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Electronic

r
[educational System:
Heathkit
s.r.u.

Educational
Systems

ELECTRONIC
COMMUNICATIONS

Copyright © 1981
Model EB-6106 First printing
Heath Company
HEATH COMPANY Not Affiliated with DC. Heath Inc.
BENTON HARBOR, MICHIGAN 49022 All Rights Reserved
595-2692 Printed in the United States of America
CONTENTS
Introduction V

Course Objectives VI

UNIT ONE — COMMUNICATIONS FUNDAMENTALS


Introduction 1-3
Unit Objectives 1-5
What is Communication? 1-6
Review of Complex Waves 1-11
Amplitude Modulation 1-19
Angle Modulation 1-33
The Communications System 1-49

UNIT TWO — AMPLITUDE MODULATION


Introduction 2-3
Unit Objectives 2-5
AM Circuits 2-7
Suppressed Carrier AM 2-25
Single Sideband 2-39

UNIT THREE — AM RECEIVERS


Introduction 3-3
Unit Objectives 3-5
AM Detectors 3-6
Tuned RF Receiver 3-16
Superheterodyne Receivers 3-19
Analysis of a Communications Receiver 3-40
IV

UNIT FOUR — ANGLE MODULATION


Introduction 4-3
Unit Objectives 4-5
Angle Modulation Transmitters 4-6
Demodulators 4-15
Receivers 4-27
FM Transceivers 4-36

UNIT FIVE — PULSE MODULATION


Introduction 5-3
Unit Objectives 5-5
Analog Pulse Modulation 5-6
Digital Pulse Modulation 5-23
Multiplexing 5-33

UNIT SIX — ANTENNAS


Introduction 6-3
Unit Objectives 6-5
Transmission Lines 6-7
Antennas 6-27
Radio Propagation 6-53

UNIT SEVEN — COMMUNICATIONS SYSTEMS


Introduction 7-3
Unit Objectives 7-5
FM Stereo Broadcasting 7-6
Television Broadcasting 7-17
Data Communications 7-31
INTRODUCTION
Throughout time, the one thing that has distinguished humans from
other animals is their ability to exchange ideas and other information.
That is, humans can communicate. It is this capability that has played
perhaps the biggest part in the development of civilization. In fact, as our
civilization continues to grow, so must our communications capacity.
The two are intertwined.

Itcomes as no surprise then that one of the first applications of the "new"
field of electricity was to extend our communications range. This was
accomplished through the use of wires and telegraphy. Messages were
sent by turning electrical currents on and off in accordance with a tele-
graph code. This system gradually evolved into the telephone system
where the electrical currents are varied at an audio rate. Thus, the spoken
word can be conveyed between two distant points. However, the tele-
phone system still required wires, which limited its capabilities. Thus,
the next development was to be "wireless" communications in the form
of radio. This greatly extended the communications range and, for the
first time, included ships at sea and remote areas of the world. Actually, it

allowed instantaneous and virtually world-wide communications for the


first time.

"Wireless" or radio communications represented a significant advance-


ment. And, as a result, a whole branch of electronics was established.
This course is intended to teach you the fundamental and advanced

techniques of that branch: electronic communications.

You can get a good idea of the scope of this program by reviewing the
"Course Outline" that follows. As you can see, this is a very comprehen-
sive course with a substantial amount of material. To enliven and enrich
your learning experiences, seven experiments are included. These will
reinforce the theory of communications with practical applications.
VI

COURSE OBJECTIVES
When you have completed this course, you should be able to:

1. Discuss and explain the characteristics, advantages, and disadvan-


tages of amplitude, frequency, and pulse modulation.

2. Analyze the composition of complex waveforms.

3. Describe the operation of AM and FM transmitters and receivers.


4. Construct basic communications circuits.

5. Explain the differences between frequency and phase modulation.

6. Discuss both frequency and time division multiplexing as methods


of increasing communications channel capacity.

7. State the characteristics and uses of transmission lines.

8. Explain electromagnetic radiation and propagation and list the


various types of propagation.

9. Discuss the basic types of antennas including dipoles, verticals,


and directional arrays.

10. Discuss basic communications systems including television and


FM stereo broadcasting, and data communications.
Unit 1

COMMUNICATIONS
FUNDAMENTALS
1-2 UNIT ONE
Communications 1-3

INTRODUCTION
This unit discusses the fundamentals of communications. It examines, in
communications systems, information, transmitters, receivers,
detail,
more importantly, modulation. A thorough understanding of
noise, and,
modulation is of the utmost importance, because without it, radio com-
munications would be impossible. For this reason, a large portion of this
unit is devoted to the discussion of the three different types of modula-
tion.

The "Unit Objectives" on the next page state, in a concise manner, the
goals of this unit. Review this list now and be sure you can satisfactorily
complete all the objectives before taking the unit exam.
1-4 UNIT ONE
Communications 1-5

UNIT OBJECTIVES
When you have completed this unit, you should be able to:

1. Define: communication, information, modulation, and carrier.

2. List at least two reasons why audio information can't be transmit-


ted at its original frequency.

3. List the characteristics of a sine wave that can be varied.

4. Name the two basic types of modulation.

5. State the difference between the time and frequency domains.

6. Identify the frequency composition of complex waves including


square and sawtooth waves.

7. Measure the percent of modulation when given an AM waveform.


8. Find the frequencies of AM sidebands when given the carrier and
modulating frequencies.

9. Find the bandwidth of an AM signal.

10. Define angle modulation.

11. List the characteristics of frequency modulation.

12. State the difference between phase and frequency modulation.

13. Find modulation index, sideband distribution, and bandwidth of


an angle modulated signal.

14. List the parts of a basic communications system.

15. Define transducer and noise.

16. Name the four types of noise.

17. List the major radio frequency bands.

18. State the function of a transmitter and receiver.


1-6 UNIT ONE

WHAT IS COMMUNICATION?
Communication is defined as "a process by which information is ex-

changed." In electronics, it is the transmission and reception of informa-


tion. Likewise, information is defined as "the communication of know-
ledge or intelligence." For the purposes of this course, it is defined as any
electrical signal representing data. Thus, the purpose of any communica-
tions system is to convey or transfer information from one point to
another.
Communications 1-7

Information Transfer

Our most fundamental methods of transferring information are speech


and, as in this case, writing. Communication of the written word de-
veloped from hand-carried letters and newspapers to the mail system,
telegraph, and now electronic mail. Spoken communications evolved
from face-to-face contact into telephone and radio communication. All of
these steps were taken in an effort to increase the communications dis-
tance and speed.

The most significant advance in increasing communications range was


radio. Basically, the audio or sound waves are converted to an electrical
signal then into audio waves and transmitted to a distant receiving
station. However, if the audio signal is transmitted at its original fre-
quency, a number of problems present themselves. First to be efficient,
the transmitting antenna must be at least 1/4 to 1/2 wavelength long. This
means that for a 3,000 Hz signal, the antenna would have to be at least 25
kilometers long! Second, even if the antenna problem was solved, only
one station could transmit at a time. This is because all stations would be
operating on the same "audio" frequencies. And third, transmission
systems at these frequencies are very inefficient.

All of these problems can be solved by using a higher frequency signal as


a "carrier" for the audio information. In essence, the speech signal is

transferred to a much higher frequency for transmission, then converted


back to audio frequencies by the receiver. The former is called modula-
tion, while the latter is demodulation.
1-8 UNIT ONE

Modulation

In the process of modulation, some characteristic of a high frequency sine


wave is varied accordance with the information or modulating signal.
in
This signal may be an audio waveform, a digital pulse train, a television
picture, or any other form of information. The important consideration is
that it is transferred to a higher frequency for efficient transmission.

As mentioned before, the modulated high frequency sine wave is called


the carrier. The mathematical expression for an unmodulated sine wave
or carrier is:

e = A sin (tut + </>)

Where

e = instantaneous value of the wave


(voltage or current)

A = maximum amplitude

(o = angular velocity [2irf)

t = time

(f>
= phase angle

This equation shows that there are three characteristics of the wave that
can be varied or modulated. These are: amplitude (A), angular velocity or
frequency (&>), and phase angle (<£).

Types of Modulation

Since, three characteristics of the sine can be varied, itwave carrier


follows that there are three types of modulation. These are amplitude
modulation (AM), frequency modulation (FM), and phase modulation
(PM). However, in practice, it is very difficult to distinguish between
phase and frequency modulation. Therefore, these two types of modula-
tion are grouped together under the title of Angle modulation. Thus,
there are two basic types of modulation: Amplitude and Angle. The next
section will discuss both of these in detail.

.
Communications 1-9

Self-Review Questions

1. Define communication. —

2. Define information.

3. List at least two reasons why audio information isn't transmitted at

its original frequency. _

4. What is modulation?

5. What is a carrier?

6. What three characteristics of a sine wave can be varied or mod-


ulated?

7. List the two basic types of modulation.


.

1-10 UNIT ONE

Self-Review Answers

1 Communication is the transmission and reception of information.

2. Information is any electrical signal representing data.

3. Audio information isn't transmitted at its original frequency be-


cause the antennas would have to be much too long, there would be
severe interference between stations, and the transmitters would be
very inefficient.

4. Modulation is the process of varying some characteristic of a high

frequency sine wave in accordance with an information signal.

5. A carrier is a high frequency sine wave that is modulated so it can


"carry" the information signal.

6. The three characteristics of a sine wave that can be varied are:

1. Amplitude
2. Angular velocity or frequency
3. Phase angle.

7. The two basic types of modulation are amplitude and angle.


Communications 1-11

REVIEW OF COMPLEX WAVES


In the first section, the need for modulation and its basic principles were
established. Now, before continuing the study of modulation, a quick
review of complex waves and frequency domain analysis is needed.

Frequency domain analysis stems from the any periodic


fact that
waveform is made up of sine waves. A periodic wave is one that has the
same waveshape from one cycle to the next. All the waveforms shown in
Figure 1-1 are periodic waves if they have the same shape on each and
every cycle. Any waveform of this type can be formed by the superimpos-
ing of a number of waves which have certain amplitude, phase, and
frequency characteristics. Similarly, regardless of how a particular
periodic waveform is formed, it can be proven that it consists of sine
waves in a specific phase, frequency, and amplitude relationship.

Figure 1-1
A sine wave (A), square wave (B), and sawtooth wave (C).
1-12 UNIT ONE

Sine Wave
Since the sine wave is the basis of all waveforms, let's examine it in more
detail. Figure 1-lA is a time domain representation of the sine wave. It
shows the wave as it would appear on an oscilloscope display, which is a
graph of time versus amplitude.

Figure 1-2 shows the wave as it would appear on a spectrum analyzer


display. This device analyzes the wave's frequency domain. As you can
see, it gives us a graph of frequency versus amplitude. It is actually a bar

graph showing all the frequency components contained within a wave.


Since this sine wave has a period of 1 ms, its frequency is 1 kHz. There-
fore, a bar representing the wave's amplitude appears at 1 kHz. There are
no other frequency components, since this is a pure sine wave.

H 1 1- -• » • 1 1 • 1 1 1 1-
5 6 7 8 9 10 11 12 13 14

FREQUENCY (KHZ)

Figure 1-2
A spectrum analyzer display of the frequency domain.
Communications 1-13

Square Wave

Figure 1-3A shows one cycle ofa 1 kHz square wave. Since this is a
periodic wave, this cycle is repeated over and over again. This waveform
is made up of a large number of sine waves. More specifically, a perfect
square wave is composed of a fundamental frequency and an infinite
number of odd harmonics.

The result of adding the first harmonic or fundamental and the third
harmonic is shown The addition of the fifth harmonic is
in Figure 1-3B.
shown in Figure 1-3C. Note that the resultant waveform is already ap-
proaching a true square wave. In fact, as more odd-order harmonics are
added, the closer the waveform gets to becoming a "perfect" square wave.

Figure 1-3
The composition of a square wave.
1-14 UNIT ONE

Itshould now be a simple matter to plot the frequency domain of a square


wave since we know that it is composed of odd-order harmonics. Figure
1-4 shows the results. It illustrates the correct amplitude for all the
harmonics out to 13 kHz. For a perfect square wave, the chart would
extend to infinity.

•OO
5 6 7 8 10 11 12 13 14
FREQUENCY (KHZ)

Figure 1-4
The frequency domain analysis of a square wave.
Communications 1-15

Sawtooth Wave

Figure 1-5 illustrates that a different type of waveformdeveloped when is

the harmonic content is changed. Here both even and odd-order har-
monics are added. The result is a sawtooth wave. Figure 1-5C shows that,
with just the first through fourth harmonics present, the resultant
waveform comes very close to the perfect sawtooth.

PERFECT
SAWTOOTH
V

Figure 1-5
The superimposing of sine waves to form a sawtooth wave.
1-16 UNIT ONE

The frequency domain analysis of a sawtooth is shown in Figure 1-6. Note


that all harmonics are present, both even and odd-ordered.

The sawtooth and square wave are but two examples of the unlimited
variety of complex waveforms. The important concept is that sine waves
can be combined in an infinite number of ways to produce an infinite
variety of waveforms. By the same token, any waveform can be broken
down into its sine wave components.

>oo
1 2 3 4 5 6 7 8 9 10 11 12 13 14

FREQUENCY (KHZ) ^

Figure 1-6
The frequency spectrum of a sawtooth wave.
Communications 1-17

Self-Review Questions

8. What is the difference between the time and frequency domains?

9. Any complex waveform is composed of

10. A square wave is composed of

11. A sawtooth wave is composed of


1-18 UNIT ONE

Self-Review Answers

8. The time domain shows a waveform's time-versus-amplitude dis-


an oscilloscope does. The frequency domain shows a
play, just as
waveform's frequency-versus-amplitude display, which is similar
to a spectrum analyzer.

9. Any complex waveform is composed of sine waves.

10. A square wave is composed of a fundamental and an infinite


number of odd-order harmonics.

11. A sawtooth wave is composed of a fundamental and an infinite


number of both even and odd-order harmonics.
Communications 1-19

AMPLITUDE MODULATION
As stated earlier, with amplitude modulation, the carrier's amplitude is

varied in accordance with the modulating signal. There are several cir-

accomplish this and they will be examined in the next unit.


cuits that
Right now,we wish to limit our discussion to the characteristics of the
modulated waveform itself.

The AM Waveform
Figure 1-7 shows a very simple AM circuit. Here, a radio frequency
carrier is applied at "A" and The
the modulating audio tone at "B".
circuit consists of a nonlinear device such as a diode or transistor. The
two signals "mix" in this circuit and produce the AM waveform shown at
"C". Notice that both the negative and positive peaks of the output
waveform correspond exactly to the modulating tone's waveform.

,A

CARRIER

NONLINEAR c
DEVICE

>
B

MODULATING
TONE

Figure 1-7
The basic method of obtaining amplitude modulation.
1-20 UNIT ONE

The amplitude and frequency of the modulating tone determines the


shape of the output waveform or the modulation envelope. For example,
Figure 1-8A shows a high amplitude audio signal. The resultant mod-
ulated waveform is shown in Figure 1-8B. On the other hand, Figure 1-8C
shows a low amplitude, higher frequency audio signal. The modulated
waveform is in Figure 1-8D.

MODULATING
SIGNAL

Figure 1-8
Examples of how the modulated waveform varies with the
modulating signal.

^
J 1-21
Communications

Percent of Modulation

The waveforms of Figures 1-8 B and D are said to have different degrees of
modulation. The degree of modulation is normally expressed as a percen-
tage from to 100. However, it is also known as the modulation factor
which varies from to 1. An unmodulated carrier like that shown in
Figure 1-9A has 0% modulation. For comparison purposes, let's assume
that the carrier has a peak-to-peak amplitude of 40 volts as shown.

Figure 1-9B shows the same carrier modulated to 100%. Here, the
amplitude of the modulated waveform falls to zero volts for an instant
during each cycle of the modulating wave. Also, the amplitude increases
to 80 volt peak-to-peak once during each cycle of the modulating wave.
The average peak-to-peak amplitude is still 40 volts.

In Figure 1-9C the carrier shown'modulated to 50%. The peak-to-peak


is

amplitude varies from 60 volts to 20 volts. However, the average peak-


to-peak amplitude is still 40 volts.

The equation for determining the percent of modulation is

Percent of modulation = x 100

For example, in Figure 1-9C,

%= x 100
'-'max ' *-*min

%= 60V - 20 V
X 10°
60V + 20V
B 10 °

%= 4?-^x 100
80 V

%= 0.5 x 100 = 50

C 50

-MAX
60V

Figure 1-9
Measuring the percent of modulation.
"

1-22 UNIT ONE

Generally, it is desirable to keep the percent of modulation high. For a


given transmitter power, a high percent of modulation will produce a
stronger audio tone in the receiver. The reason
can be visualized
for this
from Figure 1-10. Since the AM
receiver recovers just the modulation
envelope of the transmitted wave, it is easy to see that the higher mod-
ulated waveform in "B" will produce a louder signal than that in "A".

MODULATION
ENVELOPE

RECOVERED
AUDIO

RECOVERED
AUDIO

Figure 1-10
The relative amplitude of the recovered audio depends on
the modulation percentage.
Communications 1-23

While it is a good idea to keep the percent of modulation high, overmodu-


lation must be avoided. Overmodulation is shown in Figure 1-1 1C. It
occurs when the amplitude of the modulating signal is too high com-
pared to the unmodulated carrier. Obviously, the minimum amplitude of
the carrier is zero volts. It cannot drop below this level regardless of how
high the modulating signal is. If the modulating signal is too high, it will
cause the carrier to cut off for a portion of each cycle. As a result, part of
the envelope will be distorted. That is, the envelope will not be an
accurate representation of the modulating wave.

Figure 1-11 A shows the high modulating waveform. The unmodulated


carrier is shown in Figure 1-llB. The modulated waveform, shown in
Figure 1-1 1C, cuts off for a portion of each cycle. At the receiver, the
envelope is detected, and since it is distorted, the detected waveform is

also distorted. The detected envelope is shown in Figure 1-1 ID.

A
MODULATING
SIGNAL

B
UNMODULATED
CARRIER

OVERMODULATED
WAVEFORM

DETECTED
MODULATING
SIGNAL

PART OF
NEGATIVE
HALF-CYCLE
IS CLI PPED

Figure 1-11
Overmodulation causes severe distortion in the received
signal.
1-24 UNIT ONE

Sidebands

In the first section of this unit, you learned that any complex waveform
can be broken down into its component sine waves. The same is true for
an amplitude modulated waveform such as that shown in Figure 1-12A.
This wave is a 1 MHz carrier modulated by a 10 kHz sine wave. At first
glance, you might say that the wave is composed of a 1 MHz sine wave
and a 10 kHz sine wave. However, we apply the waveform to both a 10
if

kHz bandpass filter and a 1 MHz bandpass filter as shown in Figure


1-12B, we would see an output only at the 1 MHz filter. This shows that
there is no 10 kHz signal present in the modulated wave, the modulation
envelope only represents the audio signal. But we do know that some
other sine wave components must exist in this complex wave.

If we had a tunable bandpass filter, or a spectrum analyzer, we could


search the spectrum and determine what other frequencies are contained
within the modulated signal. Doing this, we would find a signal at 1.01
MHz and another signal at 0.99 MHz. These two signals are called
sidebands.

They can be extracted from the modulated waveform by using sharply


tuned filters as shown in Figure 1-1 2C.

The higher frequency (1.01 MHz) is called the upper sideband. Its fre-
quency always equal to the carrier frequency plus the modulating
is

frequency. That is:

Upper sideband = fc + fm

Where:

fc = carrier frequency
fm = modulating frequency

In our example:

Upper sideband = 1 MHz + 10 kHz


= 1.01 MHz
The lower frequency (0.99 MHz) is called the lower sideband. Its fre-
quency is equal to the carrier frequency minus the modulating frequency.
In this case, it is:

Lower sideband = fc - fM

= 1 MHz - 10 kHz
= 0.99 MHz
<

Communications 1-25

10kHz

1MHz

10kHz
BAND PASS
FILTER

B I

1MHz
BAND PASS
FILTER

10kHz
BAND PASS
FILTER

O—
1MHz
BAND PASS
FILTER

l.OlMHz
BAND PASS
FILTER

0. 99MHz

BAND PASS
FILTER

Figure 1-12
Frequency domain analysis of an AM wave.
1-26 UNIT ONE

If we were to carefully observe the filter outputs of Figure 1-12C, we


would find that the sideband and carrier amplitudes do not vary. In fact,
we would find that the carrier amplitude never varies whether it is
modulated or not. You might ask, if the individual frequency compo-
nents do not change, how can the modulated waveform change to follow
the modulating signal? Figure 1-13 gives a detailed look at exactly what
happens. It shows that the constant amplitude sidebands are at different
frequencies and, therefore, are in phase and out of phase with one another
at various times. For example, at point A, they are exactly in phase with
each other. At this time, the constant amplitude carrier is also in phase.
The result is a high amplitude peak in the modulated waveform. Now
observe the wave relationships at point B. Once again, the sidebands are
in phase with each other. However, they are 180° out of phase with the
carrier wave. The result is a low point, or trough, in the modulated
waveform.

From this analysis, you can see that the shape of the modulation envelope
isdependent on the sidebands. And, the sidebands are in turn dependent
on the modulating signal. That is, the frequency of the sidebands deter-
mines their phase relationship and, therefore, the peaks and troughs, or
frequency, of the modulation envelope. The sideband amplitude will also
determine the envelope's amplitude, or percent of modulation. This is
because they will be either adding to or subtracting from the constant
amplitude carrier. This illustrates an important fact about amplitude
modulation: The modulating intelligence or information is contained
only in the sidebands.
Communications 1-27

SIDEBANDS UPPER SIDEBAND

LOWER SIDEBAND

CARRIER

MODULATED
WAVEFORM

Figure 1-13
The phase relationships of an AM wave.
1-28 UNIT ONE

The frequency spectrum charts of Figure 1-14 will further illustrate this
point. Since these are voltage diagrams, the sideband amplitudes shown
will add or subtract directly from the carrier to produce the modulated
envelope. For example, Figure 1-14A shows the sideband amplitudes as
being exactly one half that of the carrier. This is the condition for 100%
modulation, because when all signals are in phase, the waveform
amplitude will be twice the carrier and when the sidebands are out of
phase with the carrier, the waveform amplitude will be zero.

Figure 1-14B shows a 50% modulated signal. Note that the carrier
amplitude remains the same while the sideband amplitudes have de-
creased. The frequency of the sidebands has also changed. Since the
sidebands are further from the carrier, the modulating frequency has
increased. This is shown in the modulated waveform to the right.

The result of a square wave modulating signal is shown in Figure 1-14C.


In this case, since the modulating signal is actually the fundamental and
all odd-order harmonics, there is a sideband for each sine wave in the
modulating signal.
1

Communications 1-29

CARRIER
/
SIDEBANDS

FREQUENCY MODULATED
WAVEFORM

(9

FREQUENCY

I
J i_
H
1
FREQUENCY

Figure 1-14
The sideband spectrum of AM waves.
1-30 UNIT ONE

Bandwidth

It is readily apparent from the frequency spectrum charts shown in Figure


1-14 that, with amplitude modulation, the transmitted signal is actually a
band of frequencies rather than just the carrier. The carrier contains no
information. If we transmitted or received just the carrier, no information
would be conveyed. In AM systems, both the carrier and the sidebands
must be transmitted and received.

The bandwidth of an AM signal extends from the lowest sideband fre-


quency to the highest sideband frequency. Therefore, the bandwidth is

always twice the highest modulating frequency. Thus, if the highest


modulating frequency is 15 kHz, then the bandwidth will be 30 kHz. In
the case of a complex modulating wave, such as a square wave, the
bandwidth is twice the highest harmonic contained in the wave. How-
ever, each AM transmitter has bandwidth limitations above which it
cannot go. In this case, the transmitter itself would limit the maximum
bandwidth.
Communications 1-31

Self-Review Questions

12. Find the percent of modulation for the waveforms shown in Figure
1-15.

A.
B.
C.

13. List the three frequency components of an AM wave.

14. The intelligence in an AM signal is contained in the

15. What is the bandwidth of an AM signal when the highest modulat-


ing frequency is 6 kHz?

Figure 1-15
What is the percent of modulation for these waveforms.
1-32 UNIT ONE

Self-Review Answers

12. The percent of modulation of the waveforms shown in Figure 1-15


is

- g-
A. % - I-
&max + £min
x 100

- 40
% = 120 x 100
120 + 40

80
% =tt^-X 100
160

% =0.5 x 100 = 50%

160 - 20
B- % =—777^ 7^-x 100
160 + 20

% =T7777 X 100
180

% =0.778 x 100 = 77.8%

C. Overmodulation

13. The three frequency components of an AM wave are the carrier,


upper sideband, and lower sideband.

14. The intelligence in an AM signal is contained in the sidebands.

15. The bandwidth is twice the highest modulating frequency or 12


kHz.
Communications 1-33

ANGLE MODULATION
There are three characteristics of a sine wave that can be modulated:
amplitude, angular velocity or frequency, and phase angle. The last two
are similar in that the angle, velocity or phase, is being modulated. They
are also similar in that the modulation of one indirectly causes the other
to vary. Therefore, they are grouped together under the heading of Angle
Modulation.
1-34 UNIT ONE

Frequency Modulation

Figure 1-16 shows a frequency modulated or FM waveform. The informa-


tion or modulating waveform is shown in Figure 1-16A, while the un-
modulated carrier is shown in Figure 1-16B. With FM, the modulating
signal changes the frequency of the carrier rather than its amplitude. The
resulting frequency modulated waveform is shown in Figure 1-16C.

Figure 1-16
The modulating unmodulated carrier
signal (A), the (B),
and the frequency modulated waveform (C).
.

Communications 1-35

At time T the modulated waveform is at its center frequency. As the


,

modulating signal swings positive, the frequency of the carrier is in-


creased. The carrier reaches its maximum frequency when the modulat-
ing signal reaches its maximum amplitude, at time Tt .

At time T2 the modulating signal returns to and the carrier returns to its
,

center frequency. After T2 the modulating signal swings negative. This


,

forces the carrier below its center frequency. The carrier again returns to
its center frequency when the modulating signal returns to volts at time
T4 Between times T 4 and T 8 the modulating signal repeats its cycle. As a
. ,

result, the carrier again shifted in frequency. It swings first above then
below its center frequency. Notice that it returns to its center frequency
each time the modulating signal passes through volts.

The carrier changes equally above and below its center frequency. The
amount of frequency change is called the frequency deviation. For exam-
ple, let's assume that a carrier continuously swings from 100 MHz, down
to 99.9 MHz, back to 100 MHz, up to 100.1 MHz, and back to 100 MHz.
The frequency deviation is ±0.1 MHz or ± 100 kHz.

The rate of determined by the frequency of the


frequency deviation is

modulating signal. For example, if the modulating signal is a 1 kHz audio


tone, the carrier will swing above and below its center frequency 1000
times each second. A 10 kHz audio tone will still cause the carrier to
deviate ±100 kHz; but this time at the rate of 10,000 times each second.
Thus, the frequency of the modulating signal determines the rate of
frequency deviation but not the amount of deviation.

The amount that the carrier deviates from its center frequency is deter-
mined by the amplitude of the modulating signal. A high amplitude
audio tone may cause a deviation of ± 100 kHz. A lower amplitude tone of
the same frequency may cause a deviation of only ± 50 kHz.

Thus, the frequency-modulated waveform has the following characteris-


tics:

1. It is constant in amplitude but varies in frequency.

2 The rate of carrier deviation is the same as the frequency of the


modulating signal.

3. The amount of carrier deviation is directly proportional to the


amplitude of the modulating signal.
1-36 UNIT ONE

Phase Modulation

In phase modulation (PM), the phase angle is


carrier's instantaneous
made to vary in proportion to themodulating signal's amplitude. How-
ever, an instantaneous phase change cannot occur without a correspond-
ing frequency change. Therefore, phase modulation also produces carrier
frequency deviation. In fact, frequency modulation cannot occur without
a certain amount of phase variation. Thus, any variation in frequency or
phase produces both FM and PM; the two must occur simultaneously.

A comparison of phase and frequency modulation is shown in Figure


1-17. Careful analysis of thetwo modulated waveforms shows them to be
essentially identical. The difference is that, with phase modulation, the

carrier frequency deviation is proportional to the modulation signal's


phase change as well as its amplitude. For this reason, in phase modula-
tion, maximum deviation occurs when the modulating signal crosses the
zero axis. In other words, the greatest phase change occurs when the
modulating wave changes polarity from to - and from - to +.
-I-

This is shown in Figure 1-17 when, at time T2 , the modulating wave


crosses the zero axis. The result is that the PM wave is deviated to its

lowest frequency. At time T4 when


, the modulating signal changes from
— to +, the PM wave is deviated to its maximum frequency. Thus, the
modulating signal, while phase modulating the carrier, also indirectly
frequency modulates it. For this reason, phase modulation is sometimes

referred to as indirect FM.

As stated earlier, with phase modulation the amount of carrier deviation


is proportional to the modulating signal's phase change. This is why
maximum deviation occurs when the modulating wave crosses the zero
axis.However, consider what happens when the modulating frequency
increases. For example, what happens if the modulating frequency in-
creases from 100 Hz to 1000 Hz? At 100 Hz, the signal's phase changes
100 Hz x 360° or 36,000 degrees per second. At 1000 Hz, the phase
change will increase by a factor of 10 to 360,000 degrees per second.
Since the carrier's deviation is proportional to phase change, its devia-
tion will increase as the modulating frequency increases.

Thus, the primary difference between PM and FM is:

1. With PM, the carrier frequency deviation is proportional to


both the modulation frequency and amplitude.

2. With FM, the frequency deviation is proportional to the mod-


ulation amplitude regardless of the signal's frequency.
Communications 1-37
J

MODULATING
SIGNAL

CARRIER

FREQUENCY
MODULATION

PHASE
MODULATION

Figure 1-17
The differences between phase and frequency modulation.
1-38 UNIT ONE

LOW-PASS
NETWORK
AUDIO
INPUT X PHASE
MODULATOR
FREQUENCY
MODULATION
I

HIGH-PASS
NETWORK
AUDIO
INPUT
1~
1/ ^ FREQUENCY
MODULATOR
PHASE
MODULATION

B
Figure 1-18
Conversior between
i PM and FM.

Actually, at a single modulating frequency with the same deviation, FM


and PM are indistinguishable. The only difference is that as the modulat-
ing frequency increases, the PM waveform's deviation increases. With
this in mind, we can design a simple RC network to convert from PM to
FM or vice versa.

Figure 1-18A shows this concept in practice. Here, a low pass network
attenuates the modulating signal as its frequency increases. The resulting
output is then applied to a phase modulator. (The actual modulator
circuits are discussed in a later unit). The output is an FM wave. This is
because, as the modulating frequency increases, the low pass network
correspondingly reduces its amplitude. The end result is that, as the
deviation increases, due to an increase in the modulating frequency, it is

also decreased due modulating signal amplitude. The two


to the lower
changes offset one another. Therefore, the output deviation remains
constant regardless of modulating frequency. Thus the output is actually ,

an FM signal.

The reverse is also possible. Figure 1-18B shows


network a high pass
used to obtain phase modulation from a frequency modulator. The high
pass filter attenuates the low frequencies and allows the high frequencies
to pass. Therefore, the output wave's deviation will increase as the
modulating frequency increases. The output, thus, has the characteristics
of phase modulation.

It can be concluded from the foregoing that the actual differences be-
tween FM and PM are moot. Therefore, we shall disregard them for the
rest of this discussion.

Communications 1-39

Modulation Index
In of modulation is measured as a percentage from 0% to
AM, the degree
100% or as a modulation factor from to 1. In angle modulation, the
degree of modulation is measured by the modulation index. The equation
for modulation index is:

m =-?u

Where

id = The frequency deviation

fm = The modulating frequency

While the modulation factor in AM is limited to a decimal between and


1, it must be emphasized that the modulation index in angle modulation

can reach quite high numerical values. For example, the maximum devia-
tion in FM broadcasting is 75 kHz. If a 1 kHz audio signal causes full
deviation, the modulation index is:

75 kHz
m= TkHz-
= 75

Another measure of angle modulation is the deviation ratio. This is the


ratio of the maximum deviation to the maximum audio frequency, thus, it
isa total system measurement rather than the instantaneous measure-
ment of modulation index. Using the FM broadcast system as an example,
maximum deviation is 75 kHz and the maximum audio frequency is 15
kHz. The deviation ratio is:

f„
deviation ratio =
f
Lm max

75 kHz
15 kHz
= 5
1-40 UNIT ONE

Sidebands

One of the distinct differences between amplitude and angle modulation


is the number of sidebands. Of course, AM has only two, the upper and
lower sidebands. In angle modulation, the number of sidebands is
theoretically infinite. This is because the numerous frequency shifts
produced by the modulating signal cause the generation of many addi-
tional frequencies. Fortunately, many higher order sidebands contain an
insignificant amount of energy and can, therefore, be disregarded.

In AM, the sidebands add to or subtract from the constant amplitude


carrier which modulation envelope. However, in angle
results in the
modulation, the waveform remains at a constant amplitude regardless.
This means that, as sideband number, amplitude, or distribution change,
the carrier must also change to keep the resultant waveform's amplitude
constant. This interelationship between carrier and sidebands is orches-
trated by the modulation index. That is, the modulation index determines
the number of significant sidebands, their amplitude, and the carrier's
amplitude.
Communications 1-41

A graph of modulation index versus carrier and sideband amplitude is


shown in Figure 1-19. You can see that, with a modulation index of zero,
the carrier amplitude is one and there are no sidebands. As the modula-
tion index increases, more sidebands are added, their amplitude in-
creases, and the carrier decreases. This happens until, at an index of 2.4,
the carrier disappears entirely. This is known as the first carrier "null",
since it also occurs at still higher modulation indices. At an index of 3.1
the carrier's amplitude is -0.3. This indicates that it is 180° out of phase
with the components above the zero axis.

VI

IS

II <w.

• 7
&*
Of

IS
_
1.4 •>

13 4jg
12
« j!
*>
I 1
i
&* g T* o
swrt^
Tth" SIDtBM D STH sipe»*»j
I
ITH SIDEBM10
-II

-12

-I J

-14
IS Li 2 3 1 3 5

MODULATION INDEX

Figure 1-19
Graph of sideband and carrier amplitude in an
angle modulated signal.
1-42 UNIT ONE

SIDEBANDS
MODULATION CARRIER
INDEX 1st
2nd 3rd 4th 5th 6th 7th 8th 9th 10th 11th 12th 13th 14th 15th 16th
PAIR

0.00 1.00
0.25 0.98 0.12
0.5 0.94 0.24 0.03
1.0 0.77 0.44 0.11 0.02
1.5 0.51 0.56 0.23 0.06 0.01
2.0 0.22 0.58 0.35 0.13 0.03
2.5 -0.05 0.50 0.45 0.22 0.07 0.02
3.0 -0.26 0.34 0.49 0.31 0.13 0.04 0.01
4.0 -0.40 -0.07 0.36 0.43 0.28 0.13 0.05 0.02
5.0 -0.18 -0.33 0.05 0.36 0.39 0.26 0.13 0.05 0.02
6.0 0.15 -0.28 -0.24 0.11 0.36 0.36 0.25 0.13 0.06 0.02
7.0 0.30 0.00 -0.30 -0.17 0.16 0.35 0.34 0.23 0.13 0.06 0.02
8.0 0.17 0.23 -0.11 -0.29 -0.10 0.19 0.34 0.32 0.22 0.13 0.06 0.03
9.0 -0.09 0.24 0.14 -0.18 -0.27 -0.06 0.20 0.33 0.30 0.21 0.12 0.06 0.03 0.01
10.0 -0.25 0.04 0.25 0.06 -0.22 -0.23 -0.01 0.22 0.31 0.29 0.20 0.12 0.06 0.03 0.01
12.0 -0.05 -0.22 -0.08 0.20 0.18 -0.07 -0.24 -0.17 0.05 0.23 0.30 0.27 0.20 0.12 0.07 0.03 0.01
15.0 -0.01 0.21 0.04 0.19 -0.12 0.13 0.21 0.03 -0.17 -0.22 -0.09 0.10 0.24 0.28 0.25 0.18 0.12

Figure 1-20
Table of sideband and carrier distribution for several
modulation indices.

