the Adder’s g control to obtain a 4Vp-p output.
3. Disconnect the patch lead to the Adder module’s B input.Note: This removes the signal on the Adder
module’s B input from the set-up’s output.
4. Adjust the Adder’s G control to obtain a 4Vp-p output.
5. Reconnect the patch lead to the Adder’s B input.
The gain of the Adder module’s two inputs will now be nearly the same. Now:
6. Vary the Phase Shifter module’s Phase Adjust control left and right and observe the effect on the
envelopes of the set-up’s output.
7. Adjust the Phase Shifter module’s Phase Adjust control to make the envelopes as “flat” as possible.
The phase difference between the two messages will now be nearly 90°.
8. Tweak the Adder module’s G control to see if you can make the output’s envelopes flatter.
9. Tweak the Phase Shifter’s Phase Adjust control to see if you can make the output’s envelopes
flatter still.
Once the envelopes are as flat as you can get, the gain of the Adder module’s two inputs are very close to
each other and the phase difference between the two messages are very close to 90°.
Use the scope’s Horizontal Position control to examine the viewable length of SSB signal closely.
1. Locate the Tuneable Low-pass Filter module and set its Gain control to about the middle of its travel.
2. Turn the Tuneable Low-pass Filter module’s soft Cut-off Frequency Adjust control fully.
Modify the assembly of connections as in figure 7-5-1:
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Figure 7-5-1: SSBSC Demodulation Connections
The above assembly can be shown in block diagram as in Figure 7-5-2. The Multiplier module and the RC
Low-pass Filter on the Utilities module are used to implement a product detector which demodulates the
original message from the SSB signal.
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Draw the observed waveform:
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LAB 7 ASSESSMENT
Date: _________________
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Frequency Modulation of Signals
7.1 Apparatus
EMONA 101 Trainer Kit
7.2 Introduction
A disadvantage of the AM, DSBSC and SSB communication systems is that they are susceptible to picking
up electrical noise in the transmission medium (the channel). This is because noise changes the amplitude of
the transmitted signal and the demodulators of these systems are affected by amplitude variations.
As its name implies, frequency modulation (FM) uses a message’s amplitude to vary the frequency of a
carrier instead of its amplitude. This means that the FM demodulator is designed to look for changes in
frequency instead. As such, it is less affected by amplitude variations and so FM is less susceptible to noise.
This makes FM a better communications system in this regard.
There are several methods of generating FM signals but they all basically involve an oscillator with an
electrically adjustable frequency. The oscillator uses an input voltage to affect the frequency of its output.
Typically, when the input is 0V, the oscillator outputs a signal at its rest frequency (also commonly called
the free-running or center frequency). If the applied voltage varies above or below 0V, the oscillator’s output
frequency deviates above and below the rest frequency. Moreover, the amount of deviation is affected by the
amplitude of the input voltage. That is, the bigger the input voltage, the greater the deviation.
Figure 8-2-1 shows a simple message signal (a bipolar square wave) and an un modulated carrier. It also
shows the result of frequency modulating the carrier with the message.
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3. The message alternates above and below 0V, the signal’s frequency goes above and below the
carrier’s frequency. (Note: It’s equally possible to design an FM modulator to cause the frequency to
change in the opposite direction to the change in the message’s polarity.)
Before discussing FM any further, an important point must be made here. A square wave message has been
used in this discussion to help you visualize how an FM carrier responds to its message. In so doing, Figure
7-2-1 suggests that the resulting FM signal consists of only two sine waves (one at a frequency above the
carrier and one below). However, this isn’t the case.
This highlights one of the important differences between FM and the modulation schemes discussed earlier.
The mathematical model of an FM signal predicts that even for a simple sinusoidal message, the result is a
signal that potentially contains many sine waves. In contrast, for the same sinusoidal message, an AM signal
would consist of three sine waves, a DSBSC signal would consist of two and an SSBSC signal would consist
of only one. This doesn’t automatically mean that the bandwidth of FM signals is wider than AM, DSBSC
and SSBSC signals (for the same message signal). However, in the practical implementation of FM
communications, it usually is.
Finally, when reading about the operation of an FM modulator you may have recognized that there is a
module on the Emona Telecoms-Trainer 101 that operates in the same way - the VCO module. In fact a
voltage-controlled oscillator is sometimes used for FM modulation (though there are other methods with
advantages over the VCO but in this lab we will be focusing FM through VCO).
