Sai Jonin Book

EC310 Notes
Version 2018.2
BOOK 3 of 3
Department of Electrical and Computer Engineering

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Contents
Part III: Wireless ..................................................................................................................................... 381
Chapter 19: Communications Systems, EM Spectrum, and Signals .................................................. 383
CH. 19 Problems ............................................................................................................................. 389
Security Exercise 19 ....................................................................................................................... 391
Chapter 20: Intro to Modulation ......................................................................................................... 401
CH. 20 Problems ............................................................................................................................. 411
Chapter 21: Analog to Digital Conversion ......................................................................................... 421
CH. 21 Problems ............................................................................................................................. 431
Chapter 22: Digital Modulation .......................................................................................................... 441
CH. 22 Problems ............................................................................................................................. 451
Chapter 23: Power Gain and SNR ...................................................................................................... 461
CH. 23 Problems ............................................................................................................................. 469
Chapter 24: Antennas and Propagation............................................................................................... 477
CH. 24 Problems ............................................................................................................................. 491
Chapter 25: Electronic Warfare .......................................................................................................... 499
Appendices .............................................................................................................................................. 511
Basic Linux Commands ...................................................................................................................... 513
The Linux File System ........................................................................................................................ 515
Brief Primer on gdb ........................................................................................................................... 519
File Syntax .......................................................................................................................................... 525
Performing Base Conversions on the T1-nSpire CAS Calculator ...................................................... 527
Authorship Notes
The following faculty members developed content, reviewed the course material and offered many helpful suggestions which
greatly improved the course: Agur Adams, Chris Anderson, Jessie Atwood, Jay Benson, Justin Blanco, Dane Brown, Audrey
Callanan, Gregory Coxson, Rita Doerr, Kevin Fairbanks, Rob Ives, Ryan Kelly, Richard Kopka, Matt Rehberg, Nicholas
Rosasco, John Roth, Keneth Templin, Patrick Vincent, William Ortiz, Ryan Whitty, Jennie Wood, Currie Wooten.
Acknowledgements
Beth Haneke offered expert advice throughout the process and greatly assisted with editing, formatting, and preparing the
manuscript for print.
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Part III: Wireless
In this, the final module of the course, you will be introduced to how digital information,in the form of bits, is moved from
one location to another through free space−that is, without using wires or cables. However, while the ability to move
information through free space makes communication more convenient, it also makes communication more susceptible to
eavesdropping or jamming. Therefore, we will also explore the vulnerabilities of wireless communication.
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Chapter 19: Communications Systems, EM Spectrum, and Signals
Objectives:
(a) Describe the four components of a communications system and the impact on security of using free space as a
communication medium.
(b) Identify communication applications for various bands of the electromagnetic spectrum ranging from extremely low
frequency (ELF) to extremely high frequency (EHF).
(c) Explain the basic properties of a sinusoidal electromagnetic signal (period, frequency, wavelength, phase, and amplitude)
and describe their mathematical relationship.
(d) Define and calculate bandwidth of transmitted signals.
(e) Plot simple (sinusoidal) electromagnetic signals in the time and frequency domains; interpret time- and frequency-domain
plots to determine the associated signals.
Connection to Cyber Security
This chapter marks the beginning of the third part of EC310. In Part I: The Host, we examined how data are stored and accessed
in memory at the machine level and examined the resulting threats against a specific computer, focusing on the buffer overflow
attack. In Part II: Networks, we concentrated on understanding how the Internet works and how networks are just as important
and vulnerable as the individual host computers that reside on them. In Part III: Wireless, we will gain an appreciation for
communicating in an environment without physical connections to every computer, router, etc. in the network, leading up to
how wireless communication systems can be hacked.
1. Communication Systems
The purpose of a communications system is to transmit information over a distance. This “information” could be audio (such
as speech or music), video, sensor data (temperature, pressure), or other data (e.g., text, stock prices, photos, etc.). “Over a
distance” may mean from here to the other side of the world via a satellite, or from one computer to another in a network, or
from your computer’s CPU to its RAM.
Any communications system consists of the following basic components, which are shown in the following figure. There are
four main components:
 Transmitter – converts information into an electronic form suitable for the channel
 Channel – the physical medium through which an electronic signal travels
o e.g., wire, fiber-optic cable, free space (i.e., air), water (sonar)
 Receiver – converts the received signal back to a usable form
 Noise – undesired, random corrupting energy
The information is passed to the transmitter which in turn transmits it into the communication channel. The receiver produces
a “recovered” information signal, which may not be the same signal that was transmitted. This is because a significant,
though undesired, occurrence in all communication systems is noise, which is random energy that enters the system and
interferes with (corrupts) the transmitted message. If the noise is strong enough, the information signal may not get through at
all. You’ve all heard what noise sounds like, for example on a telephone (we sometimes refer to it as static). If the static is
very powerful you will only hear a small portion (or none) of the words that are spoken to you. This relationship between the
useful signal and corrupting noise that impacts it will be formalized in chapter 23.
Noise can be divided into two broad categories:
 External noise is noise introduced into the transmission channel from outside sources. Examples include:
o Industrial noise arising from man-made electrical sources (e.g., motors, generators, switches)
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Chapter 19: Communications Systems, EM spectrum, and Signals
o Atmospheric noise due to naturally occurring disturbances in earth’s atmosphere (e.g., lightning)
o Extraterrestrial noise due to solar and cosmic activity.
 Internal noise is noise introduced by the electronics inside the receiver itself. Examples include:
o Thermal noise
o Semiconductor noise
For the third block of this course, we will focus on communications systems in which our channel or medium is free space.
Free space can refer to a perfect vacuum (as you might recall from physics), or to the atmosphere (as opposed to transmission
through a wire or other material). Signals that propagate in free space are often referred to as “wireless” or “over-the-air”
signals, and all signals in free space are part of the electromagnetic spectrum. With wireless routers and satellites part of
almost every network, especially in military applications, understanding the electromagnetic spectrum is critical to cyber
security.
2. Data Flow
Communications systems range from the very simple to the complex. And with complexity comes cost. Systems which have
a requirement to both transmit and receive may have a transceiver which is capable of both. In designing a communications
system, it is important to consider the data flow required between devices so that they include the correct components, but are
not unnecessarily complex. Communication between two devices will be in one of three modes: simplex, half duplex, or full
duplex.
2.1 Simplex Signals transmitted in one direction. Only one of two devices on a link can transmit. Examples of simplex
devices are keyboards and televisions. Remember from chapter 1, a key board is an example of an input device. It cannot
accept any outputs. Similarly, your television (before smart TV!) does not transmit any information back to the cable
company. The simplex mode can used the entire capacity of a communications channel to send data in one direction.
2.2 Half Duplex Both stations may transmit, but not at the same time. Military Radios are an example of duplex devices.
Half duplex mode works in cases where there is no need for communication at the same time and the entire capacity of the
channel can be used in one direction at a time.
2.3 Full Duplex In full duplex mode, both stations can transmit and receive simultaneously. One common example of full
duplex communication is the telephones network. Both users can talk and listen at the same time however the capacity of the
channel but must be divided in the two directions.
3. Electromagnetic Spectrum
The electromagnetic spectrum is the range of all possible frequencies of electromagnetic waves. The spectrum is broken into
regions/ranges and classified by frequency and/or wavelength. The frequency (f ) of an electromagnetic wave is a measure of
how rapidly it oscillates. Frequency is measured in Hertz (1 Hz = 1 cycle/sec).
The period (T) of an electromagnetic wave is the length of time required to complete one cycle. The period is measured in
seconds, and is the reciprocal of the frequency in Hz (T = 1/f). Wavelength (λ) is the physical distance between the peaks of
one cycle of a transmitted wave as it moves through the medium, and is measured in meters (m). The following plots show an
EM wave’s voltage as a function of time (left plot), and as a function of distance (right plot).
For electromagnetic waves traveling in air (or vacuum), we will assume that they travel at the speed of light (c) which is
roughly 3 x 108 m/s. The wavelength is inversely proportional to the frequency, and is related to the speed of light by:
𝑐
𝜆=
𝑓
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Practice Problem 19.1

What is the wavelength of an FM radio station whose broadcast frequency is 101.1 MHz?
Solution:

What is the frequency of a signal whose wavelength is 8 cm?
Solution:
3.1 Frequency Bands The specific bands of frequencies in the EM spectrum is shown in the following figure. In this course,
we are concerned with communications in the frequency ranges from ELF to EHF.
To transmit signals effectively, the characteristics of the transmission medium and the EM wave behavior in the frequency
bands are considered and matched.
For example, a communication system transmitting to a submarine will have water as the transmission medium. Lower
frequency, longer wave length waves from VLF band do not attenuate as quickly in water as higher frequency bands, so the
transmission frequency will be selected from the VLF band.
You should be familiar with the frequency ranges for communications from ELF to EHF.
 Extremely low frequency (ELF) 30 Hz to 300 Hz. Power line frequencies and low end of human audio.
 Voice frequency (VF) 300 Hz to 3000 Hz. Typical range associated with human voice.
 Human hearing 20 Hz to 20 kHz. (You may try a demo at https://www.youtube.com/watch?v=qNf9nzvnd1k)
 Very low frequency (VLF) 3 kHz to 30 kHz. Used for communications with submerged submarines.
 Low frequency (LF) 30 kHz to 300 kHz. Long range radio navigation.
 Medium frequency (MF) 300 kHz to 3000 kHz. AM radio and long range communication.
 High frequency (HF) 3 MHz to 30 MHz. Known as “short wave”, used by two-way radio.
 Very high frequency (VHF) 30 MHz to 300 MHz. Radio communications and FM radio.
 Ultra high frequency (UHF) 300 MHz to 3000 MHz. TV, military and cell phones.
 Super high frequency (SHF) 3 GHz to 30 GHz. Microwave. Satellite communications and radar.
 Extremely high frequency (EHF) 30 GHz to 300 GHz. Satellite communications.
3.2 Bandwidth The range of frequencies contained in a signal is its bandwidth. Bandwidth is the amount of the frequency
spectrum occupied by a signal regardless of where it is in the spectrum. It is the difference between the upper and lower
frequency limits of the signal.
If a signal occupies the range of frequencies between approximately 300 Hz and 3000 Hz. The following figure demonstrates
that for that signal, it’s bandwidth would be 2700 Hz.
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3.3 Federal Communications Commission (FCC)

The electromagnetic spectrum is crowded; everyone wants some bandwidth. The FCC was established by the
Communications Act of 1934 to regulate interstate and foreign communication. The FCC:
 Allocates bands of frequencies for specific uses

 Sets limitations on broadcast power
 Monitors broadcasts to detect unlicensed operations and technical violations
 Auctions spectrum usage
The FCC controls which portions of the EM spectrum are used for various purposes (e.g. FM radio, AM radio, broadcast TV,
satellite communications). The FCC also makes sure that transmissions do not interfere with each other (two transmitters
physically close to each other transmitting in the same frequency range can destroy each other’s signals). For example,
Washington D.C. can have an FM station that transmits at 101.1 MHz (the FM station called FM101), but Baltimore cannot
have an FM station that transmits at 101.1 MHz because it is too close to the Washington D.C. station (approximately 35
miles away).
Because the spectrum is a non-renewable resource in a society that is increasingly connected it is incredibly precious. To give
you an idea of its value, 400 MHz of spectrum was auctioned by the FCC in 2015 and sold for $44.9 billion dollars!
Typical bandwidths:
 AM Radio Station – 10 kHz

 FM Radio Station – 180 kHz
 Broadcast TV Station – 6 MHz
4. Signals as a Function of Time and Frequency

Recall that the purpose of a communications system is to transmit information over a distance. The block diagram for a
communication system is again shown below.
Thus far, we’ve covered that during the final section of this course we’re going to focus on free space as our channel or
medium, which means we’re considering the electromagnetic spectrum. Why do we care?
Information can be in various forms. We transmit information in the form of a signal.
4.1 Time Domain (Sinusoidal Wave) Earlier in this chapter, we discussed some basic properties of sinusoidal
(electromagnetic) waves. A sinusoidal voltage waveform can be expressed mathematically in the following way:
1
vm (t )  Vm sin(2 f mt   m ) fm 
Tm
This equation is plotted in the following figure. Note that a cosine is a sine wave with a phase shift of π/2 radians (which is
90°).
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Amplitude (Vm) – distance from average to peak (in volts)

Peak-to-peak Voltage (Vpp) – distance from maximum value to minimum value (in volts)
Period (Tm) – time to complete one cycle (in seconds)
Frequency (fm) - number of cycles in one second (in Hz)
Phase (m)– Left/right shift with respect to the t = 0 axis (in radians)
A sinusoidal wave is one way to represent the sound the tuning fork makes as a function of time. This is referred to as its
“time domain” representation. If the amplitude of the signal is 2 Volts, then the equation for the tuning fork signal could be:
vm (t )  2 cos(2 440t )
This signal can also be represented in terms of its frequency content (i.e., which frequencies are present in the signal) in the
“frequency domain.”
4.2 Frequency Domain (Frequency Spectrum) To display a signal in the frequency domain, we determine the frequency
content of the signal (which can be done using Fourier theory or, for this class, when the signals we will analyze are
composed of sinusoids it can be done by inspection). The frequency content is then displayed on a plot of magnitude vs.
frequency. (magnitude is the absolute value of amplitude). Since our tuning fork is a very simple tone with a single frequency
component of 440 Hz and an amplitude of 2V, the frequency domain plot looks like this:
Both the time-domain (sine or cosine wave) and the frequency-domain displays represent the important characteristics of the
tuning fork as far as a communication system is concerned– they’re just different ways to express the same signal. For
communication engineers, the primary interest is what portion of the frequency spectrum does the signal occupy and how
strong is the signal (magnitude); for our purposes, phase offset (if present) is not part of the frequency plot,, so a sine or
cosine with the same amplitude (positive or negative) and any phase offset have the same frequency plot.
Suppose we had a slightly more complicated signal. Suppose
𝑣𝑚 (𝑡) = 2𝑠𝑖𝑛(2𝜋440𝑡) − 3𝑠𝑖𝑛(2𝜋900𝑡 + 𝜋⁄4) + 5𝑐𝑜𝑠(2𝜋1100𝑡 + 2𝜋⁄7)
In this case, there are three sinusoids (i.e., there are three frequencies in
the signal) so the frequency plot will have three spikes, at the three
frequencies given, with heights corresponding to the magnitudes of the
amplitudes given. Again, the phases given are not considered in this
plot.
Part of the benefit of a frequency domain representation is that certain
signal attributes, like bandwidth, are easy to visualize. For instance, in
the above graph, you can quickly see the bandwidth is 1100 Hz – 440 Hz
= 660 Hz.
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CH. 19 Problems
1. What is the purpose of a communications system? Draw and explain the components.
2. What part of the electromagnetic spectrum (frequency range) is visible to humans?
3. Find 5 major uses of the UHF band (Use a book or the Internet to find your answer).
4. Calculate the frequency of signals with the following wavelengths:
(a) 30 m
(b) 2 km
(c) 8 cm
5. AM Radio
(a) What is the frequency range used by commercial AM radio broadcast stations?
(b) What is the bandwidth (BW) occupied by each station?
6. Given the sine wave below, answer the following questions:
4
3.75
3.5
3.25
3
2.75
2.5
2.25
2
1.75
1.5
1.25
1
0.75
Amplitude (V)
0.5
0.25
0
-0.25
-0.5
-0.75
-1
-1.25
-1.5
-1.75
-2
-2.25
-2.5
-2.75
-3
-3.25
-3.5
-3.75
-4
0 0.01 0.02 0.03 0.04 0.05 0.06 0.07 0.08 0.09 0.1 0.11 0.12 0.13 0.14 0.15
time (msec)
(a) What is the period of this signal?
(b) What is this signal’s amplitude?
(c) What is the frequency of this signal?
(d) In which range of the electromagnetic spectrum would this signal be classified?
(e) What is the wavelength of this signal?
(f) Sketch this signal in the frequency domain.
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7. Given the following equation for a signal, sketch the frequency plot. Put your frequency axis in kHz.
𝑣𝑚 (𝑡) = −18𝑠𝑖𝑛(2𝜋44,000𝑡) + 13𝑠𝑖𝑛(2𝜋150 × 103 𝑡 − 6𝜋⁄11) + 7𝑐𝑜𝑠(2𝜋30 × 104 𝑡 + 𝜋⁄21)
8. Given the following plot, write the equation for one signal that has this as its frequency plot (note: there is not one single
answer).
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Security Exercise 19
Part 1: Setup
This lab will introduce you to the standard lab bench setup and explore the basic characteristics of a signal. We will use the
following equipment:
 Agilent 33210A 10 MHz Function / Arbitrary Waveform Generator
 Tektronix TDS 1002B Two Channel Digital Storage Oscilloscope
 RadioShack Mini Amplifier/Speaker
 Microphone
For the next portion of this lab you will be investigating a sinusoidal signal from the function generators at your lab bench
and displaying it on the oscilloscope.
Step One: Function generator setup.
□ Turn on power to the lab bench. The power switch is on the right side of the lab bench and is labeled "120 V
OUTLETS." The switch should be in the raised position if power is on.
□ Locate the "10MHz Function/Arbitrary Waveform Generator" on the lab bench
and turn the power on. We will refer to this equipment as simply the "function generator."
□ Power on your function generator (Bottom left hand side of the front panel).
□ Select the sinusoidal function by pressing the button with the Sine wave on it. The function generator display should
indicate a small sine wave.
□ Turn any other lit buttons off.
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As the name implies, the function generator is able to generate electrical signals. For this lab we will use the function
generator to generate sinusoidal voltage waveforms.
We will set the frequency 440 Hz using the key pad method.
□ Select the Frequency (Freq) function using the soft keys under the display
screen.
□ Enter the desired frequency (440) using the key pad.
□ Enter the desired units (Hz) by pressing the button under Hz on the screen.
We will set the size of the waveform to 2.00 Vpp (volts peak-to-peak) using the key pad
method.
□ Select the Utility function and then select the Output Setup soft key. Check that
High Z is highlighted. If not, push the Load/High Z button until it is highlighted.
Push Done.
□ Select the Amplitude (Ampl) function using the soft keys under the display screen.
□ Enter the desired amplitude (10) using the key pad.
□ Enter the desired units (Vpp) by pressing the button under Vpp on the screen.
□ Press the output button. It should now be lit indicating the function generator is producing an output.
Right now your function generator is generating a 440 Hz signal that has a peak-to-peak voltage of 2VTo see the signal, we
will send the output of the function generator to an oscilloscope. Proceed to Step Two!
Step Two: Oscilloscope Setup
□ Locate the oscilloscope at the top of your lab bench and turn its power on. The power push button is located on the
top left of the oscilloscope.
The oscilloscope can receive electrical signals from two probes, channel 1 (CH 1) and channel 2 (CH 2). We have attached
adaptors to CH 1 and CH 2, so that they can receive electrical signals via our banana plug cables.
□ Locate the CH 1 input on your oscilloscope. It will have a two-input (RED / BLACK) banana plug adaptor installed.
□ Using the cables which can be found under the bench on the plastic Quad board. Connect your equipment as
follows:
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Power Outlet
The oscilloscope has the ability to measure and display two different electrical signals, but only with respect to one common
reference point. The BLACK CH 1 input provides this common reference point (ground) for both CH 1 and CH 2. Therefore,
CH2 has an adaptor for only one banana plug. We will not be using CH2 for this lab.
Step Three: Display your sine wave with the oscilloscope.
□ Press Autoset to have the scope automatically calibrate to your signal.
□ Both the horizontal and vertical position of the sine wave can be adjusted using the Position dials. The horizontal
and vertical axis scales are controlled by the Volts/Div and Sec/Div dials. Experiment with all dials to see the effects
on the sine wave.
□ Press CH 1 MENU on the oscilloscope and make the following settings.
o Coupling: AC
o BW Limit: OFF
o Volts/Div: COARSE
o Probe: 1X
o Invert: Off
□ Adjust the vertical axis to 1 Volt/Division. Bottom left hand corner of the screen will display CH1 1.00V when
adjusted correctly.
□ Adjust the horizontal axis to 500 s/Division. Bottom center of the screen will display M 500 s when adjusted
correctly.
□ Scope screen should be very similar to the image below when your scope is set up properly.
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□ With the CH 1 menu selected, adjust the position of the vertical axis zero level by rotating the vertical position knob
in the CH 1 column, so that the “1→” on the left side of the display is adjacent to the major horizontal axis (centered
vertically on the display).
Question 1: Is the sine wave displayed as a function of time or frequency? Why?
Question 2: What is the mathematical representation of this particular sine wave?
Part 2: Measuring in the time Domain with the Oscilloscope
Manual method. The first method is by counting the divisions of grid and applying the scale (volts/div for vertical, or
sec/div for horizontal). This will only provide you with approximate values with little precision.
Question 3: Fill out the table on your answer sheet using this manual measurements method. Specifically:
□ Using the vertical scale, determine the peak-to-peak voltage on CH1 which is the total voltage from positive peak to
negative peak. That is, you should count the number of vertical divisions from peak to peak, and multiply the
number of divisions by the number of volts/division.
□ From your measured Vpp, determine the amplitude of the signal (Vm).
□ Using the horizontal scale, determine the period (Tm) and then calculate the signal's frequency. That is, you should
count the number of horizontal divisions for one full cycle, and then multiply the number of divisions by the number
of microseconds (in this case) per division.
Taking measurements with cursors. The oscilloscope has time and amplitude cursors that you can move on the plot to help
take measurements of voltage, period and frequency.
□ Press the Cursor button (top middle) to view the cursor menu on LCD.
□ Then choose the Type of cursor to be Amplitude by cycling through the options on the button associated with this
menu option. Two horizontal cursors will now appear that are moveable.
□ Choose the Source to be CH 1. Two cursors are now available for you to move around the display. Push Cursor 1 to
move the first cursor, and Cursor 2 when you want to move the second cursor. The cursors are moved using the
large knob next to the green power-on light.
□ Place cursor 1 at the sinusoid’s maximum voltage, and cursor 2 at it’s minimum. The ΔV value (which is the voltage
difference between the two cursors) can be read out on the right side of the display…this is the peak-to-peak
voltage.
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□ Then choose the Type of cursor to be Time by cycling through the options on the button associated with this menu
option. Two vertical cursors will now appear that are moveable.
□ Adjust the two time cursors to allow you to measure the period of the sinusoid.
Question 4: Fill out the table on your answer sheet the using cursor measurements method.
Taking automatic measurements. The oscilloscope has the ability to take automated measurements of voltage, period and
frequency.
□ Press the Measure button (top middle) to view the measurement menu on LCD, then push the top menu box button
to highlight Source and select CH1 for Measure 1.
□ Then choose the Type of measurement to be Pk-Pk by cycling through the options on the button associated with
this menu option, and then hit button for Back option
□ Press the second to top menu box button to select Measure 2.
□ Then choose the Type of measurement to be Freq by cycling through the options on the button associated with this
menu option, and then hit button for Back option
□ You can add new measurements for all 5 buttons. Amplitude is not automatically measured but can still be
calculated from the peak-to-peak voltage as before. Record your results in Question 2.
Question 5: Fill out the table on your answer sheet using the automatic measurements method.
Question 6: The equation for a cosine wave is vm (t )  Vm cos(2 f mt   ) . Write your equation for the sinusoid based on
your measurements.
Part 3: Measuring in the time Domain with the Oscilloscope
Our scope can also provide a frequency spectrum of a signal. For this particular sinusoidal signal we know it is periodic and
has a single frequency, fm, described by the previous measurements and shown in our equation.
Now let us see how this signal is displayed in the frequency domain.
The oscilloscope performs a Fast Fourier Transform (FFT) and displays the magnitudes of the frequencies present in the
signal vs. frequency, so the horizontal scale shows frequency (in Hz) instead of time.
□ Press AUTOSET (top right) and you will see the options to display the signal in time or in frequency (FFT).
□ Select Math Menu, and cycle the soft key next to Operation until FFT is displayed.
□ Set the horizontal scale to read 50Hz per division by turning the Sec/Div knob.
At this point, you should see one large spike in the display (which corresponds to the sinusoidal signal from CH1, and also
many smaller spikes scattered throughout the frequency spectrum (this is noise). Your concern is the large spike.
Question 7: The left edge of the display is 0 Hz, and frequency increases from 0 Hz as you move to the right. Determine the
value of the frequency component ( fm= ?) by counting the number of horizontal divisions and multiplying that by the number
of Hz per division.
You can check your answer using the frequency cursor:
□ Press the Cursor button (top middle) to view the cursor menu on LCD.
□ Choose the Source to be MATH (note: this is because the FFT is a mathematical computation).
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□ Then choose the Type of cursor to be Frequency. Two vertical cursors are now available for you to move around
the display. Push Cursor 1 to move that cursor, and place it on the largest spike. The readout of frequency will be
displayed on the right side of the LCD.
Question 8: Sketch your frequency plot (as seen on the oscilloscope) and label the axes with your values. Include the smaller
noise spikes. Have your instructor check it. Note: this oscilloscope displays amplitude of the frequency content in decibels
(dB) vice volts, as in the lecture. Decibels will be discussed later. For now you can label the magnitude of the largest spike
from the measurements obtained in Questions 3-5 Do not mind the magnitude for the smaller noise spikes.
Part 4: Exploring Signal Parameters in Time and Frequency Domains
To explore how changes in amplitude and frequency affect a sine wave, both the oscilloscope and the RadioShack speaker
will be used.
□ Power on the RadioShack speaker by adjusting the volume dial on the right hand side.
□ On the Agilent Function Generator, ensure the Frequency menu is selected turn the dial clockwise to adjust
frequency of the sine wave.
□ Press AUTOSET (top right) and you will see the options to display the signal the time domain again.
□ Turn function generator dial to observe the effects.
Question 9: How does the frequency change? How is this reflected in the O-Scope display? How is the speaker audio output
affected?
Question 10: What is the highest frequency you can hear from your Radio Shack mini speaker? The lowest?
□ Next, ensure the Amplitude menu is selected on the function generator and turn the dial to adjust the amplitude.
Question 11: How are amplitude changes reflected in the O-Scope display? How is the speaker audio output impacted?
□ Reset your original sine wave by setting Voltage to 2.0 VPP and Frequency to 440 Hz.
Our scope can also provide the frequency spectrum of a signal. This particular sinusoidal signal consists of a single
frequency, f, which is currently set to 440 Hz. Now, we’ll explore how this signal is displayed in the frequency domain.
□ Change your o-scope display from time domain to frequency domain as follows:
o Select Math Menu, and cycle the soft key next to Operation until FFT is displayed.
Question 12: Now, what does the x-axis represent?
□ Adjust the horizontal scale to 50 Hz per division using the SEC/DIV dial.
Question 13: Sketch the o-scope display and describe what the sketch represents.
□ Using the dial on the Agilent Function Generator, repeat your previous step of adjusting the frequency of the sine
wave.
Question 14: How is the changing frequency reflected in the O-Scope display? How is the speaker audio output affected?
Compare and contrast to your time-domain results.
□ Using the dial on the Agilent Function Generator, repeat your previous step of adjusting the amplitude of the sine
wave.
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Question 15: How is the amplitude change reflected in the O-Scope display? How is the speaker audio output affected?
Compare and contrast to your time-domain results.
Part 5: Exploring a voice signal in Time and Frequency Domains
Since a sine wave is as simple as it gets, we’ll modify the setup to display a more complicated signal: Voice.
□ This section of the lab will continue using the o-scope and speaker, remove the function generator, and add a
microphone. Set your equipment up as indicated below:
Power Outlet
□ Use Autoset to calibrate the o-scope if desired. You will likely need to adjust the horizontal and vertical axis by
hand. 1.00 V/div and 2.50 ms are good starting points for this section.
□ Speak into the microphone to observe the real-time signal your voice makes. You may need to adjust the vertical
position of the signal and the trigger to observe a better waveform.
□ Press the Run/Stop button at various times to examine a “snapshot” of your voice signal. Record & sketch your
results.
Question 16: Experiment with talking and humming using lower and higher ranges of your voice. Summarize your findings
with regards to signal appearance and resemblance of a sine wave at both the low and high ends of your vocal range.
□ To observe the near-real-time frequency content of your voice signal, select the Math Menu button and adjust the
horizontal axis to 250 Hz per division. Continue speaking in the microphone to observe changes in the spectrum.
Question 17: How is the frequency-domain representation of voice different than that of a sinusoidal tone? Explain your
answer.
Part 6: Clean up
□ Unplug and stow the banana plug cables.
□ Turn off your oscilloscope.
□ Turn off your function generator.
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Security Exercise 19 Answer Sheet