Figure 1-20 is a table of sideband and carrier amplitudes at various

modulation indices. Although the sidebands theoretically stretch out to


infinity, any sidebands with an amplitude less than 1% of the original
carrier are insignificant and, therefore, left out. As an example, suppose
the modulation index is 0.5. From the table in Figure 1-20, a modulation
index of 0.5 means that carrier amplitude is 0.94, the first sidebands have
an amplitude of 0.24, and the second, and last significant sidebands, have
an amplitude of 0.03. This sideband distribution is shown in Figure 1-21
along with several others.

100%-q

1 1 1 1

- 50%1
3 1 1 3

12_t ?6
li ill
0.5 1.0 1.5 2.0 3.0
MODULATION INDEX

Figure 1-21
Sideband distribution for angle modulation.
Communications 1-43

Bandwidth

In angle modulation, the number of sidebands and their amplitude is

determined by the modulation index. However, the frequency of each


sideband depends on the modulating frequency. The first-order
sidebands are fr + im and f - fm The second-order sidebands are fr + 2fm
r .

and - 2im This progression continues for each higher order sideband.
f,. .

The bandwidth, therefore, depends on the number of sidebands in the


wave. This is determined by the modulating frequency and modulation

index. Since modulation index is ——


f
, frequency deviation is also a

determining factor in bandwidth.

Thus, if we know the modulating frequency and the frequency deviation,


we can easily determine the required bandwidth.

As an example, what is the bandwidth of an angle modulated signal in


which the modulating frequency is 3 kHz and the maximum deviation is
18 kHz? We must first find the modulation index.

m = j-

18 kHz
3 kHz

= 6

From the table in Figure 1-20, we find that a modulation index of 6, has 9
significant sideband pairs. Therefore, the bandwidth is

BW = fm x highest order sideband x 2

= 3 kHz x 9 x 2

= 54 kHz
1-44 UNIT ONE

Self-Review Questions

16. Angle modulation includes both and


modulation.

17. List the three characteristics of frequency modulation.

18. In phase modulation, maximum deviation occurs when the mod-


ulating signal .

19. Phase modulation is also called

20. What is the primary difference between PM and FM?

21. The output of the circuit shown in Figure 1-22 is (phase, frequency)
modulation.

AUDIO LOW-PASS PHASE


INPUT NETWORK MODULATOR
+ OUTPUT

Figure 1-22
Is the output FM or PM?
Communications 1-45

22. What is the modulation index when a carrier is deviated 35 kHz by


a 7kHz signal?

23. What is the deviation ratio?

24. A 100 MHz carrier is deviated 70 kHz by a 10 kHz audio signal.


What is the modulation index, the frequencies of the sidebands, the
amplitude of the sidebands and carrier, and the bandwidth?
1-46 UNIT ONE

Self-Review Answers

16. Angle modulation includes both phase and frequency modulation.

17. The three characteristics of FM are:

1. The waveform is constant in amplitude but varies in fre-

quency.

2. The rate of carrier deviation is the same as the frequency of


the modulating signal.

3. The amount of carrier deviation is directly proportional to


the amplitude of the modulating signal.

18. In phase modulation, maximum deviation occurs when the mod-


ulating signal crosses the zero axis.

19. Phase modulation is also called indirect FM.

20. The primary difference between PM and FM is that, in PM, the


deviation is proportional to both the modulating frequency and
amplitude, while in FM, the deviation is proportional only to the
amplitude of the modulating signal.

21. The output of the circuit shown in Figure 1-22 is frequency mod-
ulation.

22.
m =— f,
7-

_ 35 kHz
" 7 kHz

= 5

23. Deviation ratio is the ratio of a system's maximum deviation to its

maximum audio frequency.


Communications 1-47

24. The modulation index is

m = 7—

70 kHz
10 kHz

=7

From the table of Figure 1-20 for a modulation index of 7 there are
, ,

10 significant sidebands. Their frequencies and amplitudes are:

Frequency Amplitude
100 MHz Carrier 0.30 of original
carrier level

100.01 & 99.99 MHz 1st sidebands 0.00


100.02 & 99.98 MHz 2nd sidebands -0.30
100.03 & 99.97 MHz 3rd sidebands -0.17
100.04 & 99.96 MHz 4th sidebands 0.16
100.05 & 99.95 MHz 5th sidebands 0.35
100.06 & 99.94 MHz 6th sidebands 0.34
100.07 & 99.93 MHz 7th sidebands 0.23
100.08 & 99.92 MHz 8th sidebands 0.13
100.09 & 99.91 MHz 9th sidebands 0.06
100.1 & 99.9 MHz 10th sidebands 0.02

The bandwidth is

BW = fm x highest order sideband x 2

= 10 kHz x 10 x 2

= 200 kHz
1-48 UNIT ONE

The frequency spectrum for this wave is shown in Figure 1-23. The
frequency spectrum does not show phase angle, so sidebands with
negative amplitudes (180° phase shift) are shown the same as the
other sidebands.

0.4 —

0.3 —

^ 0.2 —

0.1 —

100 MHZ
FREQUENCY

Figure 1-23
Frequency spectrum for problem 24.
Communications 1-49

THE COMMUNICATIONS SYSTEM


So far in this unit, we have discussed the need for modulation and the
characteristics of the two basic types of modulation. This section is a brief
overview of the communications system itself.

The basic block diagram of a communications system is shown in Figure


1-24.It consists of an input device, a transmitter, a transmission channel
— which also introduces noise, a receiver, and an output device. Let's
discuss each of these items.

MODULATION DEMODULATION

INPUT OUTPUT
TRANSMITTER CHANNEL RECEIVER
DEVICE DEVICE

t
NOISE

Figi ire 1-24


A commun ications syste •m.

Input/Output Devices

An input or output device is nothing more than a transducer, which is the


name given to any device that converts an input of one physical form to
an output of a different physical form. For example, in radio broadcasting
the input device is a microphone and the output device is a loudspeaker.
Both of these are transducers. The microphone converts sound waves
into electrical signals, while the loudspeaker converts the electrical sig-
nals into sound waves.

The same is true for television broadcasting. The visual input device is a
TV camera which converts light into electrical signals. The output device
is the cathode ray, "picture," tube which converts the electrical signals

into light.

There are a number of other input/output devices used in communica-


tions. These include telegraph keys, teletypewriters, chart recorders,
facsimile machines, and, in telemetry systems, a wide variety of trans-
ducers. Many of these items require a complete course of study on their
own. However, they all have an electrical output or input which attaches
to the transmitters and receivers you will study in this course.
1-50 UNIT ONE

Transmitter

As stated earlier, it is usually impractical to transmit the input signal at its


original frequency. Therefore, the transmitter must transfer the input
signal to a higher frequency. It does this through the process of modula-

tion. Thus, the transmitter consists of a high frequency carrier generator


and a modulator. It may also use amplifiers to increase its power output.

Channel

The transmission channel can be any radio frequency between approxi-


mately 3 kHz and 300 GHz. It could also be any light wave frequency.
However, this course deals only with radio frequency communications.

Figure 1-25 shows the radio frequency spectrum. It is divided into several
frequency bands from VLF low frequencies, 3 kHz-30 kHz) to EHF
(very
(extremely high frequencies, 30 GHz-300 GHz). Each of these bands has
specific characteristics. For example, the medium frequency (MF) band
allows local communications during daylight hours and longer range
communications at night. The standard AM broadcast band is located
within this band. The international short-wave broadcasting band is
located in the HF band (3 MHz-30 MHz). These frequencies allow
world-wide communications and, therefore, are the best for international
broadcasting. The television channels are located in the VHF and UHF
bands. These frequencies offer local fade-free communications and,
therefore, are well suited to TV broadcasting.
Communications 1-51

1Hz ~i~

10Hz-

100HZ' AUDIO
FREQUENCIES (AF)
(15Hz-20kHz)

1kHz-

VERY-LOW FREO. (VLF)


(3kHz-30kHz)
10kHz

LOW FREQ. (LF)


(30kHz-300kHz)
100kHz
STANDARD
MEDIUM FREQ. (MF) BROADCAST BAND
(300kHz-3MHz) (540kHz-1600kHz)
1MHz
m
RADIO SHORT WAVE
'
FREQUENCIES (RF) HIGH FREQ. (HF) BROADCASTING

10MHz
1 (3kHz-3000GHz) (3MHz-30MHz) TV CHANNELS
^ 2 THROUGH 6
(54MHz-88MHz)
VERY-HIGH FREQ. (VHF)
^ FM BAND
100MHz
(30MHZ-300MHZ)
I (88MHZ-108MHZ)

V TV CHANNELS
ULTRA-HIGH FREQ.(UHF) ~~W 7 THROUGH 13

1GHz
(300MHz-3GHz) —*- U74MHz-216MHz)
TV CHANNELS
SUPER-HIGH FREQ. (SHF) 14 THROUGH 83
(470MHz-890MHz)
(3GHz-30GHz)
10GHz

EXTREMELY-HIGH FREQ. (EHF)


(30GHz-300GHz)
100GHz
T
1000GHz

Figure 1-25
The radio frequency spectrum.
1-52 UNIT ONE

Noise

Noise is defined as any unwanted form of energy tending to interfere with


the reception of wanted signals. In Figure 1-24 only one source of noise
, is

shown, not because only one exists,but to simplify the drawing. There
are basically four types of noise. These are atmospheric noise, space
noise, man-made noise, and internal noise.

Atmospheric noise is caused by electrical disturbances in the earth's


atmosphere. The most visible of these sources is lightning. Since radio
waves can propagate around the world, given the right conditions, so can
the noise generated by lightning. Therefore, atmospheric noise due to
lightning is present at all times, even though there may be no local
thunderstorm in progress.

Space noise due primarily to the sun and the billions of other stars in
is

the universe. The sun is an extremely large body at a very high tempera-
ture of over 6000°C on the surface. It therefore radiates energy over a very
broad frequency spectrum. This is a constant source of noise. However,
there are also disturbances on the sun's surface such as sun spots and
flares that generate even higher levels of noise. Since distant stars are also
suns, they too have high temperatures and, therefore, radiate noise.

Almost all of us are familiar with man-made noise. The electric motor of a
hair dryer, vacuum, etc. can literally
, wipe out reception. Other sources of
man-made noise are automobile ignitions, high voltage power line leak-
age, fluorescent lights, and other transmissions can be considered as
noise.

Internal noise is created by the active and passive devices inside the
receiver This type of noise can be divided into two categories:
itself.

thermal noise and shot noise. Thermal noise is generated by any resis-
tance and due to the rapid and random motion of the atoms and
is

electrons of which the resistance is constructed. Shot noise is present in


all active devices and is due to the random variation in the arrival of

electrons at the collector of a transistor or the drain of an FET.


Communications 1-53

Receiver

By the time the transmitted signal reaches the receiver, most of its power
has been dissipated and several forms of noise have been added. The
receiver must amplify this weak signal enough to overcome any internal
noise and also separate it from all the other transmitted signals present. It

must then demodulate this signal to recover the original information.


This is quite a difficult task. Over the years, many types of receivers have
been developed to perform these functions for specific applications. We
will discuss many of these receivers in a later unit.

Self-Review Questions

25. List the parts of a basic communications system.

26. What is a transducer?

27. What are the names and frequencies of the major radio frequency
bands?

28. What is noise?

29. List the four types of noise.

30. What is the function of a transmitter? A receiver?


1-54 UNIT ONE

Self-Review Answers

25. A basic communications system consists of an input device, a


transmitter, a transmission channel, one or more sources of noise, a
receiver, and an output device.

26. A transducer is any device that converts an input of one physical


form to an output of another physical form.

27. The major radio frequency bands are:

VLF 3kHz-30 kHz


LF 30 kHz-300 kHz
MF 300 kHz-3 MHz
HF 3 MHz-30 MHz
VHF 30MHz-300 MHz
UHF 300 MHz-3 GHz
SHF 3 GHz-30 GHz
EHF 30 GHz-300 GHz

28. Noise is any unwanted form of energy tending to interfere with the
reception of wanted signals.

29. The four types of noise are atmospheric noise, space noise, man-
made noise, and internal noise.

30. The function of a transmitter is to generate a high frequency carrier


and to modulate that carrier. The function of a receiver is to amplify

the received signal, separate it from all other signals present, and to

demodulate the signal.


Unit 2

AMPLITUDE MODULATION
2-2 UNIT TWO
Amplitude Modulation 2-3

INTRODUCTION
In the previous unit, you studied the need for modulation, its various
types, and its characteristics. This unit discusses several methods of
generating amplitude modulation. It also discusses the disadvantages of
AM and presents a more advanced modulation technique single —
sideband. There are also two experiments included in this unit to en-
hance your understanding of AM and single sideband.

The "Unit Objectives" on the next page state the goals of this unit. Review
this list now and be sure you can satisfactorily complete all the objectives
before you take the unit exam.
2-4 I UNIT TWO ,
Amplitude Modulation 2-5

UNIT OBJECTIVES
When you have completed this unit, you should be able to:

1. Name five methods of amplitude modulation.

2. Define modulator.

3. State the three purposes of a pi output network.

4. Find the required modulator power when given input power and
modulation percentage.

5. Find sideband power when given carrier power and modulation


percentage.

6. State the advantages and disadvantages of collector modulation,


series modulation, and progressive series modulation.

7. List the disadvantages of standard AM.

8. Name three types of balanced modulators.

9. Identify a double sideband suppressed carrier waveform.


2-6 UNIT TWO

10. State the difference between low-level and high-level modulation.

11. State the primary advantage and disadvantage of suppressed car-


rier AM.

12. List the advantages of single sideband over standard AM and sup-
pressed carrier AM.

13. Name two methods of generating an SSB signal.

14. List the two types of sideband filters.

15. Draw a block diagram of a filter type SSB transmitter and show the
frequencies present at each stage.

16. List three methods of sideband selection.

17. Compute the peak envelope power rating of an SSB transmitter.

18. Construct a simple AM transmitter and measure the percent of


modulation.

19. Construct and properly adjust a balanced modulator.


Amplitude Modulation 2-7

AM CIRCUITS

To obtain amplitude modulation, the audio signal and the carrier must be
combined in a stage called the modulator. This circuit is used in basically
two types of devices — signal generators and AM transmitters. A signal
generator produces a low-level AM signal for testing and alignment
purposes. This type of modulator is relatively simple. However, an AM

transmitter must produce power levels from 5 watts, for a citizen's band
(CB) transceiver, up to 2 million watts, for a broadcast transmitter. Such
high power levels require very complex modulators, primarily to im-
prove efficiency. This section discusses both types of modulators.
2-8 UNIT TWO

The Diode Modulator

A simple circuit for producing amplitude modulation is shown in Figure


2-1. The modulating signal, in this case audio, is applied at the top of Rj
while the carrier is applied at the top of R2 . The signal at the junction of Rj
and R2 is the sum of the carrier and audio. That is, the carrier is simply
riding on the audio signal. Notice that the carrier is not amplitude
modulated at this point. It is simply added to the audio signal.

R2 |Ri

.">!

....
AHlllllinnnll
R3

Fl*
-OQUTPUT

Cl
T
T

Figure 2-1
The diode modulator and its waveforms.
Amplitude Modulation 2-9

When S! is R3 when the signal on the anode


open, D! conducts through
swings positive. However, when this signal swings negative, D cuts off.
x

Thus, the signal developed across R3 will consist of positive-going pulses


as shown. Notice that the positive pulses vary in amplitude in accordance
with the audio signal.

Sj is included in the circuit merely for explanation purposes. Normally, it


is omitted and the tank circuit composed of Q
and L, is connected
directly in parallel with R3 .

The purpose of the tank circuit, which is tuned to the carrier frequency,
becomes clear when S! is closed. Each time Dj conducts, a pulse of
current flows through the tank. This causes the tank to resonate and the
flywheel action of the tank produces a negative half-cycle for each posi-
tive input pulse. The high amplitude positive pulses cause high
amplitude negative pulses, and the low amplitude positive pulses cause
low amplitude negative pulses. Therefore, each negative half-cycle will
have the same amplitude as the positive half-cycle. As you can see, the
output is an AM wave. Thus, this simple circuit produces amplitude
modulation.

The diode modulator is a good example of nonlinear mixing. That is, the
audio and carrier are mixed across the nonlinear resistance of D,. The
result of this mixing action is sum and difference products. Therefore, the
signals across Dj are the original frequencies and their sum and differ-
ence. If the carrier is 100 kHz and the audio signal is 10 kHz, the sum is
100 kHz + 10 kHz = 1 10 kHz, and the difference is 100 kHz - 10 kHz = 90
kHz. These are the sideband frequencies.

The 100 kHz carrier and 10 kHz audio signal are also present. However,
the resonant circuit of C r Li is tuned to 100 kHz and, therefore, allows
only the carrier and sidebands to pass to the output. The 10 kHz audio
signalis filtered out. Thus, the output of the diode modulator is a true
amplitude modulated signal.
2-10 UNIT TWO

The Absorption Modulator

The absorption modulator used primarily in VHF, UHF, and higher


is

frequency signal generators. Most absorption modulators use PIN diodes


because of their variable impedance characteristics. Usually, several of
these diodes are arranged in an attenuator network. This arrangement not
only allows amplitude modulation, but also a variable output level con-
trol which makes it an extremely versatile device for use in signal
generators.

Another advantage of the absorption modulator is that it does not affect


the frequency generating device or circuit. This is very important at VHF,
UHF, and higher frequencies since any voltage changes at the oscillator
may cause undesired frequency variations.

You will recall that a PIN diode is formed by placing an intrinsic (I)
semiconductor material between the P and N type materials of a junction
diode. Above 100 mHz, this causes the diode to act like a linear resistance
by conducting in both directions. However, its resistance depends on the
amount of forward bias current. Therefore, the PIN diode acts as a
current-controlled resistance. This makes it ideal for attenuator net-
works.

Figure 2-2 shows a typical attenuator. The variable resistor elements


represent PIN diodes. A DC bias current is applied to the diodes, which
varies their resistance and, thus, the network's attenuation of the RF
signal. Therefore, the steady-state DC bias current can be used to adjust
the output amplitude and function as the signal generator's output level
control. Similarly, the modulation signal can be superimposed on the DC
bias and, as a result, amplitude modulate the RF signal.

PIN

Figure 2-2
A simplified diagram of a PIN diode absorption modulator.

Mft
Amplitude Modulation 2-11

However, for 100% amplitude modulation the positive peak of the


waveform must be twice the carrier level. For this reason, the attenuator
must be set for approximately mid-range attenuation. This establishes the
carrier level and allows equal amplitude variations both above and below
this set level.

The efficiency of the absorption modulator is very low. However, at


microwave frequencies it is one of the few methods of amplitude modula-
tion available. Moreover, since it is normally used at low power levels,
efficiency is not an important requirement.

The Collector Modulator

The most commonly used AM technique is collector modulation or, in


vacuum tube modulators, plate modulation. In collector modulation, the
modulating signal is applied to the transistor's collector, in series with
the DC supply voltage.
2-12 UNIT TWO

A typical collector modulation circuit is shown in Figure 2-3. Q lf and its

associated circuitry, is a class C RF power mod-


amplifier. Note that the
ulating signal is applied to the collector through transformer T With no x
.

modulating signal present, there is approximately zero volts dropped


across the secondary of Tt Therefore, the entire V cc voltage is present at
.

the collector of Qj. However, when modulation is applied, there is an AC


voltage, Em induced in the secondary of Tj Since it is an AC voltage, it is
, .

alternately in phase and out of phase with V cc When it is in phase with


.

V cc a high positive potential is applied to the collector of Q This causes


, t
.

an increase in collector current and, therefore, a higher amplitude output


signal. This is the peak of the modulation envelope.

OUTPUT

RF INPUT

Figure 2-3
Basic collector modulation circuit.

When Em is out of phase with V cc a lower positive potential is applied to


,

Qx. This results in lower collector current and, hence, a lower output
amplitude. This is the trough of the modulation envelope.
Amplitude Modulation 2-13

Since Q, is a class C amplifier, only positive pulses will appear at its

output. To form a complete AM waveform, these output pulses must be


applied to a resonant or "Tank" circuit. The circuit of Lj, Q, and Q in

Figure 2-3 form the output tank for this circuit. You may recognize this
circuit as a pi network. A similar circuit is often used as a low pass filter in
power supplies. The circuit shown here is also a low pass filter, but it is
different from the power supply filter in two important ways: it is a
resonant tank circuit, and an impedance matching network. Let's
examine its operation in detail.

Figure 2-4A shows the basic output network. The output signal of Ch is
developed across the RF choke (RFC) and capacitively coupled to the pi
network by Q. If the schematic is rearranged, as shown in Figure 2-4B,
you can see that the combination of Q
and Q, are in parallel with Lj.
Therefore, by tuning Q
and Q, the circuit will resonate at the output
frequency. Figure 2-4B also shows that the input is across Q while the
output is taken across Q. By varying the ratio between Q
and Q, an
impedance match can be obtained between Qj and the output load. Thus,
not only is the pi network a low-pass filter, which reduces harmonic
output, but it is also a resonant tank and impedance matching network.

OOUTPUT

B
L]

Figure 2-4
Pi output tank circuit.
2-14 UNIT TWO

One problem with the collector modulation circuit of Figure 2-3 is that
100% modulation is not possible because the transistor saturates before
reaching the positive peak of modulation. Also, at the negative trough,
the base RF drive feeds through the base-collector junction capacitance,
thereby preventing zero output at the negative peak. Both of these prob-

lems can be overcome by modulating both the driver and final amplifier.
This is shown in Figure 2-5. By modulating the driver, extra drive is
provided to the final amplifier at the positive peak, allowing it to reach a
full 100% modulation. At the negative trough, the drive is reduced
sufficiently to prevent RF feedthrough.

Notice that the modulator in the circuit of Figure 2-5 is a push-pull audio
amplifier.It must deliver at least one-half of the input carrier power to

obtain 100% amplitude modulation. Therefore, a relatively high power


audio amplifier is required. The push-pull circuit delivers high power at
much better efficiency than a single-ended class A amplifier. Let's
examine exactly what the modulator power requirements are.

OUTPUT
DRIVER
AMPLIFIER

RF INPUT

AUDIO
INPUT

Figure 2-5
Collector modulation of both driver and final amplifiers.
Amplitude Modulation 2-15

As an example, suppose that V cc is 28 volts and Ic is 1 ampere. The DC


input power is:

P/jv = V cc x Ip
= 28 V x 1 A
= 28 W
The impedance of the collector circuit is:

VV rr
Zc =
Ic

28 V
1 A

= 28

This is the impedance that the audio modulating power must be de-
veloped across.

To modulate the amplifier stage 100%, the sum of the instantaneous


audio peak voltage and V cc must be twice the V cr for the positive peak and
zero for the negative trough. Since V cc is 28 V, the modulator must double
this by supplying 28 V peak to the transformer secondary. This voltage
must be developed across the 28 H collector impedance. To find the
E2
power required, you can use the formula P = -5- . However, this
K
equation applies only to effective or rms values of AC voltages. Therefore,
the peak audio voltage must be converted to rms:

E™, =EpK x 0.707

Er ms =28 V x 0.707

= 19.8 V

Now, using the power formula:

-4- .

19.8 V2
28 n

= 14 W
2-16 I
UNIT TWO

Thus, the modulator output must be 14 W to obtain 100% modulation of a


28 W RF amplifier.

From these figures you can also determine the sideband and carrier
power in the transmitted signal. Continuing our example, if the effi-
ciency of the final amplifier is 75%, the carrier output power is:

Pour =P,„xEFF%

= 28 W x 75%

= 28 W x 0.75

= 21 W
At 100% modulation, the modulator supplies 14 W
to the final amplifier.
The amount of modulator power that appears at the output is also deter-
mined by the final amplifier efficiency. Therefore,

Pout = Pin x EFF%

= 14 W x 0.75

= 10.5 W

The modulator power supplied to the output, which is divided equally


between the sidebands, is 10.5 W or one-half the carrier output power.
Amplitude Modulation 2-17
J

An equation for determining modulator power or sideband power for any


modulation percentage is:

P m2P r
Am _

Where: P m = modulator power or sideband power

m = percent of modulation

Pc = carrier power.

To determine modulator power, P c must be the carrier input power. To


find sideband power, P c must be the carrier output power.

Let's try a couple of examples. What is the sideband power for a 100-watt
carrier at 50% modulation?

Pm -
I
^ _

2
_ 0.5 x 100 W

= 12.5 W

Since sideband power is 12.5 W, the power of each sideband is 6.25 W.

What modulator power is required for 80% modulation of an RF amplifier


with an input power of 1000 W?

P. =
m 2
P

0.8 2 x 1000 W

= 320 W
2-18 UNIT TWO

Series Modulator

While collector modulation is the most commonly used technique, it


does have disadvantages. The most serious of these is the modulation
transformer itself. It is large, heavy, costly, and introduces frequency
distortion. That is, due to the inductance of the transformer, some fre-
quencies are attenuated more than others. One method that eliminates
the modulation transformer is series modulation. This technique uses a
transistor in place of the transformer.

A basic series modulator is shown in Figure 2-6. In this circuit, the


current through the final amplifier,Q lt is controlled by the series mod-
ulator, Q2. Therefore, any increase in current flow through Q> will in-
crease the output of Qj. Likewise, any decrease in Q2 current flow will
decrease the output of Qj. Thus, the series modulator amplitude mod-
ulates the RF signal present at Q and it does it through direct coupling
x

rather than transformer coupling. This eliminates any frequency distor-


tion.

AUDIO
INPUT

OOUTPUT

RF INPUT

FINAL
AMPLIFIER

Figure 2-6
A basic series modulation circuit.
Amplitude Modulation 2-19

However, series modulation has one significant disadvantage: it is very


inefficient. You will recall that at the positive peak of 100% modulation,
the voltage at the final amplifier must be doubled. As an example, if the
final amplifier has a collector voltage of 100 V and collector current of 1
A, the input power is 100 W. Inorderto 100% modulate this amplifier, the
collector voltage and current must double for the positive peak of mod-
ulation. This additional voltage must be dropped across the series mod-
ulator at 0% modulation. Therefore, at 0% modulation, the series mod-
ulator drops 100 V at 1 A. This is shown in Figure 2-7. Note that the
modulator is dissipating a full 100 W. This is very inefficient operation.
Collector modulation is much more efficient, since at 0% modulation,
there is almost no voltage dropped across the secondary of the modula-
tion transformer.

100V X 1A - 100W

100V X 1A > 100W

0% MODULATION

Figure 2-7
Power distribution at 0% modulation in a series modulator.
2-20 UNIT TWO

Progressive Series Modulator

One improves the efficiency of series modulation


circuit that greatly
while still maintaining the advantage of direct coupling is the progres-
sive series modulator. This circuit is shown in Figure 2-8. It uses two
modulator transistors and two power supplies in a special switching
configuration. At 0% modulation and during the negative troughs of
modulation, Q is turned off and, therefore, the 104 V supply is discon-
x

nected from the circuit. During the negative troughs, Q2 controls the
current and voltage applied to the final amplifier. For positive modula-
tion, Q2 saturates and Qj turns on. As soon as Q turns on, Dj is reverse
x

biased and, therefore, disconnects the 52 V supply. At this time, Q x

controls the voltage and current to the final amplifier. The 104 V supply is
now connected, which allows a full positive peak of modulation.

The advantage of this arrangement is that, at carrier-only conditions or


0% modulation, very little power is dissipated across the modulator. This
is because, at 0% modulation, Qj is off while Q2 is on and saturated. Its

saturation voltage is approximately 2 V, therefore, this leaves 50 V from


the 52 V supply for the final amplifier. Thus, at carrier-only conditions,
50 V is applied to the final amplifier. If the current is 2 A, input power is
50 V x 2 A = 100 W. However, the power dropped across modulator
transistor Q2 is only 2Vx2A = 4W. This is a signficant difference over
the normal series modulator. Thus, the progressive series modulator
offers much better efficiency while maintaining direct coupling. The
only disadvantage is the increased circuit complexity.

+ 104V

AF INPUT < + 52V

c
RF INPUT
FINAL
AMPLIFIER
-MODULATED
RF
OUTPUT

T
Figure 2-8
A simplified diagram of a progressive series modulator.
Amplitude Modulation 2-21

Self-Review Questions

1. Name five methods of amplitude modulation.

2. What is a modulator?.

3. What are the three purposes of a pi output network?.

4. How much modulator power is required for 75% modulation of a 10


kW amplifier?

5. What is the power in the upper sideband for a carrier of 100 W


when modulation is 55%?
2-22 UNIT TWO

6. What is the primary disadvantage of collector modulation?

7. List the advantage and disadvantage of series modulation.

8. How does progressive series modulation overcome the disadvan-


tage of series modulation?

9. Which of the modulators discussed in this unit use nonlinear


resistance to obtain AM?
Amplitude Modulation 2-23

Self-Review Answers

1. The five methods of amplitude modulation discussed in this unit


are:

1. Diode modulation.
2. Absorption modulation.
3. Collector modulation.
4. Series modulation.
5. Progressive series modulation.

2. A modulator is a circuit that causes modulation either in its inter-

nal circuitry or in another associated circuit.

3. The pi output network is a low pass filter, a resonant tank circuit,


and an impedance matching network.

mP 2

Pm
2

= .75 2 x 10 kW
2

= 2.8 kW

m 2
Pc
p. =
2

.55 2 x 100 W

= 15 W sideband power 7.5 W in upper sideband


2-24 UNIT TWO

6. The primary disadvantage of collector modulation is the modula-


tion transformer which is heavy, costly, and causes frequency
distortion.

7. The advantage of series modulation is that it eliminates the mod-


ulation transformer. The disadvantage is that it has very poor
efficiency.

8. Progressive series modulation uses two power supplies and a spe-


cial switching modulator that greatly improves its efficiency.

9. All of the modulators use nonlinear resistance to obtain AM. The


diode modulator uses the nonlinear diode junction. The absorption
modulator uses the nonlinear PIN diode. The other types use the
nonlinear collector-base junction of the modulated transistor
amplifier.
Amplitude Modulation 2-25

SUPPRESSED CARRIER AM
The type of amplitude modulation just discussed will be referred to in
this section as standard AM. and both
In this system, the carrier
sidebands are transmitted just as they appear at the output of the mod-
ulator. This system is used by standard AM broadcast stations and most
citizen's band radios. Its primary advantage is that it uses straightforward
and inexpensive transmitting and receiving equipment. However, it has
several disadvantages. One of these disadvantages can be overcome by
transmitting a suppressed carrier signal.

This section discusses the disadvantages of AM, methods of generating


suppressed carrier AM, and its disadvantages.

Disadvantages of Standard AM
The standard AM system has several disadvantages. The three most
important are:

1. Most of the transmitted power is in the carrier. This power is

wasted because it does not contribute to the intelligence being


conveyed.

2. The bandwidth of the transmitted signal is twice that of the


intelligence being conveyed.

3. The sidebands and carrier must have precise amplitude and


phase relationships. These relationships are difficult to main-
tainunder some conditions.

Let's discuss each of these in more detail.


2-26 UNIT TWO

CARRIER POWER

Let's assume that a standard AM transmitter produces a 100-watt output


when unmodulated. The entire 100 watts is contained in the unmod-
ulated carrier. If the transmitter is modulated at 100%, the total radiated
power will increase to 150 watts. The transmitter still produces the
100-watt carrier. The additional 50 watts represents the power in the
sidebands. Since the power of the lower sideband is equal to that of the
upper sideband, each sideband contributes only 25 watts to the total
transmitted power. When the percent of modulation is lower than 100%,
the power in the sidebands is even lower.

It is important to remember that the carrier itself does not vary in fre-

quency or amplitude. Consequently, the carrier does not contain any


intelligence. All information being transmitted is in the sidebands. The
carrier acts only as a reference frequency. It allows a modulator to convert
low frequency audio to high frequency sidebands. It also allows the
demodulator (or detector) to recover the audio information. Between the
modulator stage in the transmitter and the detector in the receiver, the
carrier serves no useful purpose. Thus, in the standard AM system, at
least two-thirds of the transmitted power is taken by a signal which
conveys no intelligence.

BANDWIDTH

In the previous section, we discussed how the bandwidth is determined


by the intelligence. When a 600 kHz carrier is amplitude modulated by a 5
kHz audio tone, a 595 kHz sideband and a 605 kHz sideband are pro-
duced. In standard AM, both the sidebands are transmitted along with the
carrier. Thus, the transmitted bandwidth is 605 kHz - 595 kHz = 10 kHz.
This is twice the modulating frequency.
Amplitude Modulation 2-27

When the modulating frequency changes, both sidebands change fre-


quency. When the amplitude of the modulating signal changes, the
power in the sidebands change. In both cases, the carrier remains con-
stant in both power and frequency. Obviously then, the intelligence is
contained in the sidebands and not in the carrier. Just as important, the
intelligence is duplicated in the sidebands. That is, if we know the
frequency of either sideband and the frequency of the carrier, we can
determine the modulating frequency.

Therefore, in the standard AM system, the bandwidth is twice as wide as


is necessary.

PROPAGATION PROBLEMS

For perfect reception in a standard AM system, the two sidebands and the
carrier must be received exactly as they are transmitted. Unfortunately,
when propagation conditions are poor, the AM signal can greatly de-
teriorate before it reaches the receiver. Moreover, each component of the
AM signal may be affected differently since they have different frequen-
cies. Thus, the standard AM system is subject to fading and interference

when propagation conditions are poor.

Balanced Modulators

The first disadvantage of standard AM, wasted carrier power, can be


eliminated by suppressing or removing the carrier before transmission.
This accomplished by using a balanced modulator. This device mixes
is

the audio and RF carrier. Then, due to its circuit configuration, it allows
only the mixing products or sidebands to pass to the output. The resultant
signal is called double sideband suppressed carrier. It can be transmitted
and received without a carrier and, thus, offers more efficient transmis-
sion than standard AM.

There are several different types of balanced modulators. Each type has
its own advantages and disadvantages. Let's discuss several of the more

common balanced modulators.


2-28 UNIT TWO

TWIN FET BALANCED MODULATOR

A relatively simple balanced modulator is shown in Figure 2-9. Here, the


RF carrier input is applied in phase to the gates of Qi and Qj. This causes
equal but opposite drain currrents to flow in the primary of T2 The result .

is when the carrier alone is applied to the circuit. Potentiome-


zero output
used to compensate for differences in the two FETs. The output is
ter R, is
monitored and Rj is adjusted for minimum or zero output with the carrier
only applied to the circuit.

RF
NPUT
T2

+V DOUBLE
DD
O- -/ SIDEBAND
OUTPUT
O

Figure 2-9
The twin-FET balanced modulator.

When audio is present, it is applied to the gate of Qj


180° out of phase with
the gate of Q> due to the coupling transformer T, This sets up a push-pull
.

type circuit for the audio. It also causes nonlinear mixing or modulation
across the FETs. The end result is that the mixing products or sidebands
are in push-pull across the primary of T2 and, therefore, pass to the
output. However, the carrier is cancelled across the primary and, thus,
does not appear at the output. The original audio is filtered out by the RF
transformer and resonant tank. Thus, the output is a double sideband,
suppressed carrier signal.
Amplitude Modulation 2-29

The amount of carrier suppression depends on how closely the FETs


match and also the other components in the system. As mentioned before,
R, can be adjusted to improve the balance. Also, the center tap on the
primary of T2 can be adjusted for optimum operation. When properly
constructed and adjusted, this circuit can suppress the carrier to -60 dB
below the sideband amplitude.

DIODE-RING BALANCED MODULATOR

A balanced modulator that uses diodes as the nonlinear elements is


shown in Figure 2-10. In this circuit, the diodes must be carefully
matched to achieve a good carrier balance or null.

AUDIO
INPUT OUTPUT

Figure 2-10
The diode-ring balanced modulator.
2-30 UNIT TWO

Figure 2-11 A shows the circuit operation when the RF input has a
polarity as shown. Diodes D, and D 2 are forward biased
and equal but
opposite currents flow in both sides of TV is no output.
Therefore, there
When the RF input polarity reverses, diodes D 3 and D 4 conduct as shown

in Figure 2-1 IB. Here again, equal but opposite currents flow in T, and
there is no output.