7.3 The Experiment
In this experiment you’ll generate a real FM signal using the VCO module on the Emona Telecoms-Trainer
101. First you’ll set up the VCO module to output an unmodulated carrier at a known frequency. Then you’ll
observe the effect of frequency modulating its output with a square wave. Finally, you’ll use a sine wave to
observe the spectral composition of an FM signal (in the time domain).
In this experiment FM modulating a square wave will be analyzed. For that purpose follow the steps below:
1. Locate the VCO module and turn its Gain control to about two thirds of its travel (about the position
of the number 2 on a clock face).
2. Set the VCO module’s Frequency Adjust control to about the middle of its travel.
3. Set the VCO module’s Range control to the LO position.
4. Now connect the setup as shown in Figure 8-3-1:
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6. Adjust the VCO module’s Frequency Adjust control so that one cycle of its output is exactly 5
1
divisions. (Note: This sets the VCO module’s rest frequency to 10kHz (proof: 10,000 ))
5 20
7. Set the scope’s Time base control to the 0.1ms/div position. (Note: This will show about ten cycles
of the VCO module’s SINE output.)
8. Now modify the setup as shown in Figure 8-3-2:
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Figure 8-3-1-1: FM-Spectral Composition of Sine Waves Connections
You should now be able to see a display that looks as shown in Figure 8-3-1-2:
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Draw the observed waveform:
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LAB 8 ASSESSMENT
Date: _________________
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Frequency Demodulation of Signals
8.1 Apparatus
EMONA 101 Trainer Kit
8.2 Introduction
There are as many methods of demodulating an FM signal as there are of generating one. Examples include:
the slope detector, the Foster-Seeley discriminator, the ratio detector, the phase-locked loop (PLL), the
quadrature FM demodulator and the zero-crossing detector. It’s possible to implement several of these
methods using the Emona Telecoms-Trainer 101 but, for an introduction to the principles of FM
demodulation, only the zero-crossing detector is used in this experiment.
8.2.1 Zero Crossing Detector
The zero-crossing detector is a simple yet effective means of recovering the message from FM signals. Its
block diagram is shown in Figure 9-2-1-1.
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Figure 9-2-1-2: Zero Crossing Detector Waveforms
Pulse trains are actually made up of sine waves and, in the case of Figure 9-2-1-2, a DC voltage. The size of
the DC voltage is affected by the pulse train’s duty cycle. The greater its duty cycle, the greater the DC
voltage.
That being the case, when the FM signal in Figure 9-2-1-2 above switches between the two frequencies, the
DC voltage that makes up the rectangular wave out of the ZCD changes between two values. In others words,
the DC component of the rectangular wave is a copy of the square wave that produced the FM signal in the
first place. Recovering this copy is a relatively simple matter of picking out the changing DC voltage using
a low-pass filter.
8.3 Experiment
In this experiment you’ll use the Emona Telecoms-Trainer 101 to generate an FM signal using a VCO. Then
you’ll set-up a zero-crossing detector and verify its operation for variations in the message’s amplitude.
To experiment with FM demodulation you need an FM signal. The first part of the experiment gets you to
set one up. To make viewing the signals around the demodulator possible, we’ll start with a DC voltage for
the message.
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Figure 9-3-1: FM Modulator Step 1
5. Set the scope’s Timebase control to view two or so cycles of the VCO module’s SINE output.
6. Adjust the VCO module’s SINE output to 10kHz. (Note: You do this by adjusting the signal’s period
1
to 100µs (recall that P ))
f
7. Set the scope’s Trigger Source control to the CH2 position.
8. Set the scope’s Channel 1 and Channel 2 Input Coupling controls to the DC position.
9. Set the scope’s Mode control to the DUAL position.
10. Connect the setup as shown in Figure 9-3-2
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The block diagram representation is of FM signal is shown in Figure 9-3-3.The Variable DCV module is
being used to provide a simple DC message and the VCO module implements an FM modulator with a carrier
frequency of 10kHz.
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Figure 9-4-1: Zero Crossing Detector Connections
Now,
5. Set the scope’s Timebase control to the 2µs/div position.
6. Adjust the Twin Pulse Generator module’s Width control for an output pulse that is 12µs long. (Note:
Generally speaking, the longer the pulse the greater it’s DC component and, in the case of the zero-
crossing detector, the greater the size of the recovered message. However, the pulses cannot be too
long otherwise the circuit’s operation breaks down due to other performance parameters of the TPG
module. In this case, 12µs is a compromise.)