Name:
Question 1:
Question 2:
Question 3:
Question 4:
Question 5:
Question 6:
Question 7:
Question 8:
Question 9:
Question 10:
Question 11:
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Question 12:
Question 13:
Question 14:
Question 15:
Question 16:
Question 17:
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Chapter 20: Intro to Modulation
Objectives:
(a) Define the term baseband signal and describe some potential limitations associated with transmitting baseband signals
directly.
(b) Discuss the role of modulation in signal transmission and the methods of modulating a sinusoidal carrier.
(c) Using a plot of an AM signal, determine Vmax, Vmin, Vm, Vc and m.
(d) Create a frequency plot for and determine the sidebands and bandwidth of an AM signal where the information signal
consists of one or more distinct sinusoids.
(e) Determine the condition for overmodulation in an AM signal, and the consequence of overmodulation.
In Chapter 19, we defined a communications system and learned that the wireless section of EC310 will focus on
communications in which the communication channel is free space.
We also learned that in order to send any information through a communications system it must be in the form of a signal
(which is the name given to the function that conveys our information), and if our communication channel is free space, it
means we’re dealing with signals carried in the Electromagnetic Spectrum (EM). Finally, we learned that signals can be
represented as a function of either time or frequency. Wireless channels have different vulnerabilities than we saw in the host
section of the course, because of the frequency of transmission. It is possible that a wireless network can be attacked like we
saw in the networks section of the course, but to see how such a cyber-attack can be carried out on a wireless network or a
wireless communication in general, you must first understand how information is carried through the wireless channel.
1. Baseband Signals
If you’re sitting in your EC310 classroom in the basement of Rickover Hall and you speak to the student next to you, will that
person be able to hear you? Will you be heard across the room? How about at the end of that loooong Rickover passageway?
Does anyone in Bancroft have a chance of hearing what you’re grumbling about in your EC310 classroom? Why not? Your
voice doesn’t travel as far as you might like it to—your voice creates pressure waves in the air, and the strength of these
waves attenuates over distance. The louder you yell the farther you’ll be heard, but this of course has its limits. As an
alternative to walking around screaming all day, you might consider speaking at a more normal volume into a microphone. A
microphone is a device that transforms sound pressure waves into electrical signals. You could then send the electrical output
of the microphone to an antenna. Then your voice frequencies would travel as electromagnetic waves (“radio waves”), and as
long as you provided enough power to the antenna, you could presumably greatly extend the geographic range of your EC310
musings. In this chapter, we’ll see that the latter approach is on the right track, but in order to be practical, it requires a bit
more finesse.
In this example our voice signal, which you’ll recall is comprised of frequencies roughly in the range between 300 Hz and 3
kHz, is what we call a baseband signal. Baseband signals are information signals at their original frequencies, typically low
frequencies. To transmit a baseband signal directly as is, we use baseband transmission…as you’ll see in shortly,
communication systems typically will upshift the frequency spectrum of baseband signals to a higher range of frequencies to
allow transmission through the atmosphere.
In general, before signals can be transmitted effectively, they must first be converted to a form that is compatible with the
communication medium. One facet of this conversion is transducing the signal from its natural physical form into an electrical
signal. For example,
 Microphones convert acoustic pressure waves (sound) into electrical signals.
 Video cameras convert light patterns into electrical signals.
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 Computer keyboards convert physical input (typing) into an electrical signals.

But, as alluded to above, even after converting your voice signal to, say, a voltage signal using a microphone, attempting to
transmit it over the air as a baseband signal is impractical. Why? Let’s look at an example that will point us in the right
direction.
Physics dictates that antenna length is intrinsically tied to the wavelength of the signal it is transmitting or receiving. To transmit
a signal through the atmosphere with an antenna efficiently, the length of the antenna must be at least a tenth of a wavelength
long.
What is the approximate length of the antenna required to transmit the sound of a tuning fork (which creates musical note A =
440 Hz)? Note: this sound must be transduced into an electrical signal first before it is transmitted.
Solution:
Wait! To transmit that lousy tuning fork signal my antenna needs to be at least 68 km? That’s over 42 miles! We’d need an
antenna that extends into the upper parts of the atmosphere for that. Clearly, that’s not going to work.
Well, I know that if I want to listen to the Navy game on the radio (because for some
reason I avoided the mandatory fun), I can tune in to AM radio station 1430 WNAV.
Recall that from Chapter 19, when referring to a commercial AM radio station, such as
1430 WNAV, that the 1430 refers to the center of its transmission frequency in kHz.
So what size antenna does WNAV use?

(a) What is the wavelength of an AM radio station whose transmission frequency is 1430 kHz?
Solution:
(b) What is the approximate antenna length if the station uses an antenna that is half the wavelength long?
Solution:
105 meters? Okay, that’s still big – it’s about 115 yards - almost an entire football field… but at least you don’t need an
antenna that reaches into outer space now. In reality,
WNAV’s antenna, pictured at right, is 117 meters.
We’re close!
The purpose of those two examples was to
demonstrate that we need to somehow get our
baseband information to a higher frequency (shorter
wavelength) in order to be able to transmit it across
our channel. Higher frequencies give us reasonable
antenna sizes plus some added benefits: first, signals
will attenuate less quickly if the higher frequencies are
well-chosen. Second, multiple people - whose voices
all occupy the same baseband frequencies - can
communicate without interfering if each transmits on
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a different higher frequency range. We can shift baseband information to higher frequencies for transmission using a process
called modulation.
2. Modulation
To overcome limitations of the communications channel and permit multiple access, information signals are impressed upon
a higher-frequency carrier signal for transmission. This process is called modulation. Now we’re dealing with two signals:
1. Original (“baseband”) information signal - frequency is too low to transmit efficiently
2. Higher frequency (“carrier”) signal - we can transmit this efficiently, so we use it to carry our information
Mathematically, the cosine wave representing the higher-frequency carrier is given by:
vc (t )  Vc cos  2 fc t  c 
Modulation is the process of varying any of three properties (amplitude, frequency or phase, underlined above) of a high-
frequency carrier using the lower-frequency information signal (baseband signal). A modulator is a component of a
communication system which achieves modulation. The three types of modulation we will focus on are:
2.1 Amplitude modulation (AM) Varying the amplitude Vc of the carrier with the info signal.
2.2 Frequency modulation (FM) Varying the frequency fc of the carrier with the info signal.
2.3 Phase modulation (PM) Varying the phase angle cof the carrier with the info signal.
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Since the intention of the “Wireless” section of EC310 is to give you a broad understanding of wireless communication
techniques rather than to make you communication engineers, we’re only going to go into more detail with amplitude
modulation (AM) in this chapter. This is NOT to say that frequency modulation and phase modulation are unimportant –
they’re very important and very widely used. The fact is that there’s only so much modulation that can be reasonably covered
in the last several chapters of this course, and AM is the easiest to visualize and demonstrate. Later, in Chapter 22: Digital
Modulation, we will again address amplitude, frequency and phase modulation to some extent as they apply to digital
communications.
3. Amplitude modulation (AM)
In amplitude modulation, the information signal is used to vary the amplitude of the carrier sine wave. For simplicity,
consider a cosine wave information signal, vm(t) (a 440 Hz tuning fork) and a cosine carrier, vc(t) (frequency 5000 Hz (or 5
kHz)).
vm (t )  Vm cos  2 f mt   Vm cos  2 440t 
The diagram of an amplitude modulation system using this information signal follows.
The AM wave (vAM(t)) is the product of the carrier (with amplitude = 1) with a modulating signal. The modulating signal is
the information signal vm(t) with an added offset, Vc. The AM signal is then given by:
vAM  t   Vc  vm (t )  cos  2 f c t   Vc  Vm cos(2 f mt )  cos  2 f c t 
Modulating Signal Carrier Modulating Signal Carrier
In the figure below, the top plot is of the information signal and the bottom is the resulting AM signal. Note that the
information signal starts at a value of zero (for approximately 2 msec), so the resulting AM signal is the unmodulated carrier
signal (meaning that the carrier is not being modulated). When the information signal is no longer zero, it starts to modulate
the carrier’s amplitude as shown.The horizontal axes are time in msec. The information signal is equal to zero at the
beginning, then changes to the tuning fork cosine wave at approximately 2 msec. Here, the carrier and information signal
parameters are: fc = 5 kHz, Vc = 10V, fm = 440 Hz, Vm = 7.5V.
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A zoomed-in plot of the resulting modulated AM signal is as follows, showing the graphical relation between Vm and Vc:
The envelope of the modulating signal (which is drawn onto the AM signal below in a dashed red line) varies above and
below the unmodulated carrier amplitude, Vc. It is the envelope that carries the information signal; the receiver must separate
the envelope from the received AM signal to recover the information that was transmitted. In this case, the envelope is in the
shape of a sine wave, which is the same as the information signal. The values of Vm and Vc are related by the modulation
index (m).
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3.1 Modulation Index The relationship between the information signal amplitude, Vm , and the unmodulated carrier
amplitude, Vc, is expressed as a ratio called the modulation index (m), defined as:
Vm
m
VC
Sometimes m is expressed as a percentage: percent modulation = m x 100%. The following figure shows the AM signal at
three different values of percent modulation: 20%, 50% and 90%. Overall, the greater the value of m, the closer the envelope
gets to the horizontal (time) axis.
We can also mathematically determine the modulation index m from the maximum and minimum values of the envelope of
vAM(t) as follows, where Vmax is the maximum value of the envelope and Vmin is the minimum value:
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Vmax  Vmin
Vm 
2
Vmax  Vmin
Vc 
2
Vm Vmax  Vmin
m 
Vc Vmax  Vmin
In order for the AM signal to convey the original signal accurately and
prevent distortion, the information signal amplitude (Vm) must be less
than the unmodulated carrier signal amplitude (Vc). Here again, the
unmodulated carrier refers to the AM signal if the information signal
amplitude is equal to 0 (Vm = 0), in which case,
vm (t )  Vm cos(2 f mt ) . The maximum usable modulation index is m
= 1.0, corresponding to 100% modulation, when Vm is equal to Vc. When
Vm is greater than Vc (that is, m > 1), overmodulation occurs.
Overmodulation, depicted below, results in distortion of the AM signal’s
envelope, and since the envelope holds the information, the recovered
information signal is also distorted.

If a carrier signal vc(t) = 9 cos(25000t) Volts is modulated by a cosine wave vm(t) =7.5 cos(2440t) V, what is the
percentage modulation of the resulting AM signal?
Solution:
Now that we have a basic understanding of how Amplitude Modulation works in the time domain, let’s look at AM in the
frequency domain.
4. AM in Frequency Domain
Recall the equation for the amplitude modulated waveform if the information signal is a single sine wave is given by: .
vAM (t )  Vc  Vm cos  2 f mt   cos  2 fc t  . We already know the frequency domain representations of the modulating signal
(vm(t)) and the carrier signal (vc(t)), but how does the amplitude modulated signal look in the frequency domain?
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To answer this question, recall the trig identity for the product of two sine waves:
cos A cos B  12 cos( A  B)  12 cos( A  B)
Applying this trig identity for product of two sine waves to the AM signal results in:
vAM (t )  Vc  Vm cos  2 f m t   cos  2 f c t 

 Vc cos  2 f c t   Vm cos  2 f c t  cos  2 f m t 
 Vc cos  2 f c t   cos  2  f c  f m  t   cos  2  f c  f m  t 
Vm Vm
2 2
1 2 3
This means that when a single sine wave information signal is used to modulate the carrier in AM, the resulting AM signal
contains three sinusoids (labeled above): one at the carrier frequency (1), one fm Hz above the carrier frequency (2), and one
fm Hz below the carrier frequency (3). For the tuning fork example, we have: fc = 5 kHz, fc − fm = 4.560 kHz and fc + fm = 5.440
kHz. The trig identity puts the amplitudes at frequencies fc − fm and fc + fm at one half that of Vm.
This means the resulting frequency domain plot for this tuning fork example looks like the following (the labels follow the
ones above):
1
3 2
The process of modulating a carrier creates an upper and a lower sideband that is apparent in the frequency plot. The lower
sideband (or LSB) is that portion of the transmitted signal that has frequency content less than the carrier frequency, and the
upper sideband (or USB) has frequency content greater than the carrier frequency. For the tuning fork example, the USB is
the 5440 Hz cosine, and the LSB is the 4560 Hz cosine. On a frequency plot of an AM signal, the lower sideband is a mirror
image of the upper sideband centered about the carrier frequency.
What is the AM signal’s bandwidth? Since bandwith is the highest transmitted frequency minus the lowest frequency
transmitted, it is (fc + fm) − (fc − fm) = 2 fm = 880 Hz. This is twice the bandwith of the information signal we started with…if
we didn’t modulate the information signal, the transmission bandwidth would have only been fm (440 Hz in this case)1. This
means that by transmitting with AM, we have doubled the required bandwith to transmit the signal. Why is this a concern?
Bandwidth is the #2 limiting factor in communications systems, and can be expensive to use… so we’re going to want to
1
The bandwidth of a baseband signal is considered to be its maximum frequency content. In this case, if the message is a
single sinusoid at a frequency of fm Hz, we say its bandwidth is fm Hz.
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send as much information as possible while occupying the minimum amount of bandwith possible. (The #1 limiting factor in
communications systems is noise, but that’s a subject for a future lesson.)
Let’s look at a slightly more complex example – suppose the information signal was comprised of two cosine waves:
vm (t )  V1 cos  2 f1t   V2 cos  2 f 2t 
What does the AM signal look like? Again, we apply the same trig identity to each sine in the information signal, resulting in:
v AM (t )  Vc  vm (t )  cos  2 f c t 
 Vc  V1 cos  2 f1t   V2 cos  2 f 2 t   cos  2 f c t 
 Vc cos  2 f c t   V1 cos  2 f c t  cos  2 f1t   V2 cos  2 f c t  cos  2 f 2 t 
 Vc cos  2 f c t   21 cos  2  f c  f1  t   21 cos  2  f c  f1  t   cos  2  f c  f 2  t   cos  2  f c  f 2  t 
V V V2 V2
2 2
The net result is that for each cosine in the message, we will wind up with two cosines in the AM signal: one will have a
frequency greater than the carrier frequency, one will have a frequency less than the carrier frequency. The bandwidth is still
equal to the highest frequency in the AM signal minus the lowest frequency. This is equal to two times the max frequency in
the information signal. That is, if fmax is the maximum of the two frequencies in the information signal (either f1 or f2), then
the AM bandwidth is BW = 2 fmax.
Suppose we want to transmit the sound of a two chime doorbell (f1=349 Hz, f2= 440 Hz) using VLF (very low frequency)
communications (let fc = 20 kHz). Each of the chimes has an amplitude of 10V, and the carrier’s amplitude is 20V. Sketch the
frequency domain representation of the transmitted signal and determine the bandwidth.
Which of the two chime frequencies determines the bandwidth?
Solution:
If a carrier signal vc (t )  20sin  2 5000t  Volts is amplitude modulated by information signal

vm (t )  4sin  2 200t   6cos  2 400t  411  , sketch the frequency plot for the resulting AM signal and calculate the
transmission bandwidth.
Solution:
An example of an even more complicated signal is the signal created by recording an oboe (the musical instrument) playing a
single note. When someone blows into the mouthpiece of an oboe to play a note, their fingers are placed over certain holes to
create the note. Because of the structure of the oboe (its length, diameter and placement of the holes for example), the sound
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the instrument makes is actually a combination of a number of tones (sinusoids) with different amplitudes. Different
instruments can all play the same note, but each instrument will sound differently because the structure of the instrument
produces different sinusoids with different amplitudes. Amplitude modulation of an oboe playing the note Concert A is
demonstrated on the next figure. In this figure, the maximum frequency present in the note is approximately 4 kHz, so the
bandwidth of the AM signal is BW = 2 fmax = 2 (4 kHz)= 8 kHz.
Other common information signals, such as voice or music, are composed of many different frequencies. AM modulation still
works the same way, but in order to compute transmission bandwidth, we again compute it as BW = 2 fmax. Again, here, fmax
is the maximum frequency content present in the information signal.
5. Demodulation
Modulation is used to upshift the frequency content of a baseband signal, to facilitate transmission (e.g., to allow a smaller
antenna). Demodulation is performed in the receiver to downshift that frequency content back to its baseband frequency. For
example, if the 440 Hz tuning fork signal were transmitted on Annapolis AM radio station WYRE 810AM (fc = 810 kHz), the
transmitted signal is at a frequency of approximately 810 kHz, which is well outside our hearing (we can hear signals with
frequency content up to approximately 20 kHz). In order to hear the tuning fork signal, our car’s radio receiver must shift the
frequency content back down to its original range (440 Hz). This is demodulation; it basically “undoes” what modulation did
to the information signal. How demodulation works is beyond the scope of this course, but you should be aware of its
importance in a communication system.
What’s the point? AM is by no means the only form of modulation (though it’s probably the easiest to work through and
visualize). The intent of this course is not to make you all communications engineers, but you do need to have enough
background in modulation to understand the implications (especially with regard to bandwidth) moving forward. You’ll see
this again in a few lessons, with digital applications.
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CH. 20 Problems
1. (a) Calculate the wavelength of signals with frequencies of 1.5 kHz, 18 MHz, and 22 GHz.
(b) Since an antenna that is needed to transmit these frequencies must be at least a tenth of the wavelength, which signal
frequency would NOT be practical for direct (i.e., baseband) transmission?
(c) Name and define a technique that could be used to transmit the frequency in part (b).
2. An AM signal is comprised of the following message signal and carrier:
vm(t) = 80 cos (2π5000t) volts

vc(t) = 100 cos (2π800,000t) volts
where vm(t) is the message and vc(t) is the “unmodulated carrier” (i.e., the output of the modulator when no information
signal is present).
(a) Find the carrier frequency, the upper-sideband and lower-sideband frequencies, and the percent modulation (m).
(b) Suppose vm(t) changes to 120 cos(2π5000t). Find the new percent modulation (m). Give the technical term for this
condition and explain the effects of this condition occurring.
3. A radio station, 1280AM, is conducting a monthly test of the Emergency Alert System. The test begins with an annoying
sound comprised of two pure tones at 853 Hz and 960 Hz. The signal being broadcast has exactly five frequency
components, i.e., the signal could be written as follows:
vAM(t) = V1 cos(2π f1 t) + V2 cos(2π f2 t) + V3 cos(2π f3 t) + V4 cos(2π f4 t) + V5 cos(2π f5 t) volts
(a) Find the five frequencies that comprise the AM signal being broadcast. Recall that transmitted signal is comprised of
the carrier frequency and two sideband frequencies for each of the emergency alert tones..
(b) Find the bandwidth for this particular broadcast.
(c) Determine which of these two emergency alert tones (853 Hz or 960 Hz) determines the bandwidth.
(d) What is the bandwidth assigned to a commercial AM radio station in the United States?
4. Musical notes can be viewed as pure tones (if we ignore the “warmth” added by any particular instrument). Pure tones
are signals that contain only one frequency. Chords are combinations of notes, such as the C-Major chord on the piano,
comprised of notes C, E, and G.
If the radio station 1280AM broadcasts the C-Major chord, it would broadcast the following seven frequencies, listed in
ascending order and annotated by note and sideband:
f LSB-G = 1,279,608 Hz
f LSB-E = 1,279,670 Hz
f LSB-C = 1,279,738 Hz
f carrier = 1,280,000 Hz
f USB-C = 1,280,262 Hz
f USB-E = 1,280,330 Hz
f USB-G = 1,280,392 Hz
Notice that in the lower sideband, the notes are in reverse order. G, the highest pitch in the chord, is always the
farthest away from the carrier frequency. The carrier frequency is exactly in the middle. Assume that the carrier
amplitude is 100V, and the voltages for the three musical notes are all 20V.
(a) Sketch this broadcast in the frequency domain (label frequencies and amplitudes).
(b) After demodulation, what frequencies would be heard coming out of a your AM radio’s speaker?
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(c) Find the bandwidth of the broadcast and determine which note (C, E, or G) sets the bandwidth.
5. The following plots of AM signals, determine Vmax, Vmin, Vc, Vm and m. Indicate Vmax and Vmin on the plots, and show your
work!
(a)
15
13
11
3
AM signal
-1
-3
-5
-7
-9
-11
-13
-15
0 0.5 1 1.5 2 2.5 3 3.5 4 4.5 5
time (msec)
(b)
20
18
16
14
12
10
8
6
4
AM signal
2
0
-2
-4
-6
-8
-10
-12
-14
-16
-18
-20
0 0.5 1 1.5 2 2.5 3 3.5 4 4.5 5
time (msec)
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(c)
15
13
11
3
AM signal
-1
-3
-5
-7
-9
-11
-13
-15
0 0.5 1 1.5 2 2.5 3 3.5 4 4.5 5
time (msec)
(d)
20
18
16
14
12
10
8
6
4
AM signal
2
0
-2
-4
-6
-8
-10
-12
-14
-16
-18
-20
0 0.5 1 1.5 2 2.5 3 3.5 4 4.5 5
time (msec)
6. Determine fc and fm for any of the AM signals in problem 5 (parts (a)-(d)). (Hint: fc and fm are the same for each case).
Using these values of fc and fm, along with your answers to problem 5, sketch the frequency content for each of these AM
signals (parts (a)-(d)).
7. For any of the plots shown in problem 5, determine the bandwidth of the transmission. Note that each of these AM
signals has a single sinusoid as the information signal. (Hint: the bandwidth is the same for each case).
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Introduction to Amplitude Modulated Signals
Part 1: Set Up
Step One: Function generator setup.
□ Turn on power to the lab bench (the switch on the right that says "120V OUTLETS.")
□ Turn on the function generator.
□ Select the sinusoidal function by pressing the button with the Sine wave on it. The function
generator display should indicate a small sine wave.
□ Turn any other lit buttons off.
□ Select the Utility function and Output Setup soft key. Load should be High Z.
□ Push Done.
□ Press the output button.
Step Two: Oscilloscope setup.
□ Turn on the oscilloscope.

□ Locate the CH 1 input on your oscilloscope. It will have a two-input
(RED / BLACK) banana plug adaptor installed.
□ Connect banana plug cables (which can be found under the bench on
the plastic Quad board) from the function generator’s output to the CH
1 input on the oscilloscope (red to red and black to black).
Part 2: Amplitude Modulated Signal in the Frequency Domain
An Amplitude Modulated (AM) signal looks somewhat like the figure that follows. This particular depiction includes the AM
signal’s envelope, and the definitions of Vm (message amplitude), Vc (carrier amplitude), Vmax (max envelope voltage) and
Vmin (min envelope voltage). The information signal (message) modulates the amplitude of the carrier.
□ On the Function Generator select the Store/Recall button and then push Recall State and then push State 3 and then
Recall State again. Make sure the output button is lit.
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□ Push AUTOSET on the Oscilloscope and you will see an AM signal.
□ Press CH 1 MENU on the oscilloscope and make the following settings.