I
4y////////^p. .JL
r p
— W//////////A

^^INPUT

™ — RF ±

Figure 2-11
Diode-ring modulator operation with only the RF carrier
applied.
Amplitude Modulation 2-31

Figure 2-12A shows the circuit action when the audio signal is applied.
To simplify the drawing, the RF input has been removed and the diodes
replaced by switches. When diodes D, and D 2 conduct, due to the RF
input, the audio signal flows through T, in the direction shown. When
the RF signal changes polarity, as shown in Figure 2-12B, diodes D and
3

B A conduct. This causes the signal through T, to change polarity. Thus,


the audio polarity at Tj is actually being switched back and forth by the
RF input.

AUDIO
INPUT

^mmmmmmzm^' D? mmmmmmmmm

Figure 2-12
Audio signal switching action in the diode-ring modulator.
2-32 UNIT TWO

To clarify matters, theaudio signal does not actually forward bias the
diodes. Instead, the RFinput switches the diodes and the audio signal
sees the forward biased diodes as a low impedance path. This is due to the
varactor action of the diodes. A forward biased diode has a high capaci-
tance and, therefore, a low capacitive reactance to the audio signal. On
the other hand, the reverse biased diodes offer a high reactance to the
audio signal.

The various waveforms for this circuit are shown in Figure 2-13. The RF
input is shown as a square wave. This allows instantaneous diode switch-
ing and thereby simplifies the drawing. Note that each time the RF input
goes negative, the output wave reverses polarity. This is duediode
to the
switching action described earlier. You can follow this throughout the
output waveform.

Notice that the output waveform's envelope "double or mirrior


is a
image" of the modulating audio. This is significantly different than the
standard AM envelope which duplicates the modulating signal. It can be
shown that this complex wave is actually composed of two equal
amplitude sine waves whose frequency separation is twice the modulat-
ing audio frequency. This corresponds exactly to upper and lower
sidebands spaced around a suppressed carrier. In short, the output is a
double sideband, suppressed carrier signal.

You might well ask: where did the sidebands come from when all the
circuit did was switch diodes on and off? Since the diodes are being
switched, they have a nonlinear resistance. Therefore, nonlinear mixing
occurs, creating the mixing products and, thus, sidebands. You will
recall that the carrier is balanced out by circuit action.

The diode-ring modulator widely used. In fact, it is often made in


is

special packages using hot-carrier diodes. These devices offer wide


operating frequency ranges and excellent carrier suppression.

There are several variations of the diode-ring balanced modulator. They


all use diodes as the nonlinear elements and operate in a similar manner.

That is, they balance out or null the carrier, and the nonlinear mixing
across the diodes creates sidebands which are passed to the output.
Amplitude Modulation 2-33

RF
INPUT

AUDIO
INPUT

l\ /I
OUTPUT A

FILTERED
OUTPUT

Figure 2-13
Waveforms in the diode-ring balanced modulator.
2-34 UNIT TWO

IC BALANCED MODULATORS

One circuit that lends itself extremely well to balanced modulator appli-
cations is the differential amplifier. A simplified diagram of a differential
amplifier isshown in Figure 2-14. Q3 acts as the current source for Qj and
Q2. If the RF input is applied to the bases of Q, and Q> in phase, current
through both transistors will be identical and the voltage difference
across the output will be zero. This is the common-mode rejection of the
differential amplifier and it has balanced out the carrier.

RF
RF
INPUT
NPUT

AUDIO
INPUT

Figure 2-14
A differential amplifier used as a balanced modulator.

The audio input is applied to the base of Q3. This upsets the circuit
balance. As a result, the audio and RF signals are mixed across Q and Q2. }

This is nonlinear mixing and, therefore, sidebands appear at the output.


However, the carrier or RF input does not. Since it is a common-mode
signal, it is rejected.

A differential amplifier must be constructed of transistors whose charac-


teristics are very closely matched. Forming the transistors on a single
chip of silicon, as is done with ICs, ensures this necessary match. There-
fore, the differential amplifier is ideally suited to integrated circuit con-
struction.

Amplitude Modulation 2-35

Figure 2-15 shows ICs that have been specifically designed for use as
balanced modulators. Figure 2-1 5A is the 1496 balanced modulator

which manufactured by Motorola, National, and Signetics. This device


is

uses a differential amplifier configuration similar to what was previously


described. Its carrier suppression is rated at a minimum of —50 dB with a
typical value of -65 dB at 500 kHz.

Figure 2-1 5B is the SL1640 balanced modulator manufactured by


Plessey. It also uses a differential amplifier configuration. Its primary
advantage is the small number of external components required. Its

carrier suppression is typically —40 dB at 30 MHz.

820 1200
—A/V # -0+12V
_1_. ±J_100UF
,K •15V

;iok

s2700
1000
>47 wv <2700
CARRIER
INPUT
o
<>HH 1496
AUDIO
INPUT -)| O OUTPUT

.100
i25UF
15V > 10 K r
>100

10K 50K
n
BALANCE

0+5V
B
CARRIER 10K O OUTPUT
BALANCE

Figure 2-15
Integrated circuit balanced modulators:
(A) the 1496 balanced modulator,
RF CARRCR
INPUT (B) the SL1640 balanced modulator.
2-36 UNIT TWO

Disadvantages of Suppressed Carrier AM


One AM is that most of the transmitted
of the disadvantages of standard
power is in the carrier. This power
wasted since it contains no intelli-
is

gence. For this reason, the balanced modulator is used to suppress the
carrier. This allows transmission of only the sidebands. However, the
other disadvantages of AM remain: excessive bandwidth and propaga-
tion problems.

The bandwidth of a suppressed carrier AM signal is still twice that of the


highest modulating signal. This is twice as much bandwidth as is
needed.

As mentioned before, for perfect AM reception, the two sidebands and


carrier must be received exactly as they are transmitted. However, in
suppressed carrier AM, the sidebands are transmitted without the carrier.
This means that the carrier must be "reinserted" at the receiver in order to
demodulate the phase relationships are critical in AM, the
signal. Since
reinserted carrier must not only be the same frequency as the original but
also exactly in phase with the original. This is the most difficult problem
to solve when designing a double sideband suppressed carrier system. It
can be solved, but unfortunately, to date, the required circuitry has
proven too complex and expensive for practical applications. Although,
double sideband suppressed carrier signals are used as part of the color
television system and the FM stereo broadcasting system.

A further disadvantage of suppressed carrier AM is that it is normally a


low-level modulation technique. This means that it is generated at a low
signal level and then must be amplified to a level suitable for transmis-
sion. Since these amplifiers must duplicate both the RF signal and the
envelope, they must operate at class B or, more frequently, class AB. This
results in low efficiency at high power levels. On the other hand, stan-
dard AM is usually a high-level modulation technique. It is normally
generated at the final amplifier or last stage of the transmitter. Therefore,
the RF carrier can be amplified using high efficiency class C amplifiers.
However, the audio signal still requires relatively low efficiency class AB
push-pull amplifiers.

As you can involved in the use of any


see, there are several trade-offs
system. However, the increased complexity of the double sideband sup-
pressed carrier AM
system limits its use to specialized applications.
Amplitude Modulation 2-37

Self-Review Questions

10. What are the disadvantages of standard AM?

11. Name three types of balanced modulators.

12. Identify the waveforms shown in Figure 2-16.

A B

Figure 2-16

13. What is the primary advantage of suppressed carrier AM? What is


the primary disadvantage?

14. What is the difference between low-level and high-level modula-


tion?
2-38 UNIT TWO

Self-Review Answers

10. The disadvantages of standard AM are that most of the transmitted


power is contained in the carrier, the bandwidth is twice that of the
highest modulating frequency, and it is subject to fading and inter-
ference.

11. The three types of balanced modulators discussed in this section


were the twin-FET modulator, the diode-ring modulator, and the
differential amplifier IC balanced modulator.

12. The waveforms in Figure 2-16 are:

A. Standard AM.
B. Suppressed carrier AM.

13. The primary advantage of suppressed carrier AM is that no power


iswasted in the carrier since it is not transmitted. The primary
disadvantage is that the receiver requires very complex circuitry to
demodulate the wave.

14. Low-level modulation is done at a low signal level and then


amplified to the level required for transmission. High-level mod-
ulation is done at the final amplifier stage.
Amplitude Modulation 2-39

SINGLE SIDEBAND
All three disadvantages of standard AM can be overcome by a system
called single sideband suppressed carrier (SSB). In this system, only one
sideband is transmitted. The carrierand the other sideband are suppres-
sed at the transmitter. This system can still convey intelligence since

either sideband contains all the information of the original modulating


signal.

This approach requires a more complicated and, therefore, more expen-


sive transmitter. The transmitter generally uses a balanced modulator to
suppress the carrier and one of several methods to suppress the unwanted
sideband. The design of the receiver is also complicated. As in double
sideband suppressed carrier, a substitute carrier must be added to the
received sideband. However, in SSB, the reinserted carrier must only be
within a few Hertz of the original carrier. This is much easier to obtain
than the precise frequency and phase requirements for double sideband
suppressed carrier AM. As a result, SSB is less complex. Also, its many
advantages over standard AM outweigh its additional cost and complex-
ity.

The first advantage of single sideband is that all the power transmitted
represents intelligence. Compare system in
this to the standard AM
which at least two thirds of the transmitted power is in the carrier. This
means that a single sideband transmitter that is radiating 50 watts can
produce the same audio signal level at the receiver as a standard AM
transmitter that is radiating 150 watts.

Single sideband also reduces the bandwidth requirements. Since only


one sideband is transmitted, the bandwidth is only one half that of a
standard AM transmitter. Thus, a given frequency range can accommo-
date twice as many SSB channels as standard AM channels.

Finally,SSB is far superior to AM when propagation conditions are poor.


SSB can be up to eight times more effective in getting through when
propagation conditions are really bad. This is due primarily to its narrow
bandwidth and the power advantages discussed above.
2-40 UNIT TWO

Filter Method SSB Transmitter

The most widely used technique for generating a single sideband signal
is the filter method. A block diagram of this type of transmitter is shown
in Figure 2-17. Here, the modulating audio is amplified and, along with
the RF carrier, applied to the balanced modulator. The output of the
balanced modulator, which is a double sideband suppressed carrier
signal, is then applied to the sideband filter. This filter is designed to pass
the desired sideband and block the unwanted sideband. The output is,
an upper sideband signal or a lower side band signal,
therefore, either
depending on the filter's passband.

MICROPHONE
AUDIO BALANCED SIDEBAND SSB
D- AMPLIFIER MODULATOR
il
FILTER OUTPUT

RF
OSCILLATOR

Figure 2-17
Filter method SSB transmitter.

SIDEBAND FILTERS

In a single sideband transmitter, the input audio is usually limited to the


250 Hz to 3000 Hz range. This is all that is required for adequate speech
intelligibilityand it narrows the signal bandwidth still further. It is
accomplished by using a simple audio filter in the input amplifiers. The
lower limit of 250 Hz results in a 500 Hz separation between the upper
and lower sidebands as shown in Figure 2-18A.
Amplitude Modulation 2-41

CARRIER
(SUPPRESSED)

LOWER UPPER
SIDEBAND SIDEBAND

500Hz

FREQUENCY-

B FILTER
PASSBAND
ATTENTUATION
u "
/
LU
/
/
Q /

\
/
»— /
/
1

0.
s
/
f
/
/
/
\
\
t
/ UPPER
SIDEBAND
\\
\\
1
1
1
1
1
1

1
1
~'
JVQd -

,/
1

I \ ^_
FREQUENCY-

Figure 2-18
SSB filter passband.

The function of the sideband filter one sideband is to fully attenuate

while allowing the other sideband to pass. Therefore, the filter must drop
from dB attenuation to approximately -50 dB attenuation within the
500 Hz frequency range that separates the sidebands. This requires a very
sharp selectivity "skirt" or drop from dB to - 50 dB. A typical sideband
filter selectivity, or response, curve is shown in Figure 2-18B. In this case,
the filter allows the upper sideband to pass and attenuates the lower
sideband.

One most widely used sideband filters uses quartz crystals to


of the
achieve the desired passband shape. Let's briefly review the characteris-
tics of crystals before continuing.
2-42 UNIT TWO

Figure 2-19A shows the equivalent electrical network for a quartz crystal.
The inductance L represents the mass of the crystal, C, represents the
piezoelectric resilience of the crystal, R
the frictional losses, and Q> is
is

the capacitance of the metal electrodes holding the crystal. In most


crystals, the frictional losses (R) are extremely small and, therefore, the
circuit Q is very high. In fact, the Q ranges from 10,000 up to 1 or 2
million. This makes them ideally suited for use in filters.

The impedance curve for a quartz crystal is shown in Figure 2-19B. This
shows that at low frequencies, the network impedance is capacitive and
high. As frequency increases, the impedance drops until it reaches fs .

This is the series resonant frequency of the crystal which is determined


by the and thickness. Going back to the equivalent circuit,
crystal's cut
the reactances of L and C, are equal, at f5 forming a series resonant circuit.
,

At a slightly higher frequency, iP the inductive reactance of L is equal to


,

Xc2 . Therefore, at this frequency, the crystal is a parallel resonant circuit.


These characteristics of the quartz crystal can be used to form a sideband
filter.

B
I
FREQUENCY
ic IMPEDANCE

Figure 2-19
(A) Equivalent circuit of a quartz crystal,
(B) impedance of the crystal versus frequency.
Amplitude Modulation 2-43

A crystal lattice filter is shown in Figure 2-20. This filter uses two
matched pairs of crystals in a special bridge network. In this circuit, X 14
and X 1B are identical. X 2A and X 2B are also identical to each other but
slightly different from X 1A —X lB The impedance diagram for the crystals
.

is shown in Figure 2-21. Note that there are two parallel resonant fre-

quencies for each crystal. This is due to the inductance present in the
circuit. They will form resonant circuits with the crystals at specific
frequencies. Let's analyze this circuit in detail.

Figure 2-20
A crystal lattice filter.

MAX-

ZERO
FREQUENCY

Figure 2-21
Impedance diagram and selectivity curve of a crystal lattice
filter.
2-44 UNIT TWO

At fj in Figure 2-21, both Xj and X 2 present equal impedances to the input


signal. Therefore, in the circuit of Figure 2-20, equal amplitude signals
are passed through X lA and X 2 ^. However, since they are from opposite
sides of the transformer, they are 180° out of phase, and cancel. The same
holds true for X 1B and X 2B . a point of maximum attenuation for the
This is

filter as shown on the response curve of Figure 2-21.

At f2 , X have equal impedances again. However, X, acts


both Xj and 2

inductively while X acts capacitively The result is that the signals arrive
2 .

at the output in phase. For example, if the signal at X lA is at 0°, since X lA

acts inductively, it will shift the signal +90°. The signal at X 2A is at +180°
due to the transformer. X 2A acts capacitively and, therefore, shifts the
signal -90°. The result is that both signals arrive at the output with a
phase angle of +90°. They are in phase and there will be zero attenuation
at this frequency. This is shown in Figure 2-21.

At a frequency midway between f2 and f3 X 2 attains series resonance


,

while X! is in parallel resonance. The signal, therefore, passes unat-


tenuated to the output through X 2A and X 2B At f3 the crystal's impe-
. ,

dances are again equal but opposite and zero attenuation occurs. Midway
between f3 and f4 crystal X, reaches series resonance and yet another zero
attenuation point occurs. At f4 again, the impedances are equal but
opposite and zero attenuation occurs. However, at f5 the impedances are
,

equal and in phase. The output signals, therefore, cancel and maximum
attenuation occurs.
Amplitude Modulation 2-45

As you can see, the crystal lattice filter makes an excellent sideband filter
with relatively steep attenuation curves or "skirts" on both sides of the
passband. However, for most applications, even steeper skirts and more
attenuation of the unwanted sideband are required. For this reason,
several lattice sections must be cascaded to achieve the required re-
sponse.

Another type of filter used extensively in single sideband transmitters is


the mechanical filter. A typical mechanical filter is shown in Figure 2-22.
The coupling coil along with the biasing magnet form an electrical-to-
mechanical or magneto-striction transducer. Any electrical signal across
the coupling coil will set up a magnetic field which will either oppose or
attract the biasing magnet. This magnetic action will cause the coupling
coil and transducer rod to move. This mechanical motion will then be
coupled to the disc resonators. These discs are cut to precise dimensions
and, therefore, will resonate at only a certain band of frequencies. When
the discs resonate, they will pass the signal on to the output transducer. If
they do not resonate, they will attenuate the input signal substantially.

Mechanical filters offer excellent characteristics and are often chosen for
the most exacting applications. However, they are limited to operation in
the 60 - 600 kHz range, whereas crystal filters can be constructed up to
10 MHz and higher.

COUPLING
ROD

BIASING
MAGNET

DISC
RESONATOR
TRANSDUCER
ROD
COUPLING
COIL

Figure 2-22
A sideband filter using mechanical disc resonators.
2-46 UNIT TWO

FREQUENCY CONVERSION

Thus far in the SSB transmitter we have an upper or lower sideband


signal present at the filter's output. However, this is a fixed frequency
signal. That is, the sideband filter is fixed-tuned and, therefore, the SSB
signal is also restricted to a single frequency. We must now convert this
SSB signal to the desired operating frequency.

Figure 2-23 shows a typical frequency conversion technique. It uses a


mixer and a heterodyne oscillator. The mixer is any nonlinear circuit and
could even be a balanced modulator. The heterodyne oscillator is a
crystal oscillator, a variable frequency oscillator, or a combination of
both. Let's examine Figure 2-23 in detail.

1 i OR4 MHz
SSB
OUTPUT

MICROPHONE
9 MHz
1
14 OR 4

1
MHz V
AUDIO J^
L> AMP
BALANCED
MODULATOR
SIDEBAND
FILTER
MIXER [^ POWER
AMP

t t
CARRIER HETERODYNE
OSCILLATOR OSCILLATOR

9 MHz 5 MHz

Figure 2-23
Complete SSB filter transmitter.

The carrier oscillator is crystal controlled at 9 MHz which is the passband


of the sideband filter. Therefore, the SSB signal, either upper or lower
sideband, is applied to the mixer at an approximate frequency of 9 MHz.
The heterodyne oscillator frequency for this example is 5 MHz, although
it may be switched or varied to obtain any desired output frequency. In
this case, the 9 MHz and 5 MHz signals are applied to the mixer and the
result is sum and difference mixing products. Therefore, either 4 MHz or
14 MHz will appear at the output. The desired mixing product is selected
by a resonant tank circuit at the output of the mixer. The power amplifier
is also tuned to the desired frequency. It amplifies the low-level SSB
signal to the needed level for transmission. Thus, once the SSB signal is
generated at a low-level fixed frequency, converted to the desired
it is

operating frequency using a heterodyne oscillator and a mixer.


Amplitude Modulation 2-47

SIDEBAND SELECTION

There are three methods of selecting either the upper or the lower
sideband. The obvious method, using two different filters, is shown in
Figure 2-24A. The only drawback to this method is the additional cost of
the second The most commonly used technique is shown in Figure
filter.

2-24B. This method shifts the frequency of the carrier oscillator. For
upper sideband, the carrier oscillator frequency is set to the lower edge of
the filter's passband. For lower sideband, it is set to the upper edge of the
passband. The carrier frequency is usually changed by switching bet-
ween two crystals.

SUPPRESSED
CARRIER
LOWER SIDEBAND UPPER SIDEBAND
FILTER PASSBAND FILTER PASSBAND

FILTER
CARRIER FOR
CARRIER FOR PASSBAND
LOWER SIDEBAND
UPPER SIDEBAND
B

Figure 2-24
Two methods of sideband selection.
»

2-48 UNIT TWO

The third method of sideband selection involves frequency inversion.


This can occur during the heterodyne or frequency conversion process.
As an example, suppose an upper sideband signal is generated with a
MHz and sideband frequencies extending to 9.003
suppressed carrier of 9
MHz. When this signal is mixed with a 15 MHz heterodyne oscillator
signal, the sum products will be a 24 MHz suppressed carrier with
sideband components out to 24.003 MHz. This is still an upper sideband
signal. However, the difference products will be a 6 MHz suppressed
carrier and sideband components down to 5.997 MHz. This is now a
lower sideband signal.

Frequency inversion can be used to select the desired sideband or one of


the other two methods. However, whichever method is chosen, the
effects of frequency inversion must be taken into account when designing
the transmitter.

Phasing Method of SSB

The only disadvantage of the filter method is the expensive sideband


filter required. To get around this, the phasing method of generating a

single sideband signal was developed. The block diagram for this system
is shown in Figure 2-25.

MICR 0PH0NE j — AUDIO


AMP
90°
PHASE BALANCED
MODULATOR
SHIFT
!

I i

SSB
OUTPUT

90°
J_ ii

RF BALANCED
pu A c r
OSCILLATOR M0CIULAT0R
S HIF r

Figure 2-25
Phasing method SSB transmitter.
Amplitude Modulation 2-49

With this system, a 90° phase network is used to split the audio
shift
signal into two components that have a 90° phase difference. The RF
oscillator signal is also split into two components that have a 90° phase
difference. One RF and one audio signal are then applied to each of the
two balanced modulators.

Both balanced modulators suppress the carrier and have a double


sideband output. These output signals are then added together and, due
to the phasing of the applied signals, one sideband is cancelled and the
other sideband is reinforced.

The disadvantage of the phasing method is that it requires considerable


and precise adjustment to operate correctly. For this reason, the filter
method is used exclusively in modern single sideband transmitters.

Linear Amplifiers

A single sideband signal is at a very low power level,


usually generated
often less than one watt. To power level, one or more power
increase its

amplifiers must be used. However, these power amplifiers must be linear;


they must amplify the SSB signal without introducing any distortion.
Otherwise harmonics and other forms of "spurious radiation" will
appear at the output and cause interference.

For linear amplification of an SSB signal, the amplifier must be biased for
class A, AB, or B operation. Class AB and B can be used because the
flywheel effect of the output tank will complete the RF waveform. How-
ever, class amplifiers will cause serious distortion and cannot be used.
C
Normally class AB or B amplifiers are used because they offer the best
efficiency.
2-50 UNIT TWO

SSB Power Ratings

The power rating for an AM transmitter is its DC input power under


"carrier only" conditions, or in other words, with no modulating signal.
However, a single sideband transmitter does not have a carrier; in fact,
with no modulating signal, there is no output. This makes power
measurements slightly more difficult with an SSB transmitter.

Figure 2-26 shows the output waveform of an SSB transmitter with a


normal speech modulating signal. The power contained in the maximum
peak of the signal is the basic transmitter rating and is called the peak
envelope power (PEP). However, envelope peaks occur only sporadically
during voice transmission. Therefore, an average power reading is much
easier to obtain. In fact, the final amplifier current and voltage meters
respond only to the average value. This is because the needles of the
meters are unable to respond to the rapid peaks and thus give an average
indication.

RATK)
PEAK AVERAGE

Figure 2-26
SSB output waveform.
Amplitude Modulation 2-51

To determine the PEP input from the average meter readings, you must
determine the ratio between peak and average. This can be very difficult
to do in the case of voice modulation because each person's voice charac-
However, a two-tone test signal may be used that
teristics are different.
consists of two audio sine waves with a frequency separation of approxi-
mately 1,000 Hz. This test signal is then applied to the input of the SSB
transmitter and the resulting output waveform will appear as shown in
Figure 2-27.

With a two-tone test signal, the ratio between PEP input and average
power input is 2:1. Therefore, the PEP rating is found by multiplying the
input power reading (I x E) by 2. The ratio for a voice-modulating signal
may be anywhere in the range of 1.5:1 to 6 or 7:1. Therefore, the two-tone
test signal gives an accurate indication of the transmitter's PEP rating.

PE
RATI0

Figure 2-27
Two-tone SSB output.
2-52 UNIT TWO

Self-Review Questions

15. What are the advantages of SSB over standard AM and double
sideband suppressed carrier AM?

16. Name two methods of generating an SSB signal.

17. What two types of sideband filters are used in a filter method SSB
transmitter?

18. What are the advantages and disadvantages of these two sideband
filters?

19. Label each block in the diagram of a filter method, single sideband
transmitter shown in Figure 2-28.

20. In Figure 2-28, if the input is a 1 kHz sine wave with the other
signals at the frequencies shown, what frequencies will be present
at point A?

1 KHz

500 KHz

Figure 2-28
Amplitude Modulation 2-53

What frequency will be present at point B for an upper sideband


output ?

What frequency will be present at point C if the sum mixing pro-


duct is used?

What frequency will be present at point C if the difference mixing


product is used?

If the difference mixing product is used, will the output be an upper


or lower sideband signal? .

21. List the three methods of sideband selection

22. With a two-tone modulating signal, the final amplifier ammeter


A and the voltmeter reads 28 V. What is the
reads 3.4 DC input
power? What is the peak envelope input power (PEP)?
2-54 UNIT TWO

Self-Review Answers

15. SSB requires a less complex receiver than double sideband sup-
pressed carrier AM. The advantages over standard AM are that SSB
requires only one-half the bandwidth, all transmitted power repre-
sents the intelligence contained in the sideband, and SSB's propa-
gation characteristics are superior.

16. An SSB signal can be generated by the filter method or the phasing
method.

17. A crystal filter or mechanical filter is generally used in the filter

method SSB transmitter.

18. The mechanical filter offers superior characteristics, however it is

limited to the 60 - 600 kHz range. The crystal filter can be con-
structed up to 10 MHz and higher.

19. See Figure 2-29.

1 KHz
V
AUDIO AUDIO BALANCED SIDEBAND POWER
INPUT MIXER
AMPLIFIER MODULATOR FILTER AMPLIFIER

•500 KHz •5 MHz

CARRIER HETERODYNE
OSCILLATOR OSCILLATOR

Figure 2-29
Amplitude Modulation 2-55

20. The frequenciesat point A will be 501 kHz and 499 kHz. This is a
double sideband, suppressed carrier signal.

For an upper sideband output, point B will be 501 kHz. If the sum
mixing product is used, point C will be 5.501 MHz.

If the difference mixing product is used, point C will be 4.499 MHz.


The output will then be a lower sideband signal.

21. The three methods of sideband selection are: (1) use two separate
filters, (2) shift the carrier oscillator frequency to either side of the
filter passband, and (3) use frequency inversion during the
heterodyne process.

22. DC input power= I x E


= 3.4 A x 28 V
= 95.2 Watts

The ratio of PEP to average power in a two-tone SSB signal is 2:1.

PEP = 2 x Average DC input power


= 2 x 95.2 W
= 190.4 Watts
2-56 UNIT TWO
Unit 3

I AM RECEIVERS
3-2 UNIT THREE
L
AM Receivers 3-3

INTRODUCTION
The original "wireless" receivers consisted solely of a simple detector.
However, with more and more transmitters operating, the receiver had to
select one signal from the many that were on the air. This required one or
more tuned circuits in conjunction with the detector. Eventually, receiv-
ers were required to pick up signals from greater distances, so one or more
tuned RF amplifiers were added to the detector to improve sensitivity and
selectivity.

However, the tuned RF receiver had several problems. So in 1918, the


superheterodyne receiver was invented by Edwin Armstrong, who also
developed FM broadcasting. The superheterodyne receiver rev-
olutionized the field of radio communications, allowing truly long dis-
tance reception for the first time. Even today, virtually all receivers,
including TV receivers, use the superheterodyne technique.

In this unit, you will study the basic types of AM detectors, the tuned RF
receiver, and the superheterodyne receiver.

The "Unit Objectives" on the next page state the goals of this unit. Review
this list now and be sure you can satisfactorily complete all the objectives
before you take the unit exam.
II

3"4 UNIT THREE


AM Receivers 3-5

UNIT OBJECTIVES
When you have completed this unit, you should be able to:

1. List four types of AM detectors and state the characteristics of each.


2. Identifythe block diagrams of a tuned RF receiver and a
superheterodyne receiver.

3. List the disadvantages of the tuned RF receiver.

4. Define receiver sensitivity and selectivity.

5. List the stages of a typical superheterodyne receiver and state the


function of each.

6. State the factors that must be considered when selecting an inter-


mediate frequency.

7. Find image frequency when given the received and intermediate


frequencies.

8. State the purpose of an AGC circuit.

9. Define several important receiver specifications.

10. Analyze the operation of a complex receiver from its schematic


diagram and draw a block diagram of the receiver.
o

3-6 UNIT THREE

AM DETECTORS
As shown an audio signal is impressed onto a carrier wave in the
earlier,
form of amplitude variations. It is then amplified and applied to a trans-
mitting antenna. As will be shown in a later unit, this modulated signal is
then radiated and propagated, and a small fraction of it is collected by the
receiving antenna. The receiver must amplify this extremely weak signal
and, since the signal is one of many collected by the antenna, the receiver
must select the desired signal while rejecting all others. Finally, since
modulation took place in the transmitter, demodulation must be per-
formed in the receiver to recover the original modulating signal. The
circuit that performs this function is called a demodulator or a detector.

The Diode Detector

The most popular AM demodulator is the diode detector. This circuit is


very simple and is used in virtually all AM receivers. Its purpose is to
recover the envelope from the AM waveform.
The diode detector shown in Figure 3-1. The switch, S lt is included
is

merely for explanation. The input to the circuit is the AM waveform that
has been selected and amplified by previous stages in the receiver. It is
applied to diode D lt which acts as a half- wave rectifier. The positive half
cycles cause Dj to conduct, developing positive pulses across R D cuts x
.
x

off the negative half cycles of the RF input. The center waveform shows
the voltage developed across Rj if Sj is open.

* <- AUDIO
OUTPUT

Figure 3-1
The Diode detector.
AM Receivers 3-7

When S is closed,
x Q
is placed in parallel with R,. Ci quickly charges

through Dj to the peak of each positive pulse. Between pulses, C attempts


x

to discharge through R^ However, the RC time constant is chosen so that


Q discharges only slightly. The result is that the voltage across Q
follows
the envelope of the AM
waveform. Thus, the output looks like the upper
envelope with a small amount of ripple. Normally, the carrier frequency
is many times higher than the envelope frequency, and therefore, the

ripple is not noticeable.

Figure 3-1 illustrates the diode detector's operation in the time domain.
Let's analyze its operation in the frequency domain.

The AM input consists of three frequency components: the carrier, the


upper sideband, and the lower sideband. These signals are applied to D!
and are mixed across its nonlinear resistance. The difference signal is the
modulating information. The next step is to separate this low frequency
signal from the high frequency RF. This is accomplished by C, which acts
as a short circuit to ground for the RF signals and a high reactance for the
audio signals.

The final step in AM detection is to separate the audio from the DC


component. This is done quite simply using a coupling capacitor such as
Q in Figure 3-2.

Figure 3-2
Complete diode detector circuit.
3-8 UNIT THREE

The Transistor Detector

Many low cost transistor receivers use a germanium diode as the detector.
However, due to the limited RF gain usually preceding the detector, a
transistor is sometimes used as the demodulator to provide additional
gain. A transistor can perform as a detector, if it is biased for class B
operation. In this way, the AM signal is both rectified and amplified at the
same time.

Figure 3-3 shows a typical transistor detector. Resistors R! and R2 form a


biasing network which sets the circuit for exactly class B operation. R3 is
the collector load resistor, while C, filters out the RF components. This
leaves the audio signal to be coupled to the output by Q.

The transistor detector offers a method of achieving additional receiver


gain. However, in anything but very low cost receivers, the diode detector
is chosen for its simplicity and excellent performance.

AUDIO
*HI o
OUTPUT

AM
INPUT

Figure 3-3
The transistor detector.
AM Receivers 3-9

The Synchronous Detector

The diode and transistor detectors recovered the modulation envelope


from the RF signal by However, with double sideband,
rectification.
suppressed-carrier AM, the envelope does not duplicate the modulating
signal. This is shown in Figure 3-4 for a 10 kHz modulating tone and a 100
kHz carrier. Due to the absence of the carrier, the two sidebands combine
to produce the envelope shown in Figure 3-4D. This envelope actually
has a frequency of twice the modulating frequency or 20 kHz. You can see
that this signal could not possibly be demodulated using a diode detec-
tor. The recovered signal would have severe distortion and be at twice the

frequency of the original modulating information. Therefore, a special


circuit called a synchronous detector must be used to demodulate a
suppressed carrier AM signal.

MODULATING
TONE
10kHz 0V

B
UPPER
SIDEBAND
110kHz

c
LOWER
SIDEBAND
90kHz

DOUBLE
SIDEBAND
WAVEFORM

Figure 3-4
When the carrier is removed, the sidebands combine as
shown above.
I o

3-10 UNIT THREE

A typical synchronous detector is shown in Figure 3-5. It consists of two


diodes rather than the single diode of the simple detector. It also has a
separate "carrier" input. This is a signal that is generated by the receiver.
As will be seen later, this "carrier" signal must not only be the same
frequency as the original suppressed carrier but must also be exactly in
phase with the original.

In Figure 3-5, the carrier input causes diodes D, and D2 to switch. When
the carrier input results in a negative potential at the top of T, and a
positive potential at the bottom, both D, and D 2 conduct. However, no
current flows to the output or G, because all current remains in the D, D 2 ,
,

and T, loop. In effect, the carrier input is balanced and, therefore does not
reach the output. On
the other half cycle of the carrier, D, and D 2 are
reverse biased and no current flows. The combinations of R,C, and I^Q
form diode loads and ensure that the diodes are switched on for only a
small fraction of each cycle. C 4 filters out any RF components that reach
the output.

1 R,

W f~
vw

OUTPUT


>

#-L
2 HI
— <

MODULATED
SIGNAL
INPUT

Figure 3-5
A synchronous detector.

The modulated signal input is of a sufficiently low level that it cannot


forward bias D, and D2 . This means that the only time the modulated
signal can reach C4 and the output is when the carrier input forward
biases the diodes. What actually happens is that the carrier input samples
small portions of the modulated signal as shown in Figure 3-6. Here,
everytime the carrier input goes positive, the diodes are turned on and the
suppressed carrier signal passes to the output. Note that the carrier
samples the positive peaks during one half cycle of the input and the
negative peaks during the other half cycle. The result is an accurate
duplication of the modulating signal at the output.
AM Receivers 3-11

You can see from the waveforms of Figure 3-6 that any slight carrier phase
or frequency variation will shift the sampling point and cause severe
distortion. For this reason, the carrier signal must be held exactly in
phase with the original carrier. This presents the most serious drawback
to suppressed-carrier AM. Elaborate circuits are required at the receiver
to generate the carrier signal in the correctphase and frequency. How-
ever, suppressed-carrier AM used in the stereo FM and color TV
is

broadcasting systems. In these systems a special synchronizing signal is


also transmitted. The receiver recovers this synchronizing signal and
uses it to generate its own carrier signal in the correct phase and fre-
quency.

..SUPPRESSED
|
\\ CARRIER
SIGNAL

h — -I
!
L — -| 1
\
(-- CARRIER

OUTPUT
WAVE

Figure 3-6
Waveforms for the synchronous detector.
3-12 UNIT THREE

The Product Detector

The simple diode detector is also unsuitable for demodulating a single-


sideband signal. This is because, with SSB, the waveform's envelope
corresponds only to the amplitude of the modulating signal, not the
frequency. Infact, if an SSB transmitter is modulated by a constant

amplitude audio tone whose frequency is varied, the SSB envelope will
be constant but its frequency will vary. Therefore, an SSB demodulator

must be sensitive to both frequency and amplitude variations.

In product detection, an internal reference signal is generated and the


SSB signal is compared with
it. This reference signal is essentially a

"substitute" carrier for the single sideband. The product detector itself is
nothing more than a nonlinear mixer. It can take on many forms but is
usually a balanced modulator circuit or a dual-gate MOSFET.

A typical product detector is shown in Figure 3-7. It uses a diode-ring


balanced modulator as the nonlinear mixer. The internally generated
carrier, which is also called the BFO for Beat Frequency Oscillator, is

applied to the center tap of T,. The incoming SSB signal is applied to the
primary of Tj The two signals are then mixed across the diode ring. Now,
.

if the carrier signal is within approximately 50 Hz of the original suppres-

sed carrier, the difference signal will be the original modulating audio.
R R2 C,C2 form an RC low pass network which removes any RF component
1

and allows only the audio signal to pass to the output. Thus, the original
information has been recovered.