7. Return the scope’s Timebase control to its previous position. (Tip: If you’re not sure, try 50µs/div.)
8. Setup the assembly as in Figure 9-4-2 to the FM modulator. (Remember: Dotted lines show leads
already in place.)
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The FM Demodulator can be shown in Figure 9-4-3. The comparator on the Utilities module is used to clip
the FM signal, effectively turning it into a square wave. The positive edge-triggered Twin Pulse Generator
module is used to implement the zero-crossing detector. To complete the FM demodulator, the Baseband
LPF on the Channel Module is used to pick-out the changing DC component of the Twin Pulse Generator
module’s output.
FM Demodulated
ZCD
signal message
To Ch.2
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8.5 Lab Outcomes
At the end of this lab you should know following:
1. Methods of frequency demodulation
2. Zero crossing detector setup
3. Frequency demodulation using zero crossing detector.
8.6 Simulation Task
1. Implement Frequency Modulation demodulation in MATLAB
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LAB 9 ASSESSMENT
Date: _________________
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Pulse Code Modulation
9.1 Apparatus
DC power Supply (726 86)
Function generator (726 96)
Oscilloscope
PCM modulator (736 101)
9.2 Introduction
PCM is the most widely used and most useful form of pulse modulations. It is a tool for converting analogue
signal to digital signal. There are two main steps in digitizing an analog signal. First step is to take the
appropriate samples of the signal and the second step is the quantization. Sampling is performed with the
help of a pulse train. Quantization usually requires other methods which are explained below:
9.2.1 PAM Signal
It is the signal which is only discrete in time and in amplitude it is constant. So, in a true sense it is not analog
and not completely discrete. Actually it is an intermediate stage. In order to convert this into a digital signal,
quantization and coding have to be performed on the signal.
9.2.2 Quantization
By quantization we mean the narrowing down of all possible signal values to finite number. Whereas the
PAM signal can assume any random signal value between the modulating limits of the PAM modulator, the
quantified signal generally demonstrates a stepped shape. The modulating range of the PCM modulator is
broken down into a fixed number of intervals. Each interval is represented by assigning one signal value
only. This is set representatively for all signal values, which occur in their respective interval.
The quantization process takes an infinite number of all possible, continuous signal values and reduces them
to a finite number. This quantization process also referred to as value discretization, always results in an
additional error. This inevitable quantization error can be considered as a noise phenomenon and is this called
quantization noise. The quantization distortion is influenced by the input signal type. In addition to this the
magnitude and number of quantization intervals also play a decisive role.
The quantization interval can be either equidistant discrete or logarithmic steps. In the case of equidistant
quantization intervals this is referred to as linear quantization. In the case of logarithmic steps this is called
non-linear quantization.
9.2.3 Encoding
Evidently a sampled, discrete value signal of this kind is unsuitable for transmission via a real, noise distorted
transmission channel. The digital signal at the output of the quantifier requires a shape better suited to the
relationships prevailing on the channel. The conversion process required for this is called encoding. Coding
constitutes the allocation of a specific mapping between the elements of two different character sets.
9.2.4 Companding
In the analog world companding is a method for improving audio reproduction by altering the dynamic range
of the signals. On outgoing transmission, it raises the amplitude of weak signals and lowers the amplitude of
strong signals. On incoming transmission, it restores the signal to its original form. Such techniques are used
in dbx and Dolby noise reduction, AM, FM and SSB radio as well as in voice and musical instrument
amplification to reduce feedback. In the digital world companding is a compression technique used in pulse
code modulation (PCM). It reduces the number of bits used to represent digital voice samples in the loudest
parts of the conversation. In telecommunication and signal processing companding (occasionally called
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compansion) is a method of mitigating the detrimental effects of a channel with limited dynamic range. The
name is a portmanteau of compressing and expanding.
5 5
Output Region
Output Region
0 0
-5 -5
-10 -10
-10 -5 0 5 10 -10 -5 0 5 10
Input Region Input Region
Note the reading in the tables and draw the graph between U1 and U2
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Now disable the three least significant bits and change the input voltage from -9.5 to + 9.5. Record the
reading in the following table and draw the graph between U1 and U2.
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U1 [V] U2 [V] Code U1 [V] U2 [V] Code
-9.5 1
-9 2
-8 3
-7 4
-6 5
-5 6
-4 7
-3 8
-3 9
-1 9.5
0
In the previous experiments you have separately recorded compressor and expander characteristics
respectively. Each of the two transmission functions, by themselves, lead to signal distortion. However, in
standard operation they are both active simultaneously. Select the non-linear mode on both the modulator
and demodulator. Record the readings and plot the graph as done before.