 Coupling: AC
 BW Limit: OFF
 Volts/Div: COARSE
 Probe: 1X
 Invert: Off
□ If a trace appears for CH2, then press the CH2 Menu button twice
to turn off the CH 2 trace, since nothing is connected to CH 2 for this lab.
□ With the CH 1 menu selected, adjust the position of the vertical axis zero
level by rotating the position knob under the vertical section in the CH 1
column, so that the “1→” on the left side of the LCD is adjacent to the
major horizontal axis (centered vertically on the display).
□ Adjust the vertical scale (Volts/Div) to enlarge the

displayed signal so that
it occupies more than half of the oscilloscope display.
□ Adjust the horizontal range (Sec/Div) so that you can

see the AM waveform similar to the one shown
above. You may have to adjust the knob for trigger
level (it is to the far right below the AUTOSET
button) to clean up the signal. Turn the knob so that
you see the trigger level (arrow to far right of screen)
rise from the center line. Hint: if you have trouble
removing the “jitter” from the graph, use the
“Run/Stop” button in the upper right hand corner of
the oscilloscope to pause the capture.
Once the display is properly adjusted, use the boxes on the oscilloscope to measure the following parameters. Record your
results in Question 1 on your answer sheet.
1. Measure Vmin and Vmax .
2. Measure the period of the carrier, Tc (higher frequency signal). You will have to adjust the horizontal
(sec/div) scale to accurately measure a the period of a cycle of the high frequency carrier signal.
3. Measure Tm, the period of the message (lower frequency signal). You will need to adjust back the horizontal
(sec/div). Note: Tm measures the period of the envelope of the AM signal.
Calculate the modulation index m, the carrier frequency (fc), the message signal's frequency (fm), the amplitude of the carrier
(Vc) and the amplitude of the message signal (Vm). Use the equations shown in Table 1 below along with your measured
values. Record your results in Question 1 on your answer sheet.
Vmax  Vmin Vmax  Vmin Vm Vmax  Vmin

Vm  Vc  m 
2 2 Vc Vmax  Vmin
1 1
fc  fm 
Tc Tm
Question 2: Determine the equation of the AM waveform displayed on the oscilloscope. Write your answer on your answer
sheet using the form below, but with numerical values replacing Vc, Vm, fm and fc:
vAM (t )  Vc  Vm cos(2 f mt ) cos(2 f c t )
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Part 3: Amplitude Modulated Signal in the Frequency Domain
Let’s look at the frequency spectrum of this signal. In class we manipulated the above equation to show us what frequencies
will show up in the frequency domain:
vAM (t )  Vc cos(2 f c t )  m cos  2  f c  f m  t   m cos  2  f c  f m  t 
V V
2 2
There are 3 frequencies in the AM signal: the carrier and its 2 sideband frequencies. So let us try to see this signal in terms of
its frequency content. Again, the oscilloscope performs a fast Fourier transform (FFT) and displays the magnitudes of the
frequencies present in the signal vs. frequency, so the horizontal scale shows frequency instead of time. The y-axis will show
values in decibels, which can be ignored for this lab.
□ Press AUTOSET and you will see the options to display the signal in time or in frequency (FFT). (NOTE: If the
FFT option does not appear when you press AUTOSET, then press the MATH MENU button.)
□ Push the button next to the FFT (or stay in FFT operation) and you will see spikes at specific frequencies.
□ Set the horizontal scale to read 12.5 kHz per division using the Sec/Div knob.
Question 3: Determine the value of the center and sideband frequencies. Use the Frequency cursor (recall that when using
the FFT to show frequency content, the source must be MATH, not CH 1 or CH 2 to use the frequency plot cursors)
Question 4: Sketch your frequency plot as displayed on your o-scope. Label the frequency axis using the frequencies of the
tones present in the AM signal. Indicate the heights of the AM signal spikes with the expected amplitude of the cosines in the
signal (note: the amplitudes are not actually displayed on the FFT on your o-scope, you just need to indicate them on your
plot.) Have your instructor check it.
Question 5: What is the bandwidth of the amplitude modulated waveform?
Question 6: Rewrite your AM signal as the sum of three cosines as in the example below:
vAM (t )  V1 cos(2 f1t )  V2 cos  2 f 2t   V3 cos  2 f3t 
filling in numerical values for V1, f1, V2, f2, V3 and f3 .
Question 7: Is this AM signal within the range of frequencies that commercial AM radio stations transmit?
Question 8: If the information signal from above contained many frequencies (including frequencies higher than the fm you
measured), how would this affect the bandwidth of the amplitude modulated waveform?
Part 4: Another AM signal
You have a message signal you would like to transmit, but do not have access to an AM transmitter. An evil user offers to
transmit your signal (for a small fee). You pay the fee. Your customers start complaining that your signal sounds terrible.
You examine the amplitude modulated signal that the evil user has generated for you:
□ On the Function Generator select the Store/Recall button and then push Recall State and then push State 4 and then
Recall State again. Make sure the output button is lit.
□ Push AUTOSET on the Oscilloscope and you will see an AM signal.
□ Eliminate the CH 2 signal (if necessary), vertically center the waveform and adjust the horizontal range and trigger
level to stabilize the AM signal.
Question 9: Explain the problem with the evil user AM signal (hint: look at the shape of the envelope).
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Name:
_____________________________________________________________________________________________________
Question 1:
Vmax Vmin Vc Vm Tc (s) Tm (s) fc (kHz) fm (kHz) m
_____________________________________________________________________________________________________
Question 2:
_____________________________________________________________________________________________________
Question 3: fc = ___________ flsb = _____________ fusb = _____________
_____________________________________________________________________________________________________
Question 4:
__________________________________
Instructor / Lab Tech
_____________________________________________________________________________________________________
Question 5:
_____________________________________________________________________________________________________
Question 6:
_____________________________________________________________________________________________________
Question 7:
_____________________________________________________________________________________________________
Question 8:
_____________________________________________________________________________________________________
Question 9:
____________________________________________________________________________________________________
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Chapter 21: Analog to Digital Conversion
Objectives:
(a) Provide examples of analog and digital communication systems.
(b) Describe the advantages of digital over analog communication.
(c) Discuss the basic steps of the analog-to-digital conversion process: sampling, and quantizing/encoding.
(d) Given an analog waveform, sampling rate, and resolution, determine the resulting quantized signal and the binary
encoded A/D output.
(e) Calculate the Nyquist sampling rate for an analog signal.
(f) Given the number of bits in an A/D process, and sample frequency, determine the generated bit rate.
(g) Describe how the number of bits used in the A/D process affects the reconstructed analog signal.
In Chapter 20, you learned about modulation, and that it is impractical to transmit signals at baseband frequencies through
free space. Modulation upshifts the frequency of transmission, to allow for smaller antennas. For an AM communication
system, the signals at various places in the system is shown below.
We could have also used FM or PM, in which case the signal that exists in the communication channel (free space) might
look like the following, depending on the information signal (left: frequency modulation, right: phase modulation).
In a digital communication system, the information is composed of 1s and 0s, and the information signal is composed of
voltage pulses that represent the 1s and 0s. Hackers can attack our system in a number of ways, such as “reading our mail” or
injecting their own information into our channel. In the digital age, cyber-attacks usually fall onto digital communication
systems. But where do the 1s and 0s come from? Chapter 21 deals with how 1s and 0s are created from an analog signal.
1. Analog Systems
When you look at the waveform below, you should notice that it is a signal that varies continuously in time and amplitude. If
we observed nature, we would see that nature produces signals like this (i.e., changes in pressure, variations in light, sounds,
etc.). Analog systems use analog electrical signals to represent these natural patterns, such as the voltage signal created from
the sound waves of a person speaking into a microphone, shown in the next figure.
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What do you think might be an example of an analog system in action? How

about an 8-track tape player playing the songs on Michael Jackson’s 8-track
album, Thriller2.
This is a great example of an analog system, but my guess is you have no idea
what an 8-track is. So, let’s list some other analog systems that that may ring a
bell: AM/FM radios, rotary telephones, cassette tape players, VCRs, broadcast
TVs, the microphone you are singing into at Bancroft’s karaoke night…
So maybe you’re thinking, “I still have no idea what that stuff is!” There’s
probably a reason for that. We don’t really use many systems that are completely
analog anymore; digital communications are more widely used.
2. Digital Systems
Let’s think for a second about comparisons between what was used in the past and what you use now:
Type of Information Past Device Present Device
Music Cassette Tape CD
Videos VHS (VCR) DVD/Blu-ray Disc
Broadcast Television Standard Definition TV High Definition TV (HDTV)
We want the same types of information but are using a different method to get them: digital systems. Digital systems use
electrical signals that represent discrete (often binary) values. The electrical signals are referred to as digital signals.
Specifically, binary baseband digital signals use two discrete voltage levels to represent binary 1 or 0 (bits), as shown in the
example plots below. Combining multiple bits into words permits us to represent more than just two things. Digital circuits
operate on digital signals, performing logic and arithmetic functions.
Interesting fact and important to the class: digital signals are not representative of signals that occur in nature. Natural signals
are analog, and must be converted into digital format to be used in a digital system.
Great! So we’re using a new method to get the same information. Is this a big deal? It is, because using digital systems offers
a number of advantages over using analog systems.
3. Digital Advantages
3.1 Relative noise immunity (What is the number one limiting factor in communications? Noise.)
Relative noise immunity is the most important advantage of digital communications
Between the transmitter and receiver, whether the system is analog or digital, noise always corrupts the transmitted signal. In
general, an analog receiver has no idea what the received signal is supposed to be after it has been corrupted by noise, but a
digital receiver only has to decide between a finite set of choices: for example, a binary digital system’s receiver must only
2
Michael Jackson’s album Thriller (released in 1982) is claimed to be the Best-Selling album of all time!
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decide at any time whether or not it is receiving a binary 0 or a binary 1. This means that receiver circuitry can be designed to
distinguish between a 0 and 1 even in the presence of a significant amount of noise. It is possible that the noise could be
severe enough that the receiver gets confused, and incorrectly decides it is receiving a 0 when it should be deciding a 1 (or
vice versa)…these are referred to as bit errors. But in general, digital systems are much better in noisy environments.
In long distance digital communications, digital signals can be stripped of any noise in a process called signal regeneration.
Consider a long distance transmission that incorporates a set of relay stations in order for the signals to move from transmitter
to receiver, such as what is used to connect the east coast of the US to the west coast.. Relay stations are needed because the
farther a signal travels, the weaker it gets; to make it to its destination, it must be amplified and retransmitted at the relay
stations.
If this was an analog system, the analog signal is received, amplified and retransmitted at each station. However, noise is now
a part of the signal, and so is also amplified at each station.
In a digital communication system, a digital signal is received (receiver decides 0s or 1s), regenerated (digital signal recreated
based on the 0s and 1s), and then retransmitted at each station. With signal regeneration, the noise can be eliminated at each
station. This can only be done in digital communication systems.
3.2 Error detection/correction Digital signal processing (DSP) techniques allow the detection and correction of bit errors.
Even if a digital signal contains bit errors, many of these errors can be fixed at the receiver through the use of error
correcting codes. Error correcting codes allow, for example, CDs with minor scratches to be played without errors. Analog
systems cannot detect or correct errors.
3.3 Easier multiplexing Multiplexing is the process of allowing multiple signals to share the same transmission channel. For
example, digital telephony allows carrying 24 phone conversations on a single wire (called a T1 line) at the same time.
Digital signal processing techniques enable this.
3.4 Easier to process and store Since computers store and use digital data, digital signals can be easily processed by
computers. Similarly, the digital format lends itself to easier storage of communication signals (e.g., smaller storage
footprint). DSP allows operations such as filtering, equalization and mixing to be done in software without the use of analog
circuits. DSP also permits data compression (transforming signals so that fewer bits are needed to represent them). An
example of DSP would be Garage Band, for you musicians, or photo editing software like Adobe Photoshop, for those with a
knack for photography.
To emphasize this again, these advantages are huge. This is such a big deal that even though communication systems used to
be exclusively analog, it is worth the billions and even trillions of dollars that the government and private sector are spending
to migrate communication systems to digital. In contrast, some disadvantages include: bandwidth is about twice that of
analog, and circuitry is more complex, but generally smaller and less expensive.
4. Conversion from Analog to Digital (A/D)
If nature produces analog signals, how do we create digital signals from them? Before we can use digital transmission, we
must convert the signal of interest into a digital format. The natural signal (e.g., speech) that we want to transmit will be
acquired using an analog device. The analog signal will be translated into a digital signal using a method called analog-to-
digital (A/D) conversion. The device used to perform this translation is known as an analog-to-digital converter or ADC.
Through A/D conversion, analog signals are changed into a sequence of binary numbers (encoded bits), from which the
digital signal is created by the transmitter. This process is depicted below.
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There are two major steps involved in converting an analog signal to a digital signal represented by binary numbers:
sampling, and quantizing/encoding.
Steps for A/D conversion:
4.1 Sampling This is a process of inspecting the value (voltage) of an analog signal at regular time intervals. The time
between samples is referred to as the sample period (T, in seconds), and the number of samples taken per second is referred to
as the sample frequency (fs, in samples/second or Hz). Basically, sampling is taking snap-shot values of the analog signal so
that you have an accurate representation of how the analog signal is changing over time.
The receiver must convert the bits it receives into sample values, and then recreate what it thinks the analog signal looks like
from the samples alone. As you might deduce from the figure below, when the samples are closer together (smaller sample
period, which means higher sample frequency), the analog signal is more accurately represented. Note that with the lower
sample rates, some of the fluctuations in the analog signal have no samples on them, so the samples are not a good
representation of the analog signal. How high does our sampling frequency fs need to be in order to accurately represent the
signal? That is, what is the minimum sample frequency for the A/D to work properly?
We could consider taking just a few samples (i.e., using a low sampling rate), which means less information to transmit to the
receiver. But if we choose that option, when we reconstruct the signal, it will likely be a terrible representation of the
original. The low sampling rate will only work well for very slowly changing (low frequency) signals. Alternatively, we
could choose the highest possible sampling rate known to man, to ensure that we can accurately capture even very fast signal
fluctuations. But the higher the sampling rate, the higher the cost of the equipment and more information must be transmitted.
In addition, if we decide to record the communications our saved files will be unnecessarily enormous.
But what is “low” and what is “high”? In other words, how exactly do I go about choosing my sampling rate? In order to
accurately reconstruct an analog signal from its samples, one must sample faster than the Nyquist sampling rate (also called
the Nyquist rate), fN, given by the formula
𝑓𝑁 = 2𝑓𝑚𝑎𝑥 , where fmax is the highest frequency component of the analog signal.
That is, the sampling frequency must be more than twice the value of the highest frequency component of the signal:
f s  f N , where fN = 2fmax
If the sample rate is not greater than the Nyquist rate, a problem called aliasing results, which can cause severe distortion of
your signal.
The Nyquist sample rate is a floor on the sampling rate, and practical systems sample greater than the Nyquist rate. Some
examples of common sample rates are:
Signal Signal frequency range Standard Sample Rate

Voice 300 Hz-3 kHz 8 kHz
Music 0-20 kHz 44.1 kHz (CD-quality)
Music 0-20 kHz 192 kHz (DVD-quality)
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Consider the signal from the oboe depicted below in time and frequency domain representations.
(a) What is the maximum frequency present in the oboe signal?
(b) Based upon the max frequency, what sampling rate must be exceeded in order to accurately reconstruct
the signal from its samples?
1 0.25
0.2
0.5
Voltage (V)
Voltage (V)
0.15
0
0.1
-0.5
0.05
-1
1 1.0005 1.001 1.0015 1.002 1.0025 1.003 1.0035 1.004 1.0045 1.005 0
0 1000 2000 3000 4000 5000 6000
Time (sec)
Frequency (Hz)
Solution:
(a)
(b)
4.2 Quantizing/Encoding Quantizing/encoding is the process of mapping the sampled analog voltage values to discrete
voltage levels, which are then represented by binary numbers (bits). This is needed because the analog sample values are real
numbers that occur on a continuum. That is, for example, if a sine wave of amplitude 1V is being sampled, the sample values
could be any value between -1V and +1V… an infinite number of possibilities. In any digital system, there is only a finite
amount of memory, so only a finite number of values can be used to represent the samples of the analog signal. Converting a
sample value from the set of infinite possibilities to one of a finite set of values is called quantization or quantizing. These
values are referred to as quantization levels.
Inputs to A/D converters are limited to a specific voltage range. For the sine wave example above, we assumed that all values
of the analog input fall within a range of -1.0 to +1.0 volts (note: this is the typical voltage range of voice or music signals on
a computer, such as in .wav or .mp3 files).
A/D systems are characterized by the number of bits they have available to perform quantization. The number of bits
determines the number of quantization levels. An N-bit A/D converter has 2N quantization levels and outputs binary words of
length N (that is, it outputs N-bit values for every sample). For example, a 3-bit A/D system has 23 = 8 quantization levels, so
all samples of a 1V analog signal that is input to this A/D will be quantized into one of only 8 possible quantization levels
and each sample will be represented by a 3-bit digital word. In general, the A/D converter will partition a range of voltage
from some vmin to some vmax into 2N voltage intervals, each of size q volts, where
vmax  vmin
q .
2N
Some common examples of A/D quantizing are digital telephony, which uses 8-bit A/D (28 = 256 quantization levels), CD
audio, which uses 16-bit A/D (216 = 65,536 quantization levels), and DVD audio, which uses 24-bit A/D (224 = 16,777,216
quantization levels).
The following figures represent conceptually how a 3-bit A/D converter converts an analog signal into bits. In these figures,
the analog signal is shown as well as the samples, with samples taken every 0.5 msec (corresponding to a sample rate of fs =
1/0.0005 sec = 2000 samples/sec). The actual analog sample voltages are shown in parentheses next to the samples. Here, the
voltage range of the signal is divided into 23 = 8 smaller voltage intervals (also called steps). These are separated by the
dashed, bold horizontal lines, and each interval is 0.25V wide:
𝑣𝑚𝑎𝑥 − 𝑣𝑚𝑖𝑛 1 − (−1)
𝑞= = = 0.25𝑉
2𝑁 8
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The value of q is more formally called the quantizer’s resolution.
Each of the voltage intervals is assigned an N-bit binary number representing the integers from 0 to 2  1 . For this example,
N
you can see that since we are using a 3-bit A/D, the intervals will be assigned binary numbers representing the integers from
0 to 7 (that is, 000, 001, 010, …, 111), starting from the bottom of the voltage range. In this case, the digital word 000 is
assigned to the voltages from -0.75 V to -1.0 V, 001 is assigned to the voltages from -0.5 V to -0.74999 V, and so on. The
figure that follows shows for each quantization interval the associated 3-bit digital word (on the left side of the plot). Any
analog sample that falls in a given voltage interval will result in those 3 bits being transmitted.
When a sample point falls within a given interval, it is assigned the corresponding binary word (this is the Encoding part of
Quantization/Encoding). For the first sample point at time 0, the voltage is 0.613 V, which means that sample is assigned a
binary value of 110. The A/D then creates a voltage signal that represents these bits, and that process continues as long as an
analog signal is input to it.
The binary representation of the above signal is:
110 101 100 011 011 100 110 110 100 010 000 000 001.
In this example, every sample produces 3 bits (that is, there are 3 bits/sample). The sample rate was 2000 samples/sec.
Multiplying these two values together results in the bit rate (Rb) produced from this A/D conversion:
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3 bits 2000 samples

Rb    6000 bits/sec (bps)
sample sec
Bitrate is the speed of transfer of data given in number of bits per second.
To the right of the plot above is the quantization level associated with each voltage interval. Any analog sample voltage that
falls in a given interval is effectively estimated to the center of its quantization level when it is desired to reconstruct the
analog signal from the received bits (a receiver may perform this). This process is referred to as Digital-to-Analog conversion
(D/A) and will be discussed briefly in the next section. For this example, the quantization level for the lowest voltage interval
is the value halfway between -.75 V and -1 V (which is -0.875 V). This means that any analog sample that fell into this range
will be represented as -0.875 V.
Alright, we’ve walked through an A/D example together; now it’s your turn. Work through the problem on the next page.
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Consider the following analog waveform. This waveform is sampled at a 500 Hz rate and quantized with a 2-bit quantizer
(i.e., A/D converter) The input range is -1.0 to +1.0 V.
(a) Circle the sample points (first sample is at time t = 0 sec).
(b) Indicate the quantization intervals and corresponding digital words.
(c) Indicate the digital word assigned to each sample point.
(d) What is the stream of binary bits generated after the A/D conversion is complete?.
(e) What is the resulting bit rate from this A/D?
amplitude (volts)
time (msec)
Here is an example of a digital voltage waveform that might have been generated from an A/D process:
This waveform could be transmitted from the receiver to the transmitter over a wire using the voltage pulses shown, but is not
suitable to transmit wirelessly through the atmosphere. We’ll get into more detail about how this is done in our next chapter
on digital modulation.
To give you an idea of how quantization affects a digital picture, here’s a comparison of quantizing the same scene with 4
bits per pixel (16 levels) and with 8 bits per pixel (256 levels). Look at a color display of this picture (such as the picture in
the pdf file of this page in the notes posted on the course website). See the difference?
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4 bits/pixel = 16 colors 8 bits/pixel = 256 colors

5. Conversion from Digital to Analog (D/A)
But how do we recover the analog information after it has been converted to digital? As mentioned earlier, the receiver
converts these N-bit digital words back into an analog signal. This process is called digital-to-analog (D/A) conversion. It is
very similar to being the reverse of the analog-to-digital conversion process. The analog signal is reconstructed by converting
the N-bit digital words into the appropriate quantization levels, and this voltage is “held” for one sample period, creating a
stair-step type signal shown below.
Good job. We’ve regenerated our original signal. How does it compare with the original? Let’s see. The reconstructed
analog signal for our 3-bit example is shown in a thick black line in the next figure, along with the 3-bit digital word that
represents each sample. The original analog signal is also shown in the continuous line, along with all of the sample points
that were on the earlier figures.
Is it close? It follows the same general shape. Even if we perform filtering to smooth out the reconstructed signal to remove
its staircase appearance (which is typical) it will still not quite be the same as the original red signal. Why? Is that the best we
can do?
6. Quantization Error (also called Quantization Noise).
There is always error introduced with the A/D process. The error is the difference between the original analog signal and the
reconstructed (stair-step) signal after A/D and D/A. The following figure is a portion of a music signal that has been
quantized with 3 bits. The upper plot shows the original analog signal along with the recovered analog signal from the A/D
process. The bottom plot is the quantization error, which is created by subtracting the recovered signal from the original
analog signal at each instance of time.
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So is it bad? It can be. The quantization error manifests as noise in the reconstructed analog signal. For digital audio signals
(music or voice), it can sound like static, which is why it is also called quantization noise. The greater the quantization error,
the louder the static, making it harder to hear the voice or music. Reiterating what was presented in Chapter 19: NOISE IS
THE NUMBER ONE LIMITING FACTOR IN COMMUNICATION SYSTEMS. In this case, if quantization is part of the
communication system (e.g., using a digital communication system to transmit analog information), then the A/D process
adds even more noise to the signal as it moves from transmitter to receiver.
So how do we reduce the quantization error and its associated noise? Quantization error can be reduced by increasing the
number of bits N for each sample. This will make the quantization intervals smaller, reducing the difference between the
analog sample values and the quantization levels. The figure below is the same analog signal quantized with 4-bits per
sample. Note the step-size is smaller than in the 3-bit plot, (½ the size), and the noise signal is approximately ½ the
amplitude of what it was with 3-bit quantization. The reconstructed signal looks much closer to the original analog signal
compared to the 3-bit A/D. It is worth noting that increasing the sampling frequency will not reduce quantization noise, only
increasing the number of quantization levels will do this.
We of course can’t use an infinite number of bits, so some quantization noise is always inevitable, but the nice thing about
the human ear/brain - sticking with the example of audio signals - is that beyond a certain number of bits for each sample, the
associated quantization noise becomes imperceptible. We just need enough bits to make the recovered signal “good enough”
(e.g., the recovered music sounds “good enough”).
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CH. 21 Problems
1. What is the greatest advantage that digital communication has over analog communication?
2. Describe the function of a regenerative repeater.
3. What is the cause of aliasing in the A/D process?
4. Why does a 5-bit quantizer produce a better approximation to an analog signal than a 3-bit quantizer?
5. A music signal has frequency content from 0 Hz up to 18.75 kHz. What sampling frequency must be exceeded for
successful A/D conversion? What is another name for the minimum sampling frequency?
6. Consider the following analog waveform. This waveform is to be sampled at a 1-kHz rate and quantized with a 3-bit
quantizer (input voltage range is -1.0 to +1.0 V).
(a) What is the resolution (q) of this quantizer?

(b) Circle the sample points on the analog waveform below. Note that the time axis is in msec.
(c) Indicate the quantization intervals and corresponding digital words.
(d) Indicate the digital word assigned to each sample point.
(e) When a receiver receives the transmitted bits, D/A is used to recover the analog signal, but the recovered signal is
not the same as the original analog signal. What is the term to describe this difference and what can be done to
minimize this difference?
1
0.75
0.5
0.25
amplitude (V)
-0.25
-0.5
-0.75
-1
0 1 2 3 4 5 6 7 8 9 10
time (msec)
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7. Consider the following analog waveform. This waveform is to be sampled at a 1.333333 MHz rate and quantized with a 3-
bit quantizer (input voltage range is -2.0 to +2.0 V).
(a) What is the resolution (q) of this quantizer?