BFO/ CARRIER

Q
l

SSB AUDIO
SIGNAL OUTPUT

Figure 3-7
Balanced modulator product detector.
AM Receivers 3-13

As you can see, the requirements for the internally generated carrier for
SSB are not nearly as stringent as they are for double-sideband
suppressed-carrier AM. In suppressed-carrier AM, the local carrier has to
be exactly the same frequency and phase as the original. In SSB, it has to
be only within 50 Hz of the original. This greatly simplifies the SSB
receiver and, along with bandwidth considerations, is the primary reason
for the superiority and widespread use of SSB instead of double-sideband
suppressed-carrier AM.

Since the local carrier is not the same frequency and phase as the original,
the audio quality of SSB does suffer somewhat. However, the demod-
ulated signal is still intelligible and, therefore, quite effective for voice
communications. Lower audio quality is a small price to pay for the many
advantages that single-sideband offers.
3-14 UNIT THREE

Self-Review Questions

1. The most popular AM demodulator is the

2. A transistor can be used as an AM detector to provide additional


receiver

3. When a transistor is used as an AM detector, it is biased for class


operation.

4. The AM diode detector is normally chosen over the transistor


detector for its and

5. What type of detector must be used to demodulate double-sideband


suppressed-carrier AM signals?

6. What are the requirements for the local carrier signal when demod-
ulating a double-sideband suppressed-carrier signal?

7. What are the primary applications of double-sideband


suppressed-carrier AM? .

8. What type of demodulator is used for single-sideband signals?

9. What are the requirements for the local carrier signal when demod-
ulating a single-sideband signal?
AM Receivers 3-15

Self-Review Answers

1. The most popular AM demodulator is the diode detector.

2. A transistor can be used as an AM detector to provide additional


receiver gain.

3. When a transistor is used as an AM detector, it is biased for class B


operation.

4. The AM diode detector is normally chosen over the transistor


detector for its simplicity and excellent performance.

5. A synchronous detector must be used to demodulate double-


sideband suppressed-carrier AM
signals.

6. When demodulating a double-sideband suppressed-carrier signal


the local carrier must be exactly the same frequency and in phase
with the orignal suppressed carrier.

7. The primary applications of double-sideband suppressed-carrier


AM are in the stereo FM and color television broadcasting systems.
8. A product detector is used to demodulate single-sideband signals.

9. When demodulating a single-sideband signal the local carrier must


be within approximately 50 Hz of the original suppressed carrier
frequency.
3-16 UNIT THREE

TUNED RF RECEIVER
If a simple diode detector was connected directly to an antenna, there
would be no means of selecting a specific signal. In fact, all signals
present at the antenna would be demodulated and the output would be a
jumbled mess. One way to select thedesired signal is to connect a tuned
RF amplifier between the antenna and detector. However, the bandwidth
of a single RF amplifier is not narrow enough to reject all the unwanted
signals. Therefore, several RF amplifiers must be used, as shown in
Figure 3-8. This decreases the bandwidth to a point where only one signal
will pass to the detector. The RF amplifiers will also increase the
amplitude of the desired signal before it is applied to the detector. Thus,
the additional RF amplifiers increase both the sensitivity and selectivity
of the receiver.

Sensitivity is a measurement weak sig-


of a receiver's ability to receive
nals. For example, if a receiver can "pick up" an extremely weak signal, it
is said to have good sensitivity. The amount of gain a receiver has

determines its sensitivity. Therefore, the more gain obtained in the RF


section, the greater the receiver's sensitivity is.

Selectivity measure of a receiver's ability to select one signal while


is a
rejecting all others. In a tuned RF receiver, this is determined by the
bandwidth of the RF amplifiers. Therefore, as the bandwidth is reduced,
selectivity is increased.

The receiver shown in Figure 3-8 has four RF amplifier stages. Its sen-
sitivityand selectivity are adequate for the AM broadcast band of 535 to
1605 kHz. However, it is extremely difficult to get all the RF amplifiers to
tune to exactly the same frequency as the dial is tuned across the band.
Another problem is that the bandwidth of the RF amplifiers increases as
the operating frequency increases. As a result, the selectivity at 1600 kHz
is not as good as 540 kHz. One other problem is the extreme
it is at
difficulty of designing RF amplifiers that provide high gain and are
tunable over the required range. Because of these and other problems, the
tuned RF receiver was abandoned in favor of the superheterodyne re-
ceiver. The next section discusses the superheterodyne receiver in detail.
ANTENNA

V 535- 1605kHz 535- 1605kHz 535-1605kHz 535- 1605kHz SPEAKER

RF RF RF RF AUDI'
-J • DETECTOR
AMPLIFIER AMPLIFIER AMPLIFIER AMPLIFIER AMPLIFIER
IER \\\

TUNING
Figure 3-8
The tuned RF receiver.
AM Receivers 3-17

Self-Review Questions

10. A tuned RF receiver uses one or more in front of the


detector.

11. What is receiver sensitivity?

12. What is receiver selectivity?

13. List the disadvantages of the tuned RF receiver.


.

3-18 UNIT THREE

Self-Review Answers

10. A tuned RF receiver uses one or more RF amplifiers in front of the


detector.

11. Receiver sensitivity is a measure of the receiver's ability to receive


weak signals.

12. Receiver selectivityis a measure of the receiver's ability to select


one signal while rejecting all others.

13. The disadvantages of the tuned RF receiver are:

1. The selectivity changes when the operating frequency


changes.

2. It is difficult to design high gain, tunable RF amplifiers.

3 It is difficult to get all the amplifiers to tune to exactly the same


frequency.
AM Receivers 3-19

SUPERHETERODYNE RECEIVERS
Most disadvantages of the tuned RF receiver disappear if the receiver is
fixed-tuned to a single frequency. The superheterodyne receiver uses this
principle by converting the selected RF signal, regardless of frequency, to
a predetermined fixed frequency where amplification and further selec-
tion take place.

Block Diagram

The block diagram of a typical, standard AM broadcast superheterodyne


receiver is shown in Figure 3-9. The RF amplifier is similar to that used in
the tuned RF receiver. It provides signal gain and a limited degree of
selection. However, the majority of the receiver's selectivity and gain are
obtained from the later stages. In fact, in some receivers, the RF amplifier
is eliminated altogether.

ANTENNA
V 535- 1605kHz 455kHz 455kHz SPEAKER
RF — IF IF AUDIO
AMPLIFIER
•>
MIXER
AMPLIFIER AMPLIFIER
DETECTOR
AMPLIFIER
HO
!
LOCAL
990- 2060kHz
OSCILLATOR
i

f ^3 L j

TUNING

Figure 3-9
Standard AM broadcast superheterodyne receiver.

The mixer and local oscillator are the key to the superheterodyne action.
They convert the selected RF signal to a fixed frequency for further
amplification. In this case, the fixed frequency is 455 kHz which is a
commonly used intermediate frequency or IF. It is called an "inter-

mediate" frequency because it is between the original radio frequency


and the detected audio frequency.
3-20 UNIT THREE

To convert the incoming RF signal to 455kHz it must be mixed or


heterodyned with a signal that is exactly 455 kHz above or below it. That
is, the difference between the selected RF signal and the local oscillator

must be 455 kHz. In the receiver of Figure 3-9 the local oscillator is tuned
above the incoming signal. Therefore, if the desired signal is at 1000 kHz,
the tuning knob would be adjusted to 1000 kHz. This would adjust the RF
amplifier to 1000 kHz and allow a relatively small band of frequencies to
be amplified and applied to the mixer. The tuning knob would also adjust
the local oscillator to exactly 455 kHz above the incoming RF signal or
1000 kHz + 455 kHz = 1455 kHz.

Both the RF and local oscillator signals are then applied to the mixer. The
mixer is any nonlinear circuit and could even be a balanced modulator.
The outputs of the mixer are the original signals and the sum and differ-
ence mixing products. Since the mixer's output and several following
amplifiers are fixed-tuned to 455 kHz, only the difference signal, in this
example 1455 kHz - 1000 kHz or 455 kHz, leaves the mixer and is
amplified.

The IF amplifiers are actually fixed-tuned RF amplifiers. There are usu-


allytuned resonant circuits on both the input and output of each stage.
This assures both high gain and narrow bandwidth which means better
receiver sensitivity and selectivity. Since they are fixed-tuned, their gain
and bandwidth are constant. There are also no tuning problems since
each stage is individually "peaked" only once, during its initial align-
ment at the factory. The selection of the intermediate frequency itself
involves several factors which are discussed later in this unit.

The amplified IF signal is then coupled to the detector where the original
modulating information is recovered. In the standard AM receiver of
Figure 3-9, the detector is a simple diode demodulator. A single-sideband
receiver would require a product detector and a BFO. The detected audio
is then amplified and applied to the speaker.
AM Receivers 3-21

Frequency Conversion Circuits

A typical receiver front-end circuit is shown in Figure 3-10. Qj is the RF


amplifier, Q2 is the mixer, and Q3 is the local oscillator. The RF amplifier

istuned to the desired input frequency by C,. As shown in the schematic,


Cj is also mechanically coupled to Q. Therefore, Q
simultaneously tunes
the local oscillator to its correct frequency. Qresonates with L^ to form
the tank for the oscillator while L2 provides the regenerative feedback.

L3couples the local oscillator signal to the emitter of Q>, while the input
RF signal is coupled to the base. These two signals are then mixed across
the nonlinear resistance of Q>. The output of the mixer is applied to the
primary of T3 which is tuned to the intermediate frequency. Thus, the
difference signal, which is the intermediate frequency, is separated and

applied to the IF stages where it is further amplified.

ANTENNA
O )r

Figure 3-10
Frequency conversion circuit.
3-22 UNIT THREE

A variation of the previous circuit is shown in Figure 3-11. However, in


this circuit, transistor Q2 performs the dual function of local oscillator
and mixer. The key to this circuit is transformer T2 Here, Lj couples the
.

collector signal of Q2 to the resonant tank of L2 and Q> C3 then couples the.

oscillations of L2Q to the emitter of Q2. The transistor amplifies this


signaland the continued feedback of L sustains the oscillations. Thus, Q2
t

functions as an Armstrong oscillator.

When a signal present at the antenna, Q, and the tuned circuit of


is C^
select and amplify the desired signal. It is then coupled to the base of Q2
and mixed with the local oscillator signal. The output of Q2 is coupled
through Li to the primary of T3 which is tuned to the intermediate
frequency. Thus, the difference signal is separated and applied to the IF
stages. This type of circuit is called an autodyne. It is usually used in low
cost portable receivers, and is often used in conjuction with the transistor
detector discussed earlier. Both circuits perform dual functions and,
thus, allow simplicity and lower cost.

ANTENNA

BIAS

Figure 3-11
Autodyne conversion circuit.
AM Receivers 3-23

A high performance mixer circuit often found on HF, VHF, and UHF
receivers is shown in Figure 3-12. It uses a dual-gate MOSFET as the

active element. The MOSFET has a high input impedance and, therefore,
causes less loading on the previous circuit. In this mixer the RF signal is
applied to gate 2 while the local oscillator signal
is applied to gate 1. This

minimizes any interaction between the two signals. The mixing action is
produced by biasing the MOSFET into its nonlinear region. The differ-
ence output signal is then separated by the tuned primary of T2 .

LOCAL
OSCILLATOR
T

Figure 3-12
A MOSFET mixer.

Intermediate Frequency Selection

There are several factors that must be considered when selecting an


intermediate frequency. Many of these factors revolve around the equa-
tion BW = F /Q. This states that bandwidth is equal to the operating
frequency divided by the circuit Q. Therefore, for good selectivity, or a
narrow bandwidth, the circuit Q must be high and the operating fre-
quency low. Thus, it is generally desirable to use a relatively low inter-
mediate frequency. Other advantages of a low IF are better circuit stabil-
ity and higher gain stages. This is due to fewer circuit problems at low
frequencies. For example, as frequency increases, circuit resistive, radia-
tion, and dielectric losses increase; stray capacitance and inductance
effects increase; and stray feedback causes instability. Therefore, low
intermediate frequencies are desirable.

Another factor to be considered when selecting an intermediate fre-


quency is the required bandwidth. For example, if an AM broadcast
receiver uses a 60 kHz IF with a Q of 60, the bandwidth will be 1 kHz. This
is hardly practical for receiving an AM signal whose bandwidth is at least

10 kHz! Therefore, the system bandwidth must be taken into account.


3-24 UNIT THREE

The final factor to be considered is image frequency rejection. The image

frequency is an undesired RF signal which, when mixed with the local


oscillator, will also produce the correct IF value. As an example, suppose
the IF is 200 kHz and the desired signal is 4.2 MHz. If the local oscillator is
tuned above the incoming signal, it must be at 4.2 MHz + 0.2 MHz = 4.4
MHz. Therefore ,when the local oscillator and RF signal are mixed, the
difference signal will be 200 kHz which is the IF. However, there is
another signal that will produce a 200 kHz IF signal. It is located 200 kHz
above the local oscillator at 4.6 MHz. Therefore, when this 4.6 MHz signal
is mixed with the 4.4 MHz local oscillator the difference is 200 kHz. Thus,

this image frequency will be amplified just as if it were the desired signal.

Let's try another example of finding the image frequency. 455


If the IF is

kHz and the received signal is 1110 kHz with the local oscillator tuned
above the RF signal, what is the image frequency? The local oscillator
frequency is 1110 kHz + 455 kHz = 1565 kHz. Therefore, the image
frequency is 1565 kHz + 455 kHz = 2020 kHz.

From the previous examples, you can see that the image frequency is

separated from the desired signal by twice the intermediate frequency.


Rejection of the image signal depends solely on the selectivity of the RF
amplifier and mixer stages. Also, the higher the IF is, the further the
image signal will be from the desired signal. Therefore, to improve image
rejection, a higher intermediate frequency should be used. This is in
conflict with the previous factors considered. Therefore, the selection of
an intermediate frequency must be a compromise.

Once all of these factors were taken into account, several "standard"
intermediate frequencies were chosen. As an example of this, AM broad-
cast receivers (535 - 1605 kHz) generally use an IF of 455 kHz. This
frequency provides excellent Q high gain and good selectivity. It is
for
also high enough to place the image frequency well outside the RF
amplifier's passband. FM broadcast receivers use an IF of 10.7 MHz. This
was chosen to provide sufficient bandwidth, up to 200 kHz, for the FM
signal. It also places the image frequency 21.4 MHz away from the desired
RF signal. Television receivers use an IF of approximately 40 MHz. This
allows the extremely wide TV bandwidth of 4 MHz to be accomodated.
AM Receivers 3-25

IF Amplifiers

An IF amplifer is actually nothing more than a fixed tuned RF amplifier.


Figure 3-13 shows a typical amplifier tuned to 455 kHz. The only differ-
ence between this circuit and a typical RF amplifier is the double-tuned
transformers Tj and T2 The resonant circuits in both primary and secon-
.

dary improve the selectivity and circuit gain. Apart from this, the circuit
is a straightforward RF amplifier with Q
providing neutralization for
added stability.

The selectivity of a receiver can be tailored to the exact requirements


using a bandpass filter as part of the IF section. This technique is used
primarily in communications receivers where narrow bandwidths are
used, in FM receivers where the high 10.7 MHz IF does not provide
adequate selectivity, and it is starting to appear in TV receivers where it

eliminates tedious tuning and provides optimum IF response.

OUTPUT

INPUT

Figure 3-13
An IF amplifier.
3-26 UNIT THREE

Automatic Gain Control

When you tune across the AM Broadcast band, it is not uncommon to hear
an extremely weak station at one point and a strong local station at the
next frequency. To compensate for these level variations, Edwin
Armstrong invented automatic volume or gain control. This circuit au-
tomatically varies the gain of the receiver to keep the volume at a rela-
tively constant level regardless of the input signal.

A simple automatic gain control (AGC) circuit is shown in Figure 3-14.


Actually it is an AM diode detector with R 2 and Q> added. These compo-
nents take the detected and filtered audio signal and filter it still further to
remove the audio signal. What remains is a DC voltage that is directly
related to the input signal amplitude. If a strong input signal is present,
the AGC voltage will be high. Just the opposite is true for a weak signal.
This voltage is then used to control the receiver gain. While a positive
AGC voltage is shown in Figure 3-14, negative voltage is also used and is
obtained by reversing the connection of D x
.

AGC
VOLTAGE
J^

Figure 3-14
A simple automatic gain control circuit.
»

AM Receivers 3-27

To see how the AGC voltage controls receiver gain, consider the block
diagram shown in Figure 3-15. Here, the DC voltage derived from the
detector is applied to both IF amplifiers and the RF amplifier. This

voltage is used to set the bias of these stages. Therefore, by varying the
stage bias, the gain of each stage can be controlled. For example, when a
strong signal is present, the AGC voltage is high and it decreases the gain
of the RF and IF amplifiers. If the signal fades or a weak signal is tuned in,
the AGC voltage is reduced and the RF and IF amplifier's gain is in-
creased. The result is a relatively constant audio output level.

You can see that the AGC circuit is a valuable addition to a receiver.
However, some modes such as SSB do not have constant
of operation
signals as with AM. This makes it difficult to derive an AGC voltage. In
this case, special AGC circuits have been developed. Although, in some
cases, the AGC circuit is turned off altogether and a manual "RF gain"
control is used.

ANTENNA

V 455kHz 455kHz SPEAKER

AMPLIFIER
1
i
MIXER
i i
IF

AMPLIFIER
1 i
IF

AMPLIFIER
1 ,
— DETECTOR
—* AUDIO
AMPLIFIER
M
LOCAL
OSCILLATOR

I I

AP r

Figure 3-15
A superheterodyne receiver with AGC.
4

3-28 UNIT THREE

An AGC circuit used with SSB receivers is shown in Figure 3-16. Since
there is no carrier present in SSB, the AGC system must derive its voltage
from some other part of the signal. The circuit of Figure 3-16 uses the
demodulated audio to obtain the AGC voltage. Here, the audio is
amplified by the operational amplifier and then rectified by the voltage
doubler circuit of DjDa and QQ
Therefore, the DC voltage across
• is Q
directly proportional to the strength of the SSB signal. However, if no
audio is transmitted this voltage drops to zero. In fact, in between words
and sentences, when the speaker pauses, C2 attempts to discharge
through Rj However, the time constant of QR! is sufficient to prevent the
.

AGC voltage from dropping during normal pauses in speech.

DETECTED
O—/w—
AUDIO

r T T
I

+v

AGC
OUT
3.0V

Figure 3-16
Audio derived AGC circuit.

JFET Qj and transistor Q> form an interface between the AGC detector and
the receiver circuits being controlled. Q/s high input impedance allows
it to amplify the DC voltage across QRj without affecting their time
constant. The output of Qj is applied to the base of Q> and when the
voltage is sufficient, it Up until this time the AGC voltage is
turns on Q>.
held at a constant 3 V by zener diode D 3 When Q> turns on, it effectively
.

turns off the zener by reducing the voltage below 3 V. Q> now controls the
AGC voltage and any increase in the detected signal at Q will cause Q> to
conduct more, which lowers its collector voltage and, hence, the AGC
voltage. Thus, as the received signal increases, the AGC voltage decreases
which reduces the receiver gain and holds the audio output relatively

constant.
» »

AM Receivers 3-29

Double Conversion Receivers

As mentioned earlier,choosing an intermediate frequency is usually a


compromise. A low IF is desired because of its better gain, stability, and
narrow bandwidth. However, a high IF is desirable to improve image
rejection. The double-conversion designed to take advantage
receiver is

of both high and low IFs by using a high "first" IF and then converting to
a low "second" IF.

A simple double-conversion receiver is shown in Figure 3-17. This re-


ceiver uses two mixers, two local oscillators, and two intermediate fre-
quencies. The first IF is selected to place the image as far away from the
desired signal as possible. In this case, the 9 MHz first IF places the image
18 MHz away from the desired signal. Therefore, the image is well
outside the bandwidth of the RF amplifier and is greatly attenuated.

ANTENNA

V 9MHz 455kHz 455kHz


SPEAKER
AUDIO
— BANDPASS 2ND
— IF IF —*
RF
AMPLIFIER
1ST
MIXER FILTER MIXER
l ,
AMPLIFIER AMPLIFIER
DETECTOR
AMPLIFIER!
H\
t
LOCAL CRYSTAL
OSCILLATOR OSCILLATOR
i 9.45! iMHz

TUNING

Figure 3-17
Double conversion receiver.

To demonstrate how the receiver operates, we must follow the signal


from the antenna to the speaker. As an example, suppose you want to
receive the 5 MHz signal of the standard time and frequency station,
WWV. The tuning knob is set to 5 MHz which tunes the RF amplifier to 5
MHz and the local oscillator to 14 MHz. This places the image frequency
at 23 MHz, well outside the bandpass of the RF amplifier.
3-30 UNIT THREE

The incoming signal is amplified and then applied to the mixer along
with the local oscillator signal. At the mixer output, the difference signal
of 9 MHz is selected by the bandpass filter and applied to the second
mixer. The bandpass filter may be an LC filter or a crystal filter for
improved selectivity. In this receiver there is no first IF amplifier since
most of the receiver's gain is supplied by the second IF.

Since a 9 MHz signal always appears at the second mixer, a crystal-


controlled "second" local oscillator is used. Its frequency of 9.455 MHz is
mixed with the 9 MHz input signal and the difference frequency, 455
kHz, is selected by the IF amplifiers.

The IF amplifiers increase the signal amplitude and apply it to the


detector. The detector demodulates the signal and recovers the original
audio. This audio signal is then amplified and applied to the
loudspeaker.
AM Receivers 3-31

Receiver Specifications

In the 1920's and 30's the most important receiver specification was
sensitivity. The receiver had to be sensitive enough to "pick up" the weak
signals of the relatively few transmitters that were "on the air." However,
in the 1940's and 50's there were many more transmitters, and receiver
selectivity became much more important. Today, there are literally mil-
lions of transmitters and, while sensitivity and selectivity are still impor-
tant, many new specifications are recognized as important receiver per-
formance measurements. Let's look at these specifications in detail.

SENSITIVITY

One of themost important receiver specifications is sensitivity. It is a


measure of the amplitude of the weakest signal that can be received, with
a specified signal-to-noise ratio. That is, the signal must be higher than
the noise level by a certain ratio.

Receiver sensitivity is actually a measure of how much signal it takes to


overcome the internal receiver noise. This is the reason for using a

specified signal-to-noise ( -z-r ) ratio. In fact, signal-to-noise ratio mea-

surements have been replaced by a more correct measurement of signal-

plus-noise-to-noise (
—C n—)
N
_i_ XT
ratio. This ratio takes into account that the

noise is still present, along with the signal. The signal is actually added to
the noise and must overcome it.

Receiver sensitivity determined by first measuring the noise output


is

with no signal present. The output from an RF signal generator is then


applied to the receiver input. The generator's output level is now in-
creased until the receiver's output increases by 10 dB. Once this is
achieved, the RF signal generator's output voltage is measured. Typical
values are 1.0 to 0.5 fiV of RF signal for a +10 dB signal-plus-noise-to-
O T
_i_ IV.

noise ( — —
r-= ) ratio. Thus, the receiver requires approximately 1.0 /xV

of signal to overcome the internal noise by 10 dB and to provide adequate


reception.
3-32 UNIT THREE

SELECTIVITY

Another important receiver specification is selectivity. This is a measure


of the receiver's ability to select a desired signal The
and reject all others.
only way to measure selectivity is to determine the receiver's bandwidth.
A typical receiver selectivity curve is shown in Figure 3-18. This shows
that the receiver bandwidth is 3 kHzatthe -6 dB point. That is, at 1.5 kHz
either side of the center of the passband, the received signal's amplitude
drops 6 dB. At 3 kHz either side of the center, the signal drops 60 dB.
Therefore, the bandwidth at -6 dB is 3 kHz, and at -60 dB, is 6 kHz. This
selectivity curve is primarily due to the bandpass filter in the IF section,
although the RF and IF amplifiers also play an important part in improv-
ing selectivity.

FREQUENCY

Figure 3-18
Typical selectivity curve.
AM Receivers 3-33

IMAGE REJECTION

Image rejection ismeasure of the receiver's ability to reject a strong


a
signal at the exact image frequency. To measure image rejection, you
connect a signal generator to the receiver and set it to the image fre-
quency. For example, if the receiver's first IF is 9 MHz, you would set the
receiver to a specific received frequency, let's say 5 MHz. This places the

image at 5 MHz -I- 18 MHz or 23 MHz. So you would set the signal
generator to 23 MHz. Then you would continue as though you were
making a receiver sensitivity measurement. You would increase the
generator's output level until you achieve a 10 dB (S + N)/N ratio. Now,
comparing the signal generator's present output level to the normal
sensitivity of the receiver will give you the image rejection.

As an example, suppose the receiver's normal sensitivity is 1 /xV for a 10


dB (S + N)/N ratio and it takes 100 /xV at the image frequency for a 10 dB
(S+N)/N ratio. The image rejection is:

Image rejection = 20 Log =-


E2

100 fiV
= 20 Log
1/xV

= 20 Log 100

= 20 x 2

= 40 dB

Thus, in this example, the image rejection is 40 dB. In other words, the
image signal will be 40 dB below the desired signal. Image rejection is
basically a measure of the selectivity of the RF amplifier.
3-34 UNIT THREE

IF REJECTION

IF rejection is a measure of the receiver's ability to reject a strong input


signal at its own intermediate frequency. For example, if the receiver's IF
is 455 kHz, an input signal at 455 kHz should be rejected.

To measure you must set the signal generator to the re-


IF rejection,
ceiver's intermediate frequency and increase its output level until you
achieve a 10 dB (S+N)/N ratio, then compare the generator's voltage
output to the receiver's normal sensitivity, just as in the image rejection
measurement. Typical IF rejection ratings are 40 to 60 dB. This means
that a signal at the receiver's IF will be 40 to 60 dB below the desired
signal.

DYNAMIC RANGE

Dynamic range is a measure of the receiver's ability to receive both very


weak and extremely strong stations without introducing distortion to
either. It is a function of the RF amplifier's linearity and the ability of the
AGC system to control receiver gain.

To measure dynamic range, you first measure the sensitivity of the


receiver for a 10 dB (S + N)/N ratio. This is the weak signal limit. Youthen
increase the input RF signal until the output audio starts to distort. This
marks the strong signal limit. The ratio between the weak signal limit and
strong signal limit in dB is the dynamic range. Typcial values range for
100 to 120 dB and greater.

m
AM Receivers 3-35

Self-Review Questions

ANTENNA

L AUDIO
AMPLIFIER
SPEAKER

<

Figure 3-19

14. Fill in the blocks of the diagram of a superheterodyne receiver in


Figure 3-19.

15. State the function of each stage in Figure 3-19.

16. What is an autodyne circuit?

17 . Dual-gate MOSFETs are often used in superheterodyne receivers as


3-36 UNIT THREE

18. What factors must be considered when selecting an intermediate


frequency?

19. What is the image frequency when the receiver is tuned to 9.77
MHz, the local oscillator is tuned above the desired signal, and the
IF is 455 kHz?

20. What are the standard IFs for the following types of receivers?

AM broadcast
FM broadcast
TV broadcast

21. An IF amplifier is actually a

22 .
What is the purpose of a bandpass filter in the IF stage of a receiver?

23. What is the purpose of an AGC circuit?

24. Why is an AGC circuit for SSB more complex than that for standard
AM?
AM Receivers 3-37

25. What is the purpose of using two IFs as in a double-conversion


receiver?

26. Define the following terms as they refer to receiver specifications.

Sensitivity:

Selectivity:

Image Rejection:

IF Rejection:

Dynamic Range:
3-38 UNIT THREE

Self-Review Answers

14. See Figure 3-20.

ANTENNA

SPEAKER
RF IF AUDIO
MIXER DETECTOR
AMPLIFIER AMPLIFIER AMPLIFIER
•m
*
LOCAL
OSCILLATOR

Figure 3-20

15. The RF amplifier selects the desired signal and provides minimal
amplification. The local oscillator and mixer convert the desired
RF signal to the intermediate frequency. The IF amplifier provides
amplification of the signal andimproves the selectivity by limit-
it

ing the bandwidth. The detector demodulates the IF signal.

16. An autodyne circuit uses a single transistor to operate as both local


oscillator and mixer.

17. Dual-gate MOSFETs are often used in superheterodyne receivers as


mixers.

18. When choosing an intermediate frequency you must consider the


bandwidth, gain, and stability required as well as the image rejec-
tion needed.

19. The local oscillator is at 9,770 kHz 455 kHz = 10,225 kHz.
-I-

Therefore, the image frequency is 10,225 + 455 kHz = 10,680 kHz.

20. The standard IFs for the following types of receivers are:

AM broadcast, 455 kHz


FM broadcast, 10.7 MHz
TV broadcast, 40 MHz

21. An IF amplifier is actually a fixed-tuned RF amplifier.


AM Receivers 3-39

22. A bandpass filter in the IF stage of a receiver shapes the frequency


response of the receiver and, therefore, determines its selectivity.

23. An AGC circuit controls the receiver's gain and, thereby, keeps the
audio volume relatively constant regardless of signal strength.

24. In standard AM, the carrier is always present and, thus, the AGC
voltage can be derived from the received carrier level. However, in
SSB there np signal present unless speech is present. For this
is

reason the AGC voltage must be derived during the speech and
held during the pauses. SSB AGC is normally derived directly from
the demodulated audio.

25. By using two IFs, the first IF can give optimum image rejection and
the second IF maximum selectivity and gain. Thus, the com-
promise in choosing a single IF is eliminated.

26. Sensitivity is a measure of the amplitude of the weakest signal that


can be received, with a specified signal-to-noise ratio.

Selectivity is a measure of the receiver's ability to select a desired


signal and reject all others.

Image rejection is measure of the receiver's


a ability to reject a
strong signal at the exact image frequency.

IF rejection is a measure of the receiver's ability to reject a strong


input signal at its own intermediate frequency.

Dynamic range is a measure of a receiver's ability to receive both


very weak and extremely strong stations without introducing dis-
tortion to either.
3-40 UNIT THREE

ANALYSIS OF A COMMUNICATIONS
RECEIVER
In this section, you will examine a sophisticated communications re-
ceiver in detail. The unit chosen is the Heathkit HR-1680 HF Amateur
Radio Receiver shown in Figure 3-21. It is designed to receive SSB and
CW (Morse Code) transmissions. An amateur receiver must perform
under the most demanding situations. This is because the amateur bands
are extremely crowded with both weak, distant stations and extremely
strong local stations. Therefore, its sensitivity, selectivity, and dynamic
range are constantly put to the test.

Figure 3-21
The Heathkit HR-1680 SSB/CW Receiver.

Specifications

Some of the specifications for the receiver shown in Figure 3-21 are given
in Table I. The first of these, frequency coverage, shows the frequency
ranges that the receiver covers. It is an amateur radio receiver covering
the HF "ham" bands from 3.5 MHz to 29 MHz.
AM Receivers 3-41

The sensitivity is rated at 0.5 /jlV for a 10 dB (S+N)/N ratio, which is


excellent. The IF selectivity is rated at 2.1 kHz at -6 dB and 7 kHz at -60
dB. This adequate selectivity for SSB operation. However, it is not as
is

good as the best mechanical filters offer, which is 2.1 kHz at -6 dB and
4.2 kHz at -60 dB. Although, cost becomes an important criteria as well,
and mechanical filters are valuable items. This receiver uses a simple
crystal filter as you will see later.

The overall gain of the receiver is an interesting specification. It shows


that, while the receiver is very sensitive, it also has enough gain so that
you can hear these weak signals. In this case, it states that for a 1.5 /xV
input, the audio output will be 0.25 W.

The dynamic range specification shows that the receiver can handle an
extremely wide range of signal levels without distortion. In this case, a
range of up to 120 dB or greater can be handled by the receiver.

The specifications also show that image rejection is 50 dB or greater and


IF rejection is 60 dB or better.

HR-1680
SPECIFICATIONS

Frequency Coverage (Megahertz) 3.5 to 4.0, 7.0 to 7.5, 14.0 to 14.5, 21.0 to 21.5,
28.0 to 28.5, 28.5 to 29.0.

Sensitivity Less than 0.5 microvolt for 10 dB signal-plus-noise


to noise ratio for SSB operation.

IF Selectivity 2.1 kHz minimum at 6 dB down,


7 kHz maximum at 60 dB down.

Overall Gain Less than 1.5 microvolt input for 0.25 watts
of audio output.

Dynamic Range 120 dB or greater.

Image Rejection 50 dB or better.

IF Rejection 60 dB or better.

TABLE I
3-42 UNIT THREE

Circuit Analysis

For this discussion, you must find the envelope of schematic diagrams
and remove diagram #1 for the HR-1680 Receiver.

First, let's follow the signal path which marked by a red dashed line.
is

The input signal first appears at the antenna and passes through the
antenna terminal to a series of "traps" on the front end circuit board.
These traps are tuned circuits which filter out or suppress certain unde-
sired signals. TC101 is a series-tuned trap that shorts out any signals at
the second IF of 3.395 MHz. L102 and C103 form a parallel resonant
circuit that traps out signals above 30 MHz. Inductors L103, L104, and
L105 form traps that attenuate any signals at the first IF of 8.395 to 8.895
MHz. The purpose of these traps is to improve both image rejection,
above 30 MHz, and IF rejection.

The input signal is then coupled to Q101 (the RF amplifier) through


either L106, L107, L108, or L109. L106 through L108 are tuned coupling
circuits that are selected by the diode switches via the band switch. The
correct circuit not only tunes the RF amplifier to the correct input fre-

quency, but also matches the low impedance antenna to the high impe-
dance MOSFET amplifier.

The RF signal is applied to gate 1 of the MOSFET while the dashed green
line shows that AGC voltage is applied to gate 2. We'll discuss the AGC
circuit in detail later. Q101 's gain can also be controlled by potentiometer
Rj, the RF gain control. It varies the MOSFETs gain by adjusting the
source voltage. The output tank circuit for the RF amplifier is chosen by
diode switches via the bandswitch. These are parallel resonant tanks
Llll, L112, L113, L114, and L115.

The signal is then coupled to gate 1 of Q102 the first mixer. A signal from
,

the HFO, or heterodyne frequency oscillator, is applied to gate 2. The


HFO is a crystal-controlled oscillator that is used to convert the incoming
signal to the first IF of 8.395 to 8.895 MHz. Actually it converts an entire
500 kHz band of signals to the first IF. However, some selectivity
of the
desired signal is provided by the preselector control which peaks the RF
amplifier's response.

The output of the first mixer is applied to an LC bandpass filter consisting


of L116, L117, and L118. This filter ensures that only the first IF of 8.395
to 8.895 MHz passes on to Q103, the second mixer. Note that the first IF
consists of only the LC filter. It is used strictly to improve image rejection
with most of the receiver's gain and selectivity coming from the second
IF.
AM Receivers 3-43

The signal is now coupled to gate 1 of Q103 the second mixer, while the
,

VFO, variable frequency oscillator, signal is coupled to gate 2. At this


stage, the exact received signal is selected. The first mixer and HFO
selected a 500 kHz band, now the second mixer and VFO select precisely
the desired signal. The difference between the VFO frequency and the
desired signal's frequency is 3.395 MHz, the second IF. L119, C153, and

C154 form a resonant circuit tuned to 3.395 MHz, which selects the
difference signal and couples it to Q207. Q207 amplifies the signal and
passes it to the crystal filter. Y203 Y204 Y201 and Y202 form a relatively
, , ,

simple crystal filter which provides adequate selectivity at low cost. The
crystal filter's response determines the receiver's selectivity.

The output of the crystal filter is coupled to gate 1 of Q205, the IF


amplifier. The dashed green line shows that AGC voltage is applied to
gate 2. Therefore, the IF amplifier's gain is varied in response to the input
signal level.

After the IF amplifier, the signal passes to the product detector, IC204,
which is an MC1496 integrated circuit balanced modulator. This is the
same IC you experimeted with in Unit 2. In this case, the RF signal is
mixed with a signal from the lower sideband BFO or the upper sideband
BFO. Both of these BFOs are crystal controlled since the RF signal is
always at the IF of 3.395 MHz. The output of the product detector is the
difference between the BFO and the RF signal, which is the demodulated
audio.