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LAB 9 ASSESSMENT
Date: _________________
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Pulse Width Modulation Demodulation
10.1 Apparatus
EMONA 101 Trainer Kit
Function Generator
10.2 Introduction
when an analog signal is periodically sampled, the sampled signal is a complex waveform that consists of the original
analog signal plus aliases (also known es images). This has been comprehensively demonstrated in another DATEx
experiment using pulse amplitude modulation (PAM) but the fact is true for all pulse modulation schemes. Another
type of pulse modulation is called pulse-width modulation (PWM) or sometimes pulse- duration modulation
(PDM). To understand PWM, let’s consider a simple method for generating it that involves a comparator with
a sawtooth waveform for one of its inputs. This is shown in Figure 1 below.
Figure 1
Recall that the comparator amplifies the difference between the voltages on its two inputs by an extremely
large amount. So, when the instantaneous analog voltage is smaller than the instantaneous sawtooth voltage,
the comparator ‘s output is logic-0. And when the instantaneous analog voltage is larger than the
instantaneous sawtooth voltage, the comparator ‘s output is logic-1. That being the case, the comparator ‘s
output is a pulse train with a pulse-width that is a function of the size of the voltage on the analog input.
Figure 2 on the next page demonstrates this by comparing the comparator ‘s output for two DC voltages on
the analog input.
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An example of the effect on the comparator’s output for a sinewave analog input is shown in
Figure 3 below.
Figure 3
At this point it‘s useful to highlight the difference between PWM and PAM signals in the time
domain. Recall that PAM produces a series of pulses with a fixed width but with an amplitude that
is proportional to the instantaneous analog input voltage. PWM on the other hand produces a
series of pulses at only one voltage (other than zero) but with a pulse-width that is proportional
to the instantaneous analog input voltage
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A comparison of PWM and PAM signals in the frequency domain shows an interesting similarity between the
two butt also a critical difference. Both signals contain a copy of the original message and they both contain
modulated carriers at each of the pulse’s harmonics. However, the modulation of these “harmonic carriers“ is
different. The mathematical analysis of PAM signals shows that their harmonic carriers are DSBSC modulated
Whereas the mathematical analysis of PWM signals (not provided here) shows that their harmonic carriers are
phase modulated. As such, the signals spectra are quite different which explains why the signals look so
different in the time domain.
Despite the differences in the spectral composition of PWM and PAM signals, the fact that they both contain
q copy of the message means that the message can be recovered in the same way (that is, by using a low-
pass filter). Of course, for this demodulation technique to work, the Nyquist Semple Rate must be observed
in the first instance.
For this experiment you’ll set up a pulse-width modulator (PWM) using the Emona DATEx comparator
module. You’ll then verify its operation and determine its Limits using a DC inputst voltage. Next you’ll
observe the PWM‘s operation for an AC message signal using a 2kHz tone then speech. Finally, you’ll
investigate the effects of reducing the effective sample rate below the Nyquist Sample Rate for the
message.
Part A - Generating the sawtooth waveform
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Plot the generated waveform in following table
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Part B - Verifying the operation of the PWM using a DC analog input voltage
The next part of the experiment gets you to verify the operation of the comparator as a
pulse-width modulator using a DC analog input voltage. You’ll also experimentally determine
the lower and upper threshold voltages of the analog input that the PWM can reliably convert.
The change to this set-up can be represented by the block diagram in Figure 11 on the
next page. The Master Signals module’s 2kHz SINE output is connected to the pulse-
width modulator’s input via the Adder module. The Adder module is used to add a DC
Question
Why does the mark-space ratio of the PWM‘s output signal become more uniform as
the size of the message is reduced*
Question
What’s the name of this distortion*
Date: _________________
11.1 Apparatus
Emona 101 Trainer Kit
11.2 Introduction
The methods of communication or modulation which we have studied up till now are
valid for analog communication. To use such techniques for digital communication we
have to use the crunches of shift keying. Information can be transmitted purely by
switching a carrier on and off. The switching of carrier is called keying. As such you
could interpret the transmission of a message using smoke or light signals as keying. It is
typical of this method that the receiver or recipient of the message cannot decode the
message, unless he has the right code. In electrical communication engineering keying
means the modulation of a harmonic carrier with a digital signal. A harmonic carrier with
the form:
𝑆𝐶 (𝑡) = 𝐴𝐶 cos(2𝜋𝑓𝐶 𝑡 + Φ𝐶 )
The above equation has three parameters, all of which can be used for modulation.