(b) Circle the sample points on the analog waveform below. Note that the time axis is in μsec.
(c) Indicate the quantization intervals and corresponding digital words.
(d) Indicate the digital word assigned to each sample point.
1.5
0.5
amplitude (V)
-0.5
-1
-1.5
-2
0 0.5 1 1.5 2 2.5 3 3.5 4 4.5 5
time (microsec)
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A/D Conversion: Aliasing and Quantization Noise

Discussion: In this chapter, you learned how analog-to-digital conversion (A/D) is accomplished in two steps. The first step
is sampling the analog signal, and the second step is quantization/encoding. Sampling must be done at some minimum rate,
depending on the frequency characteristics of the input signal. If the signal is not sampled fast enough, distortion called
aliasing will occur. Quantization/encoding is the process of mapping the sampled analog voltage values to discrete voltage
levels, which are then represented by binary numbers (bits). Since we are mapping a continuum of possible analog sample
values to a finite set of possible discrete voltage levels, the A/D process always introduces an error we call quantization
noise. The severity of the noise depends on how many bits are assigned to each sample; the more bits, the lower the
quantization noise and the better the digital result.
Objective: To provide hands on experience and further familiarize each Midshipman with the effects of aliasing and
quantization noise using CD-quality audio signals.
Part 1: Setup Windows Media Player
For this SX, we desire to view the music signals playing as if they were on an oscilloscope. Windows Media Player has an
option that provides an oscope display, but it is not the default media player for .wav files and it is not so easy to figure out
how to get the oscope display. Follow these instructions.
□ Set Windows Media Player to be the default media player for music files:
o Press the Windows key and type in “default”, choose the top matching choice (should be called “Default
Programs”)
o Under Music Player, choose “Windows Media Player”
□ On your laptop, open Windows Media Player, or if you don’t know how to find it,
o -Press the Windows key + R
o -Type: wmplayer.exe
o -Hit Enter
When Windows Media Player opens, it probably looks like this:
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Now, right click in the area to the right of the word “Music” as shown below, then select ViewNow Playing to get to the
“Now Playing” screen.
Right-click in this area
The Windows Media Player “Now Playing” window should now look like this:
Finally, right-click in the black area, then choose VisualizationsBars and WavesScope. Windows Media Player is now
set to display an oscope when music plays. To play a song with it, right click on the .wav file, then select “Play with
Windows Media Player.”
Part 2: Aliasing
To prevent aliasing from occurring, the analog input signal must be sampled at a rate at least twice its highest frequency
content. This minimum sample rate is referred to as the Nyquist rate. We will investigate aliasing distortion using some CD-
quality audio clips of some of your favorite songs (or at least some of your older professors’ favorite songs). CD quality
music is sampled at a rate of 44,100 samples/sec (44.1 kHz), which is a higher rate than twice the frequency content of music
(around 20 kHz, so the Nyquist rate is approximately 40 kHz). What if the music were instead sampled at less than the
Nyquist rate? What happens?
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In the following figure, you see the plot of a short clip from Prince’s song “1999”. This plot represents the vibrations of a
microphone when the song was originally recorded (an analog signal), sampled at 44.1 kHz. Higher frequency means faster
vibrations of the microphone.On the same plot, you see another signal which is the original signal sampled by a factor of 25
less than the 44,100 Hz rate (i.e., sampled at 44,100/25 = 1764 Hz), which is far below the minimum required sample rate to
prevent aliasing. The samples for this aliased signal are shown in circles, and a line has been drawn to connect the circles.
The aliasing comes about because by sampling with fewer samples per second, the aliased signal misses many of the rapid
vibrations of the microphone that occur between the samples we are using; so when we listen to the aliased signal, we do not
hear them. We will not hear the higher frequency sounds we expect to hear, and in face, there will be false vibrations
introduced in the signal that weren’t there in the original (referred to as the foldover of the actual higher frequency
information into lower frequency information). In fact, whatever the sample rate is that we choose to sample with (call it fs),
the highest frequency we could possibly get out of any A/D process is fs/2. In this case, sampling at 1764 Hz, the highest
frequency possible in the aliased signal would be 882 Hz.
0.3
0.2
0.1
Amplitude (V)
-0.1
-0.2
-0.3
0 0.005 0.01 0.015 0.02
Time (sec)
So what does aliasing sound like? Let’s hear some…
□ Create a folder on your laptop’s desktop called “SX21”, and open that folder in Windows Explorer.
□ Access the U:\Electrical and Computer Engineering\Cyber2\EC310\SX21 folder and copy the Windows executable
file called Aliasing.exe, and put it in your “SX21” folder. Your folder should now look something like the
following:
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□ Since we will be using a Windows command prompt and navigating to this folder, copy the absolute path to this
folder by using the mouse to click the “folder” icon indicated below, which changes your view to the following
where the absolute path is highlighted in blue. Copy the highlighted path to the Clipboard.
“folder” icon Absolute path
□ Open a Windows command prompt, and navigate to your “SX21” folder on your desktop by typing cd, then pasting
the absolute path to the SX21 folder: In the command window, you should see:
□ Run the Aliasing.exe program from the command line to see how to use it…type
.\Aliasing.exe
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Note that contrary to how we ran programs in Linux, the name of the program begins with a period and back-slant in
Windows, vice period and forward-slant. The usage of the Aliasing.exe program looks like the following:
As you see, you need to provide two inputs, the name of a .wav file and the factor by which you wish to reduce the sample
rate (called “aliasing factor”).
For example, to reduce the sample rate of a file by the group Devo called “Whip It” by a factor of 10, you’d type:
.\Aliasing.exe –i Devo-WhipIt.wav –a 10
In this case, the .wav file that has the aliasing will be called “Devo-WhipIt-alias-10.wav”, and you can hear the aliased
version using Windows Media Player by double clicking on that file in your SX21 folder.
□ Return to the SX25 folder on the U drive to find several audio clips (.wav files). These are CD-quality, so all of
them have a sample rate of 44.1 kHz.
Question 1: If CD-quality music is collected at 44,100 samples/sec, with 16-bits/sample, and there two channels of recording
(a left and right channel—it was recorded with two microphones), what is the resulting bit rate that is generated for CD
quality?
□ Choose one of the music clips and listen to it in Windows Media Player. Be sure to enlarge the Windows Media
Player Window when you play something. Then run Aliasing.exe with a factor of 5 on it, and listen to the result in
Windows Media Player.
Question 2: What music clip did you choose?
Question 3: Run the program again with progressively higher aliasing factors. At what value of the aliasing factor do you
start to notice the aliasing?
Question 4: What is the sample rate corresponding to this aliasing factor (when you started to notice the aliasing)?
Question 5: What would be the bit rate generated if we had performed the A/D using the sample rate from Question 4 (with
2 channels, 16-bits/sample)?
□ Try some extreme aliasing, using a factor of 50, or 100.
Question 6: Describe the audible effects of aliasing as the aliasing factor increases.
Part 3: Quantization Noise
Quantization noise is always present in a digitized music signal, however it may or may not be noticeable depending on how
many bits are used for each sample. CD-quality music is 16 bits per sample, and the standard for music files is that the music
signal will have values anywhere between -1 V and +1 V. This means that the range of voltages from +1 V to -1 V is divided
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into 216 = 65,536 voltage intervals, so very small changes in the analog signal’s amplitude are captured well. This is very
good quality, and the music is pleasing to the ear. Quantization noise is present, but very small so we don’t hear it.
On the other hand, if we reduce the number of bits per sample very much, we will definitely hear the quantization noise…
□ Copy the “QuantizationNoise.exe” file from the U drive.
□ Run the QuantizationNoise.exe from the command line to see how to use it…type
.\QuantizationNoise.exe
Like Alias.exe, you need to supply two inputs, the name of the .wav file and the number of bits to use in quantizing each
sample. The number of bits you choose should be an integer from 1 to 15 (since the original clip uses 16 bits/sample), and as
you reduce the number of bits, the result sounds worse.
□ Choose one of the music clips and listen to it in Windows Media Player. Then run QuantizationNoise.exe with
number of bits equal to 15, and listen to the result in Windows Media Player. Note: using a higher value of number
of bits will take the program longer to run than fewer number of bits, so be patient.
Question 7: What music clip did you choose? At 15 bits/sample, do you notice the quantization noise?
Rerun the program and reduce the number of bits per sample until you can start to hear the quantization noise.
Question 8: At what value of number of bits do you start to notice the quantization noise?
Question 9: What would be the bit rate generated if we had performed the A/D using the number of bits/sample from
Question 9 (with 2 channels, 44,100 samples/sec)?
□ Introduce some extreme quantization noise, using a number of bits of 2 or 1.
Question 10: Describe the audible effects of quantization noise as the number of bits is reduced.
Part 4: Clean Up
If desired, change your Music Player back from Windows Media Player to Groove Music:
□ Restore/change the default media player for music files:
o -Press the Windows key and type in “settings”, choose the top matching choice (should be called “Settings-
Trusted Windows Store App”)
o -Choose “Apps”, then “Default Apps”, then click on the app shown under “Music Player”. Change your
music player from whatever it is (probably “Groove Music”) to “Windows Media Player”.
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Name:
Question 1 (show calculations):
Question 2:
Question 3:
Question 4:
Question 6:
Question 7:
Question 8:
Question 10:
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Chapter 22: Digital Modulation
Objectives:
(a) Quantitatively describe the relationship between a symbol and a bit and the bit rate and the baud.
(b) Describe how digital information is conveyed using various digital modulation techniques (ASK or OOK, FSK, PSK and
QAM) and recognize their waveforms, and constellations.
(c) Calculate the bandwidth of an ASK, FSK, PSK, or QAM signal.
(d) Using a constellation diagram analyze a M-ary PSK signal to determine its symbols and bits per symbols.
(e) Discuss the effect of noise on M-ary PSK and how Quadrature Amplitude Modulation (QAM) overcomes these
detrimental effects.
1. Digital Signal Frequency Spectrum

In Chapter 21, it was mentioned that in many cases, we wished to convert analog signals into digital signals to take advantage
of the benefits of digital technologies. Samples of the analog signal were converted into bits and the bits were then used to
create a binary voltage waveform that represented the bits. If we then wanted to transmit this digital waveform through free
space, then all we need to do is connect it to an antenna, right?
No, it is not that easy. The binary voltage waveforms to which we are so accustomed are, typically, voltage pulses that
alternate between 0V (for a 0-bit ) and 5V (for a 1-bit). It just so happens that the preponderance of frequency content in
these voltage pulses is very low (a baseband signal), and just like was pointed out for voice signals (which also have low
frequency content), an antenna needed to transmit this kind of signal through free space would be impractically large.
For a large number of random voltage pulses, the frequency plot would look something like the following, where Rb is the
value of the bit rate in Hz.
For example, if the bit rate were 500 bps, then the frequency content magnitude would be equal to zero at 500 Hz, 1000 Hz,
etc. This plot of frequency content is much different than that of a signal composed of sinusoids! There are no spikes!
Nevertheless, most of the frequency content is at very low frequencies. The frequency content does continue out to an infinite
frequency, although the magnitude drops dramatically at higher frequencies. In a perfect world, we’d say the bandwidth of
voltage pulses approaches ∞ Hz, but for digital signals, we’ll use the null-bandwidth as our calculated bandwidth. The null-
bandwidth is defined as the amount of the frequency spectrum (in Hz) from the maximum magnitude (which occurs at 0 Hz)
to where the spectrum first goes to a magnitude of 0 (called a null, here at Rb Hz). The bandwidth is given by:
𝐵𝑊 = 𝑓2 − 𝑓1 = 𝑅𝑏 − 0 = 𝑅𝑏 𝐻𝑧
We must come up with a method to transmit the digital information (1s and 0s) using radio waves. Digital modulation
techniques allow this. As you recall, the goal of modulation is to upshift the frequency spectrum of the information signal to
allow transmission through free space; the transmitted signal’s frequency spectrum would then look like the following.
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Recall that, like in analog amplitude modulation, the information signal’s frequency spectrum is shifted up by fc Hz, and there
is a mirror image of the frequency content on the left side of fc. The transmission bandwidth (using the null-bandwidth
definition) is now
BW  f 2  f1  ( f c  Rb )  ( f c  Rb )  2Rb Hz
2. Binary Digital Modulation
Recall the equation for a high frequency carrier: vc(t)=Vc sin(2πfct + θ). As discussed in Chapter 20, a sinusoidal carrier can
be modulated by varying its amplitude, frequency, or phase using an information signal. So, how do we go about representing
1s and 0s with modulation? Just as we can vary amplitude, frequency, and phase of a high-frequency carrier in accordance
with an analog waveform, we can do the same with a digital waveform. Since bit values shift between 0s and 1s, digital
modulation techniques that vary the carrier’s amplitude, frequency, and phase are referred to as “shift keying.”
2.1 Frequency Shift Keying (FSK) Frequency-shift keying (FSK) is a frequency modulation scheme in which digital
information is transmitted through discrete frequency changes (shifts) of a carrier wave. The simplest form of FSK is Binary
FSK (BFSK), in which a carrier’s frequency is shifted to a low frequency or a high frequency to transmit 0s and 1s. The plot
below shows a sample FSK signal along with the associated bits.
An example of how FSK was used “back in the day” was with dial-up modems to connect your home computer to your
Internet service provider over your analog phone. With a modem, a 0-bit was represented with a lower frequency carrier of
1070 Hz and a 1-bit was represented with a higher carrier frequency of 1270 Hz. The lower frequency, binary 0, was called
the “space” frequency while the higher frequency, binary 1, was called the “mark” frequency. The terms mark/space were a
throwback to the days of Morse code or flashing light communications.
In the frequency domain, we consider FSK to be two different digital transmissions, one at the mark frequency (the higher
frequency) and one at the space frequency (lower frequency). The resulting frequency plot would look like the following,
with the carrier frequency being shifted between the mark and space frequencies.
The amount that the carrier frequency can be shifted is called the frequency deviation (Δf). To determine the bandwidth for
FSK modulation, we take a closer look at the frequency spectrum around the mark and space frequencies. We use the null-
bandwidth definition to compute the bandwidth as shown below.
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In the figure, the bandwidth effectively runs from the first null to the left of fspace to the first null to the right of fmark.
Mathematically, there are two equations that can be used to compute the bandwidth:
𝐵𝑊 = (𝑓𝑚𝑎𝑟𝑘 + 𝑅𝑏 ) − (𝑓𝑠𝑝𝑎𝑐𝑒 − 𝑅𝑏 ) = 𝑓𝑚𝑎𝑟𝑘 − 𝑓𝑠𝑝𝑎𝑐𝑒 + 2𝑅𝑏

or
𝐵𝑊 = 2(Δ𝑓 + 𝑅𝑏 )

You have an FSK transmitter using a carrier of 500 kHz sending 10 kbps and a frequency deviation of 100 kHz. How much
bandwidth do you need for your transmission?
Solution:
Of course, who still uses dial-up? What else is there?

2.2 Amplitude Shift Keying (ASK) and On-Off Keying (OOK) Amplitude Shift Keying is a form of amplitude modulation
that represents digital data as shifts in the amplitude of a carrier wave: for example, small amplitude for a 0-bit, and larger
amplitude for a 1-bit. We have seen what an ASK signal has looked like before in Chapter 21, repeated below.
The simplest digital modulation scheme is a form of ASK called on-off keying (OOK). This is analogous to Morse code. In
OOK, a carrier is transmitted for a 1-bit and nothing is transmitted for a 0-bit; this is the same as saying that the smaller ASK
amplitude is 0.
Note that in all forms of ASK, the frequency and phase of the carrier are the same for all outputs; it is the amplitude that
changes.
Sketch an OOK signal that represents the bit stream below.
1 0 0 0 1 1
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Before we continue, you need to learn some important terms that used in digital communication systems. The information is
carried in the bits that are transmitted, but we don’t actually transmit bits; we transmit waveforms that represent bits. These
waveforms are commonly referred to as symbols. On a wire, the symbols take the form of voltage pulses. In FSK and OOK,
the symbols take the form of a high frequency carrier that has its frequency or amplitude altered based on whether a 0-bit or a
1-bit is being transmitted. In these modulation schemes, the number of symbols that can be transmitted (M) is two (M = 2)
and each symbol represents one bit of data. For FSK and OOK, the time duration of a bit is the same as the time duration of a
symbol (Tb = Tsym).
We will soon see other digital modulation schemes where a symbol can represent more than one bit. In general, the number
of symbols for a modulation type is related to the number of bits associated with each symbol. If N is the number of bits per
symbol,
M  2N and N  log 2 M .
The relationship between bits and symbols for an OOK signal is shown in the next figure for an OOK signal.
Bitrate (Rb) is the speed of transfer of data (number of bits per second). Bitrate is inversely related to bit duration (Tb),
which is the time required to transmit a single bit.
1
Rb =
Tb
Baud (also referred to as Symbol Rate) (Rsym) is the number of symbols transmitted per second, and is inversely related to
the Symbol duration (Tsym), which is the time required to transmit one symbol.
1
Rsym =
Tsym
The Bitrate and the Baud (or Symbol Rate) are related by the number of bits per symbol (N).
𝑅𝑏 = 𝑅𝑠𝑦𝑚 × 𝑁
The bandwidth associated with OOK is what we have seen before, BW = 2Rb, as shown in the figure below.
As you’ll see shortly, the symbol rate (Rs) has a noted effect on the bandwidth required for transmission. In general, for all
digital modulation schemes that we will discuss (except for FSK), bandwidth is given by:
2Rb .
BW =
N
In the case of OOK, since N = 1 bits/symbol, BW = 2Rb = 2Rsym, as stated before. For example, for OOK, if the bitrate is 600
kbps, the symbol rate is 600,000 symbols/sec, and the bandwidth is 2(600,000) = 1.2 MHz.
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2.3 Phase Shift Keying (PSK) Phase shift keying (PSK) is a form of phase modulation where the carrier’s phase shifts to
one of a finite set of possible phases based on the bits that are input. For binary phase shift keying (BPSK), the carrier phase
is shifted between one of two phases (typically 0 and 180) depending on whether a 0-bit or a 1-bit is being transmitted. For
example:
0-bit: the symbol transmitted is Vc cos(2πfc t).
1-bit: the symbol transmitted is Vc cos(2πfc t +180°) = − Vc cos(2πfc t)
It is important to point out that in PSK, the amplitude of all output symbols is the same; it is the phase of the output symbols
that are different.
Up to this point we have discussed digital modulation with one bit per symbol, which means that at any time, one of two
possible symbols would be transmitted. But as mentioned earlier, it is possible to have a modulation scheme with more than
one bit per symbol; this is referred to as M-ary digital modulation.
3. M-ary Digital Modulation
Before launching into more complicated digital modulation, we’ll introduce a graphical way to relate output symbols to the
bits they represent. This is called a constellation diagram. A constellation is a plot of relative amplitude and phase of the
output symbols for a digital modulation system. Each dot describes a symbol which is represented by its polar coordinates. In
terms of phase, 0° is along the positive x-axis, and phase increases as you move counterclockwise around the x-y plane.
Relative amplitude is measured as distance from the origin of the plot. The possible output symbols are represented with
filled-in circles, and adjacent to them are the bits they represent.
For example, here are two possible BPSK systems’ constellation diagrams. In BPSK, the output symbols both have the same
amplitude (both of the symbols are equidistant from the origin), but their phases are 180° apart. There are other possible
combinations of two carrier phases that might be used (such as +90° and -90°), but the actual constellation used is not
important, as long as the transmitter and receiver use the same constellation.
Note that BPSK transmits 1 bit per symbol, so only one bit value is placed next to each symbol.
If it is desired to get the information from the transmitter to the receiver faster, we need to increase the number of bits per
second (bps) that are transmitted. The cost of increasing the bitrate (besides requiring more complex components) is that it
increases the transmission bandwidth: recall that for OOK BW = 2Rb, and from Chapter 19, that bandwidth can be expensive!
Is there a way to transmit a higher bitrate but using a smaller transmission bandwidth? The answer is yes, using M-ary digital
modulation.
In M-ary modulation, we can preserve bandwidth if we keep the symbol rate the same and increase the number of bits per
symbol. For example, instead of transmitting just 2 possible phase shifts (0˚and 180˚), we could transmit one of 4 possible
phase shifts per symbol. This is called quadrature phase shift keying (QPSK).
3.1 Quadrature Phase Shift Keying (QPSK) In QSPK, there are 4 symbols (M = 4) and there are 2 bits per symbol (N = 2
= log2M). Two of the many possible constellation diagrams for QPSK are shown in the following figure 3, and the four
symbols from QPSK Constellation #2 are shown to the right of this constellation. The carrier with a phase of 0˚ is plotted in a
dashed red line with each symbol for reference. The four symbols in the righthand constellation are: Vc cos(2πfc t +45°), Vc
cos(2πfc t +135°), Vc cos(2πfc t -135°) and Vc cos(2πfc t -45°).
3
The points in the picture for QPSK Constellation #2 are labeled using gray code where only one bit changes between
adjacent coordinates.
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The following figure is a plot of the use of QPSK constellation #2 to transmit the bit stream 0001111000110110. Also shown
is the bit duration, and the symbol duration for QPSK.
The frequency spectrum for M-ary modulation schemes is shown in the figure below, which also specifies the frequency axis
for QPSK. If the bitrate is constant, the benefit of transmitting more than one bit in a symbol can be seen in the fact that the
nulls are closer to the carrier frequency.
From the figure, it is seen that the bandwidth for QPSK is given by
 R   Rb 
BW   f c  b    f c    Rb Hz.
 2   2 
This is confirmed by the equation for bandwidth for all digital modulation schemes (except for FSK),
2R
BW = b
N
where N = 2 for QPSK. For example, if bitrate is 600 kbps, BW = 2(600,000)/2=600 kHz.
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3.2 M-ary PSK

We can further increase the number of bits per symbol by increasing the number of
possible phase shifts. The M in M-ary refers to the number of symbols. Consider the
8-PSK constellation to the right (one of many possible 8-PSK constellations4).
How many bits per symbol are transmitted? There are 8 symbols (M = 8), so N =
log2M = log28 = 3 bits/symbol. This is also evident from the diagram because the
three bits associated with each symbol appears next to the symbol.
What is the bandwidth for 8-PSK? Since N=3 bits/symbol, Bandwidth is given by
2 Rb 2 Rb
BW   .
N 3
For example, if the bitrate is 600 kbps, bandwidth for 8-PSK is BW = 2(600,000)/3
= 400 kHz.
We could further increase to 4 bits/symbol using 16-PSK. Here, M = 16 and N = 4
bits/symbol. A 16-PSK constellation is shown to the right, where each phase is
separated by 360o/16 = 22.5o. More complex M-ary PSK modulation is possible:
16-PSK, 32-PSK, etc., but it becomes more susceptible to noise as the symbols get
closer together. As a reminder, for PSK, all of the symbols have the same carrier
frequency and amplitude; it is their phase that is different. For that reason, on a
constellation diagram, all of the symbols for PSK appear on a circle about the
origin.
To demodulate any type of PSK, a receiver must determine the phase of the received symbol. For 16-PSK, the receiver must
determine the phase within 11.25˚, since the phases are separated by 22.5 o. A portion of the constellation diagram for 16-
PSK is shown to the right, indicating the wedge of phase values that separates one of the symbols
from the adjacent symbols.
Noise Effects
Recall that the number one most limiting factor in communication systems is noise. In all
transmissions, the received signal will be degraded by noise. The following figure shows a BPSK
signal and the same signal corrupted by noise. You might imagine that it is harder for a receiver to
determine the correct phase (correct symbol) that was transmitted for the noisy signal.
This noise corruption can be depicted in the constellation diagram to the

right, where the two transmitted BPSK symbols are indicated in the two
large black circles (phase = 0° and phase = 180°), and noisy received
symbols are the red and blue circles.
4
The points in this picture are labeled using gray code where only one bit changes between adjacent coordinates.
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A BPSK receiver must make a decision to determine the phase of a received signal to determine the corresponding bit. If the
noise is severe enough, a receiver might make a mistake, and
decide that it had received a 0-bit when it actually received a 1-bit.
These are called bit errors. Now, consider the same noise in the
presence of an 8-PSK signal. Is it easier for the receiver to make bit
errors?
The answer is yes, as more phases are used in PSK, the symbols are
closer together, which makes it easier for the receiver to make bit
errors (see the figure to the right). But, of course, the advantage of
more symbols is a narrower bandwidth, if the bitrate is held
constant. There is a way to use more symbols in modulation while
reducing the chances of making bit errors; by using symbols that
have different amplitudes AND phases.
3.3 Quadrature Amplitude Modulation (QAM)
In order to increase the distance between symbols in the
constellation, another option is to modulate both the amplitude and
the phase. This is called Quadrature Amplitude Modulation (QAM)
3.3.1 8-QAM An 8-QAM constellation is shown below (one of
many possible 8-QAM constellations). The eight symbols along with the 3-bit digital words corresponding to each are shown
to the right of the constellation. This system uses 2 possible amplitudes and 4 possible phases. In 8-QAM, the duration of a
symbol is three times the duration of a bit (since each symbol carries 3 bits). Note that there are both phase and amplitude
changes for each symbol. For the system with constellation shown below, the eight output symbols might be 2 cos(2πfc t
±45°), 2 cos(2πfc t ±135°), 4 cos(2πfc t ±45°), and 4 cos(2πfc t ±135°).
What is the bandwidth for 8-QAM? The same as for 8-PSK, since the bandwidth for all digital modulation types (except for
FSK) is given by
2R
BW = b
N
And it doesn’t stop there.
3.3.2 Higher level QAM signals QAM signals can be extended to have a larger number of signal symbols, which then gives
a much higher bit rate (because there are more bits per symbol). 64-QAM and 256-QAM are common in cable modems,
satellites, and high-speed fixed broadband wireless.
In 256-QAM, you find that for each symbol you are transmitting (there are 256 symbols), there are 8 bits of information.
Assuming the symbol rate remains constant, that means that for the same bandwidth, you are sending 8 times more
information when you use 256-QAM than when you use OOK, FSK, or BPSK. For 256-QAM, if the bitrate is 600 kbps, the
bandwidth is 2(600,000)/8 = 150 kHz.
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Now that’s powerful!

90˚
Using the signal constellation shown, answer the following questions.
(a) What type of modulation does this represent?
(b) How many symbols are represented (M)? 180˚ 0˚
(c) How many bits per symbol are used (N)?
270˚
(d) If the Baud is 10,000 symbols/second, what is the bit rate (Rb)?
(e) Would 16-QAM be more or less susceptible to noise than this type of modulation?

Label the modulation schemes.
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CH. 22 Problems
1. For an ASCII ‘Z,’ sketch both the On-Off Keying (OOK) binary waveform (voltage pulses) and the modulated signal,
where the amplitude of the carrier is modulated to either 10 V or 0 V and Tb = 100 ms. Hint: use the ASCII table from
Chapter 1 of the course notes to determine the bits that represent ‘Z’.
2. Given this FSK transmission where only two frequencies are used
and individual symbols are denoted by vertical lines:
(a) Draw the corresponding binary transmission (voltage pulses),
assuming that the higher frequency represents a 1-bit:
(b) Determine the bit rate (note: time axis in microsecs).
(c) How many bits per symbol could be conveyed if four different
frequencies were used to transmit data instead of two (that is, if
4 symbols were used vice 2 symbols)?
3. The following is a BPSK transmission. The dashed vertical lines separate the bits.
On this plot, a binary ‘1’ is represented by this signal:

(a) Determine the transmitted bits.
(b) Determine the bit rate.
(c) What is the bandwidth for this transmission?
4. QAM is a combination of which two types of modulation?
5. The “forward” signal transmitted to control a remotely-controlled (RC) car is captured on an oscilloscope
and displayed below.
Answer the following questions with regards to this signal:

(a) This modulation is binary, meaning that there are two possible symbols. What type of digital modulation is
being used?
(b) What is the bit rate? (Hint: Two time cursors are shown on the display as the two dashed vertical lines…these
cursors isolate a single bit. Also shown are some measurements about the time cursors below the plot and to the
right: here, ∆X = 496.3 µs is the difference in time between the two cursors).
(c) What bit sequence is represented by the O-scope display?
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6. 16-QAM can be used for higher data rate transmissions.

(a) How many bits are transmitted with each symbol?
(b) If 4 different phases and 4 different amplitudes are used in a 16-QAM modulation system, sketch a constellation
diagram that could be associated with the system (you do not need to label the bits for each symbol, just show
the symbols).
(c) If 8 different phases and 2 different amplitudes are used in a 16-QAM modulation system, sketch a constellation
diagram that could be associated with the system(you do not need to label the bits for each symbol, just show
the symbols).
(d) If the bit rate associated with either of these 16-QAM systems was 1.2 Mbps, what is the bandwidth of the
transmission?
7. For a given bandwidth system, what is the advantage and disadvantage of using a multi-symbol encoding scheme (that
is, using more than 2 symbols)?
8. A communication system transmits 100 kbps. For each of the following modulation types, determine the bandwidth of
the transmission.
(a) FSK, with frequency deviation 200 kHz.
(b) OOK.
(c) QPSK.
(d) 16-PSK.
(e) 16-QAM.
(f) 512-QAM.
9. Suppose the FCC has leased you the portion of the frequency spectrum from 1.2 MHz to 1.3 MHz for your free-space
communication system. What is the maximum bitrate you could obtain if you used the following modulation schemes:
(a) FSK, with fspace = 1.23 MHz and fmark = 1.27 MHz.
(b) ASK.
(c) BPSK.
(d) 8-PSK.
(e) 32-QAM.
(f) 256-QAM.
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Digital Modulation: OOK and FSK

Discussion: A baseband signal is not compatible with free-space communication. Therefore, we need to modulate the binary
0s and 1s. Digital modulation is different from analog modulation in that the analog carrier signal is modulated by voltage
pulses that represent 0s and 1s.
Objective: To provide hands on experience and further familiarize each Midshipman with some of the aspects of the simplest
form of Amplitude Shift Keying (ASK), known as On Off Keying (OOK), as well as Frequency Shift Keying (FSK).
Note: The binary information signal for this lab will be represented by a square wave of frequency fm as depicted in Figure 1
1 1 1
below. Hence, 𝑓𝑚 = = and; therefore, the bitrate Rb will be 𝑅𝑏 = = 2𝑓𝑚 .
𝑇𝑚 2𝑇𝑏 𝑇𝑏
Figure 1
Why does this exercise work the way it does?