The audio is now amplified by op amp IC203C. At the output, it goes on to


the function switch which selects either wide bandwidth audio or narrow
bandwidth audio. If wideband audio is selected, it goes on as shown by
the red dashed line. However, if narrow band audio is selected, the output
of IC203C continues on to IC203A and IC203B which form an active
bandpass filter. The bandwidth of the active filter is 300 Hz centered on
750 Hz. This provides additional selectivity when receiving CW signals
(Morse Code). Finally, IC202 is the audio power amplifier which drives
the speaker or headphones.
3-44 UNIT THREE

AGC Circuit

The AGC circuit in this receiver is very similar to that discussed earlier in
this unit. Its signal path is shown by the dashed green line. The demod-
ulated audio signal is taken just after the product detector, and amplified
by IC203D. The amplified audio is then rectified by the voltage doubler
formed by D205 and D206. This DC signal is applied to Q204. However,
by following the solid green line, you'll see that the mode switch selects a
specific resistor and, therefore, a specific time constant for the AGC
system. In this case, a high value resistor, 680 kO, is chosen for SSB
operation and a low value, 47 kfl, for CW operation. Thus, the AGC
circuit will hold its control voltage longer on SSB to compensate for
speech pauses. However, on CW the AGC will drop out quickly to allow
rapid transmit-receive Morse Code operation.

FET Q204 provides a high input impedance interface to the delicate RC


time constant of the AGC system. The output on the source of Q204 passes
to the base of Q203. It is also used to drive the S meter to give an
indication of received signal strength. Q203 controls the AGC line. How-
ever, for low signal levels the AGC line is held at a constant 2.8 V by the
zener diode. When the rectified AGC signal is high enough, Q203 will
turn on and take control of the AGC line by reducing the voltage past the
zener voltage. Therefore, a large input signal will turn on Q203 lowering,

the AGC line and reducing the receiver gain. It reduces the receiver gain
by lowering the gain of Q101, the RF amplifier, and Q205, the IF
amplifier.
AM Receivers 3-45

Self-Review Questions

27. Draw a block diagram of the HR-1680 communications receiver


using its schematic diagram. Be sure to include the AGC system.
3-46 UNIT THREE

Self-Review Answers

27. See Figure 3-22.

\/ 0101 Q102
1ST

8. 65
IF

mHz
0103
RF AMPLIFIER
BANDPASS >t
FIRST MIXER SECOND MIXER
FILTER

I l i i
11

HETERODYNE
VFO
OSCILLATOR

<B

2ND IF

IC204 IC203A + B
Q207 CRYSTAL 0205 IC203C
PRODUCT ACTIVE
IF AMPLIFIER FILTER IF AMPLIFIER DETECTOR AUDIO AMP FILTER

r^£ Z£
0208
LSB BFO
0209
USB BFO
IC202
AUDIO
POWER AMP <
AGC
B<- SYSTEM

Figure 3-22
Block diagram of the HR-1680 receiver.
Unit 4

ANGLE MODULATION
4-2 UNIT FOUR

...
Angle Modulation 4-3

INTRODUCTION
In the previous units, we discussed the different types of modulation, AM
transmitters, and AM receivers. In this unit, we'll continue that discus-
sion by covering angle modulation transmitters and receivers.

Recall that angle modulation refers to both phase and frequency modula-
tion. In this unit, we will discuss both phase and frequency modulators,
several types of demodulators, transmitters, receivers, and transceivers.

The "Unit Objectives" on the next page state the goals of this unit. Review
this list now and be sure you can satisfactorily complete all the objectives
before you take the "Unit Examination."
4-4 UNIT FOUR
Angle Modulation 4-5

UNIT OBJECTIVES
When you have completed this unit, you should be able to:

1. Name the two types of angle modulation and three different mod-
ulator circuits.

2. Draw a block diagram of an angle modulation transmitter.

3. Find the master oscillator frequency and the required oscillator


deviation when given the transmitter's multiplication, output fre-
quency and output deviation.

4. Name five types of FM demodulators, list their advantages and


disadvantages, and identify their schematic diagrams.

5. State the purpose of a limiter stage.

6. Draw the block diagram of a phase-locked loop.

7. State the purpose of AFC and squelch.

8. Draw the block diagrams of an FM receiver, transmitter, and trans-


ceiver when given their schematic diagrams.

9. Construct an FM generator and phase-locked loop demodulator.


4-6 UNIT FOUR

ANGLE MODULATION
TRANSMITTERS
There are basically two types of angle modulation transmitters, depend-
ing on the modulation technique used. One type of transmitter uses a
"direct" modulation process, where the master oscillator's frequency is
varied in accordance with the modulating signal. The second type ob-
tains an angle modulated wave "indirectly" by phase modulating the
master oscillator's carrier signal in a later stage of the transmitter.

Reactance Modulator

The first type of direct modulation we'll discuss is called reactance

modulation. Here, an FET or transistor stage is biased so that it presents a


capacitive or inductive reactance to the master oscillator. When an audio
signal is applied to the modulator, its reactance varies and, therefore, so
does the master oscillator's frequency.

A typical reactance modulator is shown in Figure 4-1. In this circuit, Q


and Rj are connected directly across the master oscillator's tank circuit.
The reactance of Q at the oscillator frequency is at least six times greater
than the value of Rj. Therefore, the Q Rj parallel branch appears almost
purely capacitive. Thus, the voltage applied to the gate of Q! leads the
oscillator voltage by 90° due to the action of Q
R,. This in turn causes the
RF component of Q x 's drain current to also lead the oscillator voltage by
90°. Therefore, the entire circuit appears as a capacitive reactance across
the master oscillator tank.

MASTER
OSCILLATOR
TANK

Figure 4-1
A reactance modulator.
Angle Modulation 4-7

Since the FET circuit acts as a capacitance, it controls the master oscil-
lator frequency. When the audio input is zero, the capacitive reactance is
normal value which is determined by the biasing components and
at its

Q When the audio input goes positive, the FET drain current in-
R,.
creases and the capacitive reactance of the circuit also increases. This
adds more capacitance to the oscillator tank and, as a result, the oscillator
frequency is reduced. Likewise, when the audio input goes negative, the
FET current decreases, also decreasing the capacitive reactance. This in
turn increases the master oscillator's operating frequency.

A bipolar transistor could be usedin place of the FET or even a vacuum


tube. The can also be biased to present a variable inductive
circuit
reactance by reversing the Cj R connection. One disadvantage of the
t

reactance modulator is that it cannot be used with a crystal controlled


oscillator. Therefore, some form of frequency control circuitry.
it requires
This makes an expensive and complex circuit, particularly when com-
it

pared to the varactor modulator which is discussed next.

Varactor Modulator

A typical varactor diode modulator is shown in Figure 4-2 . It consists of a


varactor diode, D 1( in a Pierce crystal oscillator circuit. R, and R2 reverse
bias the varactor and set the diode's nominal capacitance value. This, in
turn, establishes the oscillator's center frequency. When an audio signal
is applied, it alternately adds to and subtracts from the varactor's reverse
bias voltage. This changes the varactor's capacitance and, therefore, the
oscillator's frequency.

\[ O OUTPUT

Figure 4-2
A varactor modulated crystal oscillator.
4-8 UNIT FOUR

When the audio signal swings positive, the reverse bias on the varactor
increases. This decreases the capacitance of the varactor which, in turn,
causes the oscillator frequency to increase. When the audio signal swings
negative, the reverse bias decreases. This increases the capacitance, forc-
ing the oscillator frequency lower.

If the audio signal 1000 Hz tone, the oscillator frequency will swing
is a
above and below its center frequency 1000 times each second Thus, the .

rate of deviation is the same as the audio signal. When the amplitude of
the audio signal is increased, the varactor swings through a larger range
of capacitance. Consequently, the amount of frequency deviation in-
creases.

The varactor modulator is by far the most popular used to obtain


circuit
angle modulation. It offers simplicity, reliability, and the stability of a
crystal oscillator.

Phase Modulator

Indirect angle modultion is obtained by using a phase modulator, such as


that shown in Figure 4-3. The RF input signal is usually from the master
oscillator or a buffer stage. This input RF is applied simultaneously to the
gate of Qj and through Q to the output tank of Lj and Q. A phasor, or
vector, diagram of this circuit is shown in Figure 4-4. Figure 4-4A shows
the gate voltage phasor, E^, and the tank voltage phasor, E L E L is leading .

Eg by a given phase angle, 0, due to the phase shift imparted by C t In .

addition, a second RF component appears across L^ and Q: the FET


output drain voltage, ED This voltage leads Eg by 180° due to the FET's
.

phase inversion. The phasor diagram of this is shown in Figure 4-4B. The
output voltage of this circuit, E is therefore the phasor sum of ED and EL
,

as is shown in Figure 4-4B.

When an audio signal is applied to Qj , the gate bias varies and so too does
the stage gain.The result is a changing ED phasor and in turn an E phasor
that not only changes amplitude but also phase. This is shown in Figures
4-4B, C, and D. Therefore, the output of the phase modulator is both phase
and amplitude modulated. To remove the amplitude variations, the next
few stages must be operated at class C. This removes any AM, while the
phase modulation is unaffected by the clipping action.
Angle Modulation 4-9

O OUTPUT

Figure 4-3
An FET phase modulator.

The advantage of the phase modulator is that the master oscillator is


unaffected by modulation. Therefore, it offers increased frequency stabil-
ity over the direct modulation techniques. However, it is difficult to
obtain wide frequency deviations with a phase modulator. For this
reason, its is in narrow band, two-way FM communications
primary use
rather than wide band FM broadcasting. You should also recall, from
Unit 1, that a phase modulator can produce frequency modulation if the
audio input passes through a low pass network before modulation takes
place. This is accomplished in Figure 4-3 by the low pass RC network
formed by R, and Q.

E
G

B
E
G

Figure 4-4
Phasor diagrams of the phase modulator.
4-10 UNIT FOUR

The Complete Transmitter

The block diagram FM transmitter is shown in Figure 4-5. The audio


of an
signal is amplified and applied to the FM modulator. The modulator
deviates the master oscillator according to the frequency and amplitude
of the audio signal. The FM carrier is then amplified and transmitted.

ANTENNA

V
AUDIO FM MASTER POWER
AUDIO
AMPLIFIER MODULATOR J
OSCILLATOR AMPLIFIER

Figure 4-5
Simple FM transmitter.

This type of transmitter has several problems at high frequencies. For


example, in an FM broadcast transmitter, the output signal will be be-
tween 88 and 108 MHz. This is a very high frequency and the master
oscillator would have a great deal of difficulty generating a stable fre-
quency carrier. Also, at this high frequency, it is difficult to shift the
carrier the required amount without causing distortion.

Many of these problems can be solved by using a different approach.


Figure 4-6 shows a block diagram of an FM transmitter that uses a much
lower master oscillator frequency.
Angle Modulation 4-11

MHZ 24 MHZ 48 MHZ 96 MHZ

AUDIO
AUDIO
AMPLIFIER
FM
MODULATOR ,
MASTER
OSCILLATOR
FREQUENCY
MULTIPLIER
x3
FREQUENCY
MULTIPLIER
x2
FREQUENCY
MULTIPLIER
x2
POWER
AMPLIFIER
J
Figure 4-6
An FM broadcast transmitter.

Here the master oscillator frequency is only 8 MHz. To raise the frequency
to the required 96 MHz, a series of frequency multipliers are used.
Remember, a frequency multiplier is nothing more than a nonlinear
amplifier whose output is tuned to a harmonic of the input frequency.
The most common multipliers are doublers and triplers because the
efficiency drops when higher multipliers are used. Here, one tripler and
two doublers are used to multiply the 8 MHz signal by:

3 x 2 x 2 = 12

Thus, the output frequency is 8 MHz x 12 = 96 MHz.

The maximum deviation for FM broadcasting is ± 75 kHz. Therefore, the


output frequency must deviate this amount when an audio signal is
applied. However, frequency multipliers increase carrier deviation as
well as carrier frequency. For example, suppose the 8 MHz oscillator is
varied from 7.9 MHz to 8.1 MHz. After this is multiplied 12 times, the
carrier will vary from 94.8 MHz to 97.2 MHz. Thus, to obtain ±75 kHz
deviation at the output, the master oscillator must be deviated:

±75 kHz -5-


12 = ±6.25 kHz

This can be easily accomplished with a varactor modulator.

One advantage that FM has over AM is that the power amplifier can
operate in class C, which is more efficient. This is so because the FM
waveform has a constant amplitude envelope. Therefore, only the fre-
quency variations need to be reproduced accurately. Thus, a high effi-
ciency class C power amplifier is used.
4-12 UNIT FOUR

Self-Review Questions

1. Name the two types of angle modulation.

2. Name three different angle modulator circuits.

3. Draw diagram
a block of an angle modulation transmitter using a
phase modulator.
Angle Modulation 4-13

4. In theFM transmitter shown in Figure 4-7, what crystal frequency


should the master oscillator use to obtain the desired 90 MHz
output?

5. To obtain ± 54 kHz deviation at the output of the transmitter shown


in Figure 4-7, the master oscillator must be deviated ±

90MHZ
±54 KHz
DEVIATION

AUDIO
AUDIO FM
1

MASTER FREQUENCY FREQUENCY FREQUENCY POWER


V
I

MULTIPLIER MULTIPLIER
i —•» MULTIPLIER
AMPLIFIER MODULATOR I OSCILLATOR AMPLIFER
x3 x3 x2
I

Figure 4-7
1

4-14 UNIT FOUR

Self-Review Answers

1. The two types of angle modulation are direct modulation (fre-


quency modulation) and indirect modulation (phase modulation).

2. The three angle modulator circuits are: The reactance modulator,


the varactor modulator, and the phase modulator.

3. See Figure 4-8.

MASTER PHASE FREQUENCY POWER


OSCILLATOR MODULATOR MULTIPLIERS AMPLIFIERS

AUDIO
AUDIO
AMPLIFIER

Figure 4-8

4. The multiplication of the master oscillator is:

3 x 3 x 2 = 18

Therefore, the master oscillator frequency must be:

90 MHz + 18 = 5 MHz

5. Since the master oscillator frequency and deviation is multiplied


18 times, to obtain ±54 kHz deviation at the output, the master
oscillator must be deviated:

±54 kHz -r 18 = ±3 kHz


Angle Modulation 4-15

DEMODULATORS
There are many different ways to demodulate an angle modulated signal.
The methods that are easiest to understand are not used very often, but to
help you understand the more complex methods, we will discuss the
simple ones first. We will then discuss the more complex (but more
popular) circuits.

Slope Detector

The simplest FM detector is called a slope detector and is shown in


Figure 4-9A. The two parallel resonant tank circuits LjQ and L2Q, have
an overall response curve like that shown in Figure 4-9B. Notice that the
response curve does not peak at the center frequency of the IF signal.
Instead, the center frequency falls half way up the side of the curve.

Ll L2

-w-
FM AUDIO
C2 Cl
INPUT

nil OUTPUT

AMPLITUDE

FREQUENCY

Figure 4-9
The slope detector and its response curve.

When the IF signal is at its center frequency, an average amplitude signal


reaches D When the IF swings above the center frequency, it approaches
1
.

the peak of the response curve. Consequently, a higher amplitude signal


is applied to the diode. When the IF swings below the center frequency, a

low amplitude signal is applied to the diode.

In effect, the tuned circuit changes the frequency modulated IF signal to a


signal which varies in amplitude. That is, it changes the FM signal to an
AM signal. The diode then detects the AM signal just like the AM
detector discussed earlier.
M

4-16 UNIT FOUR

The disadvantage of this technique is that the response curve of the IF


amplifiers which precede this stage must also be considered. Having the
IF amplifier response curve offset is undesirable, because receiver gain is

obviously reduced. Also, acceptable linearity is very difficult to achieve


during the FM to AM conversion. Consequently, this technique is seldom
used in practical applications.

Double-Tuned Detector

AUDIO Another FM detector that is easy to understand is shown in Figure 4-10.


Dl
OUTPUT
-O
This one is called a double-tuned detector. Transformer T! has a center-
tapped secondary. The upper half of the secondary is tuned by C to x

= =Ci * Ri ==c 3
J resonate slightly above the center frequency of the IF signal. Likewise,
the lower half of the secondary is tuned by Q
to resonate slightly below

= =C2 U2 = = C4 the center frequency.

J — D2
1=—'
When the IF signal is at its center frequency, both halves of the secondary
are equally "detuned." Dj and D conduct equally as shown. Notice that
2

Figure 4-10 the voltages developed across Rj and R2 are of opposite polarity and,
The double tuned detector. therefore, tend to cancel. Consequently, if Dj and D 2 conduct equally, the
output voltage is zero.

Above the center frequency, the signal approaches the resonant fre-
quency of the upper tank circuit. Thus, a larger signal is coupled to Dj
than to D 2 ; Dj conducts harder developing a larger voltage across R x
than
is developed across R2 . Therefore, the resultant is a negative voltage.
Angle Modulation 4-17

When the carrier swings below the center frequency, the signal ap-
proaches the resonant frequency of the lower tank circuit. D 2 receives a
stronger signal than D,. D 2 conducts harder, developing a larger signal
across R2 than is developed across R v Since the voltage across R 2 is
positive with respect to ground, the net output voltage is positive. As you
can see, the output signal corresponds to the intelligence contained in the
FM input signal.

The primary disadvantage of the double-tuned detector is the precise


tuning required for each resonant circuit. Ifone of the tank circuits is
slightly out of tune, the detected audio will be distorted. For this reason a
double-tuned detector is seldom used to demodulate FM audio signals.
However, it is used to demodulate frequency-shift-keyed or FSK signals.
These are digital transmissions where a high level, or binary 1, corres-
ponds to one frequency and a low level, or binary 0, corresponds to
another frequency, which is slightly shifted from the first. Here, one
resonant circuit is tuned to the binary 1 frequency and the other is tuned
to the binary frequency. The demodulated output is a pulse train which
can be coupled to a digital computer or other peripheral device. Usually
the FSK signal is at audio frequencies, for example 1070 Hz and 1270 Hz,
and, therefore, the demodulator is also tuned to these frequencies. By
using audio frequencies, the signal can be transmitted over the telephone
lines.
4-18 UNIT FOUR

Foster-Seeley Discriminator

A more popular FM detector is shown in Figure 4-11. This one is called


the Foster-Seeley discriminator.

Rl
| ?
—\J- AUDIO
OUTPUT

Figure 4-11
The Foster-Seeley discriminator.

The input to the circuit is an IF signal which is varying ±75 kHz at the
audio rate. The output is the detected audio signal. The two diodes and
their associated RC networks operate similar to their counterparts in the
double-tuned detector discussed That is, when both diodes con-
earlier.
duct, equal but opposite polarity voltages are developed across R, and R 2 ,

the two voltages cancel, and the output is volts. However, if D, conducts
harder, the output is positive. By the same token, the output is negative
when D2 conducts harder. The audio signal can be recovered from the IF
signal if:

1. Both diodes conduct equally at the center frequency;

2. D, conducts harder above the center frequency;

3. D 2 conducts harder below the center frequency.

What determines how much each diode conducts?

By transformer action, Lj couples the IF signal to L*, and L3 As connected, .

L2 and L3 act as a center-tapped secondary. Thus, the voltage developed


across L2 [EL2 ) is 180° out of phase with the voltage developed across L 3
(E L3 ). EL2 controls the conduction of D, while E^ 3 controls the conduction
of D 2 Keep in mind that these two voltages are equal in amplitude but are
.

180° out of phase.


Angle Modulation 4-19

The IF signal is also capacitive coupled through Q to L A voltage which


4 .

we will call E L4 is developed across L4 EL4 . controls the conduction of both


Dj and D 2 .

The circuit is arranged so that EL4 is 90° out of phase with both EL2 and E t3 .

However, as you will see, this is so only when the IF signal is at its center
frequency. At the center frequency, EL4 leads E^ by 90° but lags behind EL2
by 90°.

The amount that diode Dj conducts is determined by EL2 and EL4 while ,

the amount that D 2 conducts is determined by EL3 and EL4 We have seen .

that EL2 and EL3 are equal in amplitude but are 180° out of phase. At the
center frequency, EL4 is 90° out of phase with both EL2 and E L3 Thus, at the .

center frequency, EL4 adds to both of these signals equally. Consequently,


Dj and D 2 conduct equally and the output voltage is 0.

The parallel resonant circuit is resonant at the center frequency, where X L


exactly cancels X c and the resonant circuit acts resistive. However, above
resonance, XL is larger than Xr . Thus, there is a net reactance that shifts
the phase of EL2 and EL3 EL2 . is shifted more in phase with EL4 Since . EL3 is

always 180° out of phase with EL2 EL3 is shifted more out of phase 'with
,

EL4 That is, EL4 tends to add to EL2 but tends to subtract from E^. Thus, D!
.

conducts harder than D 2 The net result is that the output swings positive
.

each time the IF signal swings above the center frequency.

Below resonance X c is larger than X L The net reactance shifts the phase of
.

EL2 and EL3 in the opposite direction. This time, EL3 is shifted more in
phase with EL4 Consequently, D 2 conducts harder, producing a net nega-
.

tive output voltage. Thus, the output swings negative each time the IF
signal swings below the center frequency.

If the modulating signal is a 1000 Hz tone, the IF will swing above and
below the center frequency 1000 times each second. The discriminator
produces an output sine wave that swings positive then negative at the
same rate. Thus, the discriminator recovers the modulating signal.

One of the disadvantages of the Foster-Seeley discriminator, as well as


the other FM detectors previously discussed, is that it is sensitive to
amplitude changes. That is, any noise spikes present on the FM wave will
be detected and passed on to the audio output as noise or "static."
However, since an FM wave does not have an envelope that must be
preserved, we can use a special circuit to clip off both the positive and
negative peaks of the wave. The frequency variations will remain but the
majority of noise impulses will be removed. This circuit is called a limiter
and it immediately precedes the FM demodulator. The limiter is respon-
sible for the noise-free reception of FM signals.
4-20 UNIT FOUR

Ratio Detector

A further improvement over the Foster-Seeley discriminator is the ratio


detector shown in Figure 4-12. This circuit is one of the most popular
types of FM detectors because it has a significant advantage. It provides
its own limiting action and requires no preceding limiter stage.

,
AGC
VOLTAGE

:±c

<> O
% AUDIO
OUTPUT

Figure 4-12
The ratio detector.

The ratio detector circuit looks somewhat similar to the Foster-Seeley


discriminator. However, closer examination shows that diode D x
is re-

versed and that the output configuration is different.

With Dj reversed, the two diodes are in series across the entire secondary.
Conduction of the two diodes is controlled by the same factors as in the
Foster-Seeley discriminator. At the center frequency, the two diodes
conduct equally. In this circuit, the voltages build up across C4 and C 5 in
series. Recall that in the discriminator, the diodes produced opposing
voltages.
Angle Modulation 4-21

The key to the unique operation of the ratio detector is Q. Q is a large


value capacitor. After several cycles of the input signal, this capacitor
charges to a voltage which is proportional to the average received signal
strength. Q is large enough to hold the voltage across R2 constant. It also
holds the voltage constant across the series combination of C4 and C 5 . It

does this even if there are momentary amplitude variations (noise). This
is the reason that the ratio detector is relatively insensitive to noise.

Q also provides a convenient AGC voltage since the voltage across it is

directly proportional to the input signal amplitude.

The voltage across C 5 plus the voltage across C4 must always equal the
voltage across Q. At the two diodes conduct
center frequency, the
equally and the voltages across the two capacitors are equal. A sample of
the voltage across C 5 is tapped from Rj. This is the audio output. This
output will be at some negative DC level.

As and D 2 alternately conduct harder as the IF


in the discriminator, D]
signal swings above and below the center frequency. When the diodes are
not conducting equally, the difference current flows through L4 When D, .

conducts harder, the voltage across C4 exceeds that across C 5 However, .

since the sum of these two voltages remains constant, the voltage across
C 5 must decrease. Consequently, the voltage at the output must also
decrease.

When D conducts harder, a higher voltage develops across C 5 Therefore,


2
.

the output voltage increases. As you can see, the output voltage swings in
step with the frequency changes in the IF signal. That is, when the
frequency increases, the output voltage increases. And, when the fre-

quency decreases, the output voltage decreases.

The primary advantage of the ratio detector is its self-limiting action. It

also provides AGC voltage.


4-22 UNIT FOUR

Phase-Locked Loop Demodulator

It has always been possible to use phase-locked loops (PLL) as FM


demodulators, but they were not practical until integrated circuits were
developed. A complete phase-locked loop, like that shown in Figure
4-13, can be purchased as a single integrated circuit. This makes the PLL
much simpler and cheaper to use. The basic loop of Figure 4-13 consists
of a phase detector, low pass filter, and voltage controlled oscillator. The
VCO or voltage controlled oscillator operates at the input frequency, in
this case, the intermediate frequency. The phase detector compares the
VCO frequency and the input frequency. It then develops an error voltage
that is proportional to the amount and direction of the frequency differ-
ence. This error signal is then coupled to the low pass filter. The filter

establishes many of the dynamic characteristics of the PLL. It determines


the frequency range over which the loop will acquire and hold its phase
lock. The filter also determines how fast the loop will respond to input
frequency variations.

ERROR VOLTAGE

IF
INPUT
n
u »
^ PHASE
DETECTOR
/, LOW- PASS
FILTER
\ ..
T
w AUDIO
"^OUTPUT
t i

VOLTAGE
LUNI KULLtL
OSCILLATOR

Figure 4-13
A phase-locked loop demodulator.
Angle Modulation 4-23

The error voltage from the low pass filter is then used to control the VCO.
For example, if the input frequency deviates upward, an error voltage is

generated by the phase detector. This voltage is filtered and applied to the
VCO. The error voltage will cause the VCO to move upward in frequency
in an exact lock with the input frequency. When the input signal is
frequency modulated, the VCO tracks the FM deviation and, as a result,
the error voltage is an exact duplicate of the modulating information.

Figure 4-14 shows a typical integrated circuit PLL. In this case, it is the
Signetics NE565 monolithic IC. In this circuit, R, and C, are used to set the
VCO's operating frequency range. Q> and the internal 3.6 kH resistor form
an RC low pass filter. The only difference between this circuit and the
block diagram of Figure 4-13 is the DC amplifier which increases the error
signal voltage.

The primary advantage of the phase-locked loop is its excellent perfor-


mance at low cost, with a minimum of components.

It also eliminates costly inductors and transformers and it greatly


simplifies tuning requirements.

LOW- PASS
FILTER

.OluF

iN°-lr AUDIO
' OUTPUT
560OJ

Figure 4-14
An IC phase-locked loop.
4-24 UNIT FOUR

Self-Review Questions

6. Name 5 types of FM demodulators.

1.

2.

3.

4.

5.

What are the advantages and disadvantages of each of the FM


demodulators named in Question 6?

8. What shown in Figure 4-15? Which terminal is used


circuit is for
the audio output? Which terminal is used for AGC voltage?

Figure 4-15
1 o

Angle Modulation 4-25

M oA
1 T t

OB
i . R

1—M — 4= C4 *

i—C 2

Figure 4-16

9. What circuit is shown in Figure 4-16? Which terminal is used for


the audio output? Which terminal is used for AGC voltage?

10. What is the purpose of a limiter?

11. What FM demodulator does not require a limiter?

12. Draw a block diagram of a phase-locked loop and label the audio
output.
4-26 UNIT FOUR

Self-Review Answers

6. The 5 types of FM demodulators are:

1. Slope detector.
2. Double-tuned detector.
3. Foster-Seeley discriminator.
4. Ratio detector.
5. Phase-locked loop.

1. The slope detector's advantage is simplicity, it's disadvantages


are reduced IF gain and poor linearity.
2. The double-tuned detector's disadvantage is the precise tuning
requirements, it's main advantage is its application as an FSK
demodulator.
3. The advantage of the Foster-Seeley discriminator is that it is

much easier to tune and align than the previous demodulators.


Its disadvantage is that it requires a limiter stage.
4. The advantage is its self-limiting action. Its
ratio detector's
disadvantage, which is shared with the previous demod-
ulators, is the use of inductors and transformers.
5. The phase-locked loop's advantage is simple circuitry and
tuning. What is its disadvantage? None!

Figure 4-15 is a ratio detector. Terminal C is the audio output and


terminal A is the AGC voltage.

9. Figure 4-16 is a Foster-Seeley discriminator. Terminal A is the


audio output and there is no directly available AGC voltage termi-
nal.

10. The purpose of a limiter is to remove amplitude variations from the


FM signal before it is demodulated. The result is noise-free recep-
tion.

11. The ratio detector does not require a limiter.

12. See Figure 4-17.

FM PHASE LOW PASS AUDIO


INPUT DETECTOR FILTER '
OUTPUT

vco
Figure 4-17

r\J*^
Angle Modulation 4-27

RECEIVERS
An FM receiver is actually very similar to the AM superheterodyne
receiver discussed previously. That is, it converts the received RF signal
to a lower intermediate frequency where the majority of gain and selectiv-
ity are obtained. The only difference is the use of an FM detector and
possibly a limiter stage.

Block Diagram

A typical FM broadcast receiver is shown in Figure 4-18. Notice the


similarity between this receiver and an AM receiver. The primary differ-
ences are the intermediate frequency, the limiter, and the FM detector.
The IF of 10.7 MHz is chosen to provide good image rejection and also to
provide the wider bandwidth required for FM broadcast signals. In FM
broadcasting, maximum transmitter deviation is ± 75 kHz. Therefore, the
receiverbandwidth must be at least 150 kHz and is usually 180 to 200
kHz. The IF of 10.7 MHz allows this while maintaining acceptable IF
gain.

88- 108MHz

V 10. 7MHz SPEAKER

AUDIO
RF AMPLIFIER MIXER IFAMPLIFER LIMITER FM DETECTOR
AMPLIFER
HI
LOCAL
,
7 - 118. 7MHz
OSCILLATOR

TUNING

Figure 4-18
An FM broadcast receiver.
4-28 UNIT FOUR

Let's follow the signal through the receiver. Assume that the receiver is
tuned to MHz. When the tuning indicator is set to this frequency, the
100
RF amplifier is tuned to 100 MHz, while the local oscillator is tuned to
110.7 MHz. The RF and local oscillator signals heterodyne in the mixer,
forming a difference signal of 110.7 - 100 = 10.7 MHz. This signal is
passed to the IF amplifier where it receives most of its amplification. The
IF amplifier also reduces the received bandwidth sufficiently to select the
correct signal while rejecting all others.

If the received carrier is deviating ±50 kHz and the local oscillator
frequency is constant, the 10.7 MHz IF signal will also deviate ±50 kHz.
Thus, the original modulation is converted to the intermediate frequency
without distortion.

LIMITER

The next stage in the block diagram is the limiter. As mentioned earlier,
this circuit is used to remove any amplitude variations, such as noise,
from the FM signal. It does this by saturating on the positive half cycle
and going into cut off on the negative half cycle. It, therefore, clips or cuts
off both the positive and negative peaks of the waveform and, along with
them, most of the noise. This limiter action is shown in Figure 4-19. Note
also that the frequency variations or original modulation is unaffected by
the amplitude clipping.

LIMITER

Figure 4-19
The removes noise spikes
limiter
and other amplitude variations.
Angle Modulation 4-29

Note from Figure 4-19 that the limiter stage must be overdriven to a
certain extent. That is, if the input amplitude is not high enough to drive
the limiter into cutoff and saturation, noise will pass on to the FM
detector and eventually the audio output. In fact, one FM receiver
specification refers to the "quieting" of the audio output. This is the
sensitivity of an FM receiver. It specifies the unmodulated input signal
required to reduce the audio noise output by 20 dB. The only way to
reduce the noise output is to amplify the input signal sufficiently to
overdrive the limiter. The limiter will clip the noise pulses from the wave
and, in the absence of modulation, provide a "quiet" audio output.

AUTOMATIC FREQUENCY CONTROL

When an FM receiver is operated at VHF, as it is in FM broadcasting, the


local oscillator must be extremely stable. For example, a frequency drift

of just 0.1% at 100 MHz will cause the IF to drift 100 kHz. This will result
in serious distortion of the demodulated wave. One way to prevent drift is
to use a crystal controlled local oscillator. However, this is impractical for
broadcast reception since you want to select many different frequencies
without buying a new crystal each time. Although, crystals are used in
two-way FM communications where only a few frequencies are required.
In FM broadcasting, an automatic frequency control circuit is usually
used to "lock" the local oscillator onto the incoming signal. Figure 4-20
shows the block diagram of an FM receiver that uses AFC.

V 10. 7MHz SPEAKER

AUDIO
RF AMPLIFIER MIXER IF AMPLIFIER LIMITER FM DETECTOR AMPLIFIER

1
LOCAL
AFC m
OSCILLATOR

L — .,-
i
.
1

TUNING
Figure 4-20
The use of automatic frequency control
in an FM receiver.
4-30 UNIT FOUR

The basic principle of AFC is to use a varactor, or a similar device, as part


of the local oscillator tank.The varactor's capacitance is controlled by an
"error" voltage that produced when the oscillator goes off frequency.
is

This error voltage is obtained from the output of the discriminator or ratio
detector. Here, if the oscillator is on frequency, the output is zero. How-
ever, if the frequency drifts to one side or the other, an error voltage is
produced. The polarity of this voltage indicates the direction of drift and
the amplitude indicates the magnitude of frequency drift.

It seems that it would be a simple matter to use this error voltage to


control the local oscillator.However, the output of the discriminator also
contains the demodulated audio wave. Therefore, before this voltage can
be used to control the oscillator, the audio component must be removed;
otherwise, the AFC action would cancel the frequency modulation. The
audio can be removed by using a low pass filter. This removes the audio
while allowing the DC and the very low frequency error signal to pass on
to the varactor.

Figure 4-21 shows a simplified AFC circuit. Qj and its associated cir-
cuitry form the local oscillator. Q is the receiver's main tuning control,
while Dj is a varactor which is in parallel with the oscillator tank. In this
circuit, the output of the FM coupled to both the audio
detector is

amplifiers and the AFC circuit. Rj and Q


form a low pass filter which
removes the audio component. Then this DC error voltage either adds to
or subtracts from the varactor bias provided by R2 R3 Rj is used to isolate .

the AGC filter from the oscillator. As the varactor's capacitance changes,
so too does the local oscillator frequency.

10. 7 MHZ
RF
IF FM
• |( — AUDIO
MIXER
AMPLIFIER DETECTOR
INPUT

R
i >

LOCAL ^ C,

OSCILLATOR

Figure 4-21
Simplified AFC circuit.
Angle Modulation 4-31

Careful observation of the AFC circuit of Figure 4-21 shows that it is

actually a variation of the phase-locked loop. In this case, the FM detector


is the phase detector, I^Q are the filter, Q t
and Dj form the VCO, and the
mixer is used to convert the two VHF signals (RF input and local oscil-
lator) to a much lower frequency and to provide the needed 10.7 MHz
offset.

AM/FM Broadcast Tuner

Examining the schematic diagram of a typical AM/FM tuner, such as that


shown in Figure 4-22, will show you the practical application of the
circuitswe've discussed so far. Remove "Schematic Diagram #2" from
the schematics packet included with this course. The signal path is
marked by a red dashed line. Let's follow it through to the audio output.

Figure 4-22
The AJ-1219 AM/FM Tuner.

From the antenna input, either 300 Cl balanced or 75 H


which can be
unbalanced, the signal coupled to the RF amplifier Q301. This FET
is

amplifier has tuned circuits on both its input and output. This improves
its selectivity and gain but lowers its stability. For this reason, L301 and

C306 couple energy from the drain tuned circuit back to the gate. The
phasing of this transformer coupling provides degenerative feedback
and, therefore, neutralizes the amplifier.