𝐴𝐶 : Amplitude
𝑓𝐶 : Frequency
Φ𝐶 : Phase
Accordingly, a distinction is drawn between the following 3 types of keying
ASK: Amplitude Shift Keying
FSK: Frequency Shift keying
PSK: Phase Shift keying
The modulating, digital signal is binary, i.e. there are only 2 states which should be
transmitted using the harmonic carrier. Under this prerequisite this means that:
ASK: Alternating between 2 amplitudes values of the carrier, e.g. changing from
0 to A
FSK: Alternating between 2 defined carrier frequencies f1 and f2.
PSK: Abrupt phase change of the carrier e.g. from Φ1 = 0° to Φ2 = 180°.
11.3 The Experiment
11.3.1 ASK Modulation
1. Use 2KHz Sine as carrier and 2KHz digital signal as message signal
2. Locate sequence generator and duel switch
3. Set up the apparatus as shown in 0000
4. Connect modulated signal to oscilloscope
Notice that the ASK signal’s carrier and the Sequence Generator module’s clock are
the same frequency (2kHz). Moreover, notice that they’re from the same source –
the Master Signals modul.
_____________________________________________________________________
Question 2. What is the ASK signal’s voltage when the digital signal is logic-0?
____________________________________________________________________
Question 1. What’s the name for the VCO output frequency that corresponds with logic-
1s and 0s in the digital data?
____________________________________________________________________
Question 2. Based on your observations of the FSK signal, which of the two is the higher
frequency? Explain your answer?
_____________________________________________________________________
____________________________________________________________________
Question 1. What happens to the BPSK signal on the data stream’s logic transitions?
____________________________________________________________________
With a choice of ASK, FSK and BPSK you might be wondering about which system
you’ll most likely see. All other things being equal, BPSK is the best performing system
in terms of its ability to ignore noise and so it produces the fewest errors at the receiver.
FM is the next best and AM is the worst. On that basis, you’d expect that BPSK is the
preferred system.
11.4 Lab Outcomes
At the end of this lab you should know following:
1. Amplitude Shift Keying- Modulation and Demodulation
2. Phase Shift Keying- Modulation and Demodulation
3. Frequency Shift Keying- Modulation and Demodulation
11.5 Simulation Task
1. Implement Shift Keyings in MATLAB
Date: _________________
12.1 Apparatus
MC 14046B (Motorola)
Oscilloscope
Function generator
Multi meter
Double power supply (DC)
Coupling board
12.2 Introduction
“Phase Lock loop is an electronically designed circuit which has linear output that is
phased locked with the phase of input signal with difference registered by Phase
Comparator controlled using Voltage Control Oscillator. The phase gets locked for
certain band of frequencies while shows non-linear behavior for other set of
Frequencies.”
Phase Lock Loop (PLL) has several applications in communication network of Radar
and GSM architecture while its major application is the demodulation of the FM
generated Signal. Based on its wide range of usage, many industries have generated
integrated circuits that have all in all solution including internal VCO, Phase Comparator
and LP filters.
In figure below the flow of processes that are taken in consideration for the design of
Phase Lock Loop are represented. It indicates 3 major steps starting from FM signal
Reception, in our case this is generated using Function Generator which is fed to VCO
section. Secondly, the signal is passed through the Low Pass Filter to take out the high
frequency components followed by gain or amplification section to get the original
signal at desirable value. This section has its advantages in advantages in real time
application and in major GSM/Satellite communication but in our experiment, it has
been avoided to reduce the complexity of the system. The Signal generated after step 2
is then fed and compared with the reference signal. Giving us linear state for certain
band of frequencies while the output behaves rather abnormally for other set of
frequencies. In doing so; we get the Acquisition and the Holding range of the Phase Lock
Loop. The important thing to consider here is that the acquisition range is always
imbedded in the holding range. The holding range is always greater than the acquisition
range.
The figure shown above is the block diagram of the Phase Lock Loop with the
Input/Reference Signal as 𝑋𝑟 (t) and the output signal from VCO to be 𝐸𝑜 (t). Both of these
signals having varying phase whose difference is measured in the Phase Detector .