The binary information signal for this exercise is represented by a square wave. A square
wave is not a typical binary signal (which is aperiodic), but a periodic signal. Fourier analysis
explains that the frequency spectrum of a periodic signal is represented in the frequency
domain with a set of discrete frequencies, versus a continuum of frequencies as shown in the
lecture, although the envelope of these discrete frequencies will follow the same shape as the
continuous ones. Moreover, these discrete signals occur at specific intervals. For a periodic
square wave with symmetric high/low time, such as that used in this exercise, these
correspond to any DC component in the square wave followed by odd multiples of its fundamental frequency. Thus, the
resultant representation of a sinusoidal carrier modulated by the square wave in this exercise, in the frequency domain has the
form as depicted below.
carrier frequency fc
fc + f (carrier plus the fundamental frequency)
null at fc + 2f
fc + 3f
fc + 5f
Because of this, measuring bandwidth from these plots can be challenging, as it does not match the theory presented in the
lecture. In an attempt to simplify measurements, a few display settings for the oscilloscope are recommended, like zoom,
which will make easier locating the nulls in the spectrum.
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Part 1: On-Off Keying (OOK)
In OOK, the amplitude of the digital signal controls the carrier signal, so that the carrier is turned on to represent a 1-bit and
turned off to represent a 0-bit.
Using your familiarity with the oscilloscope ( o-scope) and function generator from your previous labs, set up the Function
Generator with the following settings:
□ Press the Utility button and set your Output Setup to High Z.
□ Select the sinusoidal function by pressing the Sine button.
o Freq = 300 kHz (this will be the carrier frequency, fc)
o Ampl = 1 Vrms
□ Push Mod button with the following settings:
o TYPE = AM
o SOURCE = Int
o AM Depth = 100%
o AM Freq = 10 kHz (this will be the information signal representing a bit rate of Rb = 20 kbps)
o SHAPE = Square
□ Connect the function generator Output (red to red, black to black) to CH 1 of the o-scope
□ Connect the function generator Sync (red to red) to CH2.
□ Push Output button to send the signal to the o-scope.
□ Push AUTOSET on the o-scope.
□ Adjust the o-scope with CH 2 on top (square wave) and CH 1 (carrier) on the bottom using the vertical positions on
CH1 and CH2.
□ Push the Trig Menu button on the o-scope and use the following settings:
o TYPE = Edge
o SOURCE = CH 2
o SLOPE = Rising
o MODE = Auto
o Coupling = AC
Note: You may need to adjust the Trigger level arrow to stabilize your display.
□ Push CH 1 MENU to return.
□ Adjust the horizontal scale to 25 µsec per division.
□ Adjust CH 1 and CH 2 vertical scale (volts/div) so that you see a display similar to Figure 2.
Note: Your scope display should look similar to Figure 2, below, except your digital signal is a square wave, 101010…
Figure 2
Question 1: Looking at CH1 and using the time cursors, measure carrier period and then calculate the carrier frequency, fc.
Recall that the carrier is the rapidly changing sinusoid.
Question 2: Looking at CH2, measure the bit duration Tb, then calculate the bitrate, Rb.
□ Change the o-scope to display the frequency domain by choosing MATH MENU and using the following settings:
o OPERATION = FFT
o SOURCE = CH 1
o FFT zoom = X5 (then use the Horizontal Position knob to center the plot on the screen)
o 50 kHz per Division
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The o-scope display should look similar to Figure 3 below. (An Aside: Notice that the fundamental frequency
spikes“blend” with the carrier frequency.)
measure BW here
Figure 3
Question 3: Find the carrier frequency from the o-scope display (hint: use the frequency cursor).
□ Use the frequency cursors to measure the bandwidth (hint: the bandwidth is determined by the first null to the left
and right of the carrier).
Question 4: What is the bandwidth of the OOK signal when Rb = 20 kbps — that is, when fm = 10 kHz?
□ Change the AM Freq, not the carrier, to 20 kHz on the frequency generator (representing increasing your bit rate to
40 kbps).
□ Measure the bandwidth of the signal between the first pair of sidebands.
The o-scope display should look similar to Figure 4 below. (An Aside: Notice that the fundamental frequency spikes are still
visible to the side of the carrier, but the nulls are very distinct for measurement.)
measure BW here
Figure 4
Question 5: Now, What is the bandwidth of the OOK signal when Rb = 40 kbps ( fm = 20 kHz)?
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Question 6: Based on the Questions 4 and 5, as the bit rate increases describe what happens to the bandwidth of the signal?
Remember that the equation for the bandwidth of an OOK signal is
2𝑅𝑏
𝐵𝑊 =
𝑁
Your findings should be supported by this equation!
Part 2: Frequency Shift Keying (FSK)
Frequency shift keying (FSK) is another digital modulation technique in which a continuous sine wave changes frequency
when the digital bit stream changes between zero and one. The higher frequency represents a binary ‘1’ (also called mark)
and the lower frequency represents a binary ‘0’ (also called space). FSK is used primarily in low speed applications (<500
Kbps) and noisy environments where accuracy is preferred over speed.
Keep the carrier frequency the same (fc is still 300 kHz), but change the modulation mode to FSK using the following steps:
□ Use the following modulation settings on the function generator (Mod):
o TYPE = FM
o SOURCE = Int
o FREQ DEV = 200 kHz (this is frequency deviation, Δf).
o FM Freq = 10 kHz (this will be the information signal representing a bit rate of Rb = 20 kbps).
o SHAPE = Square
□ Push CH 1 MENU on the o-scope to return to the time domain.
□ Set horizontal scale to 25 µs per division.
Note: Your display should look similar to Figure 5 below, where a 1-bit is represented by a sinusoid with a frequency
higher than the carrier’s (called the mark frequency) and a 0-bit with a frequency lower than the carrier’s (called the
space frequency).
Figure 5
□ Adjust the picture on the o-scope to answer the next question by changing the horizontal range setting (sec/div) and
using the time cursors to measure the periods of the two sinusoids.
Question 7: What is the mark frequency, fmark? What is the space frequency, fspace?
□ To see the difference in the bandwidth for the FSK signal, shift to the frequency domain. Push the MATH MENU
button and use the following settings:
o OPERATION = FFT
o SOURCE = CH 1
o FFT zoom = X5 (then use the Horizontal Position knob to center the plot on the screen)
o 50 KHz per Division
□ Measure the bandwidth as shown in Figure 6. This is based off the first null to the left of fspace and the first null to the
right of fmark.
The o-scope display should look similar to Figure 6 below. (An Aside: Notice that the fundamental frequency spikes“blend”
with the spikes at both fspace and fmark.)
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measure BW here
Figure 6
Question 8: What is the measured bandwidth (hint: your answer should be much larger than your answer for the OOK
bandwidth) ? Remember that the equation for the bandwidth of an FSK signal is:
𝐵𝑊 = (𝑓𝑚𝑎𝑟𝑘 + 𝑅𝑏 ) − (𝑓𝑠𝑝𝑎𝑐𝑒 − 𝑅𝑏 ) = 𝑓𝑚𝑎𝑟𝑘 − 𝑓𝑠𝑝𝑎𝑐𝑒 + 2𝑅𝑏

or
𝐵𝑊 = 2(Δ𝑓 + 𝑅𝑏 )
Your answer should be supported by this equation!
□ Change the FM FREQ, not the carrier, to 20 kHz on the signal generator (representing Rb = 40 kbps).
□ Push the MATH MENU button on the oscilloscope and change the following setting:
o 100 KHz per Division (then use the Horizontal Position knob to center the plot on the screen)
□ Measure the bandwidth of the signal as shown in Figure 7.
The o-scope display should look similar to Figure 7 below. (An Aside: Notice that the fundamental frequency spikes“blend”
with the spikes at both fspace and fmark.)
measure BW here
Figure 7
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Question 9: What is the new bandwidth?
Question 10: Based on the above change, as the bit rate (Rb) increases, describe what happens to the bandwidth of the signal.
What can you say about the comparisons of the bandwidths for FSK as compared to OOK?
□ Turn off your equipment and clean up your lab bench.
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Name:
Question 1:
Question 2:
Question 3:
Question 4:
Question 5:
Question 6:
Question 7:
Question 8:
Question 9:
Question 10:
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Chapter 23: Power Gain and SNR
Objectives:
(a) Define gain and attenuation and describe their application to communications.
(b) Calculate power gains for single and multiple stage systems; determine power at each stage.
(c) Express power gain in dB, and power levels in dBW and dBm. Compute power gain and power from dB, dBW and dBm.
(d) Calculate signal to noise ratio (SNR) and discuss the impact of noise in a communication system.
Communication systems transmit electrical (EM) signals to convey information. The strength of a signal is based on its
electrical power, and the transmit power is an important consideration in how far a signal can be transmitted through the
atmosphere. In addition, the received power is important factor in how accurately an information signal can be recovered; if
the received power is not high enough to overcome the noise present, then information will be lost. Cyber security attacks
against wireless communication systems can take advantage of the frequencies and modulation types of the transmission
(Chapter 22), but also the power that is received by a receiver. These attacks take the form of jamming, and possibly taking
control of devices that are controlled via a wireless communication link if the received control signals from the actual
transmitter are weaker than the received power from a hacker’s signal. In 2011, Iran captured a US unmanned aerial vehicle
(UAV) while inflight, claiming that their cyber warfare unit had commandeered and safely landed the UAV. This chapter
introduces the power aspects of wireless communication.
1. Gain/Attenuation
Electrical power is measured in Watts (W), and your typical flat screen TV uses maybe 250 W while it is on, and your laptop
may use 60 W while it is running heavy-duty programs. In wireless communications, it may take an incredibly large transmit
power to cover the distance to the receiver, and even then, the power arriving at the receiver may be incredibly small. For
example, a commercial FM station may transmit 15.5 kW of signal power to reach your car’s radio, and by the time it gets to
your car’s antenna, the received power may be on the order of 1 pW (10 -12 Watts). This means that the transmitted power has
dropped by a factor of approximately 1016. Consider a satellite ground station (on Earth), transmitting to another ground
station on the other side of the Earth via a geostationary satellite, 22,300 miles away. This is an immense distance to transmit
over to reach the satellite! How about NASA’s New Horizon space probe mission currently on its way to the planet Pluto…it
will be transmitting information back to Earth nearly 4 billion miles away!
So how are you going to get your signal to travel further? Turn up the power. But modulators that produce PSK or QAM
typically do not produce signals of substantial power; instead we use devices called amplifiers to increase the power of the
modulated signals (that is, to amplify them) so that they are strong enough to cover the required distances.
The term power gain refers to the factor that the power is increased in a signal as it goes through an amplifier. The power
gain (AP) is the ratio of the output signal power to the input signal power. In a block diagram of a communication system, an
amplifier is typically drawn as a triangle (although rectangular blocks are also used), as in the following figure. To calculate
power gain (AP) where Pin is the power input and Pout is the power output, we use the equation:
P
AP  out (unitless)
Pin
An amplifier can take a modulated signal and increase its power large enough to transmit many miles, much like the above
example of a FM radio broadcasting tower transmitting at 15.5 kW.
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There are, however, some components of communication systems can also reduce the power of a signal. Reduction of the
power of a signal (signal loss) is termed attenuation. Attenuation is still computed using the equation for power gain, but a
component that attenuates has a power gain that is less than 1.0.
Putting together what we’ve learned, we have our modulated signal feeding into an amplifier that increases the power of the
signal. The signal is broadcast out of the transmitter via an antenna, where the signal is attenuated as it travels through the air
to the receiver’s antenna. Finally the significantly reduced signal is picked up by the receiver, and the receiver recovers the
information. This is depicted in the following diagram for an ASK system.

The input power of an amplifier is 6 W. The power gain is AP = 80. What is the output power?
Solution:

The input power is 15.5 kW. The power output is 10-15 W. Is this system associated with amplification or attenuation? What
is the gain (or attenuation) of this system?
Solution:
You may have noticed that there can be a large disparity in the power values between transmitter and receiver, and dealing
with incredibly large and incredibly small values in the same system is challenging. For this reason, in many cases we deal
with decibel values instead of the numeric values.
2. Decibels
As engineers, we just want our lives to be as easy as possible. So rather than work with these terribly tedious numbers, we
often convert the numbers into decibels (dB). The decibel is a logarithmic measure that provides more convenient gain and
attenuation values by changing them to a logarithmic scale. The benefit of a log scale is that it can map a very large range of
decimal values into a small range of decibel values. Consequently, small changes in decibel quantities may mean very large
changes in power (we’ll revisit this in the accompanying security exercise). To convert a decimal value X into decibel value
XdB is given by:
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X dB  10log10  X 
If X is a value greater than 1.0, then XdB will be a positive value, and if X is a value less than 1.0, XdB will be a negative value.
The decibel value of zero is negative infinity and the decibel is undefined for negative values.
For power gain (or attenuation) then:

P 
AP , dB  10 log10  AP   10 log10  out 
 Pin 
So then for the above practice problem that gave us a headache, we see:
P   0.000000000000001W 
AP , dB  10 log10  out   10 log10    191.9 dB
 in 
P  15,500W 

Convert these two power gains to decibels (dB).
AP =1000 AP,dB =
AP =0.0001 AP,dB =
A couple of very common values of power gain are 2 and ½. A power amplification by a factor of two (AP = 2) will result in
a power gain of +3 dB.
AP, dB  10log10  AP   10log10  2   3dB
An attenuation by a factor of one-half will result in a power gain of -3 dB.
AP, dB  10log10  AP   10log10  0.5  3dB
How do you find the decimal value corresponding to a decibel value? Just rearrange the dB equation from earlier and you get:
AP , dB  10 log10  AP 
AP ,dB
AP  10 10

Convert the following power gains from decibels to decimal gains.
AP,dB = 25 dB: AP =
AP,dB = -6 dB: AP =
Power gain is a ratio of two powers, Pin and Pout, each with a unit of power, usually W or mW. When taking this ratio, the
units of power cancel, and you’re taking the log of a unitless ratio. Logarithms only work with numbers, not units. In
communications, we are sometimes asked to compute the decibel value of a power level (in W or mW). In this case, you will
take the log of that power level with respect to a fixed reference power level, either 1 W or 1 mW so that the units cancel and
you’re just taking the log of a number.
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2.1 dBm The number of decibels of power relative to 1 mW. The reference power level is 1 mW and the dBm value is
expressed mathematically as:
 P   PW 
PdBm  10log10  mW  or PdBm  10log10  
 1mW   0.001W 
If the power value to compute is already in mW, the first equation can be used, and if the power value is in W, then the
second equation can be used. In this case, since 1 mW = 0.001 W, the units will cancel.
2.2 dBW The number of decibels of power relative to 1 W. The reference power level is 1 W and the dBW value is expressed
mathematically as:
P 
PdBW  10log10  W 
 1W 
In all cases, the units of power must cancel so that the resulting ratio is unitless.
Also, if given a dBm or dBW value, the power in mW or W can be found from:
PdBm PdBW
PmW  10 10 or PW  10 10

Express Pin = 2 W in decibels as both dBm and dBW.
Pin,dBm =
Pin,dBW =

Express 25 dBm in terms of mW and W.
P(in mW) =
P(in W) =
Besides compressing a large range of values into a smaller range of decibel values, another benefit from using decibels is the
mathematics involved in combining decibel terms; decibel values are added or subtracted instead of multiplying or dividing.
This is typically seen in communication systems that cascade amplifiers as in the following figure. Here, the output power
after each amplifier is computed as the product of the power into that amplifier and its power gain.
So, if we leave the gains in ratio form, then the total gain of the system will be the product of all the gains multiplied
together, and we could rewrite this cascade of three amplifiers as a single amplifier with power gain AT.
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In terms of decibels, the overall decibel gain of a cascade of amplifiers can be found as follows:
AT , dB  10log10  AT   10log10  AP1  AP 2  AP3 
Using the property of the log function that the log of a product is the sum of the logs, we have:
AT , dB  10log10  AP1  AP 2  AP 3 
 10log10  AP1   10log10  AP 2   10log10  AP 3 
 AP1, dB  AP 2, dB  AP 3, dB
Also, we could use the property of the log function that the log of a quotient is the difference of the logs to write the
following equation:
P  P  P 
AP , dB  10 log10  out   10 log10  out ,W   10 log10  in ,W 
 Pin   1W   1W 
in dBW in dBW
or
P  P 
 10 log10  out , mW   10 log10  in , mW 
 1mW   1mW 
in dBm in dBm
In this equation, the input and output powers must be in the same decibel units, either dBW or dBm. Note that the difference
between two dBm or dBW values will result in a dB value (see More on Decibels below). Applying the log of products
property to a cascaded system of amplifiers,
P 
Pout , dBm  10 log10  in , mW  AP1  AP 2  AP 3 
 1mW 
P 
 10 log10  in , mW   10 log10  AP1   10 log10  AP 2   10 log10  AP 3 
 1mW 
 Pin , dBm  AP1, dB  AP 2, dB  AP 3, dB
Here, it is okay that dB and dBm are mixed on the right side of the equation, because all of the decibel values represent
unitless numbers; it’s just that the input and output power values’ decibel values must be computed relative to 1 mW. If the
input and output powers are in W instead of mW,
Pout , dBW  Pin, dBW  AP1, dB  AP 2, dB  AP3, dB
Adding and subtracting decibels can be a much simpler operation than multiplying and dividing very large or very small
decimal numbers. A common mistake midshipmen make when dealing with decibel values is that you should NEVER,
EVER multiply or divide decibel values. Decibels are always added or subtracted from other decibels.
2.3 More on Decibels
Consider the following:

 P   Px / Ay 
10 log  x   10 log  Ay   10 log   
so...
dBm  dB  dBm
 .001   .001 
dB
dBm dBm
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and
 P   Py   Px 
10 log  x   10 log    10 log   
so...
dBm  dBm  dB
 .001   .001   Py 
dBm dBm dB
It follows then that:

dBm  dB  dBm
dBm  dBm  dB
and
dBW  dB  dBW
dBW  dBW  dB

The diagram below represents the first three stages of a typical AM or FM receiver. Find the following quantities.
(a) AT and AT,dB
(b) AP1,dB, AP2,dB, and AP3,dB.
(c) P1, P2, and Pout.
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(d) Pin,dBm, P1,dBm, P2,dBm, and Pout,dBm.
3. Noise and the Signal-to-Noise Ratio (SNR)
Recall from Chapter 19 that noise is one of the principal limiting factors in the performance of communication systems, and
that noise is added to our signal from external sources in the communication channel and also from internal (electronic)
sources within our own system’s hardware.
As we saw in Chapter 22, if significant enough, it can mask the original signal such that the signal becomes unrecoverable, or
in the case of digital modulation, that bit errors can occur. This noise effect is not much different than if an enemy were to
flood the air waves with an erroneous signal at the same frequency on which you were transmitting. If that erroneous signal
was stronger at the receiver than your signal, your signal would become unrecoverable.
How do we know the effect of noise on the signal, or the quality of the received signal in the face of noise? We use the
signal-to-noise ratio (S/N, also referred to as SNR), which is the ratio of the power of a signal to the power of the noise
corrupting that signal. A strong signal in weak noise results in a high SNR. A weak signal in strong noise results in a low
SNR. Below are four samples of a sine wave with various amounts of noise added:
The signal-to-noise ratio indicates the relative strengths of the signal and the noise in a communication system. The stronger
the signal and the weaker the noise, the higher the SNR. Mathematically, SNR is defined as:
Ps P 
SNR  and SNRdB  10 log10  s 
Pn  Pn 

The signal power at the input to a receiver is 6.2 nW and the noise power at the input to that receiver is 1.8 nW. Find SNR
and SNRdB.
Solution:
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CH. 23 Problems
1. Convert power gains of 100, 1000 and 2000 to decibel values.
2. Convert power gains of 0.01, 0.001, and 0.0005 to decibel values.
3. Convert decibel power gains of 13 dB, 33 dB, and 103 dB to power gains.
4. Three amplifiers with gains of 12.5, 4, and 20 are cascaded as shown in the following diagram (from left to right). The
input power is 120 mW. What is the overall gain and the output powers of each stage?
5. A power amplifier has an output power of 200 W and an input power of 8W. What is the power gain in decibels?
6. A power amplifier has a gain of 55 dB. The input power is 600 mW. What is the output power in W?
7. An amplifier has an output power of 5W. What is this amount of power in dBm?
8. A communication system has five stages, with gains and attenuations of 12 dB, -45 dB, 68 dB, -31 dB and 9 dB.
(a) What is the overall gain in dB (AT,dB)?
(b) The overall power gain (AT)?
(c) If the input power is 1 dBm, what is the output power in dBm?
9. The signal input power to a receiver is 6 W. The noise power is 25 mW. What is the SNR? What is SNRdB?
10. A receiver’s sensitivity is the minimum received signal power for the receiver to successfully recover the transmitted
signal. If a receiver’s sensitivity is -45 dBm, and the received power is 10 μW, will the receiver be able to recover the
transmitted signal?
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Introduction to Wireless Signals
Discussion: In the wireless section of the course so far, you have learned that we transmit information using EM waves in
free space. If we encode some meaningful data onto these waves, we can communicate without being physically tethered to
the medium. But how far away can we be? How close to the transmitter must we be in order to receive the message and
successfully recover the information?
Just like our human voice only travels so far when we shout, radio waves only travel so far from the transmitting station. At
some point, you will just be out of range.
Fortunately, we can measure the strength of the transmitter (how loud it can “shout”) and the sensitivity of a receiver (how
quiet a sound it can still “hear”) When we put these measurements on a logarithmic decibel scale and compare their utility,
we can figure out things like:
What is the optimal location for a transmitter?
Where is the best spot to get reception?
Which devices receive a better signal?
These are things you probably do with your cell phone all the time. Today we will do an experiment and see if we can answer
these questions scientifically.
Objective: To provide hands on experience and further familiarize each Midshipman with power measurements in a wireless
communication system, and the effects of distance from the transmitter to the receiver.
Part 1: Measuring Signal Strength from your Local WiFi

Set-up
Equipment required:
Your issued Laptop
Xirrus software:
□ On your laptop, check the wireless connections and you should see a list of devices. One of the devices is the
wireless Access Point (AP) in your classroom named cyber2_xx. The xx is your room number. If you do not see the
specific AP for your room, tell your instructor.
□ Once you see your cyber2 AP, start Xirrus Wi-Fi Inspector by double clicking the icon on your desktop. Let’s
explore the Xirrus Graphical User Interface (GUI) shown on the next figure. You should identify each of the
following parts on the display (identified with letters a-e) and then perform any specific instructions on your laptop.
a) Start by clicking on settings and turning “Locate Sound” to Off. Click OK.
b) In the upper left is the “Radar Display.” This shows the relative signal strength of an AP. The stronger the AP,
the closer it is to the middle of the display. It doesn’t correlate with specific direction of the AP relative to you,
but it will converge to the center as strength increases.
c) In the center top is “Connections” which lists the details of the AP you are connected to.
d) Below that is “Networks” which lists of all the AP’s you can observe with their respective data.
e) “Signal History” is a time versus signal strength (in dBm) graph of the AP you are trying to locate. Highlight
the cyber2_xx node, then right click and choose locate cyber2_xx and you should see it appear on the signal
history plot.
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Question1: For your cyber2_xx AP, write the following details down on the lower left corner of the map on the solution
page.
SSID (Service Set Identifier) - the wireless network name
BSSID (Basic Service Set Identifier) - the MAC address of the wireless interface unit
Channel - allows the carrier frequency to be separated into bands to keep from overlapping
Frequency - carrier frequency the AP is using for communications
□ Now that we have Xirrus running, we can take some measurements of the signal strength. Look at the Networks
display list in the middle (d above), find your AP, highlight it and note the dBm.
Question 2: Record the signal strength noted for your AP on the table on the last page of the lab under classroom.
Question 3: Assume you record the signal strength of some other fictional access point when you are standing next to it as -
30 dBm. Next, you walk some distance away from it and take another signal strength measurement and record it as -33 dBm.
By what factor has the signal strength dropped from measurement one to measurement two? (Hint: convert each
measurement to mW then divide measurement two by measurement one). Recall that:
PdBm
PmW  10 10
Keep this realization in mind when answering the following questions:

A SMALL CHANGE IN DECIBELS CAN MEAN A LARGE CHANGE IN POWER!
Question 4: Staying highlighted on your assigned AP, move from point to point on the map and record the signal strength
(dBm), allowing a period of time to let the value settle. Note if the dBm falls much below -90 it may drop from your list.
You can locate it again by returning closer to the classroom. Simply record -90 dBm if your AP is lost at any point on the
map.
Question 5: Convert your dBm measurements to mW and finish filling in the table.
Question 6: Observations:
a) At what locations did you receive the strongest signal?
b) The weakest signal?
c) Would you expect to stay connected to this AP in Maury Hall? Why?
d) Have an instructor check your results
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Part 2: The Hunt for an Unknown AP
□ Understanding how Xirrus reads signal strength, we will now try to locate an unknown AP using the Xirrus
program. This AP is not located in your classroom, but you should be able to pick up the signal in your hallway.
a) Try to find the AP with SSID Bad_Egg_xx ( again with xx indicating your class room).
b) Turn On the “Locate Sound” in Settings and change the polling time to 1 second. Right click on Bad_Egg_xx
in the Networks list and select Locate. This will create a ping. The closer the pings are together, the stronger the
AP’s signal.
c) Begin walking through the lab deck following your ping, dBm and Radar in a direction that makes the signal
stronger.
Question 7: Where is the AP located? What is the message that is written on the AP?
Question 8: Emissions controls in the military refers to controlling your Radio Frequency Emissions. Keeping “The Hunt”
from above in mind, why might it be important to maintain radio silence at certain times in Navy and Marine Corps?
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Name:
Question 1/2/4/5:
Question 3:
Question 6:
a)
b)
c)
d) _________________________ Instructor/Lab Tech Signature
Question 7:
Question 8:
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Chapter 24: Antennas and Propagation
Objectives:
(a) Describe the role of an antenna in a wireless communication system.
(b) Explain the difference between power gain and antenna gain, compute an antenna's gain relative to an isotropic point
source (dBi), and describe the advantages and disadvantages of a directional antenna.
(c) Interpret an antenna's radiation pattern to determine the sidelobe level and front-to-back ratio in dB, the beamwidth, and
directions from which interfering or eavesdropping antennas may lay.
(d) State/describe the important parameters of dipole, monopole and Yagi antennas: the antenna gain, beam pattern/width,
bandwidth and size.
(e) Define reflection, refraction, diffraction and scattering.
(f) Describe the characteristics of ground waves, sky waves, and space waves, and calculate the radio horizon distance for
space waves based on antenna height.
(g) Compute received power level for a wireless communication system using the Friis Free Space equation.
Recall from Chapter 12, The Physical Layer, the transmission medium or communication channel of a signal can be guided
or unguided media. If the communication channel is free space, then the signal is transmitted and received using an antenna.
Antennas serve as the bridge from the transmitter and receiver to the communication channel. Unlike guided media (wired)
transmissions, free-space is an open medium, and anyone with an antenna can collect transmitted signals or transmit their
own signals into free space. This makes free-space systems particularly vulnerable to cyber attacks involving both
eavesdropping and jamming.
Once the signal has been transmitted, it begins to attenuate. By the time the signal arrives at the receiver the received power
can be extremely low. If the received power is too low, the receiver will not be able to recover the information. In this
chapter, you’ll review some commonly used antennas, the various ways that signals can propagate through free-space, and
also how to compute the received power. Cyber attacks against wireless communication systems can take advantage of a low
received signal power to jam the transmission, or to take control of the communication link.
1. Antenna Characteristics
An antenna is a device that provides a transition between guided electromagnetic waves in electrical circuits to
electromagnetic waves in free space, and can be a length of wire, a metal rod, or a piece of metal tubing. Recall that
wavelength () and frequency (f ) of an electromagnetic wave in free space are related by the speed of light (c), where c = 3.0
x 108 m/s:
c.