Once the RF input is selected and amplified, it is applied to the mixer,


Q302. The signal from the local oscillator, Q303, is also applied to the
mixer (via the red dotted line). The output of the mixer is the difference
signal of 10.7 MHz, the intermediate frequency. This IF signal is coupled
via T302 to the 10.7 MHz ceramic filter, Fl. The ceramic filter, used to
improve receiver selectivity operates in the same manner as a crystal
filter since it exhibits piezoelectric effects similar to quartz. However, the

ceramic filter is smaller and cheaper than a comparable crystal filter.


j

4-32 UNIT FOUR

The IF amplifier, ICl, is an MC 1357 integrated circuit which is a combi-


nation IF amplifier, limiter, and FM detector. ICl is connected as an IF
amplifier and limiter. In this configuration, it can provide up to 60 dB
gain. It has two output connections — pins 10 and 14. The output from
pin 10 is applied to Q6, which is the AGC amplifier. The amplified IF
signal from Q6 is rectified by the voltage doubler circuit of D! D 2 and Q> 9
by R21 and Q5, the resulting negative DC voltage
Q> 7 After further filtering
.

is directly proportional to the input signal level. This voltage


is then used

to control the gain of the RF amplifier, Q301, by changing the gate bias.
This is shown on the schematic diagram by the dotted green line. AGC
amplifier Q14 is only used during initial alignment after the customer
builds the kit (this Stereo Tuner is sold only in kit form).

The output from pin 14 of ICl is applied to the second 10.7 MHz ceramic
filter, F2. This provides additional IF selectivity. The output of F2 is
coupled to IC2 which is also an MC 1357 integrated circuit. It too pro-
vides IF amplification and limiting. However, it is also connected as an
FM detector. Therefore, the output of IC2 at pin 1 is the demodulated
audio signal. This audio signal is applied to Q3 which is an audio
amplifier. Both the DC and audio output of IC2 are coupled to the AFC
circuit as indicated by the dashed green line. In the AFC circuit, the audio
component is removed by the low pass filter composed of R28 and Cg. The
remaining DC voltage, which is directly proportional to the frequency
drift, is used to control the bias of D301 which is a varactor. Steady-state
,

bias of D301 is provided by R30 8 and R309 for its cathode and R22 and R^ for
its anode. Potentiometer R, is used to set the local oscillator for correct

dial calibration. The AFC voltage either adds to or subtracts from D301's
anode bias voltage to provide automatic frequency control.

In FM "Stereo" broadcasting, the output of the demodulator contains


frequency components as high as 53 kHz. This is due to the multiplexing
techniques used to transmit both the right and left audio signals. This
wide range by Qj and applied to IC3, which is
of frequencies is amplified
the stereo demodulator. This circuit separates the demodulated audio
into right and left channels and also, from pin 6, turns on Q> and the
stereo lamp to indicate when a stereo signal is being received. FM stereo
broadcasting is discussed in detail in a later unit.

Qs and Q 7 are audio amplifiers for the right and left channels respectively.
LiL 5 and LgLj form low pass filters and traps to prevent any signals above
15 kHz from appearing at the audio output. Since this is a tuner, there are
no audio power amplifiers; instead, a separate power amplifier must be
used to drive the loudspeakers. You might like to look over the AM
section of the tuner as a review of the previous unit.

i-
Angle Modulation 4-33

Self-Review Questions

13. Why is 10.7 MHz the standard intermediate frequency in FM re-

ceivers?

14. How can a limiter remove noise from an FM signal without causing
distortion?

15. What is the purpose of AFC and why is it needed?

16. Draw a block diagram of the FM portion of the tuner in schematic


diagram #2.
4-34 UNIT FOUR

Self-Review Answers

13. A 10.7 MHz IF provides good image rejection and a sufficiently


wide bandwidth to accomodate the ±75 kHz deviation of FM
broadcasting.

14. Since an FM signal does not have an envelope, both positive and
negative amplitude peaks can be removed, along with any noise,
without causing distortion to the wave's frequency deviation.

15. The purpose of AFC is to stabilize the local oscillator frequency


and, in effect, lock it on to the incoming signal. It is needed,
primarily, because the local oscillator is operating at VHF. Anytime
a variable LC oscillator operates at VHF or higher, it normally
requires some form of stabilization, in this case, AFC.

16. See Figure 4-23.


Angle Modulation 4-35

ex. x
o

Figure 4-23
,

4-36 UNIT FOUR

FM TRANSCEIVERS
Intwo-way radio communications, the transmitter and receiver are often
combined into a single unit called a transceiver.

The hand-held VHF transceiver shown in Figure 4-24 contains every-


thing needed for communication. It has both transmitter and receiver;
microphone, loudspeaker, antenna, battery power supply, and even a
Touch-Tone® key pad for dialing a telephone through a special base
station. In the following text, we will analyze the schematic diagram of
this transceiver.

Schematic Analysis

Remove "Schematic Diagram #3" from the packet of schematics in-


cluded with this course. We'll analyze the receiver firstand then the
transmitter. The receiver signal path is marked by a red dashed line.
Figure 4-24
A handheld FM transceiver. The input RF signal appears at the antenna jack and passes through L41
and it associated capacitors. They form a low pass filter that reduces
harmonic radiation by the transmitter and protects the receiver from
overload by very strong UHF signals. The signal then passes through
diodes D 6 and D 9 In the receive mode, these diodes are forward biased by
.

the receiver's positive supply voltage through R 5 and Ri 25 Since they are .

forward biased, the RF signal passes through unattenuated. However D„


which leads to the transmitter, is reverse biased and it prevents signals
from entering the transmitter.

From the diode switches, the signal passes to Q, the, first RF amplifier. It

is then coupled to Q2, an additional RF amplifier. Together these two


amplifiers provide 28 dB of power gain. They also provide sufficient
selectivity to limit reception to a 2 MHz bandwidth. In this case, the
frequencies of interest are the amateur radio band from 144 MHz to 148
MHz. The RF amplifiers must be tuned to the desired 2 MHz portion of
this band.

^Registered Trademark, American Telephone & Telegraph Co.

I^^BB
Angle Modulation 4-37

The output of the RF amplifiers


applied to gate 1 of the mixer, Qj. The
is

local oscillator signal, the red dotted line, is applied to gate 2. These two
signals are mixed and their difference signal is applied to crystal filters
FLl and FL2. These filters determine the IF selectivity and, therefore, the
receiver's bandwidth.

Let's follow the local oscillator signal path. Q9 is the crystal oscillator;
SW202 can be set to select any one of 8 different crystals. D3 is a varactor
which is used only during transmit to provide frequency modulation.
The crystal shown at Yl is at 15.1377 MHz. Not only is Q 9 the oscillator
but it also provides a frequency multiplication of 3. This is because the
collector load of Q9 , L^ and C^, is resonant at the third harmonic of the
crystal frequency. Therefore, the output of Q 9 is 15.1377 x 3 = 45.4131
MHz. This signal is then coupled to a 45 MHz bandpass filter formed by

Qe, Q7, Qs, and L> 3 Qio amplifies the 45 MHz signal and provides a
.

further multiplication by 3 through its output tank of L> 4 and Q,. Thus,
the output of Q 10 is 45.4131 x 3 = 136.2393 MHz. Q u provides further
amplification, filtering for the 136 MHz signal, and buffering for the
oscillator chain. When the local oscillator signal is applied to the mixer,

Qa, the RF input signal must be 136.2393 MHz + 10.7 MHz = 146.9393
MHz to provide a difference signal of 10.7 MHz that will be amplified by
the IF section. The crystal oscillator's frequency can be shifted slightly by
coil L 13 to set the received and transmitted frequencies to exactly the right
channel (146.940 MHz).

The output of IF amplifier Q4 is applied to IC, which is a CA3089. This


integrated circuit an IF amplifier, FM detector, and audio amplifier.
is

Crystal filter FL 3 forms part of the FM detector circuit.

The audio output from pin 6 of IC, is applied through the volume control
to pin 7 of IQ>, the audio power amplifier. IQ> is used to drive the speaker
or earphones, if used.

There one more part of the receiver which we must discuss. This is the
is

squelch circuit. The squelch circuit is used to "cut off" the audio
amplifier when no signal is present. If the audio amplifier was not cut off
during no-signal conditions, the noise present would be amplified and
applied to the speaker. This can be extremely annoying. The squelch
circuit monitors the FM detector. If no signal is present, it cuts off the
audio amplifier and there is no output. If a signal is present, the squelch
circuit turns on the audio amplifier and the signal is heard through the
loudspeaker. Let's analyze this particular squelch circuit in detail.
4-38 UNIT FOUR

The path of the squelch signal is shown by the solid red line. Pin 13 from
ICj is a voltage level that corresponds to the relative received signal
strength. This voltage is applied to Q5 which compares it to the voltage
coming from the squelch control. The collector voltage of Qs is applied to
IC3B which is wired as a comparator. At Q 5 if the signal level voltage at
,

the emitter exceeds the squelch voltage at the base, the transistor will be
cut off and the
collector voltage will be high. This high positive voltage
will cause the IC3B comparator to switch, giving a high voltage at pin 7,
its output. This high voltage will cut off Qe and prevent a "turnoff"

voltage from being applied to pin 5 of IQ, the audio amplifier. Therefore,
as long as the received signal level is above the squelch control setting,
the audio amplifier will be turned on and operating.

However, if the input signal is very low or nonexistent, Q5 will be turned


on and a low voltage will be applied to the comparator, IC3B. This will
cause the comparator output to go low and, therefore, turn on Q6 With Q6 .

turned on, a positive voltage will be applied to pin 5 of IQ turning off the
,

audio amplifier.

Connected to the squelch circuit is a special "battery saver" circuit. D 5 (at

the output of IC3B, the comparator) switches this circuit on and off. IC3A
forms an 8 Hz astable multivibrator with the output waveform as shown.
When a received signal is present, the comparator output is high. D 5 is
forward biased, and, therefore, places a high voltage on pin 2 of IC3A.
This shuts off the multivibrator and also the battery saver circuit since a
signal is being received. However when no input signal is present, the
comparator's output is low. This reverse biases D 5 which
, allows the
astable to operate. The output waveform is used to key Q 7 on and off.
Since Q7 power supply voltage to the receiver, it is keyed on
controls the
eight times each second and remains on for only a few milliseconds. This
allows the receiver to monitor the input while consuming the least
amount of valuable battery power. When a signal is detected, the squelch
circuit turns off the astable and applies a low voltage to Q 7 This holds Q 7 .

on and, thus, applies continuous power to the receiver.


Angle Modulation 4-39

Power to the transmitter is off unless the push-to-talk button is depressed,


which turns on Q« and applies 12.6 V to the transmitter. When power is
applied, D 7 is forward biased and shuts off the receiver audio amplifier,
IC2. Also D 8 is forward biased, which shuts off the battery saver circuit.
This turns off Q 7 and, therefore, removes power from the receiver. In
addition, D n is forward biased, which connects the antenna to the trans-
mitter, and D 6 D 9 are reverse biased, disconnecting the receiver. D and D 2 l

also switch; these diodes determine whether the local oscillator power is
provided by the receiver voltage line or the transmitter line.

The input audio to the transmitter starts at the microphone. This is shown
by the solid green line. IC4A provides voltage gain, while IC4B provides
both gain and limiting. Above approximately 4 V peak-to-peak, IC4B will
clip the output signal, therefore, limiting the maximum transmitter de-
viation. Output waveforms for both amplifiers are shown on the schema-
tic. The output of IC4B is applied to the low pass filter formed by R 86 R 87 , ,

Q5, and Cge. The filter will remove any high frequency components or
harmonics from the clipped audio signal. R88 is the deviation control and
it sets the maximum transmitter deviation. The modulator output is

applied to D 3 a varactor. The varactor's changing capacitance frequency


,

modulates Q9 the crystal oscillator.


,

You can follow the transmitter signal through the local oscillator to the
transmitter mixer, Q
by the dashed green line. At the transmit mixer,
13 ,

the output of the offset oscillator is mixed with the local oscillator signal.
In simplex operation, which means transmission and reception on the
same frequency, the offset oscillator signal is 10.7 MHz. Therefore, the
output of the transmit mixer is 10.7 MHz + 136.2393 MHz = 146.9393
MHz, which is exactly the same as the received signal.

The can also generate an 11.3 MHz signal or a 10.1 MHz


offset oscillator
signal. These outputs will provide transmitter frequencies, respectively,
600 kHz above or 600 kHz below the receiver frequency. This is done to
allow operation with repeater stations whose input and output frequen-
cies are separated by 600 kHz. A repeater station will receive an input
signal and retransmit it at a different frequency, with high power, from a
central or more favorable location. In this way, it can greatly extend the
communications range of hand-held transceivers such as this one.
4-40 UNIT FOUR

The output of the transmit mixer, which is at the correct operating


frequency, is applied to a bandpass filter composed of L3, , Cge, C^, L32 , C 99 ,

and C 10 i. This filter insures that the correct mixing product is selected.
The transmit predriver, Q, 4 and amplifier, Q 15 provide additional gain
, ,

and filtering before the signal reaches the driver, Q 16 The input voltage to
.

Q 16 can be varied by Rj 14 This adjustment


. sets the input and output power
levels for the transmitter.

The output of the driver is applied to the power amplifier, Ql 7.


Its output

is approximately 2.5 watts. This signal coupled to the antenna through


is

a low pass filter and wave trap. C 123 C 124 L41 and C 126 form a low pass filter
, , ,

to attenuate harmonics. L41 and C 125 form a wave trap tuned to the second
harmonic. They provide additional attenuation of this signal.
Angle Modulation 4-41

Self-Review Questions

17. What is the purpose of a squelch circuit?

18. Refer to schematic diagram #3. What frequency crystal should be


used at Yl to receive a 148 MHz signal?

19. In schematic diagram #3 the output deviation is normally ± 5 kHz.


,

How much deviation is required at Q 9 and D 3 the crystal oscillator,


,

in order to maintain ±5 kHz deviation at the output?

20. From schematic diagram #3, draw the transmitter block diagram.
Label the frequencies present between each block, assuming a Yl
crystal frequency of 15 MHz.
4-42 UNIT FOUR

Self-Review Answers

17. The purpose of a squelch circuit is to turn off the receiver's audio
output when no RF input signal is present.

18. The must generate a frequency that


local oscillator chain is 10.7
MHz lower than the desired signal. Therefore,

148 MHz - 10.7 MHz = 137.3 MHz

The must generate a 137.3 MHz signal. Since


local oscillator chain
the crystal frequency is multiplied by two tripler circuits, the total
multiplication is 3 x 3 = 9. Therefore, the crystal frequency is:

137.3 MHz - 9 = 15.2556 MHz

19. Since the crystal oscillatoris multiplied by 9, so too is the fre-

quency deviation. Therefore, the required deviation at the crystal


oscillator is:

±5 kHz + 9 = 555.6 Hz

20. See Figure 4-25.


Angle Modulation 4-43

1- tr Of a; LO
LU
on
014 MPLIFIE
Ql 1—
RANSMI REDRIVE

to
s
POWER 017

1— Q. •< <t

TRANSMIT MIXER 013 OFFSET OSCILLATOR Q12 —o-^o |Q|


|l

Figure 4-25
4-44 UNIT FOUR
Unit 5

PULSE MODULATION
5-2 UNIT FIVE
Pulse Modulation 5-3

INTRODUCTION
In amplitude and angle modulation, some characteristic of the carrier —
amplitude, frequency, or phase —
is continuously varied in accordance

with the modulating information. However, in pulse modulation, a small


sample is made of the modulating signal and then a pulse is transmitted.
In this case, some characteristic of the pulse is varied in accordance with
the sample of the modulating signal. The sample is actually a measure of
the modulating signal at a specific time.

There are several types of pulse modulation systems. Three of the more
common types are: pulse amplitude modulation (PAM), pulse duration
modulation (PDM), and pulse position modulation (PPM). In each of
these systems, a characteristic of the pulse —
such as amplitude, dura-
tion, or position— is continuously varied in accordance with the mod-

ulating signal. This type of pulse modulation, where a pulse character-


istic is continuously varied, is called analog pulse modulation.

Another type of pulse modulation is pulse code modulation (PCM),


which is digital pulse modulation. With PCM, the modulating signal is
sampled and then quantized. In quantization, each sample is assigned a
specific numerical value according to its amplitude. This numerical
value is then represented by a group of pulses of equal amplitude and
duration. The absence or presence of pulses represent the modulating
signal's value in the binary number system. This system has many advan-
tages and, therefore, has many applications in modern communications.

You'll study both analog and digital modulation in this unit as well as
multiplexing. This is a technique used to send many different communi-
cations channels over a common circuit.
5-4 UNIT FIVE
J 5-5
Pulse Modulation

UNIT OBJECTIVES
When you have completed this unit, you should be able to:

1. List the two broad categories of pulse modulation.

2. Name the two types of analog pulse modulation.

3. State the sampling theorem.

4. Define Nyquist rate, PAM, PTM, PDM, and PPM.

5. State the advantage PTM has over PAM.

6. Define digital pulse modulation, quantization, and quantization


noise.

7. List the three basic types of digital pulse modulation and describe
each one.

8. Define multiplexing.

9. Name the two basic types of multiplexing and define each one.

10. Define frame, commutator, and decommutator as they apply to


multiplexing.
5-6 UNIT FIVE

ANALOG PULSE MODULATION


So far, we have divided pulse modulation into two broad categories:
analog and digital. In analog pulse modulation, the indication of the
sample amplitude is continuously variable, while in digital pulse mod-
ulation a numerical code is sent that indicates the sample amplitude.

In this section, you'll study analog pulse modulation in detail. You'll find
that it can be subdivided into pulse amplitude modulation and pulse time
modulation. Furthermore, you'll see that pulse time modulation includes
pulse duration, pulse position, and pulse frequency modulation.

However, before you can study the various types of pulse modulation, we
must discuss sampling and the sampling theorem.

^a^m^m
Pulse Modulation 5-7

Sampling

Sampling is just what the name implies. Rather than transmitting the
entire modulating waveform, a periodic sample is taken and only the
sample is transmitted. If enough samples are sent, the wave can be
reconstructed at the receiving end. An example of sampling is shown in
Figure 5-1. Here, a switching signal periodically energizes a relay and
thereby connects the modulating signal to the output. If the sampling
signal's frequency is high enough, sufficient information will appear at
the output to accurately reproduce the modulating waveform. In fact, all
that is needed to demodulate the output signal is a low-pass filter. This
filter will remove the pulses and extract only the original information.

The critical part of thissystem is the sampling frequency. The minimum


frequency is defined by the sampling theorem:

The sampling frequency in any pulse modulation system must


be equal to, or exceed, twice the highest signal frequency in
order to convey all the information of the original signal.

MODULATING SIGNAL

^vAJ
OV

\in w
OUTPUT

+V

0V J
1
SAMPLING SIGNAL 1
Figure 5-1
A simplified sampling system.
5-8 UNIT FIVE

\ -> f

B
f
m 2f 3f 4f 5f
m m m m m

GUARD BAND

nr^r^, f
m
i

f
s
-f
m
f
s
f
s
+ f
m
2f
s

cxrxrACoo
f -f f f 2f
c 3f

Figure 5-2
The effects of sampling frequency (f.) on the
signal's frequency spectrum.

The reasons for this are shown in Figure 5-2. Figure 5-2 A shows the
bandwidth of a modulating signal with a maximum frequency of fm This .

signal is then sampled at a rate of 2 fm and the resulting frequency


spectrum is shown in Figure 5-2B. At 2fm there is a suppressed carrier
with both upper and lower sidebands spaced around it. The lower
sideband extends down to fm since it is 2fm - fm = fm The upper sideband .

extends to 2fm + fm = 3 fm Since the sampling signal is a pulse train, its


.

harmonics theoretically extend to infinity as shown by carriers at 4 fm etc. ,

In practice, the bandwidth is much less but still extends out quite a ways.

To demodulate the signal of Figure 5-2B, it is necessary to have an ideal


low-pass filter with a cutoff frequency of fm . If the filter's cutoff was
infinitely sharp at fm , it would pass the original modulating signal and
nothing else.
Pulse Modulation 5-9

Since ideal low-pass with infinitely sharp cutoffs are hard to come
filters

by, a sampling frequency of 2fm is never used. Instead, a frequency


slightly greater than 2fm is used. The result is shown in Figure 5-2C.
When the sampling frequency
(fs ) is greater than 2fm a guard band ,

appears between fm and the lower end of the first lower sideband. This
guard band allows the use of a practical low-pass filter. However, the
wider the guard band is, the greater the bandwidth of the pulse mod-
ulated signal becomes.

One example of a pulsemodulation system is the telephone network.


Here, the voice bandwidth is limited by filters to 3.3 kHz and the sam-
pling frequency is 8 kHz. This is well above 2fm and results in a guard
band offs - 2fm or 8kHz - 6.6kHz = 1.4 kHz. This allows relatively simple
low-pass filters to be used as demodulators. However, the guard band is
not so wide as to unduly increase the signal bandwidth.

The situation shown in Figure 5-2D occurs when the sampling frequency
is less than 2fm Here, the overlap between the modulating signal and the
.

lower sideband of fs means that no amount of filtering will allow an exact


recovery of thefull modulating signal.

Figure 5-2 proves the sampling theorem since it shows that, in principle,
the original signal can be recovered when fs = 2 fm (Figure 5-2B). It also
shows why the minimum sampling frequency is 2 im (Figure 5-2D). This
minimum sampling frequency is called the Nyquist rate. Figure 5-2C
proves that increasing fs above the Nyquist rate provides a guard band
which greatly eases the problems of filtering. However, increasing fs also
increases the signal bandwidth. Therefore, a compromise must be
reached that allows sufficient guard band for filtering at a minimum
signal bandwidth.
5-10 UNIT FIVE

Pulse Amplitude Modulation

The simplest form of pulse modulation is pulse amplitude modulation


(PAM). In PAM, the amplitude of the pulse varies in proportion to the
amplitude of the signal. This is illustrated in Figure 5-3. The modulating
signal is shown in Figure 5-3A and the sampling signal in Figure 5-3B.
Figure 5-3C shows a dual-polarity PAM signal. This results if the
waveform in "A" is centered on a zero-volt axis or, in other words, is a
true AC wave. If a DC level is added to the modulating signal, single-
polarity PAM results, as shown in Figure 5-3D. In this case, sufficient DC
level is added to ensure that the pulses are always positive. Likewise, a
negative DC voltage could be used to obtain negative pulses.

A hs

/U U
^n IT
A
\
h A

UuuLluuJ
Figure 5-3
Examples of dual polarity

and single polarity PAM.


Pulse Modulation 5-11

One practical method of generating aPAM signal is shown in Figure 5-4.


This circuit uses a 4016 integrated circuit CMOS switch. Basically, is an it

FET logic switch. When the sampling pulse goes positive, the switch
closes and the modulating input appears across R, and the output. When
the sampling pulse drops to zero, the switch opens and the output is zero.
As shown, the circuit provides dual-polarity PAM. However, single-
polarity PAM can be achieved by adding R and R2 These resistors form a
x
.

voltage divider that adds a DC level to the input signal. The result is that
the input AC wave now varies around a positive DC reference rather than
a zero-volt reference.

The demodulator for a PAM signal is merely a low-pass filter. It removes


the sampling signal and its harmonics, and passes the original modulat-
ing signal. However, the rolloff of the filter must be steep enough to pass
the highest modulating frequency and to fully attenuate the lowest sam-
pling frequency component. That is, the filter's cutoff must fall well
within the guard band of the particular PAM system.

When pulse amplitude modulation is used, it is sent over cable or wire, or


it can be used to modulate a radio frequency carrier. When this is done,

the PAM signal normally frequency modulates the carrier rather than
amplitude modulate the carrier. However, PAM is not used very often to
transmit information, since it is more susceptible to noise interference
than other forms of pulse modulation.

+v

MODULATING i

INPUT I
PAM OUTPUT

SAMPLING
INPUT

Figure 5-4
One method of obtaining PAM.
5-12 UNIT FIVE

Pulse Time Modulation

In pulse timemodulation (PTM), the modulating signal is sampled, just


as it is in PAM. However,
in PTM, the amplitude of the sample is
indicated by a timing variation of the modulated pulse, rather than an
amplitude variation. The variable timing characteristic may be the dura-
tion, position, or frequency of the pulses. Therefore, there are three basic
types of PTM: pulse duration modulation, pulse position modulation,
and pulse frequency modulation. This section will discuss the first two in
detail. However, pulse frequency modulation has no significant practical
applications and is, therefore, omitted.

In PTM, only the particular timing characteristic is modulated. The


amplitude of each pulse remains constant. Therefore, amplitude limiters
can be used at the receiver to remove most noise pulses or spikes. For this
reason, PTM has the same advantage over PAM that frequency modula-
tion has over amplitude modulation. Thus, PTM is used much more often
than PAM in communications applications.

PULSE DURATION MODULATION

This type of PTM is also called pulse width or pulse length modulation,
however, pulse duration modulation (PDM) is the preferred term. There
are three different classifications of PDM: symmetrical PDM, leading
edge PDM, and trailing edge PDM. These are shown in Figure 5-5 along
with the sine wave modulating signal.
J 5-13
Pulse Modulation

SAMPLING POINTS

TIME

lttl

n_j~L
Figure 5-5
The three types of pulse duration modulation.

Figure 5-5A shows a symmetrical PDM waveform. Here, the modulating


signal is sampled and both the leading and trailing edges of the pulse are
varied in accordance with the sample amplitude. When the sample is a

high positive voltage, the pulse duration increases, and when it is a high
negative voltage, the pulse duration decreases. When the modulating
wave is at zero, the pulse is at its average or reference duration. The
spacing between the center of the pulses remains constant, as shown.

Leading edge PDM is shown in Figure 5-5B. In this type of PDM, the
sample amplitude varies the leading edge of the pulse. The trailing edge
of each pulse is fixed and, therefore, the spacing or timing between each
pulse's trailing edge is constant.

Figure 5-5C shows trailing edge PDM. Here, the sample amplitude varies
the trailing edge of the pulse, with the leading edge remaining fixed.
5-14 UNIT FIVE

SAMPLING INPUT

CLIPPING
LEVEL
/

jinnjui
PDM OUTPUT

MODULATING INPUT

Figure 5-6
Method of generating a PDM signal.

A typical PDM generator is shown in Figure 5-6. There are two inputs: a
triangular waveform at the sampling frequency, and the modulating
input, which is a sine wave. These two signals are summed and the
output is a triangle wave superimposed on the sine wave as shown. This
wave is then applied to a clipper which removes everything below the
clipping level shown. What remains is a triangle wave of varying
amplitude and duration. This wave is then amplified and limited by
several successive stages until all the pulses have the same amplitude,
and both the leading and trailing edges are vertical. The result is a PDM
wave. In this case, since the sampling wave is a triangle wave, the output
is symmetrical PDM.

Leading edge PDM is generated using a sawtooth sampling wave that has
a sloping leading edge and a vertical trailing edge. In this way, the
resulting pulse has its leading edge modulated. Likewise, trailing edge
PDM is generated using a sawtooth sampling wave with a vertical leading
edge and a sloping trailing edge.
Pulse Modulation 5-15

PAM WAVEFORM OUTPUT WAVEFORM

JUl
INPUT SIGNAL

_r~LTLr~L
HOLDING
CONSTANT DIODE
INPUT SAMPLING LOW-PASS
CURRENT
SOURCE
-M- CIRCUIT FILTER
-> OUT

CHARGING
CAPACITOR
T L ft

SAMPLING SIGNAL

DUMP
CIRCUIT

CHARGING DUMP DUMP SIGNAL


TIME CIRCUIT
B
^ "ON" I
I

z
ULSE ^^=3 SAMPL ING
DURATION T)ME

Figure 5-7
A PDM/PAM conversion demodulator.

The simplest method of demodulating a PDM wave is to use a low-pass


filter. However, technique can sometimes produce severe distortion.
this
Therefore, a more elaborate but lower distortion system is usually used.
One possible system is shown in Figure 5-7A. This circuit converts the
PDM signal to PAM and then uses a low-pass filter to demodulate the
PAM wave. This results in much lower distortion.

In Figure 5-7A the PDM input signal is applied to a constant current


source. When the input goes positive, it turns on the constant current
source, which then linearly charges C x
.
Q continues to charge until the
input signal drops back to zero. At this time, the current source is turned
off and the voltage is held constant across Q by the holding diode which
prevents any discharge. The sampling on by a circuit is then turned
sample pulse which is derived from the input PDM signal. As soon as the
sample is taken, the dump circuit discharges the capacitor and prepares it
for the next pulse. The voltage waveform that appears across C, is shown
in Figure 5-7B. The output of the sampling circuit is a PAM signal as
shown. This is applied to the low-pass filter and the original modulating
signal is recovered.
5-16 UNIT FIVE

MODULATING S IGNAL

A !

REFERENCE PULSES

I
I

I i

PPM PULSES

JUlLOillUlUUL
B

Figure 5-8
Pulse position modulation.

PULSE POSITION MODULATION

The next form of PTM is known as pulse position modulation (PPM).


With this form of PTM, both pulse amplitude and duration remain con-
stant while the position of the pulse, relative to a reference pulse, is
varied in accordance with the modulating signal.

Figure 5-8B shows a typical PPM


waveform. The modulating signal and
the reference pulses are shown in Figure 5-8A. Note that when the
modulating signal goes positive, the output pulse lags the reference pulse
by an amount of time proportional to the sample amplitude. Similarly,
when the modulating signal goes negative, the output pulse now leads
the reference pulse by a proportional amount.

One method of generating a PPM signal is shown in Figure 5-9. This


method uses the modulating signal to first generate a PDM waveform
such as that shown in Figure 5-10. In this case, it is trailing edge PDM,
where the leading edge is fixed and the trailing edge is modulated. The
circuit of Figure 5-9 effectively "synchronizes" a fixed duration pulse to
this modulated trailing edge. It does this by first differentiating the PDM
signal,with the resultant waveform as shown in Figure 5-10. The positive
spikes are then clipped off since they are fixed in time. The remaining
negative spikes are synchronized to the trailing edge and are, therefore,
position modulated. This clipped waveform is then used to trigger a
monostable multivibrator which generates a fixed duration pulse for each
input trigger. The duration of the pulse is determined by the monosta-
ble's RC time constant.
Pulse Modulation 5-17

_tul -HV ~~rT~ n_n_


MODULATING PDM
Jl
DIFFEREN- V
1
a
^ MONOSTABLE
CLIPPER MULTI- PPM OUTPUT
SIGNAL GENERATOR TIATOR
VIBRATOR

Figure 5-9
A pulse position modulation circuit.

SAMPLING POINTS

TIME

n_n_n__n n_

REFERENCE PULSE
POSITION

Figure 5-10
Waveforms for the PPM generator.
5-18 UNIT FIVE

Figure 5-11 shows a typical PPM demodulator. The PPM signal is first

applied to a limiter stage to remove any noise and to effectively regener-


ate the signal. It is then sent to a circuit that removes the synchronizing
pulses sent by the transmitter. These pulses are used to lock the local
reference pulse generator to the correct phase and frequency to demod-
ulate the PPM signal.The reference pulses and the PPM signal are then
applied to a flip-flop. Both these waveforms are shown in Figure 5-12.
Note that the incoming PPM pulse lags the reference pulse by an amount
proportional to its As shown, in Figures 5-11 and
position modulation.
5-12, the reference pulse turns on the flip-flop and the PPM pulse turns it
off. Therefore, the output of the flip-flop is a PDM wave whose pulse

duration is directly proportional to the position of the input pulse. The


resulting PDM signal is then converted to PAM as described earlier. The
PAM wave is then coupled to a low-pass filter to recover the original
modulating signal.

MODULATING
PPM PDM PAM INFORMATION

OFF,
PPM PDM/PAM LOW-PASS
ON FLIP-FLOP OUTPUT
INPUT CONVERTOR FILTER

SYNCHRON- REFERENCE <J^] REFERENCE


IZING PULSE PULSES
CIRCUITS GENERATOR

Figure 5-11
One method of demodulating a PPM signal.
Pulse Modulation 5-19

INCOM ING PPM WAVE TIME

n
i
n
i
n__n__n__n__n_ I
'

i
i

LOCAL REFERENCE PULSES

_ni_TLi_n.
i i
Jl fL_n_n.
i
i

i l
i I i
i

i I
i
I i
i

FLIP-FLOP OUTPUT

n_rL_
PAM S IGNAL

ji n.

OUTPUT WAVE

Figure 5-12
Waveforms for the PPM demodulator.
5-20 [uni
UNIT FIVE

Self-Review Questions

1. What are the two broad categories of pulse modulation?

2. What are the two basic types of analog pulse modulation:

3. What is the sampling theorem?

4. What is the Nyquist rate for a modulation signal whose highest


frequency is 8 kHz?

5. What is PAM? What are the two different types of PAM?


Pulse Modulation 5-21

6. A filter can be used to demodulate a PAM


signal.

7. What is PTM? What are the three basic types of PTM?

8. What advantage does PTM have over PAM?

9. What is PDM? List the three types of PDM.

10. What is PPM?


5-22 UNIT FIVE

Self-Review Answers

1. The two broad categories of pulse modulation are analog pulse


modulation and digital pulse modulation.

2. The two basic types of analog pulse modulation are pulse


amplitude modulation and pulse time modulation.

3. The sampling theorem states:


The sampling frequency in any pulse modulation system must be
equal to, or exceed, twice the highest signal frequency in order to
convey all the information of the original signal.

4. The Nyquist rate is twice the highest modulation frequency. In this


case it is 8 kHz x 2 = 16 kHz.

5. PAM is pulse amplitude modulation. The two different types are


dual-polarity PAM and single-polarity PAM.

6. A low-pass filter can be used to demodulate a PAM signal.

7. PTM is pulse time modulation. The three basic types of PTM are
pulse duration modulation, pulse position modulation, and pulse
frequency modulation.

8. PTM has the advantage over PAM in that it is relatively immune to


noise. This is because it can pass through a noise limiter stage
without distortion.

9. PDM is pulse duration modulation. The three types of PDM are


symmetrical PDM, leading edge PDM, and trailing edge PDM.

10. PPM is pulse position modulation.


Pulse Modulation 5-23

DIGITAL PULSE MODULATION


Pulse amplitude modulation and pulse time modulation are both suscep-
tible to noise distortion. With PAM, it is quite obvious that once noise is
present it is virtually impossible to remove it or to regenerate the signal.
However, with PTM, a limiter can be used to remove most of the noise.
Still, in PTM, the arrival of the leading and trailing edges of the pulses

must be recorded precisely to accurately demodulate the signal. Since


these pulse edges have finite rise and fall times, they too can be distorted
to some degree by noise. For these reasons and others, digital pulse
modulation (DPM) was developed. In this type of pulse modulation,
groups of pulses are transmitted that represent binary numbers that
correspond to the modulating voltage levels. Here, demodulation of the
transmitted signal does not depend on the amplitude, duration, or posi-
tion of the pulses, but only on their presence or absence. Under these
conditions, it is relatively easy to recover pulses, even in the presence of
high noise and interference levels. It is also a simple matter to regenerate
this type of pulsemodulation. As a result, digital pulse modulation is

used almost exclusively in long distance communications.


5-24 UNIT FIVE

Pulse Code Modulation

Pulse code modulation or PCM is the major form of digital pulse modula-
tion. In PCM, the modulating signal
sampled, just as in other forms of
is

pulse modulation. The sample amplitude is then converted into a binary


code and transmitted as a stream of pulses.

In the other forms of pulse modulation, the sample amplitude is con-


verted directly into pulse amplitude, duration, or position. However, in
PCM, since the amplitude must be transmitted as a specific number out of
a limited range of numbers, the sample amplitude must first be quan-
tized. That is, each sample amplitude must be converted to the nearest
standard amplitude or quantum. For example, suppose the PCM system
has a total signal amplitude range of 7 volts and each 1-volt level corres-
ponds to a specific binary code. Therefore, for this system each quantum
or standard level is 1 volt. This is shown in Figure 5-13 for a sine wave
signal with 8 quantum steps from OV to 7V. Note that the quantizing
waveform is actually a form of P AM although it is limited to the quantum
,

steps and is not continuously variable.


Pulse Modulation 5-25

You'll notice that the first sampling point in Figure 5-13 is approximately

3.3V. Since there is no quantum level at this voltage, it is represented by


the nearest level, which is 3V. This occurs at many places on the
waveform. This error or distortion is called quantizing noise. It is noise
because the errors are random. This is because the difference between the
quantum level and the actual signal at any instant is completely unpre-
dictable. The obvious method of reducing quantizing noise is to increase
the number of quantum levels until the noise level is acceptable. How-
ever, increasing the number
of levels increases the transmission
bandwidth, so a compromise must be made between acceptable quantiz-
ing noise and bandwidth.