Mathematically the two signals can be represented as
𝑋𝑟 (t) = 𝐴𝑐 cos[ω𝑐 (t) +Ф(t)]
Where Ф(t) and θ(t)] are the phase of the input and output signal respectively
The phase difference is calculated and then the accumulated signal is given to Low Pass
Filter (LPF) which removes the high frequency component of 2 ω𝑐 (t). In some of the
application a Notch filter may be required to cancel out a single unwanted frequency and
its ripple if there are any, generally notch is used to avoid electrical interference which
could be of 50Hz or 60 Hz. Similarly, there could be cascaded Low Pass Filters to
stabilize the output but that causes computational complexity of the system. The output
signal from the LPF is indicated by 𝐸𝑑 (t) which is proportional to the phase difference as
indicated in the expression 1.3. 𝐸𝑑 (t) Will behave linearly for small phase differences
while Ad is associated with the multiplier. The two major components i.e. Phase
Comparator and LPF are shown in Figure 1.3.
With the wide range of applications of Phase Lock Loop, Integrated Circuits are made
that have all encompassing solution. All entities that are there for Solution of PLL are
embedded in a single IC. We have used MC14046B Motorola IC in out experiment and
its layout is shown in Figure 3.1[1].
In the given integrated circuit of Motorola there are two phase comparators. First phase
comparator is using a Simple X-OR gate and its output is taken at pin 2 while more
Complex gating is employed for phase comparator with output at Pin13 as shown in
Figure 3.1. An additional voltage divider is also fabricated before the phase comparator
and the input of voltage divider is subjected to reference signal from Function Generator.
The phase difference is calculated by using Oscilloscope phase difference options of
channel 1 and channel 2 connected with Reference Signal and VCO signal respectively.
The overall layout of the design used is represented in the Figure 3.4 while the actual
implementation on Fabrication Board is shown in Figure 3.5.
Phase Lock Loop was approached in systematic manner to better understand the overall
system. The major tasks of it are defined as under.
Voltage Controlled Oscillator (VCO) is fabricated as mentioned in the section 2.2. VDD
is set at Fix peak to peak voltage of 10V while the 𝑉𝐶𝑂𝑖𝑛 is supplied with the varying
Voltage. Table 4.1 indicates the voltage with respect to frequency while Figure 4.1 is
tabular representation of the findings. Frequency is found using Oscilloscope and
Figure4.2 shows the representation of that.
Volts Frequency
[V] [kHz]
0 204.1
1 210.1
2 256.8
3 304.4
4 352.1
5 394.4
6 436.6
7 476.5
8 511.5
9 514.1
10 517.1
600 12
500 10
400 8
300 6 Frequency(kHz)
Volts
200 4
100 2
0 0
1 2 3 4 5 6 7 8 9 10 11
Volts Pin9 VD
[V] [V]
0 0.7
1 0.99
2 2.04
3 2.99
4 4.03
5 4.99
6 6.01
7 7.04
8 8.04
9 8.31
10 9.52
Now by using the Phase Comparator operation of the integrated Circuit the Phase is
compared using the Phase Comparator Operation of Oscilloscope. Now by using the
procedure explained in section 2.4 Phase Difference is compared for both Comparator 1
and Phase Comparator 2 and are represented in Table 4.3 and the graphical representation
of it is shown in Figure 4.3.
600 200
180
500
160
Frequenc
140
400 y(kHz)
120
300 100
Comp. 1
80
200
Phase
60
Diff.
100
40 Comp. 2
20 Phase
0 0 Diff.
1 2 3 4 5 6 7 8 9 10 11
Now by varying the reference frequency the hold in range and the acquisition range are
found out. The phase gets locked after reaching the threshold and gives us one acquisition
value and similarly the second acquisition value is obtained. Acquisition and Hold in
range are indicated in Figure 4.4 while Figure 4.5 shows the system in locked phase.
This experiment is building block of the Modulation System and deals with the
application of Phase Lock Loop. The circuit is designed using Fabrication board and
making use of Motorola IC MC1404B. Linear behavior of frequency with respect to
voltage at 𝑉𝐶𝑂𝑖𝑛 is observed with increasing voltage until the voltage of VDD is
approached and after that it stabilizes itself. Through phase comparator the phase
difference between reference signal and the VCO signal is generated. For comparator 1,
phase increases until 180 degree and after that the holding range is vanished, but the phase
locks itself better when the comparator 2 with complex gating is employed. The
acquisition range is embedded within the holding range.
Date: _________________