f
The length of an antenna is usually expressed in terms of the wavelength () of the frequencies being transmitted.
 Low frequencies imply long wavelengths, hence low frequency antennas are very large (for example, the towers
across the Severn River are used for the VLF Submarine Broadcast, 30 kHz and are hundreds of feet high).
 High frequencies imply short wavelengths, hence high frequency antennas are usually small (for example, the Dish
Network transmission frequency from the satellite to your satellite dish is 12 GHz, and the antenna is approximately
1 cm long)
Antennas are dual function, meaning that an antenna designed to transmit a certain frequency can also receive that frequency.
When selecting an appropriate antenna for a communication system, there are four key criteria that must be evaluated:
antenna gain, antenna beam pattern/beamwidth, antenna bandwidth and physical size.
1.1 Gain Because antennas can focus power in certain directions, we say that an antenna can have a gain. Antenna gain
determines how concentrated the transmitted power is in a particular direction (usually the direction of maximal radiation).
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Isotropic antennas are theoretical antennas that have no directionality, and radiate their power equally in all directions.
Consider the figure that follows. On the left is an isotropic antenna, located at the center of the sphere. The power it transmits
is spread equally in all directions, in a spherical shape. If it transmits 1 W, that 1 W will be spread over the surface of the
sphere. The power density is uniform across the sphere so as you move farther from the antenna, the received power per unit
area drops dramatically. On the right is a directional antenna. If this antenna also transmits 1 W, that power is spread over a
much smaller surface area, as indicated, so that in the direction the antenna is pointing, the reduction in power is much less as
you move farther from the antenna. The antenna gain is a measure of power transmitted by a directional antenna in the
direction it is pointing relative to that transmitted by an isotropic (uniform) source.
The mathematical definition of antenna gain is
Radiated power density at distance x from directional antenna

G
Radiated power density at distance x from isotropic antenna
If we convert this to decibels, because we are comparing relative to an isotropic antenna, it is common to use dBi instead of
dB. To compute antenna gain in decibels, we have
GdBi  10log10  G 
(dBi).
Similarly, to convert from dBi to ratio we use

GdBi
G  10 10 (unitless).
Light can be used as an analogy to antenna gain. Imagine a single light bulb in a lamp. The light bulb sends light equally in
all directions similar to how an isotropic antenna sends radio waves equally in all directions. When we put the lightbulb in a
flashlight, the design of the flashlight focuses light in a single direction. This the lamp and flashlight were next to a wall, the
lamp would illuminate more of the room, while the area the flashlight illumined would be much brighter.
This is similar to how a directional antenna focuses radio waves in a particular direction. Higher gain means a stronger signal,
making communication over longer distances possible. Antenna gain can be thought of as how much brighter the wall is with
the flashlight versus how bright it was with only the light bulb.
Or conversely, using a directional antenna we could communicate the same distance with less transmit power than if we had
used an isotropic antenna. This is appealing to us because in many military applications “getting” more transmit power is not
an option. Marines holding handheld radio are limited to the power available in the batteries, a maximum of approximately
2W.
A related characteristic of transmitting stations in a wireless communication system is the Effective Isotropic Radiated
Power (EIRP), which is the product of the transmit power and the antenna gain:
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EIRP = Pt Gt (Watts)
Here, the subscript t indicates that this is transmitter power and transmit antenna gain. In decibels,
EIRP is the amount of power that an isotropic antenna would have to transmit to achieve the same received power as a
directional antenna at the same distance.
To better explain this, let us return briefly to our flashlight analogy. I no longer have the flashlight. If I want to illuminate the
wall with the same intensity as the flashlight did, the EIRP is how much power I would now need to send into the lamp to do
so.
An antenna with directional gain has some advantages over an isotropic antenna. These include:
 Because energy is only sent in the desired direction, the possibility of interference with other transmitters at
or near the same frequency is reduced.
 More focused power results in increased gain, which means that less transmit power is required.
 Controlling the direction of the beam can help prevent eavesdropping since you must be in the beam in order
to receive the signal.
 A narrow beam can reduce the likelihood of detection in a covert setting for the same reason as was just
discussed.
However, directional antennas don’t work well in mobile situations (imagine keeping your cell phone pointed at a cell tower
as you’re driving past it) and they can be physically large if gain is big.

A radio station has an EIRP of 25 kW and a transmit power of 1.73 kW. What is the gain of the antenna?
Solution:
1.2 Beam Pattern Beam pattern is a diagram that shows specifically what direction(s) the antenna favors. You can think of a
radiation pattern being created by having an antenna radiate a constant power (say 1 W, although any power will do), and
then with a power meter, walk in a complete circle 1 km (or any other constant distance) from the antenna and record the
power received at each point along the circle. The result will look something like the following figure:
An example radiation pattern is shown in this figure in red. Note that antenna beam patterns can be in both azimuth and
elevation. Here the azimuth is shown. In this pattern, relative bearings are shown with 0° being the direction the antenna is
pointing. In this figure, each circle represents a change in received power of 3 dB, and the maximum power is along the 0°
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bearing. There are six lobes of transmitted power showing. The mainlobe is oriented towards 0°, the direction the antenna is
pointing. There are four sidelobes, oriented towards ±60° and ±120°, and a backlobe, oriented towards 180° (directly away
from where the antenna is pointing).
In many cases, the mainlobe’s maximum value will be defined as 0 dB, and the power levels at all other points on the pattern
are the number of dB less than the max; this is a measure of power relative to the max power.
This form of a radiation pattern is only one of many that could be used; sometimes the rings are not in dB, sometimes they
represent power density (W/m2), or power (dBW or dBm), etc. But the general features of the beam pattern will be similar. In
actuality, antennas radiate in 3-dimensions but the radiation patterns we will focus on are 2-dimensional, like the one shown
above.
From the radiation pattern, a few new terms that describe the properties of the antenna come about. The sidelobe level (SLL)
is a measure of the strength of the sidelobes compared to the mainlobe in decibels. The sidelobe level is measured from the
peak of the main lobe to the peak of the largest sidelobe. Mathematically,
SLLdB  Gmainlobe ( dB )  Gsidelobe ( dB ) .
For the antenna with radiation pattern on the previous page, the largest sidelobes are at ±60°, so SLLdB = 0 dB – (– 16 dB) =
16 dB. Similarly, the front-to-back ratio (FBR) is a measure of the strength of the mainlobe to the strength of the back lobe in
decibels. Mathematically,
FBRdB  Gmainlobe ( dB )  Gbacklobe (dB )

.
For the antenna with radiation pattern on the previous page, the backlobe is at –17 dB, so the front-to-back ratio is FBRdB = 0
dB – (– 17 dB) = 17 dB.
Finally, note that the radiation pattern has some bearings that are not a part of any lobe, for example ±35°. These are called
nulls of the pattern, and at these bearings, no power is transmitted from this antenna (or perhaps a miniscule amount), nor can
this antenna cannot receive signals from these bearings.
1.3 Beamwidth Beamwidth is based on the relative bearings where transmitted (or received) power is reduced by a factor of
½ (or -3 dB, since 10 log10 (½) = -3) from the direction of max power. We call these points on the diagram the -3 dB (or half-
power) points. The beamwidth is the angle that subtends these points. The following figure shows the beamwidth
computation for the above beam pattern; the beamwidth is 20°. A narrow beamwidth (small angle) means the antenna is very
directional.
1.4 Bandwidth Bandwidth determines the range of frequencies that the antenna is best suited for. Broadband signals (that is,
signals with a very wide bandwidth) transmit more data at a faster data rate, but broadband antennas are harder to
design/build. An antenna is normally designed for a certain transmit frequency, but can be used successfully for a range of
frequencies around that. For example, if you are a Marine and have a handheld VHF radio, it will come with an antenna
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which has been optimized for the center frequency of the VHF spectrum. The radio itself is capable of operating anywhere
within the VHF spectrum and the antenna has a bandwidth that will work plus or minus the center frequency and perform
within acceptable tolerance.
1.5 Physical Size Physically larger antennas have a higher gain and narrower beamwidth, but are much harder to conceal.
Also, the system using the antenna may introduce its own constraints (e.g., no one wants to mount a 6 meter dish on the roof
of their car). Antennas radiate most effectively when their length is directly related to the wavelength of the transmitted
signal. Most antennas have a length that is some fraction of a wavelength. One-half and one-quarter wavelengths are most
common.

Consider the antenna with this radiation pattern:
1. What is the beamwidth of this directional antenna?
2. What is the sidelobe level?
3. What is the front-to-back ratio?
4. Will a station transmitting bearing 90° interfere with me? Will

I interfere with it?
5. Will a station bearing 240° be able to eavesdrop on my communications?
6. Suppose the receiver I am communicating with (at 0°) requires that the signal received be at least 1 pW. Will I have to
transmit more power or less power using this antenna than if I were using an isotropic antenna? Why?
2. Dipole Antenna
One of the most widely used antenna types is the half-wave dipole. A dipole antenna is two pieces of wire, rod, or tubing that
are one-quarter wavelength long at the operating frequency connected to a voltage source (these are the poles). The antenna is
formed by placing these poles at a 90° angle from the transmission lines that are carrying the signal to be transmitted. This is
depicted in the figure below. The most efficient radiation of EM waves comes when the total length of the antenna is λ/2
long, which is why the antenna is called the half-wave (λ/2) dipole antenna.
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The radiation pattern for a horizontally oriented dipole antenna is shown below (on the left). The dipole is the heavy black
line segment. The scale is not in dB, but this is the general shape. It is bidirectional, in that there is a backlobe that is as large
as the mainlobe, both emanating perpendicular to the orientation of the dipole. If the dipole is oriented vertically, the
radiation pattern is omnidirectional in the azimuth, as shown on the right.
The 3-dimensional radiation patterns for the horizontally and vertically mounted dipole are shown on the next figure. Note
that the 2-dimensional patterns above are cutaways of the 3-dimensional patterns.
Major Parameters for the Dipole Antenna:
1. Gain – A dipole has a gain of G = 1.64, or GdBi = 2.15 dBi.

2. Beam Pattern/Beamwidth – A dipole mounted vertically has the 2-dimensional beam pattern in the azimuth plane
shown in the figure on the previous page, and a -3 dB beamwidth of 78º. The vertically mounted antenna has an
omnidirectional pattern in the azimuth (energy is spread equally in all directions).
3. Bandwidth – A dipole typically has a bandwidth that is ~25% of the center frequency of transmission. A dipole has a
center frequency equal to λ.
4. Physical Size – A dipole has a physical size equal to λ/2, where λ is the wavelength of transmission.

A transmitter feeds a half-wave dipole antenna with 100 watts of power. Calculate the effective isotropic radiated power
(EIRP).
Solution:

How long would a dipole antenna be for commercial radio station AM 1100?
Solution:
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3. Monopole Antenna
The quarter-wave (λ/4) monopole antenna, also called a Marconi antenna, is

widely used. It’s characteristics are similar in to a vertically mounted dipole
antenna, except that the monopole is connected to a ground plane (such as
the earth), and uses it as a as a type of electrical “mirror” to reflect
transmitted or received energy upwards to contribute to the upper part of the
radiation pattern. Effectively, the ground plane acts as the “missing” half of
a dipole antenna. The 3-dimensional radiation pattern for the vertically
mounted λ/4 monopole is shown in the following figure (on the left), and a
slice of the pattern (2-dimensional pattern in the vertical direction) is shown
on the right.
Major Parameters for the Monopole Antenna

1. Gain – A monopole has a gain of G = 1.45, or GdBi = 1.6 dBi.5
2. Beam Pattern/Beamwidth – A monopole has an omnidirectional pattern in the azimuth (energy is spread equally in all
directions), and a -3 dB beamwidth of 45º in the vertical plane.
3. Bandwidth – A monopole typically has a bandwidth that is ~10% of the center frequency. A monopole has a center
frequency equal to λ.
4. Physical Size – A monopole has a physical size equal to λ/4.

The ballistic submarine USS Alaska has gone on alert. They must stream a floating wire monopole antenna to get their alert
signal. If the alert signal is transmitted at 30 kHz, how far should they stream their antenna? (note: the antenna being
streamed is a straight wire)
Solution:
4. Yagi (Yagi-Uda) Antenna
The Yagi-Uda was developed in Japan in 1926 by Professor Hidetsugu Yagi and his student Shintaro Uda. Their basic
concept and structure is still used across a wide variety of modern antenna designs, and the Yagi-Uda is still the “go-to”
antenna for high gain at VHF and UHF frequencies. There was a time when every home in America was equipped with a
Yagi antenna, on their roof to allow reception of broadcast television.
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The gain of the quarter-wavelength monopole is very highly dependent on the construction of the associated conducting
ground plane. A monopole antenna would have a gain +3dB higher than a dipole (or about 5.15 dBi) due to its radiation
pattern if a good ground plane was used. A good ground plane would be of infinite size in theory, but about a wavelength in
radius for practical purposes. Since wavelength-radius planes are still impractical, a more "typical" monopole antenna gain
might be less than that of a dipole, although many manufacturers do not list them due to the various factors affecting it.
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A Yagi antenna is composed of a driven-element (a dipole antenna) and

multiple parasitic elements. A driven-element is one that is connected
electrically to the transmitter. Parasitic elements are not connected electrically,
but are placed in the vicinity of the driven element to either side. These
parasitic elements (known as reflectors and directors) will resonate with the
electric field produced by the dipole. Reflectors are longer than the dipole
antenna, are all placed on one side of the dipole, and reflect the transmitted EM
waves back towards the dipole antenna. Directors are shorter in length than the
dipole, and “direct” EM waves from the dipole and reflectors to form the
mainlobe. Judicious spacing of the parasitic elements will allow us to produce
constructive interference and “push” energy in the forward direction, giving
the Yagi-Uda good gain. The effect of directors and reflectors is:
 More parasitic elements means higher gain and narrower beamwidth.
 Adding more directors is more effective than adding more reflectors.
 The greater the number of directors, the higher the gain and the
narrower the beamwidth.
 However, we get diminishing returns as more elements are added.

Most Yagi antennas have 1 reflector and 1-20 directors.
Here is a sketch of a Yagi-Uda with one director and one reflector, the simplest design This is called a three-element Yagi,
and has a gain of approximately 5 dBi.

What is the length of the driven element of a Yagi antenna at 290 MHz?
Solution:
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A manufacturer of Yagi antennas is the L-Com Global Connectivity corporation (www.l-com.com). Here’s an example
pattern of one of their 900 MHz Yagi antennas (model HG906YE-RSP). The driven element (dipole) has a cable connected
to it. This Yagi has 1 reflector and 2 directors, and a gain of 6 dBi. The horizontal beamwidth is 100°, vertical beamwidth
60°.
Compare that to their model HG914YE-RSP antenna, which has 1 reflector, 11 directors, and 14 dBi of gain. The horizontal
beamwidth is 31°, vertical beamwidth 28°.
All else being equal, which is the antenna with the higher gain a “better” antenna? Well, unfortunately all else is not equal.
The 6 dBi Yagi is only 14.2 inches long while the 14 dBi Yagi is a whopping 60 inches (that’s 5 full feet) long, almost six
times the length of the 6 dBi antenna. If constrained by size, the 6 dBi antenna may be the better choice.
Although the Yagi antenna does a good job at directing (and receiving) energy from the forward direction (in the main lobe),
its sidelobes are fairly large in comparison.
Major Parameters for the Yagi Antenna
1. Gain – A Yagi’s gain is directly proportional to the number of parasitic elements, with typical gains in the range of 5-
20 dBi (You will never be asked to calculate the gain of a Yagi antenna based on its dimensions).
2. Beam Pattern/Beamwidth – A Yagi is a directional antenna that transmits energy in a main lobe, but with fairly high
side lobe levels. The beamwidth is dependent on the number of parasitic elements, with more elements resulting in a
narrower beamwidth.
3. Bandwidth – A Yagi is typically very narrowband, with a bandwidth ~5% of the center frequency. A Yagi will have a
center frequency equal to λ, where  is based on the length of the dipole radiator.
4. Physical Size – A Yagi’s dipole radiator has a physical size equal to λ/2, but the length is determined by the number of
directors. Directors are typically spaced in half-wavelength increments.
You should be familiar with the four major parameters for the following three antennas:
Dipole
Monopole
Yagi-Uda
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5. Wireless Propagation
If the antenna provides the transition from guided to unguided media then it naturally follows that we will be concerned about
free space propagation. Recall from Chapter 12 that we discussed the physical layer, guided and unguided transmission
media. There are many cases where free space is the appropriate choice a communication system. Perhaps you are using
military radios for tactical purposes and constantly on the move. Alternatively, perhaps, you live in a remote or rural area and
it’s simply not cost efficient to run cables to distant homes. For these reasons, we must be aware of the behavior of EM
waves in free space.
Propagation is the means by which a signal moves from Point A to Point B. It sounds simple, but it is the most fundamental
and challenging aspect of wireless communications. In a wired system (such as an Ethernet network), propagation is not
really a concern per se. However, wireless transmission requires a fundamental understanding of how electromagnetic waves
move through the atmosphere. The challenges of propagation in free-space include the fact that the transmitter and/or
receiver may be moving, obstacles in the path of propagation, a path that is not necessarily a straight line, and a signal that
takes various paths to get to the receiver.
In general, we can think about radio frequency propagation in two broad categories: large scale and small scale. Our
emphasis in EC310 is on understanding large scale propagation (longer distance), although many engineers have devoted
their entire careers to understanding and modeling small scale propagation. In fact, you’ve likely experienced a small scale
propagation issue numerous times without ever realizing it. The classic example would be driving down the highway while
talking on your cell phone and experiencing a dropped call.
Large Scale Propagation – The behavior of the radio channel over large distances (100s or 1000s of wavelength of
distance). Received power is directly related to distance between Tx and Rx, and is stationary with respect to time.
Small Scale Propagation – The behavior of the radio channel over a small local area (1-10 wavelengths of
distance) and/or of small time durations. Received power fluctuates rapidly based on position, speed, direction of
travel, etc. of the mobile.
5.1 Large Scale Propagation Large scale propagation is affected by four physical phenomena:
 Reflection – the bouncing of EM waves off of surrounding objects, such as vehicles, buildings, etc.
 Refraction – the bending of EM waves as they travel through mediums of different material
 Diffraction – bending of EM waves around objects
 Scattering – diffuse re-radiation of EM waves off rough (smaller than the signal’s λ) objects
Let’s look at these briefly one at a time.
5.1.1 Reflection Reflection occurs when a transmitted EM wave strikes a conductive object (such as a metallic object) on its
path to the receiver. As you recall from physics, in reflection, if the object is flat, the angle of reflection is equal to the angle
of incidence.
5.1.2. Refraction When an EM wave passes from one medium to another, the EM wave’s path can change direction (bend).
In wireless communications, we see this when EM waves directed towards the sky go up into the ionosphere, and eventually
bend back down to earth as depicted in the figure below.
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5.1.3. Diffraction Diffraction is the bending of EM waves around objects in their path, even behind them to some extent.
Consider a transmitter and receiver where an object is blocking the direct line-of-sight path between them. The signal can
diffract around the object such that the signal can get to the receiver even though it is shadowed. Note that the more deeply
the receiver is shadowed, the lower the received power, and in some cases, the receiver may not be able to receive any signal.
The concept of diffraction is illustrated below.
5.1.4. Rough Surface Scattering Sometimes called diffuse scattering or diffuse reflection, scattering happens when an EM
wave impacts a rough surface and is re-radiated in many directions at much reduced power levels.
So those are the basic physical phenomena propagation. What happens when we add in a real earth and a real atmosphere?
The earth and the earth’s atmosphere have the greatest impact on signals in the VLF – HF range (3 kHz – 30 MHz). It’s not
that the earth and atmosphere don’t affect signals at higher frequencies, it’s just at those higher frequencies other factors
come into play and dominate the effects of the earth/atmosphere. Let’s look at what happens to these lower frequencies first
before moving on to the higher frequencies.
Frequency Range Frequency Range Propagation Mode

VLF (Very Low Frequency) 3 kHz – 30 kHz - Ground waves
LF (Low Frequency) 30 kHz – 300 kHz - Ground waves
MF (Medium Frequency) 300 kHz –3 MHz - Ground waves, sky waves at night
HF (High Frequency) 3 MHz – 30 MHz - Sky waves
VHF (Very High Frequency) 30 MHz –300 MHz - Space waves
UHF (Ultra High Frequency) 300 MHz – 3 GHz - Space waves
SHF (Super High Frequency) 3 GHz – 30 GHz - Space waves
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EHF (Extremely High Frequency) 30 GHz – 300 GHz - Space waves

5.2 Modes of Propagation
For VLF-EHF communications, there are three basic modes a radio wave can travel from the transmitter to a receiving
antenna:
 Ground wave – EM waves that travel close to the surface of the earth
 Sky wave – EM waves that travel up into the atmosphere and then bend back to earth
 Space wave – EM waves that travel in a straight line (direct line-of-sight or LOS)
The mode of propagation is determined by the frequency (carrier frequency) of the radio wave. But what determines the
frequency of the transmission? The properties of how the signal will propagate in the communications channel. In this way,
the logic is circular, yet complimentary.
For example: If you have a ground station which is communicating with a satellite the carrier frequency will need to pass
through the ionosphere. Frequeices that propagate via group or sky waves would not be suitable for this communication
channel. The properties of the communications channel (that includes the ionosphere in this case) dictate the acceptable
carrier frequencies and mode of propagation.
5.2.1 Ground Wave Propagation A ground wave is a radio wave that travels along the earth’s surface (also referred to as a
surface wave). A ground wave must be vertically polarized; that is, the antenna must be oriented vertically.
Lower frequencies travel efficiently as ground waves because they are diffracted by the surface of the earth. Ground waves
thus follow the curvature of the earth and can travel beyond the horizon, for hundreds of miles. Ground wave propagation is
strongest in the LF and MF frequency ranges. Ground wave propagation constitutes the main propagation mode for signals in
the frequency range from 30 kHz – 3 MHz.
5.2.2 Sky Wave Propagation Sky waves are radiated by an antenna into the upper atmosphere where they are reflected or
refracted back to earth. The air molecules of the ionosphere are subject to severe radiation from the sun. Ultraviolet radiation
causes the molecules to ionize, or separate into charged particles, positive and negative ions. This separates the upper
atmosphere into different layers (or mediums) that promote reflection or refraction.
The direction of reflection depends on the angle at which the radio wave enters the atmosphere and the different degrees of
ionization of the layers, as well as the frequency of the transmission.
5.2.3 Space Wave Propagation A space wave refers to the radio wave that travels directly in a straight line from the
transmitting antenna (LOS). These waves are not refracted, and do not follow the curvature of the earth. The chief limitation
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of a space wave is that it is limited to line-of-sight distances. The range of space wave propagation is limited by the curvature
of the earth and height of the antennas above the earth’s surface.
If an antenna has a height h above the surface of the earth, the distance, d, to the radio horizon (which is the maximum range
for space wave communications from that antenna) is given by the formula
d  2h .
Important: In this formula, the height of the antenna is in feet, and the distance to the horizon is in miles. That is, if you plug
in the antenna height in feet, the resulting distance value will be in miles.
The next figure demonstrates the maximum distance that two stations can be apart and still conduct line-of-sight
communication. This figure shows one antenna of height h1 and a second antenna of height h2. The maximum separation at
which they can still communicate by line-of-sight is given by:
dtotal  2h1  2h2 .

What is the longest line-of-sight communication range between a transmitter whose transmitting antenna is 350 feet high and
a receiver whose receiving antenna is 25 feet high?
Solution:
Now that we’ve covered all the glories of Large Scale Propagation in real-world environments, it behooves us to look at the
most basic way we can transmit energy from Point A to Point B in an environment devoid of terrain, mountains, buildings,
ground, or atmosphere. Such an environment is known as Free Space, and conveniently, wireless propagation in such an
environment is known as Free Space Propagation.
5.3 Free Space Propagation Let’s consider the following scenario. You have a brand-new iPhone (or Samsung phone as the
case may be), have just signed up for a super-fast LTE plan, and would like to upload a photo, surf the web, browse
Facebook, or just plain make a phone call. To make that happen, your phone has to transmit that information over the air to
the nearest LTE cell tower (cost: $5 Million, that’s why your phone bill is $100/month), which happens to be 5 miles away.
Question: Will your signal make it to the tower and will it have sufficient power to “close the link” and allow you
communicate? Or will you suffer the fate of a cellular “dead zone”? That depends on the amount of signal power that is
received.
Recall from Chapter 24 the discussion of antenna gain. An antenna has gain if it can focus its transmitted power (or can
receive power) in a certain direction, as opposed to an isotropic antenna that radiates (or receives) power equally in all
directions (in a spherical shape). This led to the term effective isotropic radiated power (EIRP), which is the amount of power
an isotropic antenna would have to radiate in order to match the power that a directional antenna radiates in the direction it is
pointing. To figure out how to compute received power, let’s consider how an isotropic antenna radiates in a spherical shape.
As EM waves move away from the isotropic antenna, the sphere gets larger and larger, until it touches our receive antenna.
The transmitter transmits a constant power, however, the power density is going to decrease as the distance from the transmit
antenna increases. Power density is the amount of power received per unit area (W/m2). The power density that reaches the
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receive antenna is going to be based on the surface area of a sphere, where the distance between the transmitter and receiver
(d) is the radius of the sphere. Since the surface area of a sphere of radius d is given by:
Asphere  4 d 2
the power density (Pd) at the receiver in units of W/m2 is:
Pisotropic
EIRP PG
Pd    t t .
antenna
Ae 4 d 2 4 d 2
Now, the last thing we need to do is to turn that power density into the actual received power. Power density is power per unit
area, so what is the “area” we are interested in? Since we are receiving the signal on an antenna, the “area” of interest is the
area of the receive antenna. The derivation of the effective area of an antenna is beyond the scope of the course, but it is
mathematically defined as:
G 2
Ae  r .
4
Finally, we can put all this together and determine the equation for received power, which is received power density (W/m 2)
multiplied by effective area (m2):
t t Gr 
Gr  2 PG 2
PG
Pr  Pd  Ae  t t
 
4 d 2 4  4 d 
2
where the variables are defined as:
Pr Received power (W or mW)

Pt Transmitted power (W or mW)
Gt Transmit antenna gain (unitless)
Gr Receive antenna gain (unitless)
λ Transmission wavelength (m)
d Distance between transmitter and receiver (m)
This is known as the Friis Free Space Equation. It is fundamental to understanding how received power is reduced as a
function of distance for wireless communications.
Important note: in this equation, there are NO decibel terms! The two most common mistakes made when using this equation
is using dB values instead of linear values, and failing to get the wavelength/distance units correct. If you’re given a problem
that includes dB values for any of the terms, take the values out of decibels!
Let’s go back to our cell phone example.