QUANTIZING
WAVEFORM

V
SAMPLING PULSES

Figure 5-13
A quantized sine wave.
5-26 UNIT FIVE

111
7V-
110,110 110, 110
6V

101 101
5V-

4V-
on
Oil Oil
3V-
Figure 5-14 010 010
Coding the quantized wave. 2V-
001 001
IV-
000 000,000.000
0V-

After quantization has occurred, each sample must be coded as a binary


number before can be transmitted as PCM. Figure 5-14 shows the
it

results of coding the quantized waveform from Figure 5-13. Since there
are only 8 quantum levels, they can be represented by a 3-bit binary word,
with 000 2 representing OV and 111 2 representing 7 V. Once the quantiz-
ing waveform is coded, each sequential sample is transmitted as a pulse
code. A table comparing the quantizing level, binary number, and pulse
code is shown in Figure 5-15.

QUANTIZING BINARY PULSE


LEVEL NUMBER CODE

OV 000

IV 001 n

2V 010 n

3V 011 n n

4V 100 n

5V 101 J1 n

6V 110 Jl n

7V 111 JLJUL
Figure 5-15
Three bit PCM.
Pulse Modulation 5-27

111
7V -i-
110
6V--
QUANTIZED WAVE
101
5V--
100
4V--

3V--
on
010
2V--
001
1V--
000
ov-L-

Figure 5-16
A PCM waveform.

JUUUUUL JUUUl
000 011 101110
Jl
111100010 001
Jl

PCM WAVE

Figure 5-16 shows the resultant PCM wave for a typical quantized wave.
In practice, synchronizing pulses are required to insure that the receiver
decodes the correct pulses. Also, in practical PCM systems, an 8-bit
binary word is often used instead of the 3-bit word shown here. The 8-bit

word provides 256 quantizing steps, which allows much better reproduc-
tion of the modulating signal with very little quantizing noise.

A simplified diagram of a PCM transmitter is shown in Figure 5-17. It

consists of two sections. The first section is an analog-to-PAM converter


that performs the quantizing process. The second section is a PAM-to-
PCM converter that does the binary coding. The total system is actually
an analog-to-digital converter.

QUANTIZATION A CODING

% INPUT

O M
ANALOG
TO PAM
CONVERSION
I. PAM

PCM
TO
-Jui_juuL_n_n_

OUTPUT

ANALOG TO DIGITAL
CONVERSION

Figure 5-17
A PCM transmitter.
5-28 UNIT FIVE

A PCM receiver is shown in Figure 5-18. It is made up of a PCM-to-PAM


converter and a low-pass filter to convert that PAM back to the original
modulating signal. It is, in essence, a digital-to-analog converter.

As mentioned earlier, the primary advantage of PCM is its much better


immunity to noise and interference. For example, a typical PCM trans-
mission can be sent over a communications channel having a signal-to-
noise ratio of 2 1 dB with minimal error. In fact, the error would be just one
pulse missed or decoded improperly every 17 minutes. If the signal-to-
noise ratio is improved to 23 dB, the error rate drops to one error every
four months! To achieve this low error rate in an AM system would
require a signal-to-noise ratio of 60 to 70 dB.

nn nnn n n

INPUT
o- ^.
PCM
TO
PAM
K LOW
PASS
FILTER OUT

DIGITAL TO ANALOG
CONVERSION

Figure 5-18
A PCM receiver.
Pulse Modulation 5-29

Other DPM Systems

There are many other digital pulse modulation systems that are varia-
tions of PCM or slightly different systems. Two of the more common
types are differential PCM and delta modulation.

Differential PCM is very similar to ordinary PCM. However, in this


system, each binary word indicates the difference in amplitude between
the present sample and the previous one. It specifies whether the change
is positive or negative and the magnitude of the change. Therefore, the
PCM code needs amplitude of the sample
to indicate only the relative
rather than the absolute amplitude as in normal PCM. This system takes
advantage of the redundancy of most modulating signals, such as speech.
That is, each sample is related to the previous sample; large variations
from one to the next are unlikely. For this reason, it takes fewer bits to
indicate the relative change in amplitude than the absolute amplitude.
Thus, differential PCM has a smaller bandwidth than normal PCM. How-
ever, complications in encoding and decoding presently outweigh any
bandwidth advantage. As a result, differential PCM has few applications.
5-30 UNIT FIVE

Delta modulation is basically a 1-bit differential PCM system. In the


is sent per sample and its
simplest form of delta modulation, just one bit
whether the signal is larger or smaller than the previ-
polarity indicates
ous sample. An example of this is shown in Figure 5-19. Figure 5-19A
shows the quantized wave and Figure 5-19B shows the delta modulation
pulse train.

This system has several advantages: greatly simplified encoding, decod-


ing and quantization. However, it cannot handle rapid amplitude varia-

tions between samples unless the sampling frequency is very high.


Therefore, delta modulation requires a higher sampling rate and larger
bandwidth than a comparable fidelity PCM system. Its advantages
somewhat offset this disadvantage and, as a result, its use is increasing.

QUANTIZED WAVE

B
+
ov

^rirW^ \
DELTA MODULATION

Figure 5-19
Delta modulation.
Pulse Modulation 5-31

Self-Review Questions

11. What is digital pulse modulation?

12. Define quantization.

13. What is quantizing noise and how can it be reduced ?

14. List the three basic types of digital pulse modulation and describe
each one.
5-32 UNIT FIVE

Self-Review Answers

11. In digital pulse modulation, groups of pulses are transmitted that


represent binary numbers that correspond to the modulating vol-
tage levels.

12. Quantization is the process of converting each sample amplitude to


the nearest standard amplitude.

13. Quantizing noise is the error or distortion created by converting


sample amplitudes to the nearest standard amplitudes.

14. PCM code modulation transmits a binary word represent-


or pulse
ing the amplitude of each sample.

Differential PCM transmits a binary word representing the


amplitude difference between the previous sample and the present
sample.

Delta modulation transmits a single bit per sample whose polarity


indicates whether the signal is larger or smaller than the previous
sample.
Pulse Modulation 5-33

MULTIPLEXING
Multiplexing is the process of transmitting several separate information
channels over the same communications circuit simultaneously without
interference. There are two basic types of multiplexing: time division
multiplex (TDM) and frequency division multiplex (FDM).

Time Division Multiplex

In TDM, several information channels are transmitted over the same


communications circuit simultaneously using a time sharing technique.
As an example, PAM waveforms can be generated that have a very low
duty cycle. This means that if a single channel is transmitted, most of the
transmission time would be wasted. Instead, this time is fully utilized by
transmitting pulses from other PAM signals during the intervals. A
PAM-TDM waveform for three channels is shown in Figure 5-20. The first
pulse is a synchronizing pulse which is used at the receiver in demultip-
lexing. The second pulse is amplitude modulated by channel 1 the third ,

by channel 2, and the fourth by channel 3. This set of pulses is called a


frame. Four complete frames are shown in Figure 5-20.

CHANNEL 1

CHANNEL 3

CHANNEL 2

CHANNEL 1

CHANNEL 3

- CHANNEL 2

CHANNEL 2
CHANNEL 3

Figure 5-20
Three channel time division multiplex using
single polarity PAM.
5-34 UNIT FIVE

TRANSMITTER RECEIVER
A

SYNCHRONIZING O ^CHANNEL 1

CHANNEL 1
O
TRANSMISSION
CHANNEL
CHANNEL 2 O
^CHANNEL 2

CHANNEL 3 O
> CHANNEL 3

TRANSMITTER
CLOCK

Figure 5-21
A very simplified diagram of a TDM transmitter
and receiver.

A simplified diagram of a TDM transmitter and receiver is shown in


Figure 5-21. In the transmitter, a rotating switch, called a commutator,
connects each channel to the output in sequence. The speed of the
commutator is set by the transmitter clock. Note that a synchronizing
channel is also sent. This locks the receiver clock in phase with the
transmitter clock.

Once the TDM signal reaches the receiver, it is demultiplexed, or de-


commutated, by another rotating switch. This switch routes the pulses to
the correct channel. It also routes the sync pulse to the receiver clock to
keep the transmitter and receiver synchronized. In practical TDM trans-
mitters and receivers, the rotating switches are replaced by electronic
switches.

The primary advantage of TDM is that several channels of information


can be transmitted simultaneously over a single cable, a single radio
any other communications circuit. Also, any type of pulse
transmitter, or
modulation may be used in TDM. In fact, many telephone systems use
PCM TDM.
-
Pulse Modulation 5-35

Frequency Division Multiplex

Frequency division multiplex or FDM is normally an analog technique


and, therefore, doesn't use pulse modulation. However, it is included

here to complete the topic of multiplexing.

FDM, like TDM, is used to transmit several information channels over the
same communications circuit simultaneously. However, in FDM, each
channel uses a different band of frequencies to avoid interference. Thus,
the distinction between FDM and TDM becomes apparent. In TDM, each
channel occupies the entire frequency spectrum for only a fraction of the
time. In FDM, each channel continuously occupies a small fraction of the
transmission spectrum.
5-36 UNIT FIVE

Let's examine a simplified FDM system. Figure 5-22 shows the block
diagram of a 5-channel FDM transmitter. Each channel is a telephone
voice circuit whose bandwidth is limited to 3.5 kHz. Channel 1 is trans-
mitted just as it is, so it occupies the spectrum from approximately 100 Hz
to 3500 Hz. Next, a 4 kHz pilot carrier is transmitted. It is used at the
receiver in the demultiplexing of the FDM signal.

0-3. 5kHz
CHANNEL 1 O-

PILOT 4kHz 4kHz


CARRIER
OSCILLATOR
T
X2 X4

BALANCED LSB 4. 5-8kHz


CHANNEL 2Q >
MODULATOR FILTER

8kHz RADIO
LINEAR 0-20kHz TRANSMITTER
^
^ ADDER OR
BALANCED USB 8-11.5kHz|
CHANNEL 30 > i ^> CABLE
MODULATOR FILTER

BALANCED LSB 12. 5-16kHz


CHANNEL 4
MODULATOR FILTER

16kHz

BALANCED USB 16-19. 5kHz


CHANNEL 50 >
MODULATOR FILTER

Figure 5-22
An FDM transmitter.

_.
Pulse Modulation 5-37

10 12 14
FREQUENCY (kHz)

Figure 5-23
The output spectum of the FDM transmitter.

Channel 2 is applied to a balanced modulator along with an 8 kHz


subcarrier, which is derived from the pilot carrier. The output of the
balanced modulator is applied to a lower sideband filter. The result is a
lower sideband, suppressed-carrier signal that extends from 4.5 kHz to 8
kHz. Channel 3 is also applied to a balanced modulator along with the 8
kHz subcarrier. However, it becomes an upper sideband signal extending
from 8 kHz to 11.5 kHz. Likewise, channels 4 and 5 become lower and
upper sidebands, respectively, on a 16 kHz subcarrier.

The output frequency spectrum of this transmitter is shown in Figure


5-23. Here, channel 1 occupies the spectrum from 100 Hz to 3.5 kHz. It is
shown as having a lower amplitude at the higher frequencies to represent
average speech content. The pilot carrier is at 4 kHz and channel 2 is from
4.5 kHz to approximately 8 kHz. Note that channel 2's frequency spec-
trum is inverted. This is because it is a lower sideband signal. The same is
true for channel 4. On the other hand, channels 3 and 5 are not inverted
since they are upper sideband signals. This composite FDM signal is then
used to modulate a radio transmitter, which may use any form of modula-
tion, or it is sent on a telephone line or cable.
^

5-38 UNIT FIVE

The block diagram of a 5-channel FDM receiver is shown in Figure 5-24.


Here, each specific frequency band is separated by filters. This demod-
ulates channel 1 immediately. The 4 kHz pilot carrier is also separated
and used to synchronize an oscillator. It, in turn, provides the necessary
subcarriers for demodulation. Channels 2 through 5 are each separated
and applied to a balanced modulator along with the appropriate subcar-
rier. The audio channels are then separated from the balanced mod-
ulators' outputs by low-pass filters. Thus, each channel has been recov-
ered.

One disadvantage of FDM is that the transmission amplifiers and other


circuits must be extremely linear. This is because any nonlinearity will
cause harmonic distortion and cross modulation. For example, any har-
monics from channel 1 will appear in all the other channels, to a certain
extent. Also, if channels 2 and 3 mix, their cross modulation products
will appear in channels 4 and 5 as well as channel 1. This can cause
severe distortion. With a TDM signal, this distortion still occurs, but
since the signal occupies the entire spectrum at one time, it does not
interfere with the other channels.

0-3. 5kHz
-» CHANNEL 1

FILTER
4kHz 8kHz 16kHz

4kHz SYNCHRONOUS
-^ X2 -t^ X2 1
FILTER OSCILLATOR

3. 5kHz
4. 5-8kHz BALANCED
LOW PASS •CHANNEL 2
FILTER MODULATOR FILTER

RADIO 3. 5kHz
8-11. 5kHz BALANCED
FDM Q RECEIVER LOW PASS 'CHANNEL 3
INPUT OR CABLE FILTER MODULATOR FILTER

16kHz BALANCED 3. 5KkHz


12. 5-
>— LOW PASS CHANNEL 4
FILTER MODULATOR
FILTER

16-19. 5kHz 3.5kHz


BALANCED
LOW PASS CHANNEL 5'
FILTER MODULATOR
FILTER

Figure 5-24
A 5-channel FDM receiver.

-.
Pulse Modulation 5-39

Self-Review Questions

15. What is multiplexing?

16. What are the two basic types of multiplexing?

17. What is TDM?

18. A complete set of TDM pulses from one sync pulse to the next is

called a

19. The rotating switches in a TDM transmitter and receiver are called
the and the , respectively.

20. What is FDM?

21. What is the primary disadvantage of FDM?


5-40 UNIT FIVE

Self-Review Answers

15. Multiplexing is the process of transmiting several separate infor-


mation channels over the same communications circuit simultane-
ously without interference.

16. The two basic types of multiplexing are time division multiplex
and frequency division multiplex.

17. TDM is time division multiplex. In TDM, several channels are


transmitted simultaneously using a time sharing technique.

18. A complete set of TDM pulses from one sync pulse to the next is

called a frame.

19. The rotating switches in a TDM transmitter and receiver are called
the commutator and decommutator, respectively.

20. FDM is frequency division multiplex. In FDM, each channel oc-


cupies a different band of frequencies to avoid interference.

21. The primary disadvantage of FDM is that it requires extremely


linear amplifiers to avoid cross modulation and harmonic distor-
tion.

l^HH
Unit 6

ANTENNAS
6-2 UNIT SIX
Antennas 6-3
J

INTRODUCTION
An antenna, usually a wire or collection of wires, converts high fre-
quency current into electromagnetic waves for transmission and vice-
versa for reception. In previous units on transmitters and receivers,
antennas were not discussed but were assumed to exist at the output of
transmitters and at the input of receivers. This unit discusses antennas
and antenna systems.

There are three main elements in the antenna "system." These are the
antenna itself, the transmission line, and the transmitted radio wave's
path. The transmission line transfers the RF energy to the antenna which
then radiates an RF wave. The atmosphere then influences, to a certain
degree, the path that the radio wave follows. This unit discusses all three
of these important items in detail.

The Unit Objectives listed next state exactly what you are expected to
learn from this unit. Study this list now and refer to it often as you study
the text.
6-4 UNIT SIX
Antennas 6-5

UNIT OBJECTIVES
When you have completed this unit, you will be able to:

1. State the primary purpose of a transmission line.

2. State the difference between balanced and unbalanced transmis-


sion lines.

3. State what determines the characteristic impedance of a transmis-


sion line.

4. Write the formulas for standing-wave ratio.

5. Define reflectometer.

6. State the characteristics of resonant transmission lines.

7. List the advantages and disadvantages of balanced and unbalanced


transmission lines.

8. Define antenna, field strength, end effects, and the mirror image
principle.

9. Identify a half-wave dipole from its voltage and current distribu-


tion.

10. Identify the radiation pattern for a half-wave dipole.

11. List the radiation fields present in an electromagnetic wave and


state which field determines the polarization.

12. State the input impedance of a dipole.


6-6 UNIT SIX

13. Find the length of a half- wave dipole for a given frequency.

14. Identify the radiation pattern for a quarter- wave vertical antenna.

15. Find effective radiated power when given antenna gain, transmis-
sion line loss, and transmitter power output.

16. Name the two types of antenna arrays.

17. List the elements used in a Yagi antenna.

18. List two examples of driven arrays.

19. Name the three broad classifications of radio wave propagation and
describe each one.

20. Define critical frequency, maximum usable frequency, and op-


timum usable frequency.

21. State the primary cause of fading.

22. Name the various layers of the ionosphere.

23. List the variations of the ionosphere and their causes.

24. List the two main types of ionospheric disturbances.

25. Find the space wave radio horizon when given antenna height.

26. Calculate the required antenna heights to achieve a given com-


munications distance.

27. List thetwo common methods used to extend VHF, UHF, and
higher frequency communications range.
Antennas 6-7

TRANSMISSION LINES
The primary purpose from a
of a transmission line is to transfer energy
source to the load. A good example is the transmission line used to
transfer RF energy from a transmitter to the antenna. Lines are also used
to interconnect equipment such as transmitters to power amplifiers,
receivers to speakers, etc. When the energy being transferred is DC or
audio, the lines present no special problems. At these frequencies, they
usually appear as short circuits and, for most purposes, can be ignored.
However, at higher frequencies, transmission lines take on very peculiar
characteristics that cannot be ignored. These characteristics are primarily
due to the signal's wavelength, the distance the wave travels in one cycle.
This section discusses these characteristics in detail.

The Balanced Transmission Line

Most of us are familiar with the "TV twin lead" type of transmission line
shown in Figure 6-1 A. This is the type of lead-in wire used to connect an
antenna to a television receiver. This type of transmission line is known
as a balanced line. Both wires of the line carry RF current and the current
in each wire is 180° out of phase with the other wire. (Shown in Figure
6-lB). Therefore, the current in this type of transmission line is balanced
with respect to electrical ground. On the other hand, with an unbalanced
transmission line, one wire would be at ground potential, while the other
wire would carry the RF current.

Jz^a

^ Vszm
\

B I

Figure 6-1
Balanced transmission line.
6-8 UNIT SIX

The equivalent circuit of a balanced transmission line is shown in Figure


6-2. Inductance is present because, whenever current flows through a
conductor, a magnetic field produced around it. Since a potential
is

difference exists between the two conductors of the transmission line,


due to the 180° phase difference, an electrostatic field exists between the
conductors. Therefore, some small amount of capacitance also exists. The
equivalent circuit shown in Figure 6.-2 represents the ideal transmission
line. In a practical line, a series resistive element is present due to the
resistance of the conductors. Also, a parallel resistive element is present
due to leakage in the dielectric material that separates the two conduc-
tors. The inductive, capacitive, and resistive properties of the transmis-
sion line present a specific impedance to any AC waveform.

Figure 6-2
Balanced line equivalent circuit.

Ifan RF generator is connected to an infinitely long transmission line, a


forward traveling wave of RF energy will start at the generator and travel,
at the speed of light, down the line. This is shown in Figure 6-3A. This
applied RF energy causes a specific amount of current to flow in the
transmission line and a resulting potential difference or voltage to appear
across the two conductors. If you now take current and voltage measure-
ments, with an AC meter, at several points on this line, you will find that
the amplitude of voltage and current is constant at all points. That is, the
applied RF energy causes a specific current and voltage to appear at all
points on the transmission line. A graph of voltage and current on this
transmission line is shown in Figure 6-3B.
Antennas 6-9

oo

VOLTAGE

B CURRENT

LENGTH OF LINE
+ OO

Figure 6-3
An infinite transmission line.

Since voltage (E) and current (I) are constant at all points, impedance (Z)

is also constant, proven by the Ohm's law equation(Z= — ). Therefore,

the infinite length of transmission line has a constant impedance. This is

known as its characteristic impedance. Every transmission line has a


specific characteristic impedance which is due to its physical construc-
tion. The formula for the characteristic impedance of a balanced trans-
mission line is:

Z =276 x log —
r

where Z = characteristic impedance in ohms

d = center-to-center distance of
separation of conductors

r = radius of conductors (using the same


unit of measurement as with d)
6-10 UNIT SIX

Figure 6-4 shows a balanced transmission line with measurements d and


r indicated.

The following example shows how to find the characteristic impedance


of a typical balanced transmission line. The distance between conductors
is 1.22 cm and the conductor radius is 0.1 cm. Using the formula for
characteristic impedance:

Z =276 x log —
r

1.22
Z = 276 x log
0.1

Z =276 x log 12.2

Z =276 x 1.086

Z =300 a

Thus, the characteristic impedance of this particular line is 300 O.

Z • 276 x LOG 7
Q

Figure 6-4
Balanced transmission line dimensions.

•-L
Antennas 6-11

If you connect an infinitely long length of this 300 Cl line to an RF


generator, the input impedance that the generator will see is 300 Cl. In
fact, the impedance at all points on the line
is 300 ft. Therefore, if you cut

the line to some and connect a 300 ft resistor across the


practical length
far end of the line, the generator will still see a 300 ft input impedance.
You are, in essence, replacing the lost portion of the line with a 300 ft
resistor. Since the resistor has the same impedance as the lost portion of
the line, the previous conditions are not upset. Therefore, the transmis-
sion line behaves just as though it were The forward
infinitely long.
traveling wave of RF energy travels the length of the line and is com-
pletely dissipated in the 300 ft resistor. This condition exists whenever a
transmission line is correctly terminated; that is, whenever the load
impedance matches the characteristic impedance of the transmission
line. In the example given, the 300 ft load resistor matched the 300 ft

transmission line.

Standing Waves

If a transmission line has infinite length or is terminated in its charac-


teristic impedance, all the power applied to the line at one end is ab-
sorbed or dissipated by the load at the other end. However, if a transmis-
sion line is terminated in an impedance other than its characteristic

impedance, some (but not all) of the applied power is absorbed by the
load. The remaining power is reflected.

When a transmission line is incorrectly terminated, the power that is not


absorbed by the load is sent back toward the generator. The amount of this
reflected power is directly proportional to the amount of impedance
mismatch at the load. For example, if the line is terminated by a short
circuit or an open circuit, none of the power is dissipated in the termina-
tion and all of it is reflected back to the generator. As the termination
impedance approaches the line's characteristic impedance, less and less
power is reflected, until finally, when the impedances match, no power is
reflected.

When power is applied to a transmission line, a voltage and current


appear whose values depend on the characteristic impedance (Z ) and
the applied power. The voltage and current waves travel to the load (ZJ
and, if ZL = Z
the load absorbes all the power and none is reflected. The
,

only waves present at this time are the forward traveling waves of current
and voltage from the generator to the load.
6-12 UNIT SIX

REFLECTED WAVES
S3
OPEN

Figure 6-5
Forward and reflected waves.

However, if Z ;
is not equal to Z ,some power is absorbed and the rest is
reflected. Therefore, one set of current and voltage waves is traveling
toward the load and another set is traveling back to the generator. This is
shown in Figure 6-5. These two sets of traveling waves, going in opposite
directions, will alternately add to and subtract from one another due to
their relative phase differences. This in turn sets up a pattern known as
standing waves along the line. The standing wave pattern for an open-
circuited line is shown in Figure 6-6. You can see that stationary voltage
and current minimums and maximums have appeared. Compare this
voltage and current pattern to that of a correctly terminated transmission
line shown in Figure 6-3 . In the infinite or correctly terminated line, there
are no reflected waves to interfere with the forward waves. How do these
standing waves develop?

Figure 6-6
Standing wave pattern for an open circuited line.

iL
Antennas 6-13

For the moment, consider only the forward traveling voltage and current
waves. At the load, the current is zero and the voltage is maximum
because the load is an open circuit. This is shown in Figure 6-6. The
reflection that occurs at the open both voltage and current.
circuit affects
The voltage now starts travelingback to the generator, unchanged in
phase, but the current is reflected with a 180° phase inversion. Therefore,
at a point exactly a quarter-wavelength (1/4 A) from the load, the voltage is

permanently zero as shown in Figure 6-6, because the forward and


reflected voltage waves are exactly 180° out of phase. This occurs because
the reflected wave has had to travel a distance of 1/4 A + 1/4 A or 1/2 A
farther than the forward wave. Since 1/2 A is equal to a phase change of
180°, the two waves cancel and voltage is zero at this point. The current
wave has also traveled an extra 1/2 A; however, since it underwent a 180°
phase inversion on reflection, its total phase change is 360°. Therefore,
because the forward and reflected current waves are in phase, reinforce-
ment takes place and a current maximum occurs at exactly the same point
as the voltage minimum.

Figure 6-6 shows that 1/2 A from the load, there is a current minimum
(zero in this case) and a voltage maximum. This occurs because the
forward and reverse voltage waves are now in phase, voltage has traveled
a distance ofone wavelength (360°) to return to this point. At this same
place, the currentwaves cancel, because of the additional 180° phase
inversion that occurred on reflection. Figure 6-6 shows that these
maximum and minimum points repeat every half-wavelength and re-
verse every quarter-wavelength.
6-14 UNIT SIX

Figure 6-7
Standing wave pattern for a short circuited line.

Note that this condition is permanent and is determined by the load.


Figure 6-7 shows the standing-wave pattern for a short-circuited trans-
mission line. Here, the current at the load is maximum and the voltage is
zero because the load is a short circuit. At a quarter-wavelength from the
load, current is zero and voltage is maximum. At a half- wavelength from
the load, current is maximum and voltage is zero. Thus, the standing-
wave pattern for a short-circuited line is just the opposite of that for an
open-circuited line. This shows that the load does indeed determine the
standing-wave pattern.
Antennas 6-15

Standing Wave Ratio

The ratio of maximum current to minimum current or maximum voltage


to minimum voltage along a transmission line is called the standing-
wave ratio (SWR). This ratio is a measure of the mismatch between the
load and the transmission line. For example, the SWR is equal to 1 when
the load is perfectly matched, and equal to °° when the load is a short or
open circuit. If the transmission line is terminated in a purely resistive
load, the standing-wave ratio is found using the formula:

SWR =^ or SWR =-^-


R/. ^o

(Whichever is greater than 1)

Where R, = Load resistance

Z = Characteristic impedance

The standing-wave ratio is always equal to or greater than one. Regard-


less of whether R, is half as large or twice as large as Z , the ratio of a
voltage maximum to a voltage minimum is 2:1. Therefore, the degree of
mismatch is the same in both instances.

The higher the SWR is, the greater is the mismatch between line and load.
Also in practical transmission lines, power loss increases with SWR.
Therefore, a low value of SWR is always sought, except when the line is
used as a tuned or resonant circuit.

Transmission Line Resonant Circuits

A transmission line that is terminated in its characteristic impedance has


an SWR of 1 and is called a non-resonant or flat transmission line. It is
non-resonant because it presents a constant load impedance to the
generator regardless of the operating frequency. On the other hand, a
transmission line with an SWR greater than 1 presents a variable load
impedance that is dependent on the operating frequency. Therefore, a
line with an SWR greater than 1 is called a resonant transmission line.
6-16 UNIT SIX

zs^sz z z^gzzz SHORT

A 300 MHj

B 600 MH2

Figure 6-8
A resonant transmission line.

Figure 6-8 shows a 25 cm length of transmission line that is terminated


with a short circuit. Therefore, the S WR is infinite. Figure 6-8A shows the
standing-wave pattern that develops on the line when a 300 MHz signal is
applied. The 25 cm length of line is 1/4 k at 300 MHz because

3 x 10* meters/second (velocity of wave)


k (meters) =
frequency (Hz)

3 x 10 8 3 x 10 8
k= = 1 meter
300 MHz 3 x 10* Hz

Therefore, 1/4 \ is 0.25 m or 25 cm at 300 MHz. Notice that the input of the
line in Figure 6-8A has maximum voltage and minimum or zero current.
This corresponds to an open circuit and it is in fact the input impedance
of the 1/4 k shorted transmission line. The quarter-wave line actually
inverted the impedance of the load. If the load had been an open circuit,
the input impedance would have been a short circuit. Since these condi-
tions are only true when the line is exactly 1/4 k long, if the input
frequency changes, so does the input impedance. Because the transmis-
sion line is frequency dependent, it is said to be resonant. In fact, the 1/4 k
shorted line acts exactly like a parallel resonant circuit. That is, at its

resonant frequency or the frequency at which it is exactly 1/4 k long, its


input impedance is infinite or open. At all other frequencies, its input
impedance becomes either capacitive or inductive depending on
whether it is above or below its resonant frequency.

'
Antennas 6-17

Transmission lines are used as both parallel and series resonant circuits
in VHF, UHF, and microwave equipment. They not only offer an ex-
tremely high Q but often are the only practical method of obtaining a
resonant circuit at these high frequencies. This is because, as frequency
increases, the inductors and capacitors required become smaller and
smaller. When happens, circuit Q drops and bandwidth increases.
this
The only way to improve this is to use transmission line resonant circuits.

To be sure you understand this important concept, consider Figure 6-8B.


Here is the same 25 cm length of transmission line. However, since the
input frequency has been doubled to 600 MHz, the line appears as a
half-wavelength. This accounts for the different standing-wave pattern.
Note that the input now has maximum current and minimum or zero
voltage. Therefore, the input is a short circuit for a 1/2 \ shorted transmis-
sion line. Since this only occurs at the frequency at which the line is
exactly 1/2 k, it acts as a resonant circuit. In fact, the 1/2 k shorted line acts
like a series resonant circuit. Notice that the 1/2 k line duplicates the
terminating impedance at its input. Therefore, the input of a 1/2 k open
transmission line will also be an open.

To summarize, impedance. Therefore,


a 1/4 k line inverts its terminating
the input of a shorted 1/4 k line is an open and the input of an open 1/4 k
line is a short. Thus, a shorted 1/4 k can be used as a parallel resonant
circuit and an open 1/4 k line can be used as a series resonant circuit.
Conversely, a 1/2 k line duplicates its terminating impedance. Therefore,
the input of a shorted 1/2 k line is a short and the input of an open 1/2 k
line is an open. Thus, a shorted 1/2 k line can be used as a series resonant
circuit and an open 1/2 k line can be used as a parallel resonant circuit.

A correctly terminated transmission line has a constant input impedance


regardless of frequency or line length. For example, if a 300 H transmis-
sion line is terminated by a 300 H resistor, its input impedance will be
300 fl no matter what the frequency or the length of the line. It is for this
reason that a matched condition, indicated by an SWR of 1, is always
sought when installing a transmission line. The only time a high SWR is
desired is when the line is being used as a resonant circuit as shown

previously.
6-18 UNIT SIX

Figure 6-9
Reflectometer or SWR meter.

A\ A
Antennas 6-19

SWR Measurement

Standing-wave ratio is measured by a reflectometer or an SWR meter. A


typicalSWR meter is shown in Figure 6-9. This unit measures the ratio
between forward and reflected power and displays it on a calibrated
meter.

Figure 6-10 shows that the SWR meter is connected in series in the
transmission line, between the transmitter and the load or antenna. SWR
is measured by first tuning the transmitter for maximum output. The
SWR sensitivity control is then set for full-scale meter deflection. The
meter is then switched to measure SWR and the standing-wave ratio is
read directly.

The meter is actually indicating forward power when the SWR sensitivity
control is set for full scale. Then the meter is switched to measure
reflected power. The amount of reflected power is indicated; however,
the meter is calibrated to read the standing-wave ratio directly. This is a
quick and easy method of checking SWR, with a relatively low-cost
instrument.

SWR LOAD
TRANSMITTER OR
METER
ANTENNA

Figure 6-10
An SWR meter is connected in series in the line between
transmitter and load.
6-20 UNIT SIX

Figure 6-11
A directional wattmeter.

A meter that measures the actual amount of reflected and forward power
is shown in Figure 6-11. It is a directional wattmeter and, with the flick of
a switch, you can measure forward and reflected power. A perfectly
matched condition, SWR = 1, is indicated by zero reflected power. For
other values of reflected power, the standing wave ratio is determined
from the chart shown in Figure 6-12. You can use this chart to find the
SWR when forward and reflected power are known. For example, if the
forward power is 5 watts and reflected power is 0.2 watts, find these lines
on the chart; the point where they intersect indicates the standing-wave
ratio. In this case, they intersect on the diagonal line that represents an

SWR of 1.5.
i '

Antennas 6-21

STANDING WAVE RATIO CHART


0.5
J
r1^
1.0
^W^
\6
1. 5
i

|C
2.0 kr
2. 5
Sit
3.0 y i9

4.0 ^
/N 6
~ 5.0
oo

k^
oc 10
y
O 15 y X*H
Q_
Q
^Tx
w i
. _
OH 20
< 25 ^
^/

o 30
V r *

40 gp
50
y 7
S ff

/
100
&
150 !?

200

300

400
500
CVJ rri
O LT\
O
r^ i— C\J CO LT\ f»~ O
o CD
o O O o o •—

REFLECTED POWER (WATTS)

Figure 6-12
Standing wave ratio chart.
6-22 UNIT SIX

Coaxial Transmission Lines

Figure 6-13A shows a coaxial type of transmission line, which is one of

the most commonly used types an unbalanced


of lines. line
It is since it
consists of a shield that is at ground potential, and a center conductor.
The center conductor "carries" the RF energy and the shield prevents this
energy from being radiated into space. Thus, losses due to radiation are
kept to a minimum. With a balanced line, radiation is minimal due to the
balanced currents. The radiated fields created by each line are 180° out of
phase and, therefore, cancel.

A coaxial line has a specific characteristic impedance just as a balanced


line does. The formula for the characteristic impedance of a coaxial line
is:

Z = 138 x log-r

Where Z = ()
characteristic impedance in ohms.

D = inside diameter of shield.

d = outside diameter of the


center conductor.

These dimensions are shown in Figure 6-13B.

?SmS^^S0f*^
SHIELD

CENTER
CONDUCTOR
f
B -®%4
i D

V 138 LOG 4
d

Figure 6-13
Coaxial transmission line.
Antennas 6-23

Comparison of Coaxial and Balanced Lines

When transmission were discussed earlier, the lines


line characteristics
were assumed to be perfect. However, practical transmission lines
exhibit power losses. These losses are due to the resistance of the conduc-
tors, leakage in the dielectric material that separates the conductors, and

in some cases, radiation. Figure 6-14 shows the normal attenuation of


three typical transmission lines versus the operating frequency. RG-58/U
cable is a small diameter 52 ft coaxial cable, RG-8/U cable is a larger
diameter 52 ft coaxial cable, and 214-022 cable is a 300 ft balanced
transmission line. You can see that the balanced transmission line offers
the lowest attenuation; however, this advantage is offset to some degree
by its installation requirements.

NORMAL ATTENTUATION VS FREQUENCY

3
CO]
& ft

Jfr
$T \,
\J
10 ,
c<
$
oA^.
TV^£
$r *&
vV
v\*

$ £
^^ b*
X
2 ' v
10

01

11

n?
3 4 5 i 7 8 10 20 30 40 50 EO 10 100

FREQUENCY IN MEGAHERTZ

Figure 6-14
Attenuation in transmission lines.
6-24 UNIT SIX

When you install a balanced transmission you must prevent it from


line,
collecting moisture. Any moisture that accumulates on the line will
change the dielectric between the two lines, which changes its charac-
teristic impedance. The result is reflections and additional line losses.
You must also keep the line clear of other conductors such as down-
spouts, rain gutters, and metal towers. The line must be spaced at least
two to three times the conductor separation from any other conductors.
For example, if the distance between the two wires of the transmission
line is 1/2", the line must be spaced 1" to 1-1/2" away from other conduc-
tors. This requires standoff insulators.

Coaxial line, on the other hand, has an outside shield that is at ground
potential, so you can mount anywhere. You can tape it
it directly to any
metal object without adverse Although moisture can effect co-
effects.
axial cable if it enters the cable, it has little effect when proper care is
taken to prevent moisture buildup on the inside of the cable. Thus, while
balanced line offers the lowest attenuation, coaxial cable is widely used
because it's easy to install.
Antennas 6-25

Self-Review Questions

1. What is the primary purpose of a transmission line?

2. What is the difference between a balanced and an unbalanced


transmission line?