Your cell phone transmits at a power level of 500 mW, with an antenna gain of 2.0 dB. The cell tower has an antenna gain of
8.0 dB, and is a distance of 5 miles away. For LTE, you’re transmitting at 700 MHz. Will your signal make it to the tower
and will it have sufficient power to “close the link” and allow you communicate? Or will you suffer the fate of a cellular
“dead zone”? (note: 1 mile = 1.609 km, and consider −105 dBm as the minimum power required to be able to “close the
link”)
Solution:
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CH. 24 Problems
1. What are the 4 engineering factors associated with the design of antennas?
2. a) Calculate the length, in meters, of a dipole antenna that is designed to receive a station at AM 800 on the dial of an
AM radio.
b) Calculate the length, in meters, of a monopole antenna that is designed to receive the FM station at 107.1MHz.
3. Given the following radiation pattern, where each ring represents a 1 dB change in power, what is the beamwidth? The
sidelobe level? The front-to-back ratio?
4. Given the following radiation pattern, where each ring represents a 2 dB change in power, what is the beamwidth? The
sidelobe level? The front-to-back ratio?
5. The power applied to an antenna with a gain of 4 dB is 13 W. What is the EIRP?
6. What does it mean for an antenna to have directivity, and what are the advantages and disadvantages of a directional
antenna?
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7. Name and describe the three basic elements in a Yagi antenna.
8. The length of the driven element in a Yagi antenna is 900 mm; what is its operating frequency?
9. The mainlobe of an antenna has a maximum gain value of +18 dB at its peak point of forward direction. The same
antenna has a gain of −5dB at the peak point of its rear lobe. Determine the front-to-back ratio of the antenna.
10. Why do radio waves not always follow a straight line? Explain how large scale propagation phenomena are both useful
and harmful.
11. A ship-to-ship marine-band VHF radio operates at 156 MHz and is limited to a maximum of 25 watts. The signal
propagates via space propagation, so it is limited in range to direct line-of-sight. A Coast Guard transmitting station on
shore has a monopole antenna that is 350 feet tall.
(a) If a ship is 35 miles (56,315 m) away from the CG station, how high must the ship’s monopole antenna be mounted
to ensure reception?
(b) Using the Friis Free-Space equation, calculate the received power at the ship.
(c) If someone is standing in a life raft with a hand-held VHF radio (assume a monopole antenna held up to a height of
6’), what is the maximum range from which they could contact the ship in part (a) of this problem?
12. In a certain communication link, the transmitted power (Pt) is 5 W and along the way to the receiver, the reduction in
power is 100 dB. What is the received power in mW?
13. Suppose you’re in Paris using the European cell phone system (called GSM), and a cell tower there has an antenna gain
of 10.0 dB. For GSM, your phone transmits at 900 MHz with a power level of 500 mW, and an antenna gain of 5.0 dB.
How far from the tower could you be while maintaining the capability to communicate? (Cellular “dead zone” occurs
when Pr < -110dBm.)
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Drivers start your engines.
Today we looked at various ways radio waves propagated through space, air. For this lab, we will be using radio control (RC)
cars, our communication system, to evaluate the propagation of electromagnetic waves as they traverse through space. Now
we have gone out of the way to purchase the best radio control cars in the world. That’s right! Only the best for you guys. We
acquired Ferraris, Audi R8, Lamborghinis, Camaros, etc. Don’t they look so pretty?
The cars that you have available to you today operate at a couple of different frequencies.
Question 1: Examine the cars and write down the frequencies at which the cars operate on your answer sheet.
Question 2: Based on the frequencies you just determined for the cars in your classroom and what you learned in class,
which propagation mode is used to control these cars? (i.e. ground wave, sky wave, space wave)
Question 3: Why won’t the other two propagation modes work?
Question 4: What are the wavelengths of the frequencies associated with the RC cars? Show your work and record your
wavelengths on the answer sheet.
Now that you know the wavelengths associated with the frequencies, what is the theoretical distance you expect the cars to
travel? You need some information to calculate the distance using the Friis Free Space Equation. The gain for the
transmitting antenna is -8 dBi. The gain for the receiver antenna is also -8 dBi. When using fully charged batteries the power
of the transmitter (PT) is 10 dBm. The minimum power necessary at the receiver (PR) to control the car is -50 dBm.
Rearrange and use the Friis Free Space equation to determine the distances for both the high- and low-frequency car.
Question 5: Show your work and record your expected distances on the answer sheet.
You now have your theoretical maximum distance both RC cares will travel. It’s time to take measurements and see how
accurate they are. Measure how far the lower frequency car will go. Make sure your measurement is in meters. Drive from
the front of the classroom to the back and around back benches, not out of the classroom. Remember the distance be should a
straight line to the car, not the path is takes. So how far did it travel?
Question 6: Record the experimental distance for the lower frequency car on the answer sheet.
You should’ve noticed that the car didn’t go nearly as far as you calculated. Why?
Think back to the equation you used to calculate the distance. What did we say about the equation? It needs to be used in free
space without obstructions. That means no terrain, mountains, buildings, ground, or atmosphere. In the classroom, there are
desks, lab equipment, people—all obstacles. So the Friis Free Space equation isn’t going to provide an accurate distance.
When we have all this furniture and equipment that can interfere with the signal, they will reflect the signal, diffract the
signal, and/or scatter the signal. Remember that:
(1) Reflection occurs when energy (or the signal) reflects off a large (relative to the λ) conductive surface.
(2) Diffraction occurs when energy bends around objects.
(3) Scattering occurs when EM waves strike a rough surface (smaller than λ) and re-radiates the EM wave in many
different directions.
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As the signal is affected by all the lab equipment, people, etc., the signal at the receiver is a combination of many variations
of the original signal. This variation leads to a reduced signal strength and the RC car not travelling as far. So how are you
going to determine how far the higher frequency car should go?
Let’s use the Log-Normal model. This model is widely used to not only predict coverage for a particular mobile user (i.e. the
RC car), but also for predicting the interfering signal power that the mobile user will experience from other Radio Frequency
sources (i.e. the cell phones in your pocket). A description of the log-normal model begins with the definition of path loss.
Path loss is the amount by which the transmitted signal has dropped by the time it gets to a receiver at distance d away.
Usually computed in decibels, path loss as a function of distance d is defined as:
PLdB (d )  Pt ( dB )  Pr ( dB ) (d )
The log-normal equation is usually computed in decibel form:

d 
PLdB (d )  Pt ( dB )  Pr ( dB ) (d )  PLdB (d 0 )  10  n  log10  .
 d0 
In this equation, the variables are:
d Distance from transmitter to receiver in meters

d0 A reference distance, usually 1 meter
n Path loss exponent (unitless), with value depending on
the environment where the communication occurs
PLdB(d) Path loss at distance d (in dB)
PLdB(d0) Path loss at reference distance d0 (in dB)
The value of PLdB(d0) is usually calculated with the Friis Free-Space equation or measured empirically. Note that antenna
gains, wavelength, etc. are embedded in the model (in PLdB(d0) and n) parameters. Changing the configuration means we will
end up with different model parameters and different results.
Values for path loss exponents have been tabulated for a number of environments, and a few representative values are given
in the following table.
Path Loss Exponents for Different Environments
Environment Path Loss Exponent, n

Urban Area 2.7 to 3.5
Dense Urban Area 3 to 5
In Building with Line-of-Sight 1.6 to 1.8
In Building Obstructed 4 to 6
Factory Floor Obstructed 2 to 3
Retail Stores 1.8 to 2.5
To predict the distance of the higher frequency car, you need a few pieces of information. Inside a building with obstructions
(i.e. your classroom), you would expect a path loss exponent (n) between 4 and 6. For your classroom, use 4 as the path loss
exponent (n = 4). At the max distance, you would expect a Pr of −50 dBm, and your Pt is 10 dBm (use the difference in Pt
and Pr to determine PLdB(d)). The last piece of information you need to make your calculation work is: at d0 = 1 meter, the
path loss for 40 MHz is 10 dB (that is, PLdB(d0) = 10 dB). Use the path-loss equation above to compute the distance d, given
these parameters.
Question 7: Show your work and record the new expected distance (d) for your higher frequency car on the answer sheet.
Question 8: Now go drive the higher frequency car. Drive from the front of the classroom to the back and around back
benches, not out of the classroom. How far did it go in meters? Remember the distance should be a straight line to the car, not
the path is takes.
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You did it. So now you can calculate, at least for a RC car, the distance a radio wave will travel. But be aware that if you
change the configuration, (i.e. you go into the hall) you will have different model parameters and therefore different results.
One last test of you mathematical skills. Calculate the distance the higher frequency car will travel if you were outside. In this
case, use a path loss exponent (n) of 2.6. All other parameters are the same.
Question 9: What is the expected distance (d) for your higher frequency car? Record on the answer sheet.
Watch the Youtube video, http://www.youtube.com/watch?v=YzbxHcp3UEk.
Question 10: Were you correct (roughly)? ____________________________
Your final test. Using either car, place the car against the wall inside the classroom next to the door. Go outside the door
where you can no longer see the car. Just on the other side of the wall should be fine. Try to move the car using the radio
controller.
Question 11: Did it move? Why or why not?
As we discussed in the chapter, it’s important not to consign the large scale propagation phenomena to “a good thing” or “a
bad thing.” We certainly saw in this Security Exercise that reflection, refraction, diffraction and scattering negatively impact
the theoretical maximum distance our RC can travel. However, RC cars use Space Waves and should be limited to line of
sight transmissions. The same reflection, refraction, diffraction and scattering cause the EM waves to reach our RC car
beyond line of sight and control the car in Question 11. The key requirement with wireless propagation is to understand how
your signal’s propagation will be affected in the communication channel, model it and plan accordingly so your signal is still
received at the distant end.
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496

Name:
Question 1:
Low frequency = _____________ High frequency = _____________
Question 2:
Question 3:
Question 4:
Wavelength (low frequency car) = _____________ Wavelength (high frequency car) = _____________
Question 5:
Distance (low frequency car) = _____________ Distance (high frequency car) = _____________
Question 6:
Experimental distance for low frequency car =
Question 7:
Question 8:
Question 9:
Question 10:
Question 11:
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Chapter 25: Electronic Warfare
Objectives:
(a) Define Electronic Warfare and provide an example of each of the three Electronic Warfare categories: Electronic Defense,
Electronic Warfare Support and Electronic Attack.
(b) Define Jamming to Signal ratio (J/S) and calculate the necessary power to jam an emitter.
Warfare involves offensive and defensive operations.
In the Host Module, we learned that an adversary can attack our host computer by employing a buffer overflow exploit. To
counter this attack, we have several defensive actions at our disposal; for example, we can avoid the C library functions that
are notorious for inviting buffer overflows, we can use a non-executable stack, a canary can be used to detect an attempt to
overwrite a stored return address, etc. Recall also that, aside from formal attack operations and defensive responses, an
adversary might attempt to look for flaws in our host software. For example, an adversary might enter a ridiculously long
value when prompted to enter something, as a test to see if it can make the program behave erratically.
In the Network Module, we learned that an adversary can attack our network using either a false route injection attack or a
wide-area BGP route-hijacking attack. To defend against false route injection, we can use an OSPF authentication
mechanism, or we might selectively set up passive interfaces on router ports. To defend against a wide-area BGP route-
hijacking attack, we can use judicious filtering at Autonomous System borders, or we might attempt to authenticate routing
information against an Internet Routing Registry, or we can attempt to receive some cryptographic assurance of the routing
information we receive by using the Resource Public Key Infrastructure. Recall also that, aside from formal attack operations
and defensive responses, an adversary might attempt to perform "network reconnaissance" by using Wireshark, nmap or
various network utilities.
Not surprisingly, we find in the Wireless Module that the electromagnetic spectrum can also be used for offensive and
defensive operations, as well as for "reconnaissance" operations. In the context of wireless systems, these attack, defensive
and reconnaissance operations are termed electronic warfare. The jamming and taking over of communication links are two
of the ways that cyber attackers exploit wireless communications.
1. Electronic Warfare (EW)
The term Electronic Warfare (EW) refers to any action involving the use of electromagnetic energy to attack an adversary or
to otherwise control the electromagnetic spectrum. EW includes three major subdivisions: electronic attack, electronic
defense, and electronic warfare support. We'll discuss each of these in turn, starting with electronic warfare support.
1.1 Electronic Warfare Support Electronic warfare support refers to those actions that are taken to search for, intercept,
identify, and locate sources of radiated electromagnetic energy for the purpose of target identification, or for the planning
and conduct of future operations. Phrased another way, electronic warfare support entails gathering knowledge about the
enemy through the use of the electromagnetic spectrum.
We discussed an example of electronic warfare support in Security Exercise 23. Recall that in that lab you wandered the
hallways of the Rickover lab deck in search of a wireless access point. This was, at heart, an electronic warfare support
operation—you were attempting to locate a radio emitter of interest. In the lab, your only goal upon locating the emitter was
to note the funny message placed next to it. In a more realistic scenario, the data gathered from an emitter could produce
intelligence concerning the user (friend or foe?) and their location.
Suppose you can pick up an adversary’s radio transmission. How could you determine the direction it is coming from? If you
used a directional antenna like a Yagi you could determine a compass bearing in the direction of the emitter. If you get a
compass bearing from three locations you could plot the bearings on a map and get a fix. This was actually one of the early
means for ships to fix their position by electronic means, via the Omega or Loran C navigation systems, which were
operational until shut down in favor of GPS.
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1.2 Electronic Defense Electronic defense includes those actions taken to protect personnel, facilities, and equipment from
an adversary's use of the electromagnetic spectrum to attack us. It should be noted that in DoD literature the term "electronic
defense" is often termed "electronic protection", since in defending ourselves, we are protecting ourselves. Years ago, the
term for electronic defense was electronic counter countermeasures –ECCM, and before that, the preferred term was
electronic protective measures.
We discussed an example of electronic defense in Security Exercise 23. Sure enough, you were simply wandering the
hallways in search of a wireless access point. But in an analogous fashion, an adversary can home in on the transmissions of a
ship, a submarine, an aircraft, or forces in the field. To prevent an adversary from using the electromagnetic spectrum to
locate our transmitter, we will often limit radio communications to the minimum necessary. Thus, emissions control is a form
of electronic defense.
Another form of electronic defense is the use of stealth technologies (shapes with low radar cross-sections, non-metallic
materials, radar-absorbent coatings) to protect aircraft and ships from radar detection.
The definition of electronic defense is broadened to also include not only the actions we take to defend ourselves, but also the
actions that we take to protect our own ability to attack the enemy. This can lead to some confusion. For example, if we
launch an infrared homing missile against an enemy, we are engaging in electronic attack. If our enemy sees the incoming
missile and launches flares in an attempt to divert it, it would engage in electronic defense and electronic attack. But if we
counter its flares by using flare-rejection technology on our infrared homing missile, we are also engaged in electronic
defense, since the flare-rejection technology protects our ability to attack! Think of the great exam questions!
1.3 Electronic Attack Electronic attack involves the use of electromagnetic energy to attack personnel, facilities, or
equipment with the intent of degrading, neutralizing, or destroying an enemy's combat capability. The preeminent example of
electronic attack is jamming.
Jamming – Jamming is the transmission of an electromagnetic signal that disrupts an adversary's communications.
Consider the picture below. An enemy transmitter is sending information to an enemy receiver at a certain frequency, say f.
The enemy transmitter and receiver are separated by a distance dS. Meanwhile, our hero is a distance dJ away from the enemy
receiver.
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Our hero's goal is to transmit another signal—a jamming signal—at the same frequency—f—that the bad guys are using. The
jamming signal will target the bad guy's receiver. So, the intent is to have two signals arrive at the bad guy's receiver: the
signal sent by the bad transmitter, and the midshipman's jamming signal. The midshipman's goal is to have her jamming
signal be of sufficient power to override the signal from the bad transmitter, thus preventing the bad guys from
communicating.
It is important to note that what we are jamming is the receiver, not the transmitter. As an analogy, imagine trying to yell
something to someone across Worden Field. If a third person comes along and wants to prevent you from communicating,
what would be more effective: to yell in the ear of the person trying to relay a message or to yell in the ear of the person
trying to hear the message? The latter would be more effective.
I know what you are saying: Where's the math?
For the jammer, the object is that the received jamming power at the Bad Guy receiver be greater than the received signal
power from the Bad Guy transmitter. Using the Friis equation, the received jamming power (PJ) in terms of the jammer’s
EIRP is (rearranging the equation a little):
EIRPJ  Gr   2 EIRPJ Gr  2  EIRPJ  Gr  2 

PJ     2 
 4 d J   4   d J2  d J   16 
2 2 2
Similarly, the received signal power from the Bad Guy transmitter is:
 EIRPS  Gr  2 
PS    2 
 d S   16 
2
If we divide the received jamming power by the received signal power, we create the jamming-to-signal ratio (J/S), a term
similar to a signal-to-noise ratio:
 EIRPJ   Gr  2 
 EIRPJ 
   2  
 d J   16 
2
d2    EIRPJ   d S 
2
 J
J PJ
    2 
S PS  EIRP   G  2   EIRPS   EIRPS  d J 

S

r
2   
 S      dS
2

2
d 16
Note that the wavelenths cancel since in order for our jamming to be effective, our jamming signal must be the same
frequency as the transmitted signal. This equation is usually used in terms of decibels, so for power in watts;
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J
   EIRPJ ( dBW )  EIRPS ( dBW )  20log10  d S   20log10  d J  ,
 S dB
and for power in milliwatts:
J
   EIRPJ ( dBm )  EIRPS ( dBm )  20log10  d S   20log10  d J  .
 S dB
A J/S ratio greater than one (or a positive dB value) will mean the received jamming signal is stronger than the received Bad
Guy transmitter signal. Note that in these J/S equations, the distances to the jammer and to the signal must be in the same
units (e.g., meters).
Finally, an important assumption this equation makes is that the receiver has an omnidirectional beam pattern. This means the
receiver will accept transmissions equally from all directions. If this were not so, then the equation above would need to take
the receiver’s beam pattern into account.

You are located 5500 meters from the omnidirectional receiver you are jamming. The transmitted signal that you are
jamming originates 9500 meters from the receiver. The signal transmitter’s EIRP is 15 dBW. Assuming both the transmitter
and jammer have line of sight, what EIRP dBW must you transmit to jam the receiver with a J/S of 5 dB? How many watts is
this?
Solution:

You can transmit an EIRP of 25 Watts with your jammer. The transmitted signal you are jamming originates 8500 meters
from the omnidirectional target receiver. The signal transmitter’s EIRP is 15 dBW. Assuming both the transmitter and
jammer have line of sight, how close must your jammer be to the target receiver to achieve a (J/S) dB of 3 dB?
Solution:

Is there a possibility that our jamming scheme would not work if the Bad Guy Receiver was not omnidirectional? Explain.
Solution:
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Basics of Electronic Warfare
We devoted an entire third of this course to learning about wireless communications systems and the associated
considerations, from modulation to gain to antennas and signal propagation. Why? Because “Cyber” doesn’t exist solely in a
single computer or a closed network. You can have a significant impact by using Electronic Warfare as an enabler for Cyber
attacks. See: http://breakingdefense.com/2013/04/adm-greenert-wireless-cyber-em-spectrum-changing-navy/
Now we’re going to put all that knowledge to the test and apply your cyber skills in a wireless environment.
Set-up.
Equipment required:
□ Agilent InfiniiVision MSO7032A Oscilloscope
□ RF Explorer Handheld Spectrum Analyzer
□ Telescoping Antenna w/ BNC connector
□ RC Vehicle
□ BK Precision 4064 Arbitrary Waveform Generator & accessories
(Instructor will set up)
Part 1: Data Collection
Communications System. For this Security Exercise, we’ll explore the entire communications system employed by a Radio
Control (RC) vehicle… And then we’ll exploit it!
Answer the questions that follow to examine the RC vehicle’s communications.
Note: These images resemble the models in your classroom enough to give you the general idea.
We can’t all have Ferraris, after all!
Question 1: Which image above (left or right) most closely represents the transmitter?
Question 2: Where is the receiver located?
Question 3: What type of channel does this communications system involve?
Question 4: What do you expect your “information” to be in this case?
Question 5: What will happen when the “information” is recovered at the receiver?
Question 6: What type of antenna does the transmitter use?
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Question 7: What would you expect the beam pattern of this antenna to look like?
Question 8: Do the transmitter or receiver give any indication of carrier frequency? If so, what is fc?
To verify the carrier frequency of the transmitted signal, use the RF Explorer Handheld Spectrum Analyzer.
1. Attach Nagoya NA-773 telescoping antenna to SMA-Right connector on top of RF

Explorer.
2. Switch on RF Explorer
3. Press MENU until Frequency Menu is displayed and set/verify the following settings:
(a) Module: 15-2700M
(b) Freq Span: 000.112
(c) Center Freq: ( 27.145 | 40.680 | 49.859 ) choose correct frequency for
your car.
4. Press MENU until ATTENUATOR MENU is displayed and verify the following
settings:
(a) Calculator: Normal
(b) Top dBm: -030
(c) Bottom dBm: -110
(d) Iterations: 016
(e) Offset dB: +000
(f) Units: dBm
5. Press MENU until CONFIG MENU is displayed and verify the following settings:
(a) Backlight: Max
(b) Contrast: 3
(c) USB Bauds: 500 Kbps
(d) Draw Mode: Vectors
(e) Marker: Peak
(f) DSP: AUTO
6. Press RETURN to see the main spectrum display
7. Transmit from RC vehicle controller and verify signal displays on the spectrum analyzer.
Question 9: What is the carrier frequency? Draw the signal in the frequency domain.
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Part 2: Jamming
Now that we have some basic intel, what could happen if your instructor was to transmit a signal at the carrier frequency?
The answer: It depends!
In lecture, we learned that the effectiveness of electronic attack/jamming is dependent upon the jamming-to-signal Ratio
(J/S). The J/S depends upon both the power output from the jammer and the transmitter, as well as the distance of the jammer
and the transmitter from the receiver. In this security exercise, our scenario looks like this:
The J/S depends on the received signal power at the car and the received jamming power at the car:
æ Jö æ PJ ö
çè ÷ø = ç ÷ = PJ (dBm) - PS(dBm)
S dB è PS ø dB
Generally, if the J/S ratio is greater than 1 (or 0 dB), jamming will be effective.
□ Play time! Drive your vehicle around the classroom.
Question 10: What two conditions (with regards to frequency and received power) must exist for jamming to be effective?
Get your instructor’s signature to continue.
□ Your instructor will generate a 20 dBm frequency modulation (FM) signal at the carrier frequency.
Question 11: What is your instructor’s target?
□ While your instructor is transmitting the jamming signal, experiment! Attempt to control the RC car with its
transmitter at different distances from both the jammer and the RC car.
Question 12: When your instructor transmitted a jamming signal, were you still able to control the RC car? When could you
control it? When could you not?
Question 13: Use the RF Explorer Handheld Spectrum Analyzer to draw the jamming signal in the frequency spectrum.
How does this change if you transmit while standing next to the Spectrum Analyzer?
Question 14: How could you increase the range of the jammer? (How is jamming range dependent on signal power?)
Part 3: Reverse Engineering
So now we know the carrier frequency and the effects of transmitting a higher signal power on that frequency, but if we want
to make a bigger impact, we need to know more about the RC car’s signal. What does the transmitted signal look like? What
type of modulation does it use? How do controls work? To accomplish this, we’re going to look at the signal using the
Agilent Technologies “InfiniiVision” MSO7032A 350 MHz Oscilloscope.
505
First, some initial set-up for the O-Scope. For instructions on setting up and operating the InfiniiVision O-Scope, refer to SX-
25 Equipment Setup Instructions, Part 3: Reverse Engineering posted with the equipment at the lab station.
Once you have the Channel configured, it’s time to Capture the Signal using the Trigger section of the O-Scope.
□ Press the Edge button

□ Set Source = 1
□ Ensure the antennas are extended on the O-Scope and the Remote if applicable
□ Holding the RC car transmitter close to the O-Scope,
send the Forward signal by driving the car forward.
□ Capture the signal by pressing the Run/Stop button
If done correctly, your O-scope display should look similar* to this:
* Captured signal may vary – that’s ok for now!
Question 15: What type of digital modulation does this car use?
Question 16: What pattern of 0s and 1s does the transmitted signal represent?
To be able to control the RC car, we want to be able to do more than just drive it forward. How does the signal change for
reverse, left, or right?
Think about the controls – how many different signals do you expect to control the car? In addition to driving forward, the
car can operate in reverse, as well as turning left and right… and any combination thereof! There are actually 8 different
combinations of signals, but in the interest of time we’re only going to worry about four: Forward, Reverse, Forward &
Right, and Forward & Left. Here’s the catch: the chips that process the signal and control the vehicles motion aren’t
necessarily wired the same way in every car, so you need to identify which control operation each transmitted signal
represents!
Examine each transmitted signal by repeating the process you just followed to Capture the Signal on page 4:
□ Press “Run/Stop” button until “Run” lights-up Green
□ Transmit desired signal, one at a time:
o Forward
o Reverse
o Forward AND Right (This is different from the signal to pivot the wheels to the right only!).
o Forward AND Left (This is different from the signal to pivot the wheels to the left only!).
□ As each of the four signals is displayed on the O-Scope screen press “Run/Stop” once, Stop will light Red
Question 17: Match the transmitted signals with the operations they represent by circling the correct response. The signals
can be distinguished by the number of 1s being transmitted after the 4 large sync pulses.
Forward or Reverse or Forward-Right or Forward-Left?? ( # of 1’s: 10)
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Question 18: Now that you’ve identified the modulated signal that controls the car, could you determine the baseband binary
signal (voltage pulses) that are used for each control function? The block diagram for an OOK signal’s generation is shown
below.
We now know the bits that are transmitted to control the forward, turning, and reverse motions of the RC car. We also know
that we can’t transmit the baseband binary signal, so we need to modulate it on a high frequency carrier. If we could
reproduce these control signals and transmit by some other means than the car’s remote, do we need the remote to drive the
RC car? Let’s find out!
Part 4: The Hook
You may have noticed that each transmitted signal consists of 4 wide “sync” pulses followed by a trail of 0’s and 1’s.
On the oscilloscope, the control signal will be displayed as seen in the figure that follows.
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In Question 17 you matched the waveform to the driving direction. For example, in the image above represents
01110111011101110101010101010101010101110 in binary. Which is the signal for “forward”
Given our reverse engineering efforts, it is possible to write a program which reproduces that same RC car control signals.
Access the U:\Electrical and Computer Engineering\Cyber2\EC310\SX25 and find the RCcode.exe program. Execute the
program by double clicking on it and follow the prompts to enter the number of ones corresponding to each direction.
Press and hold your arrow keys to simulate driving your vehicle.
Question 19: What do you hear? What type of signal is being generated?
□ Bring your laptop to your instructor and get ready to drive!
Question 20: Do you need the car’s transmitter to control the car? What just happened? What is now controlling the car?
Question 21: List some examples of how this might be significant in a military setting. Need ideas? Check this out!
http://www.engr.utexas.edu/features/humphreysspoofing.
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Name:
Question 1:
____________________________________________________________________________________________________
Question 2:
____________________________________________________________________________________________________
Question 3:
____________________________________________________________________________________________________
Question 4:
____________________________________________________________________________________________________
Question 5:
____________________________________________________________________________________________________
Question 6:
____________________________________________________________________________________________________
Question 7:
____________________________________________________________________________________________________
Question 8:
____________________________________________________________________________________________________
Question 9:
____________________________________________________________________________________________________
Question 10:
__________________________
Instructor/Lab Tech Signature
____________________________________________________________________________________________________
Question 11:
____________________________________________________________________________________________________
Question 12:
____________________________________________________________________________________________________
Question 13:
____________________________________________________________________________________________________
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Question 14:
____________________________________________________________________________________________________
Question 15:
____________________________________________________________________________________________________
Question 16:
_____________________________________________________________________________________________
Question 17:
Forward or Reverse? # of 1’s ______ Forward or Reverse? # of 1’s ______
Forward-Right or Forward-Left? # of 1’s ______ Forward-Right or Forward-Left? # of 1’s ______

____________________________________________________________________________________________________
Question 18:
____________________________________________________________________________________________________
Question 19:
____________________________________________________________________________________________________
Question 20:
____________________________________________________________________________________________________
Question 21:
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Appendices
511
512
Basic Linux Commands
The information below presumes you are familiar with the Linux file system and that you understand how to refer to files and
directories using absolute and relative pathnames. You should also be familiar with the commands cd (change directory) and
ls (list). This prerequisite information is available in the handout named The Linux File System available as a link under the
Resources tab on the EC310 course website.
We will now address some basic file and directory operations.
To copy a file from one location to another, we use the cp command. For example, in the file system below
if instructor was in his home directory, and wanted to give bob a copy of the file named hwk4, placing it directly in
bob’s home directory, he would type
cp cyber/hwk4 ../bob
the item to be copied where the copy is to be placed
To create a file from scratch, you can open the file using nano, as you’ve done for all the programs you’ve entered in
EC310.
To move a file, we use the command mv command. In the file system shown above, if user instructor wanted to move
the file named spoofing to place it under the cyber directory, he would enter:
mv spoofing cyber
To view a file, we use the command cat followed by the filename (if it is right in your working directory) or cat followed
by the absolute or relative pathnames (if the file is not right in your working directory).
To remove a file, use the command rm followed by the file name (or pathname as applicable).
We can, in like manner, create a new directory using mkdir followed by the directory name (if we want to place the
directory right under the working directory) or mkdir followed by the pathname where we want to place the new directory.
Finally, we can delete directories by using the command rmdir followed by the directory name or pathname as
appropriate. Note that you can only remove an empty directory, so to delete a directory you must first delete all of its
contents.
513
You should realize that we have only scratched the surface of using the Linux bash shell. You should not place "Linux
System Programmer" on your resume. But what you have learned, along with your use of the gcc compiler, the gdb
debugger and permission management (covered in Lessons 8 and 9) is more than enough for EC310.
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The Linux File System
1. Introduction All users of a Linux OS have an account name (also referred to as a user name or a login name) and a
password. When your Linux account is created, you are also given a home directory where all of your files and folders will
reside. Your home directory has the same name as your account name.
You may be wondering: Hey, I’m using Linux in EC310 and I was never asked for an account name and password while
logging on? That is because your textbook author (Jon Erickson) has set up your VMware software to provide Linux
“already open” for you. We have, however, changed your account name to midshipman since that is, after all, your first
name.
You have been entering commands using the bash shell as your command line interface. Every time you have entered a
command such as:
gcc –g smith_2_1.c
or
nano smith_2_1.c
you have entered that command at the bash shell’s prompt. The bash shell’s prompt for ordinary users is the dollar sign.
Before the prompt, you will see your account name and your computer's name.
Your account name Your computer's name The prompt
There is one additional item in the picture above that you may have noticed: the tilde symbol (~). The tilde is an abbreviation
for your home directory. When you log in, you are placed by default in your home directory.
If you wander up to a computer and notice that someone is logged on, and you see:
then the user whose account name is joe has logged in but has forgotten to log out. Shame on him. Too bad.
If you ever forget who you are, even though your account name is staring you in the face, you can enter:
whoami
as shown below:
In Linux, just as with Windows, there are files. And in Linux, just as with Windows, there are directories (in Windows
terminology, these are referred to as folders), which hold files (or other directories).
A Linux system (like a Windows system) may support multiple users. In such cases, each user is given its own home
directory. When you logon, you are automatically placed in your home directory. When Joe logs on, he is automatically
placed in his home directory. Your home directory is the natural location for any directories or files that you create. You can
leave your home directory and move to other directories. Whatever directory you find yourself in, that directory is termed
your working directory.
A typical Linux file system (also called a directory structure) might look like this:
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At the very top is the root directory, denoted /. The root directory contains all directories and files.
2. Absolute Pathnames Every file can be referenced by its absolute pathname, which starts at the root directory and
traipses down the inverted tree structure, with each entry also separated by a forward slash. For example, the absolute
pathname for the directory joe is:
/home/joe
Note that in an absolute pathnames, the slash (/) character has two different meanings. The first slash always refers to the
root directory. Any other slashes that may be present simply serve as separators.
Since absolute pathnames can be long, a few shortcuts are provided:
 To specify a directory or file in your current directory, you can use just the name of the directory or file.
 A tilde serves as an abbreviation for your own home directory.
 A tilde followed by another user’s name serves as an abbreviation for that other user’s home directory
In the Linux command line, preceding the prompt, you are also provided with an indication of your current working
directory. If you are in your home directory (which, as an absolute path name for our classwork in EC310, might be
something like /home/midshipman ), this will appear as just a tilde since, recall, a tilde serves as an abbreviation for your
own home directory.
3. Relative Pathnames Whereas absolute pathnames always start from the root, relative pathnames start from your current
location (i.e., your working directory). The notation relies on the use of two dots (..) to serve as an abbreviation for the
immediate parent of the current directory.
As an example, in the picture above, if your working directory is instructor, the relative pathname of the home directory
is simply: .. On the other hand, if your working directory is bob, the relative pathname of the directory hacking would be
../instructor/hacking . In other words, to get from bob to hacking, you first must go up one directory to home
(the parent directory, represented by the two dots), then from home you go down one directory to instructor, and then
down to hacking.
Another shortcut is also available for use in relative pathnames. A single dot (.) can be used as a shorthand notation for your
current working directory.
4. Listing Files You can list the contents of the working directory by using the ls command. For example, if, in the picture
above, your working directory was instructor, then the command ls would yield the results:
hacking cyber spoofing
You can list the files in a different directory by typing ls followed by the absolute or relative pathname of the directory you
are seeking information about. For example, if your working directory was instructor and you entered ls ../bob ,
the result would be:
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acme fleet
5. Changing Your Working Directory To change your working directory to another directory, simply enter the command
cd followed by the directory you wish to change to. For example, if your working directory was instructor, you could
change your working directory to bob by entering:
cd ../bob
When you change your working directory, the command line will update to indicate your new working directory. For
example, if I am the user named midshipman and I change my directory to a sub-directory named work, I will see as my
new prompt:
Working directory changed to ~/work
If you find yourself lost in the file system, you can instantly reset your working directory back to your home directory by
simply typing cd by itself. You may have already noticed that we have changed our working directory at the start of most
security exercises by typing:
cd booksrc
6. The root User Every Linux system has a special user named root. The root user is the great-and-all-powerful system
administrator of the Linux system. The root user can access any file on the system, including the files of individual users. The
root user can read the files of all users, can write over any files, and can delete any files. The root user can load any software
onto the system (e.g., programs). The root user owns the system.
The dream of all hackers is to somehow become the root user. In Linux, the root user has a special prompt, the pound sign
(#). If you walk up to a computer and see this:
that means the root user has logged in and left the computer unattended. That would be bad.
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518
Brief Primer on gdb
Getting started. Assume our C program is named test.c. The program is shown below.
#include<stdio.h>
int main( )
{
int a = 2;
int b = 1000;
char x = '$' ;
char phrase[4] = "Fun" ;
printf( "Yes");
printf("No");
}
To run the debugger on the compiled version of test.c, always start by entering:
gcc –g test.c
gdb –q ./test.exe
list
If your source code is more than 10 lines, you may have to hit enter again, to list the next 10 lines. We see this:
midshipman@EC310-VM:~ $ gcc -g test.c
midshipman@EC310-VM:~ $ gdb -q ./test.exe
Using host libthread_db library "/lib/tls/i686/cmov/libthread_db.so.1".
(gdb) list
1 #include<stdio.h>
2 int main( )
3 {
4 int a = 2;
5 int b = 1000;
6
7 char x = '$' ;
8
9 char phrase[4] = "Fun" ;
10
(gdb)
11 printf( "Yes");
12
13 printf("No");
14 }
(gdb)
The line numbers shown on the left can be very useful for setting breakpoints. For example, if I wanted to run the program
but have it pause right between the two final printf statements, I would enter:
break 12
run
(You have already seen us set a breakpoint at main by entering break main… this is the same idea.)
Looking at Memory Based on our program, we should have the following items stored in memory:
2
1000
'$'
"Fun"
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The strings "Yes" and "No" are also in memory somewhere, but we'll concentrate just on the integers 2 and 1000, the
character '$' and the string "Fun".
If I were to look into memory, I would see this (where all values are hexadecimal):
Looking at this section of memory, it may not be obvious where items are stored. Here is where the integers 2 and 1000, the
character '$' and the string "Fun" are placed:
520
The string "Fun"
Consulting the handy-dandy ASCII table, we see:
Character ASCII hexadecimal value
F 46
u 75
n 6e
So, sure enough, there it is at memory location bffff808. You should also note that the NULL terminator appears as the
character immediately following the 'n' in "Fun" .
Looking at the memory on the bottom of the previous page, try to guess what will be displayed by each of the following
commands. (The answers immediately follow.)
(a) x/xb 0xbffff808

(b) x/xh 0xbffff808
(c) x/xw 0xbffff808
(d) x/xs 0xbffff808
Answers:
(a) 46 (displays a byte)

(b) 7546 (displays two bytes)
(c) 006e7546 (displays four bytes)
(d) "Fun" (displays as a string)
For (b) and (c), note the annoying little-endian.
We can specify the number of units we wish to have printed out by placing a number after the slash. For example, looking at
the memory on the bottom of the previous page, try to guess what will be displayed by each of the following commands. (The
answers immediately follow.)
(a) x/xb 0xbffff808

(b) x/2b 0xbffff808
(c) x/3b 0xbffff808
(d) x/4b 0xbffff808
(e) x/2h 0xbffff808
Answers:
(a) 46 (displays a byte)

(b) 0x46 0x75 (displays two bytes)
(c) 0x46 0x75 0x6e (displays three bytes)
(d) 0x46 0x75 0x6e 0x00 (displays four bytes)
(e) 0x7546 0x006e (notice that each half-word is presented in annoying little-endian)
If we suspect that characters are being stored, we can ask that the display be presented as characters by specifying the c
format. Looking at the memory on the bottom of page 584, try to guess what will be displayed by each of the following
commands. (The answers immediately follow.)
(a) x/c 0xbffff808

(b) x/2c 0xbffff808
(c) x/3c 0xbffff808
(d) x/4c 0xbffff808
Answers:
(a) 70 'F'
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(b) 70 'F' 117 'u'
(c) 70 'F' 117 'u' 110 'n'
(d) 70 'F' 117 'u' 110 'n' 0 '\0
Suppose we thought an integer was stored at address 0xbffff808. We could check this be entering:
x/dw 0xbffff808
If we do this, we see:
0xbffff808: 7238982
Can you guess where on Earth this value 7238982 comes from?
522
Answer:
We saw earlier that entering: x/xw gave us 006e7546. If we convert the hexadecimal value 006e7546 to a decimal
integer, we find its value is 7238982.
The character $
Looking at the bottom of page 584, we see the $ character is stored at location bffff80f.
Looking at the memory on the bottom of page 584, try to guess what will be displayed by each of the following commands.
(The answers immediately follow.)
(a) x/xb 0xbffff80f

(b) x/c 0xbffff80f
(c) x/db 0xbffff80f
(d) x/s 0xbffff80f
Answers:
(a) 0x24
(b) 36 '$'
(c) 36
(d) $�\003
Note that 0x24 equals 3610, and that that the last item is gibberish because a string is not stored in this location.
The integer 1000
So, first, we should convert the decimal value of 1000 to hexadecimal. If we do this, we find it is equal to 0x3e8.
With reference to the bottom of page 584, answer the following questions.
(a) Why is 1000 stored in four bytes if it only needs two bytes?
(b) Presuming this value does take four bytes, and thus is equal to 0x000003e8, why is it not stored with the leading
two zeros at the "top" memory locations?
Answers:
(a) All integers are stored in four bytes, even if fewer are needed.
(b) Little endian, little endian, little endian.
Looking at the memory on the bottom of page 584, try to guess what will be displayed by each of the following commands.
(The answers immediately follow.)
x/xb 0xbffff810
x/xh 0xbffff810
x/d 0xbffff810
x/2c 0xbffff810
Answers:
(a) 0xe8
(b) 0x03e8
(c) 1000
(d) gibberish
523
So… do you think you get it? To see, try this:
Your friend types x/d 0xbffff813 and sees that the result is 512. Explain!
524
File Syntax
In this text we have stepped back on level of abstraction from the actual C program. We think about the program
notetaker.c not from the C programming syntax and mechanics but first from the perspective of the programmer writing
the program. “What do I want the program I write to do?”
Now let’s take a closer look at the code that made this all happen. How this code accomplishes what is intended, including
opening and writing to files is not testable, but it is worthwhile seeing how it works. In the program, we will see that we can
use the C language to open files, read from files, write to files, save files to disk, and even run Linux commands from inside
the program. Here is the code for notetaker.c. The lines are numbered (not part of the program) so that we can refer to
specific lines in the description.
1. #include<stdio.h>
2. #include<stdlib.h>
3.
4. int main( int argc , char *argv[ ] )
5. {
6. FILE *fp; // new kind of pointer--a file pointer
7. char buffer[100];
8.
9. if (argc != 2)
10. {
11. printf("\nPlease provide an input string in double quotes.\n\n");
12. exit(0);
13. }
14.
15. fp = fopen("/tmp/notes","a+"); // open "stuff" to append notes
16. if ( fp == (FILE *)NULL ) // ensure file pointer is not NULL
17. {
18. printf( "\nFailed to open file called \"/tmp/notes\".\n");
19. exit(0);
20. }
21.
22. strcpy(buffer, argv[1]);
23. fprintf(fp,"%s\n",buffer); // prints input string to the file
24.
25. system("chmod g-r /tmp/notes"); // remove read perm. for group
26. system("chmod o-r /tmp/notes"); // remove read perm. for others
27.
28. fclose(fp);
29. }
Let’s look a little closer at how this program is able to open and write to the file /tmp/notes.
To begin, line 2 (#include <stdlib.h>) allows us to use a new function in the program that is part of the “Standard
General Utilities Library.” This is the system command, which will be described later.
On line 6, we are introduced to how the program refers to a file that the program will use. This is done with a “file pointer,”
and we use a file pointer as shown here. Each file that is used in a program must have a unique file pointer.
Lines 15-20 open the file, and make sure that opening the file was successful. When using the fopen function to open the
file, we must give it the name of the file in the first argument, and tell it how we want to open in the second argument. What
are the things we can do to a file once it is open (i.e., what are the options for the second argument?):
“r” Open a file for reading
“w” Create / writing
“a” Append
“b” Open a binary file
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“rb” Open binary file for reading
“+” Open file for read/write
“r+” text update (reading and writing)
“w+” text update, discarding previous content (if any)
“a+” text append, reading, and writing at end
For this program, we chose the “a+” option because we will want to append text at the end of whatever other text is in the file
already. If the file doesn’t already exist, then fopen will create it.
It is always a good idea to check if the file was able to be opened, so that the program doesn’t crash unnecessarily. No more
space on the system or not having permission to save a file in that directory would cause the file not to open.
Lines 22-23 add the new note to the file that has been opened. Line 22 copies the input string (argv[1]) into the buffer
array, and line 23 uses fprintf() to actually write the array into the file. The fprintf function is used exactly like
printf for files, but we must tell it which file to print to using a file pointer. The newline character (\n) is added to the
note we want to write into the file inside the fprintf function, so that each note a user adds is added on the next line.
Lines 25-26 use the system command, which allows us to run Linux commands inside a C program. In this case, we are
using the Linux chmod command to adjust the permissions on the /tmp/notes file. By default, the operating system will
give read permission for a file just created to all users on the system. Since we want only the owner of /tmp/notes to be
able to read the notes users have entered, we must change the permissions. Here, we use chmod to remove the read
permission for both the owner’s group and others on the system.
Finally, in line 28, after we open a file and have completed any operation within the file we use fclose to “close” the file,
which saves the file to the hard drive, allowing it to be used for other things.
526
Performing Base Conversions on the T1-nSpire CAS Calculator
Performing base conversions on your TI-nSpire calculator is relatively straightforward. Becoming proficient at using your
calculator this way will be useful to you throughout the course, especially as we begin to perform complex mathematical
operations in different number bases.
1. Setup
To begin, ensure that your calculator is set up in the “Auto” calculation mode and the “Decimal” base system. Your
calculator is probably already set up this way, but check it just to be sure.
To check these settings use the following key sequence:
 Press [home]
 Press 5: Settings
 Press 2: Document Settings
Your screen should now look similar to this:
Use the navigation pad to move down to the Calculation Mode setting.
This is the navigation pad
If you get to the Calculation Mode setting and it’s not set to “Auto”, then push the navigation pad to the right to
bring up the menu of Calculation Mode options. Use the navigation pad to move up or down to select “Auto”, then
press the enter key.
Repeat this procedure to ensure the Base is set to decimal.

Once the Calculation Mode and Base are properly set, repeatedly press down on the navigation pad until OK is
highlighted and press the enter key.
527
If you see the dialog box below, press the enter key to select OK.
2. Conventions
Now that your calculator is set up properly, the following conventions apply:
 Decimal numbers are typed with no special notation (e.g. 2015)

 Hexadecimal numbers are typed by preceding the number with “0h”. That’s a zero, not letter “O” (e.g. 0h3EA)
 Binary numbers are typed by preceding the number with “0b”. Again, it’s zero, not letter “O” (e.g. 0b100101)
3. The Conversion Operator
Conversion among different bases is handled through the use of the “conversion operator”. It’s a single character that
looks like a right-facing sideways triangle: ► To access this character you need to bring up the symbol palette on your
calculator. You do that by pressing and releasing the ctrl key, followed by pressing and releasing the symbol
palette key. The symbol palette will then pop-up. The keys and the palette are identified in the picture on the next
page.
ctrl key
symbol palette key
conversion operator
symbol palette
528
You select the conversion operator by pressing the enter key. The operator will then show up in your display.
That seems like a lot of steps for a single character, but it’s not too bad. Once you’ve selected the operator for the first
time, the nSpire calculator remembers that position on the symbol palette. Hereafter, to select the conversion operator
you simply press these three keys one after the other:
cntl, symbol palette, enter
For the remainder of this document, I’ll use ► as shorthand for, “press cntl, symbol palette, enter.”
4. Performing Conversions
Now we’re ready to put this to good use. Let’s jump right in using some of the examples from the first lecture. I’ll use
this notation, [enter], to mean, “press the enter key”.
Problem: Express the binary number 0b110110 as a decimal number.

Type: 0b110110 [enter]
Solution: 54
Because the calculator is set up to perform Auto Calculations in the Decimal Number System, we don’t have to use the
conversion operator in this example. Simply typing a number in binary (with the leading “0b”) and pressing enter will
tell the nSpire to output the result in decimal. Note, if you leave off the leading “0b” and type this: 110110
[enter], the nSpire will output 110110. That’s because the calculator assumes you entered: One hundred ten
thousand, one hundred and ten (the decimal number).
Problem: Convert the decimal number 148 to binary.
Type: 148 ► base2 [enter]
Solution: 0b10010100
Here we learn how to tell the nSpire that we want to convert to some other base besides decimal by typing the characters
“base2”. Note, the nSpire will capitalize the “b” in “base2” for you and redisplay your input as: 148►Base2
Problem: Express the hexadecimal number 0x3CB as a decimal number.

Type: 0h3cb [enter]
Solution: 971
Remember, absent any specified base the nSpire defaults all results to decimal. Also, you will usually see hexadecimal
numbers written with the “0x” prefix in this class. Just remember that the nSpire expects to see “0h” as the prefix for
hexadecimal numbers.
Problem: Convert the decimal number 2576 to hexadecimal.

Type: 2576 ► base16 [enter]
Solution: 0hA10
Problem: Convert the hexadecimal number 0x13F to binary.

Type: 0h13f ► base2 [enter]
Solution: 0b100111111
Problem: Convert the binary number 0b110101001 to hexadecimal.

Type: 0b110101001 ► base16 [enter]
Solution: 0h1A9
Simple calculations are great, but real power comes in more complex calculations. That’s because you can write
statements with calculations on the left side of the conversion operator symbol (►). Let’s look at an example: Problem:
529
A section of memory beginning at address 0xD213AC53 contains 438 bytes of data (one byte of data per address line).
The address space looks like this:
Address Data
0xD213AC53 45
0xD213AC54 88
0xD213AC55 3C
0xD213AC56 E2
… …
… …
… …
What hexadecimal memory address comes next after the end of the 438 bytes of data?
Type: 0hd213ac53 + 438 ► base16 [enter]

Solution: 0hD213AE09
For these examples I used “0x” in the problem statement and “0h” in the calculations on purpose. Remember, the nSpire
needs to see the “0h” prefix to represent hexadecimal numbers, but you’ll often see it written in problems as “0x”. If you
make a mistake and type 0xd213ac53 it will be obvious because the nSpire will assume you want to multiply zero (0)
times xd213ac53 and will show you the result equals zero (0).
After a little practice you’ll get the hang of it and find that you can perform very complex calculations in different bases
quite easil
530

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EC310 Notes

Department of Electrical and Computer Engineering

Connection to Cyber Security

Practice Problem 19.1

Practice Problem 19.2

3.3 Federal Communications Commission (FCC)

 Allocates bands of frequencies for specific uses

 AM Radio Station – 10 kHz

4. Signals as a Function of Time and Frequency

Amplitude (Vm) – distance from average to peak (in volts)

2. What part of the electromagnetic spectrum (frequency range) is visible to humans?

4. Calculate the frequency of signals with the following wavelengths:

(b) What is the bandwidth (BW) occupied by each station?

6. Given the sine wave below, answer the following questions:

(a) What is the period of this signal?

(b) What is this signal’s amplitude?

(c) What is the frequency of this signal?

(e) What is the wavelength of this signal?

(f) Sketch this signal in the frequency domain.

𝑣𝑚 (𝑡) = −18𝑠𝑖𝑛(2𝜋44,000𝑡) + 13𝑠𝑖𝑛(2𝜋150 × 103 𝑡 − 6𝜋⁄11) + 7𝑐𝑜𝑠(2𝜋30 × 104 𝑡 + 𝜋⁄21)

 Agilent 33210A 10 MHz Function / Arbitrary Waveform Generator

 Tektronix TDS 1002B Two Channel Digital Storage Oscilloscope

 RadioShack Mini Amplifier/Speaker

Step One: Function generator setup.

□ Locate the "10MHz Function/Arbitrary Waveform Generator" on the lab bench

□ Turn any other lit buttons off.

□ Enter the desired frequency (440) using the key pad.

□ Enter the desired amplitude (10) using the key pad.

Step Two: Oscilloscope Setup

Step Three: Display your sine wave with the oscilloscope.

□ Press Autoset to have the scope automatically calibrate to your signal.

□ Press CH 1 MENU on the oscilloscope and make the following settings.

Question 1: Is the sine wave displayed as a function of time or frequency? Why?

Question 2: What is the mathematical representation of this particular sine wave?

Part 2: Measuring in the time Domain with the Oscilloscope

□ Press the second to top menu box button to select Measure 2.

Part 3: Measuring in the time Domain with the Oscilloscope

You can check your answer using the frequency cursor:

Part 4: Exploring Signal Parameters in Time and Frequency Domains

□ Turn function generator dial to observe the effects.

Question 12: Now, what does the x-axis represent?

Part 5: Exploring a voice signal in Time and Frequency Domains

□ Unplug and stow the banana plug cables.

□ Turn off your oscilloscope.

□ Turn off your function generator.

Security Exercise 19 Answer Sheet

Connection to Cyber Security

 Computer keyboards convert physical input (typing) into an electrical signals.

Practice Problem 20.2

Practice Problem 20.3

vAM (t )  Vc  Vm cos  2 f m t   cos  2 f c t 

Practice Problem 20.5

If a carrier signal vc (t )  20sin  2 5000t  Volts is amplitude modulated by information signal

2. An AM signal is comprised of the following message signal and carrier:

vm(t) = 80 cos (2π5000t) volts

vAM(t) = V1 cos(2π f1 t) + V2 cos(2π f2 t) + V3 cos(2π f3 t) + V4 cos(2π f4 t) + V5 cos(2π f5 t) volts

(b) Find the bandwidth for this particular broadcast.

Introduction to Amplitude Modulated Signals

Step One: Function generator setup.

Step Two: Oscilloscope setup.

□ Turn on the oscilloscope.

Part 2: Amplitude Modulated Signal in the Frequency Domain

□ Push AUTOSET on the Oscilloscope and you will see an AM signal.

Question 3: fc = _____ flsb = _ fusb = _______