3. The characteristic impedance of a transmission line is determined


by its

4. What are the formulas for determining the standing-wave ratio of a


transmission line?

5. When a transmission line is terminated in its characteristic impe-


dance, the SWR is

6. What is the input impedance of a 1/4 \ short-circuited transmission


line at its resonant frequency?

7. 1/4 k and 1/2 \ transmission lines can be used as


circuits.

8. A reflectometer is used to measure

9. List theadvantages and disadvantages of balanced and unbalanced


or coaxial transmission lines.
6-26 UNIT SIX

Self-Review Answers

1. The primary purpose of a transmission line is to transfer energy


from a source to the load.

2. A balanced transmission line has two parallel conductors in which


the current in each conductor is 180° out of phase with the current
in the other conductor. An unbalanced line has a center conductor
that carries the current while the other conductor is a shield that is

at ground potential.

3. The characteristic impedance of a transmission line is determined


by its physical construction.

4. The three formulas for determining the standing-wave ratio of a


transmission line are:

F
F
L-'minimiu

^mn.rimui
SWR
'minimttn

SWR =|^or|^-

5. When a transmission line is terminated in its characteristic impe-


dance, the SWR is one.

6. The input impedance of a 1/4 \ short circuited transmission line is


infinite or an open circuit at its resonant frequency.

7. 1/4 A and 1/2 X transmission lines can be used as resonant circuits.

8. A reflectometer is used to measure standing-wave ratio.

9. The advantage of a balanced transmission line is low attenuation.


The disadvantage is the installation requirements of keeping it free
of moisture and away from other conductors. The advantage of
coaxial lines is that they are easier to install. The disadvantage is
higher attenuation than balanced lines.
Antennas 6-27

ANTENNAS
After a transmitter generates an RF must be some method of
signal, there
radiating this signal into space. There must also be some method for a
receiver to intercept or "pick up" the signal. The antenna fulfills both
requirements.

An antenna, generally made of metal, (often a wire or collection of wires)


converts high frequency current into electromagnetic waves for trans-
mission and does just the opposite for reception. Transmitting and re-
ceiving antennas have different functions, but they behave identically.
That is, their behavior is reciprocal.

Antenna Principles

Figure 6-15 shows a 1/4 A open-circuited transmission line. You can see
that the forward and reflected waves combine to form a standing-wave
pattern on the line. This was discussed previously, but it wasn't men-
tioned that not all the forward energy is reflected by the open circuit.
Actually, a small portion of the RF energy escapes from the system and is
radiated. This happens because the RF energy traveling toward the open
circuit is required to undergo a violent change, a phase reversal, when it
reaches the open. A small portion of the forward wave cannot do this, and
it actually "shoots" past the open circuit into free space. The amount of

waves escaping the transmission line is very small, for two reasons. First,
if the surrounding space is considered as a load for the transmission line,

a mismatch exists, and very little power is dissipated in this "load."


Second, since the two wires are close together and 180° out of phase, the
radiation from one will cancel that from the other.

Figure 6-15
Open wave transmission
circuited quarter line.
6-28 UNIT SIX

To develop this "open" into a better radiator, we must "enlarge" the open
circuit, or spread the two wires. When the wires are spread out, the
radiation is less likely to cancel. Also, the radiating transmission line is

better coupled to the surrounding space because more power is "dissi-


pated" or radiated. Moreover, because the wires are spread out, waves
traveling along the line find it increasingly difficult to reverse phase.
Thus, everything points to an increase in radiation.

Radiation can be increased even further by spreading the wires until they
are in line, as shown in Figure 6-16A. The electrostatic or electric field
and the electromagnetic field of the line are now fully coupled to the
surrounding space. Therefore, the maximum amount of radiation results.
This type of antenna is called a dipole. The original transmission line was
1/4 k long, but when it is spread out, the overall length is 1/2 A. Therefore,
the antenna shown in Figure 6-16A is a half-wave dipole.

The current and voltage standing-wave patterns on the half-wave dipole


are shown in Figure 6-16. Notice that both ends appear as opens and,
therefore, have a voltage maximum and current minimum. The center of
the antenna, which is the feed point, has a current maximum and voltage
minimum. As a result, the input impedance of the half-wave dipole is
low. While the voltage and current distribution indicates that the input
impedance should be zero ohms, in actuality, it is 73 ohms. This is due to
the energy lost to radiation. This radiated energy is not reflected back to
the input and, therefore, complete cancellation does not occur. For this
reason, input impedance rises to 73 ohms as the 1/4 X open line is spread
out into a dipole.

Figure 6-16
The half wave dipole.
Antennas 6-29

Antenna Radiation

Since a dipole has high voltage potentials at both ends, an electrostatic or

electric field exists between these points. This field is just like the one
that exists between the plates of a capacitor. However, in the case of an
antenna, the electric field is not confined to the area between the plates;
instead, it is radiated into space. The electric field around a half-wave
dipole is shown in Figure 6-17.

Note also from Figure 6-17 that a magnetic field exists around the an-
tenna. This is due to the current flow in the antenna. Since current is
maximum at the center of the antenna, so is the magnetic field. Both the

magnetic and electric fields are radiated as an electromagnetic wave.


This wave propagates or travels outward from the antenna and will
continue traveling even after the current and voltage are removed from
the antenna.

The farther the electromagnetic wave


travels from the antenna, the
weaker becomes. The amplitude of this wave or radiation field is
it

evaluated in terms of the voltage it will induce across a wire. This is


known as its field strength.

The field strength at any location depends on the distance from the
transmitter and on the radiated power. The strength varies inversely with
distance. For example, if the distance from the antenna is doubled, the
field strength will be reduced by one-half. Conversely, if the distance is
cut in half, the field strength will be doubled. Likewise, if the radiated
power is increased, the field strength at any location will also increase.
However, since field strength is indicated by the voltage induced in a
wire, any power increase must be converted to a corresponding voltage
increase. Therefore, since power is proportional to voltage squared
2
F
(P= _ ) t
it follows that the increase in field strength is proportional
R
to the square root of the power change. Thus, doubling the radiated
power will increase the field strength by V2~ or 1.414. Conversely, to
increase the field strength by 2, it is necessary to increase the radiated
power by 4. ELECTRIC FIELD
/

Figure 6-17
Magnetic and electric field around a half wave dipole.

GNETIC FIELD
6-30 UNIT SIX

Antenna Polarization

As discussed earlier, the electromagnetic wave radiated from any an-


tenna has two component fields: the electric field and the magnetic field.
These fields are perpendicular to each other and both are perpendicular
to the direction of propagation or travel. This is shown with vectors in
Figure 6-18. In this case, the electric field vector is horizontal. Therefore,
the wave is said to be horizontally polarized. If the pattern were rotated
90°, the electric vector would be vertical and the wave would be vertically
polarized. Thus, the electric field vector determines the polarization of
the wave. As you will see later, a horizontal antenna produces horizontal
polarization, while a vertical antenna produces vertical polarization. The
importance of this is that a horizontally polarized wave will induce
maximum voltage in a horizontal antenna. Theoretically, a horizontally
polarized wave will induce zero volts in a vertical antenna. However, in
practice, at HF and VHF, this seldom occurs because of wave changes that
take place during propagation.

MAGNETIC FIELD
DIRECTION

ELECTRIC FIELD
DIRECTION

DIRECTION
OF
PROPAGATION

Figure 6-18
Polarization of an electromagnetic wave.
Antennas 6-31

The Dipole Antenna

The most commonly used type of antenna is the half- wave dipole. As its

name implies, it is 1/2 X long at the operating frequency. The formula for
wavelength is:

_ 3 x 108
A —
f

Where X = wavelength in meters

f = frequency in hertz

This formula gives the wavelength in free space. However, with an


antenna, the wave developed on a wire and end effects must be consi-
is

dered. End effects make an antenna appear electrically as though it were


about 5% longer than its actual physical length. This is due to a capaci-
tance that exists between the ends of the antenna. The formula for finding
the length of a half-wave dipole that is supported by insulators at each
end is:

wo
1/2
v
X in feet
r
= ?468

This formula gives the length in feet and takes end effects into account.

Find the length of a half-wave dipole for 7 MHz.

1/2 X in feet = 468


fMHz

468
1/2 X in feet =

1/2 X in feet =66.9 feet

Thus, a half-wave dipole for 7 MHz is 66.9 feet long.


6-32 UNIT SIX

A typical dipole installation is shown in Figure 6-19. Notice that the


antenna is fed at the center with a 75 ft balanced transmission line. This
gives a good match to the antenna's 73 ft input impedance.

A dipole can also be "fed" with a coaxial transmission line. The center
conductor is connected to one side of the dipole and the shield to the
other side.

However, since the coaxial line is unbalanced and the dipole balanced,
having been developed from a 1/4 A balanced line, some inefficiency will
result. The current and voltage distribution on the antenna will be upset
and RF current will actually flow on the coaxial shield. This shield
current may result in radiation from the cable, which is undesirable.

Figure 6-19
Half wave dipole installation.
Antennas 6-33

To correctly couple a coaxial line to a dipole antenna, a balun is used. A


balun is a balanced to unbalanced RF transformer. A typical balun is

shown in Figure 6-20. This particular balun also acts as a center insulator
for the dipole. As shown, the wires connect to the dipole and the connec-
tor is for the coaxial cable. Baluns are constructed for specific impedance
ratios, just like conventional transformers. When 50 O or 75 O coaxial
cable is used to feed a dipole, an impedance ratio of 1:1 is used.

By far, the most important property of an antenna is its pattern of radia-


tion. In the case of a transmitting antenna, this is a graphical plot of the
field strength radiated by the antenna in different angular directions.
This same radiation pattern also indicates the receiving properties of the
antenna. This is because the receiving and transmitting properties of an
antenna are reciprocal.

Figure 6-20
A 1:1 balun. Photo courtesy Palomar Engineers.
6-34 UNIT SIX

Figure 6-21 A shows a 3-dimensional view of the radiation pattern for a


horizontal half-wave dipole. Notice that maximum radiation occurs
broadside to the antenna and minimum radiation occurs off the ends of
the antenna. A polar plot of the radiation in the horizontal plane is shown
in Figure 6-2 IB. The radiation pattern for a vertical half- wave dipole is

shown in Figure 6-22A. Note that, in this case, maximum radiation


occurs in all directions in the horizontal plane. Minimum radiation
occurs directly above and below the antenna. The polar plot for the
horizontal plane is shown in Figure 6-22B.

HORIZONTAL
AERIAL

POLAR PLOT
B

Figure 6-21
Horizontal half-wave dipole radiation pattern.
Antennas 6-35
J

MINIMUM
RADIATION
t

B POLAR
PLOT

Figure 6-22
Vertical half wave dipole radiation pattern.
6-36 UNIT SIX

A dipole antenna can also operate on harmonics of its fundamental


frequency. That is, a half-wave dipole can operate on its second har-
monic, where it functions as a full-wave dipole. The current distribution
for a full-wave dipole is shown in Figure 6-23A. Note that minimum
current occurs at the center of the antenna. This indicates that the center
of the antenna is a high impedance point. If a low input impedance is
desired, the antenna feedpoint must be moved to either of the current
maximums which are located 1/4 k from either end. The radiation pattern
for a full-wave dipole is shown in Figure 6-23B. Note that the pattern has
split up into four major lobes. This is due to the current distribution on
the dipole.

Figure 6-23
The full wave dipole.
Antennas 6-37

A half-wave dipole can also operate on its harmonic as shown in


third
Figure 6-24.The current distribution for the resulting 1-1/2 k antenna is
shown in Figure 6-24A. Note that, here, a current maximum occurs at the
center of the antenna. Therefore, the low impedance feedpoint remains at
the center of the antenna. The radiation pattern for the 1-1/2 \ antenna is

shown in Figure 6-24B. Note that there are still four major lobes, but now,
there is an additional minor lobe broadside to the antenna.

The harmonic operation of dipoles can be put to good use by designing an


antenna for use on two harmonically related frequencies. A good exam-
ple is an antenna designed for the 7 MHz and 21 MHz amateur radio
bands. It is MHz. Thus, when operated at 21
cut to a half- wavelength at 7
MHz, it wavelength
operates as a 1-1/2 antenna. Also, since both the 1/2 k
and 1-1/2 k antennas have a low impedance center feedpoint, due to a
current maximum at the center, they offer a good impedance match to a
75 fi or 50 ft transmission line.

Figure 6-24
1-1/2 wave dipole.
6-38 UNIT SIX

The Vertical Antenna

In cases where vertical polarization is required, the antenna must be


vertical. However, at low frequencies, the height of a vertical half- wave
dipole can become prohibitive. For example, at 3.5 MHz, a vertical
half- wave dipole would be 134 feet high and at 1.8 MHz, it would be 263
feet high.

This would seem to indicate that a low frequency vertical antenna would
be impractical. However, if a quarter-wavelength vertical antenna is
constructed above a perfect ground, it have the same characteristics
will
as the half- wave vertical dipole. This is possible because a perfect ground
will produce a "mirror image" of the quarter wave, thus, producing the
effect of a half- wave antenna. The image results from the reflected radio
waves as shown in Figure 6-2 5 A. The voltage and current distribution for
a quarter- wave vertical is shown in Figure 6-2 5B In order for this antenna
.

to operate correctly, the ground must be a perfect conductor. If it is not,


considerable power will be lost in the resistance of the grounding system.
When an antenna is constructed over soil of poor conductivity, such as
rocky or sandy soil, an artificial ground system must be used. This
consists of several 1/4 A copper wires extending outward from the base of
the antenna in as many directions as possible. These wires are referred to
as radials and usually four radials are the bare minimum. When only four
radials are used, they should be perpendicular to each other.

VOLTAGE
1/4
ANTENNA

GROUND B

Figure 6-25
The wave vertical.
quarter

k. ii
Antennas 6-39

The radiation patterns for a quarter-wave vertical antenna are shown in


Figure 6-26. In the horizontal plane, the antenna radiates equally well in
all directions. It is omnidirectional and gives good "coverage" in all

directions. In the vertical plane, the radiation is directed low toward the
horizon. This gives it excellent, long distance propagation characteris-
tics. Wave propagation is discussed in detail in the next section.

The input impedance of the quarter- wave antenna is around 36 O when it


is used with a perfect ground. This is a good match for a 50 H coaxial

transmission line. Also, since the vertical antenna is unbalanced because


one side is ground, unbalanced coaxial cable is a perfect feeder. The
center conductor is connected to the quarter- wave antenna, which is
insulated from ground, and the shield is connected to ground or the
radial system.

HORIZONTAL
RADIATION
PATTERN

Figure 6-26
Radiation patterns for the quarter wave vertical antenna.
6-40 UNIT SIX

Antenna Arrays

In many situations, the direction of the radiated wave must be restricted


within some specific limits. This is done to eliminate or reduce interfer-
ence to or from other stations. It also results in more efficient operation,
since any power radiated is directed towards the specific receiving sta-
tion. Therefore, very little (if any) power is wasted.

When the radiation pattern of an antenna system is essentially in one


direction, it is unidirectional. Contrast this with the half-wave horizontal
dipole which is bi-directional and the quarter wave vertical which is

omnidirectional. The measure of a unidirectional antenna's directional


properties or "directivity" is its beam width. This is measured from the

radiation pattern as the angle between the two points on either side of
maximum radiation where the field strength drops 3 dB. A typical uni-
directional radiation pattern is shown in Figure 6-27. In this case, the
beam width is 50°.

BEAM WIDTH
50°

Figure 6-27
Measuring bandwidth of a unidirectional antenna.
Antennas 6-41

An antenna system with good directivity or narrow beam width has the
advantage of providing directional gain. Since the radiated power is
concentrated within a relatively small beam, the field strength within
this beam is much greater than would be obtained from an omnidirec-
tional antenna. Therefore, the effective radiated power (ERP) of the
transmitter is increased by the directional gain of the antenna. Direc-
tional gain is the ratio of the power required to produce a given field
strength at a given location using a reference antenna, compared to the
power required to produce the same field strength with a directional
antenna. Reference antennas used are usually half-wave dipoles or
quarter- wave vertical antennas. Very high gains are possible, particularly
at VHF and UHF where electrically large antenna systems can be built in a
relatively small space.

Effective radiated power or ERP can be calculated if transmitter output


power, transmission line loss, and antenna gain are known. For example,
if the transmitter output power is 100 W, transmission line loss is 10 W,

and antenna power gain is 10; ERP is:

100 w Transmitter output


- 10 w Line loss
90 W Input to antenna
x 10 Antenna power gain
900 W Effective radiated power

This shows how a directional antenna greatly increases the effective


radiated power.

Directivity or directional gain is obtained by using combinations of two


or more antenna elements to form an antenna array. Depending on the
method used to excite the additional elements, antenna systems are
classified as parasitic or driven arrays.
6-42 UNIT SIX

PARASITIC ARRAYS

An antenna element that is not connected to the transmission line will


develop a voltage through induction. It is called a parasitic element.
Figure 6-28 shows a parasitic element located 1/4 A from a dipole, which
is called the driven element. Both elements are 1/2 \ long and, therefore,

are resonant at the operating frequency. The radiation pattern of the


dipole, by itself, is bidirectional and is maximum broadside to the an-
tenna. If you consider the energy traveling toward the parasitic element,
it travels 1/4 \. Therefore, before it reaches the parasitic element, it has
gone through a 90° phase change. As the wave cuts the parasitic element,
a voltage is induced that is 180° reversed with respect to the wave that
induced it. As a result, current flows through the element and the element
radiates. In the direction beyond the parasitic element, the field radiated
by it is opposite to the field from the driven element. Therefore, the two
fields cancel, and radiation in However, the
this direction is negligible.
parasitic element also radiates toward the driven element. By the time
this field reaches the dipole, it has gone through a further 90° phase
change. Therefore, the total phase change is 360°, and the wave arrives in
phase with the energy from the driven element. Thus, radiation toward
the driven element is reinforced and maximum radiation results. The
direction of maximum radiation is called the forward direction, and the
parasitic element is called the reflector.
Antennas 6-43

The radiation pattern obtained with parasitic arrays depends on the


magnitude and phase of the current in the parasitic elements. These
factors depend on the length of the parasitic elements and on the spacing
between the driven and parasitic elements. Optimum reflector operation
is obtained with a spacing of 0.18 to 0.2 A and with an element length

approximately 5% longer than 1/2 A. Also, if the parasitic element is


approximately 5% shorter than 1/2 A and the spacing is reduced to about
0.1 A radiation in the direction of the parasitic element will be reinforced.
In this case, the element is called a director.

REFLECTOR

Figure 6-28
A half wave dipole driven element with a parasitic reflector.
6-44 UNIT SIX

A 3-element parasitic array is shown in Figure 6-29A. This is known as a


Yagi antenna, and it is named
one of its inventors. Figure 6-29B
after
shows the radiation patterns of the driven element alone, with the reflec-
tor added, and with the director added. You can see that the directivity
improves with each additional element. Still sharper directivity and
therefore, increased gain can be achieved by adding more directors. An
example is shown in Figure 6-30. This is a 6-element Yagi designed for
440 MHz. Notice that it is oriented for vertical polarization.

3 ELEMENT YAGI

MAXIMUM
RADIATION
A ' DRIVEN ELEMENT
J WITH REFLECTOR
DIRECTOR

DRIVEN ELEMENT
DRIVEN ELEMENT
ALONE

REFLECTOR
. '

Figure 6-29
A three-element Yagi and its radiation pattern.
Antennas 6-45

Figure 6-30
A six element Yagi for 440 MHz. Photo courtesy Cushcraft
Corp.
6-46 UNIT SIX

DRIVEN ARRAYS

When all the elements of an antenna system are fed by the transmission
line, it is called a driven array. An example of a driven array is shown in
Figure 6-31 A. This is called a collinear array since all the elements are
placed end-to-end or in line. Note that the transmission line is attached to
each element. Also, the current in each element is in phase, since the
transmission line length to each element is The result is a radiation
equal.
pattern such as that shown in Figure 6-3 IB. The dotted line shows the
pattern obtained from a single dipole. The dashed line shows the in-
creased directivity obtained with two elements. Finally, the solid line
shows the pattern with four elements. Normally these collinear arrays are
oriented vertically as pictured in Figure 6-32. This gives omnidirectional
coverage but it directs the energy down towards the horizon and thus
gives extended coverage at VHF and UHF.

1/2 X DIPOLE ELEMENTS

I NPUT

Figure 6-31
A collinear driven array.
Antennas 6-47

y pr

Figure 6-32
A four element collinear array. Photo courtesy Cushcraft
Corp.
6-48 UNIT SIX

Another form of driven array is shown in Figure 6-33. This is a log-


periodic array since the element lengths are related logarithmically.
Also, each element is fed through a special phasing network. This an-
tenna offers good directivity and gain. However, its greatest advantage is
its very wide bandwidth. While most Yagi arrays operate over a very

narrow band of frequencies, the log-periodic array can operate over a


much wider range of frequencies. For example, the antenna shown in
Figure 6-33 gives high gain and an SWR less than 2:1 from 7.5 MHz to 30
MHz. The best that most Yagi arrays can do is to cover a 1 or 2 MHz band
in this same frequency range.

Figure 6-33
A log-periodic array. Photo courtesy Granger Associates.

i ^\k
Antennas 6-49

Self-Review Questions

10. A device that converts high frequency current into electromagnetic


waves is a/an

11. The antenna shown in Figure 6-34 is a


The solid line represents the dis-

tribution and the dashed line represents the


distribution.

Figure 6-34

12. What is the input impedance of a half-wave dipole?

13. The radiation from an antenna consists of a/an


fieldand a/an field. Both of these fields are
radiated as an wave.

14. The amount of voltage induced in a wire by an electromagnetic


wave is determined by the wave's

15. Antenna polarization is determined by the direction of the


field vector.

16. An antenna supported by insulators appears approximately 5%


longer, electronically, than its actual physical length. This is due to
6-50 UNIT SIX

17. Find the length of a half- wave dipole for 4 MHz.

18. With a half-wave dipole, maximum radiation occurs


to the antenna and minimum radiation
occurs

19. The operation of a vertical quarter-wave antenna is based on the


principle and requires a good ground
system.

20. A vertical quarter- wave antenna has an


radiation pattern in the horizontal plane.

21. Ifantenna power gain is 5, transmission line loss is 3 watts, and


transmitter output is 29 watts, what is the effective radiated power
(ERP)?

22. The two types of antenna arrays are the array


and the array.

23. A 3-element Yagi is an example of a array. Its


three elements are the , the
., and the To increase direc-
tivity and forward gain several can be added to
the array.

24. The collinear and log-periodic antennas are examples of


arrays.

25. The primary advantage of the log-periodic over a Yagi antenna is its
Antennas 6-51

Self-Review Answers

10. A device that converts high frequency current into electromagnetic


waves is an antenna.

11. The antenna shown in Figure 6-34 is a half- wave dipole. The solid
line represents the current distribution and the dashed line repre-
sents the voltage distribution.

12. The input impedance of a half-wave dipole is 73 O

13. The radiation from an antenna consists of an electric field and a


magnetic field. Both of these fields are radiated as an elec-
tromagnetic wave.

14. The amount of voltage induced in a wire by an electromagnetic


wave is determined by the wave's field strength.

15. Antenna polarization is determined by the direction of the electric


field vector.

16. An antenna supported by insulators appears approximately 5%


longer electrically, than its actual physical length. This is due to
end effects.
6-52 UNIT SIX

r 468
17. 1 2 A in feet =-

468
1/2 X in feet =
4 MHz

1/2 A in feet =117 feet

18. With a maximum radiation occurs broadside


half-wave dipole. to
the antenna and minimum radiation occurs off the ends.

19. The operation of a vertical quarter- wave antenna is based on the


mirror image principle and requires a good ground system.

20. A vertical quarter-wave antenna has an omnidirectional radiation


pattern in the horizontal plane.

21. 29 W Transmitter output


—3 VV Line loss
26 W Input to antenna
x5 Antenna power gain
130 \V Effective radiated power

22. The two types of antenna arrays are the parasitic array and the
driven array.

23. A 3-element Yagi an example of a parasitic array. Its three


is

elements are the reflector, the driven element, and the director. To
increase directivity and forward gain, several directors can be
added to the array.

24. The collinear and log-periodic antennas are examples of driven


arrays.

25. The primary advantage of the log-periodic over a Yagi antenna is its
wide bandwidth.
Antennas 6-53

RADIO PROPAGATION
The energy radiated from a transmitting antenna travels into space in
many directions. As the distance from the antenna increases, the energy
field spreads out and the field strength decreases. This phenomena was
discussed earlier. However, the path, or paths, by which the signal
reaches the receiving location also affects the field strength. There are
three broad classifications of the signal path. These are: the ground wave,
the space wave, and the sky wave.

Ground Waves

The ground wave is a radio wave that travels along the surface of the
earth. In the low frequency (LF) and medium frequency (MF) bands, this
is the predominant mode of propagation. These longer wavelengths tend

to follow the curvature of the earth and actually travel beyond the hori-
zon. This is shown in Figure 6-35. However, as the frequency increases,
the ground wave is more effectively absorbed by the irregularities on the
earth's surface. This is because, as frequency increases, hills, mountains,
etc., become significant relative to the transmitted wavelength. For
example, at 30 kHz the wavelength is 10,000 meters or 6.2 miles. There-
fore, even mountains are relatively insignificant compared to the
wavelength. Thus, very little ground wave attenuation is experienced at
this frequency. On the other hand, at 3 MHz the wavelength is 100 meters.
This is short enough that hills, trees, and large buildings break up and
absorb the ground wave.

Figure 6-35
The ground wave.
6-54 UNIT SIX

One way of greatly improving ground wave coverage is to use vertical


polarization. In fact, with horizontal polarization, the electric field is
parallel to the earth's surface, and any ground wave is effectively short-
circuited by the conductivity of the earth. However, since ground wave
propagation is limited to the low and medium frequency bands, con-
structing full-size quarter- wave antennas does present problems. For this
reason, and others, there are very few communications services in the low
frequency band. Most ground wave communications are in the medium
frequency bands where the antenna sizes are more practical.

Space Waves

When the transmitted signal is increased above 4 or 5 MHz, the usable


ground wave signal exists for only a few miles. At these frequencies and
above, particularly at VHF and UHF, it is possible to transmit further
using the space wave. When the transmitting and receiving antennas are
within sight of each other, the signal is considered a space wave or direct
wave. This is shown in Figure 6-36. Of course, space wave propagation is
limited to line-of-sight distances. Therefore, the range can be increased
by increasing the height of both transmitting and receiving antennas.
Another form of space wave propagation is transmission to and from
satellites and aircraft. Here, the height of the antenna is increased sub-
stantially and, thus, the propagation range is greatly increased. Space
wave propagation is used primarily in the VHF, UHF, and higher fre-
quency bands.

TRANSMITTER RECEIVER
SPACE WAVE

Figure 6-36
Space wave or direct wave propagation.
Antennas 6-55

Sky Waves

From approximately 30 to 250 miles above the earth's surface, ultraviolet


radiation from the sun causes the air particles to ionize into free electrons,
positive ions, and negative ions. This region is called the ionosphere.
Any electromagnetic waves that enter this region will be bent or refracted
much the same waves are bent when traveling through different
as light
mediums, such as aprism or lens. The amount of refraction depends on
several factors. Among these factors are the frequency of the wave, the
density of the ionized region, and the angle at which the wave enters the
ionosphere. If all of these factors are just right, the wave will be bent or
refracted enough to return to the earth as shown in Figure 6-37. This
space wave that has been refracted back to earth by the ionosphere is
known as a sky wave. Almost all propagation on the high frequency (HF)
bands is by sky wave. Also, nighttime long distance medium frequency
propagation is by sky wave. Since the ionosphere is responsible for the
behavior of the sky wave, it will be discussed in detail.

REFRACTION OF WAVE
DUE TO IONOSPHERE

Figure 6-37
The sky wave.
I

6-56 UNIT SIX

The Ionosphere

Sky waves that return to earth from the ionosphere come from different
heights above the earth depending upon frequency and on the time of
refraction. This phenomenon shows that several layers of ionization
exist. This is because the different gases which make up the earth's

atmosphere ionize at different pressures and are also affected differently


by ultraviolet radiation and cosmic ray bombardment from the sun. The
number of layers present, their heights above the earth and the amount
they bend or refract the sky wave all vary, depending upon several
interrelated factors. These factors vary from hour to hour, day to day,
month to month, season to season, year to year, and even decade to
decade. The major influence on the ionosphere is radiation from the sun.
For example, solar radiation increases the density of the ionospheric
layers and lowers their effective height. This accounts for the hourly and
seasonal variations of the ionosphere. Other solar and magnetic distur-
bances also produce changes in these layers.

Figure 6-38 shows the various ionospheric layers. The lowest layer,
called the D layer, exists only in the daytime. This layer is the furthest
from the sun and, therefore, its ionization is relatively weak. For this
reason, the D layer does not affect the direction of travel of radio waves.
However, the ionized particles do absorb appreciable energy from the
electromagnetic wave. Thus, when the D layer is present, it attenuates the
sky wave on its way up to the other layers and on its way down.

300

F
2
SUMMER
200

100

MILES

~~ 1 1 1 1 1 1 1 1 1 1~
0200 0400 0600 0800 1000 1200 1600 1800 2000 2200 2400
HOURS LOCAL TIME

Figure 6-38
The layers of the ionosphere.
Antennas 6-57

The D layer also completely absorbs medium frequency signals. There-


fore, it limits MF signals to ground wave propagation during the day.
When the D layer disappears at night, MF signals can travel much further
via sky wave propagation.

The next layer is the E layer which extends from about 55 to 85 miles
above the earth's surface. It has a maximum density at noon, but is only
weakly ionized at night, if at all. The last layer is the F layer, which is
extremely variable. At night, it is a single layer that is approximately 110
to 250 miles above the earth. However, in the daytime, during maximum
radiation from the sun, it splits into two layers called the F! and F2 layers.
The F! layer ranges from 85 to 1 55 miles. The F 2 layer, which is the closest
to the sun, is even more variable and ranges in height from 90 to 185 miles
on a winter day, to as high as 155 to 220 miles on a summer day.

Sky Wave Propagation

This section and defines several terms that are used to describe
lists

characteristics of the ionosphere and sky wave propagation. The first of


these terms is virtual height. Figure 6-39 shows that as the wave is
refracted, it is bent down gradually rather than sharply. However, below
the ionized layer, the waves follow paths that are exactly the same as if
they were "reflected" from a surface located at a greater height, called the
virtual height of the layer.

Figure 6-39
The virtual height of an ionized layer.
6-58 UNIT SIX

The virtual height of the ionospheric layers can


be calculated from the
time it takes a pulse of RF energy, directed vertically upward, to be
returned to earth. However, if the signal frequency is raised above a
certain value, it will not be refracted enough to return to earth. This
critical frequency is defined as the highest frequency that is returned to
earth when transmitted in a vertical direction. The critical frequency
value depends on the condition of the ionosphere and, therefore, changes
from hour to hour, day to day, etc.

If the angle of radiation is lowered from vertical, the wave will travel
longer through the ionized layer and will, therefore, be refracted to a
greater degree. This means that signals above the critical frequency can
be returned to earth. However, there is a limitation. For a given frequency
there is a critical angle beyond which the signal will not be refracted
enough to return to earth. This is shown in Figure 6-40. Note also that as
the radiation angle decreases, the distance the wave travels over the earth
increases. This distance is known as the skip distance. The skip distance
can be maximized by using the lowest radiation angle possible and by
using the highest frequency that will be refracted at that angle.

The highest frequency be used for transmission between two


that can
locations is called the maximum usable frequency (MUF). Actually, a
lower frequency could be used, as it would also be refracted. However, as
the signal frequency is lowered, the energy absorbed in the ionosphere
increases rapidly, and the signal level is drastically reduced. In fact, there
is a lowest usable frequency (LUF) , below which the RF signal is absorbed
entirely by the ionosphere. Therefore, maximum received signal level is

obtained when operating at the maximum usable frequency.

Unfortunately, the maximum usable frequency is constantly changing


due to the effects of solar radiation. Therefore, for the most reliable sky
wave communications, the optimum usable frequency is used. This
frequency is far enough below the MUF that it isn't affected by minute-
to-minute solar fluctuations and, thus, provides reliable sky wave com-
munications. Since MUF, LUF, and optimum frequencies change hourly,
daily, and monthly, it may seem that sky wave communications is a
hit-and-miss proposition. However, there are charts available that predict
the MUF, LUF, and optimum frequencies for every hour of the day, over
any path on earth, during a given month. These predictions are based on
elaborate solar observations and can be used to optimize sky wave com-
munications.
asj6-59
Antennas

Figure 6-40
Effects of lowering the radiation angle.
6-60 UNIT SIX

MULTIPLE-HOP TRANSMISSION

Figure 6-4 1 shows that if the signal that has returned to earth at location A
has sufficient strength, can be reflected by the ground back up to the
it

ionosphere. It is then refracted again and it will return to earth at location


B. Here again, if it has sufficient strength, it will be reflected again and
continue to a more distant location. This is known as multiple-hop
transmission and it is by no means limited to two or three hops.

Under optimum conditions, the maximum distance that a single-hop


transmission can cover is about 2000 miles. The limiting factor is the
radiation angle which, of course, cannot be reduced below the horizon.
Therefore, for long distance coverage multiple-hop propagation is neces-
sary.However, remember that each hop increases the attenuation of the
signal.Thus, attenuation from the ionosphere and the ground reflection
points is the primary factor in determining just how far a sky wave will
travel.

IONIZED LAYER

m@£$M

Figure 6-41
Double-hop transmission.
Antennas 6-61

FADING

When sky wave propagation is used, signal strength will increase and
decrease periodically. Sometimes, the changes in signal strength are
small and the receiver's automatic gain control (AGC) will compensate
for it. Other times, the signal may be lost completely. This is known as
fading and is caused by multiple-path reception and changing iono-

spheric conditions. For example, consider the condition shown in Figure


6-42. Here, the signal is received via both one and two hop paths. In this
case, the lengths of the paths and the phasing of the signals will vary with
ionospheric conditions. Because the signal received at any instant is the

phasor sum of all the waves received, alternate cancellation and rein-
forcement will occur if the path lengths vary over 1/2 wavelength. Thus,
changing ionospheric conditions which affect path lengths will cause
fading.

It is also possible to receive the same signal via both sky wave and ground
wave propagation. If they do not arrive in the correct phase, fading will
result. This occurs primarily in the medium frequency band.

'^1$^^

Figure 6-42
Multiple path reception.
6-62 UNIT SIX

Ionospheric Variations

Since the existence of the ionosphere depends on solar radiation, it

follows that any variations in this radiation will influence the iono-
sphere. The earth's rotation and its revolution around the sun also affect
the amount of solar radiation reaching the earth and therefore, they too
influence the ionosphere.

The regular variations of the sun, earth, and hence the ionosphere which
are more or less predictable are divided into the following categories:
diurnal, seasonal, geographical, and cyclical. Diurnal variations are the
hour-to-hour changes in the various layers caused by the rotation of the
earth around its axis. and F layers follow a
Diurnal variations in the D, E, x

regular pattern and are dependent on the sun's elevation above the
horizon. Ionization in these layers increases from a very low level at
sunrise, to a maximum at noon, and then decrease toward sunset. For all
practical purposes, these layers disappear at night. Typical hour-to-hour
changes in the reflection capability of the various layers, as indicated by
the critical frequency, are shown in Figure 6-43. Notice that ionization in
the F 2 layer rises steeply at sunrise. It decreases after sunset, but does not
disappear during the night.

SUNRISE SUNSET

CRITICAL "*Xj 2
:

FREQUENCY
ImHz)

^""^Fi

,^E

0000 0400 0800 1200 1600 2000 2400 LOCAL TIME

Figure 6-43
Diurnal variations of the ionosphere.
Antennas 6-63

Since the position of any point on earth relative to the sun is constantly
changing as the earth moves in its year-long orbit around the sun, iono-
spheric properties change also. These are called seasonal variations, and
a typical variation in the F 2 layer is shown in Figure 6-44. Note that since
the earth is closer to the sun in winter, it receives a higher level of
radiation, and the critical frequency reaches its highest level.

CRITICAL
FREQUENCY 6
ImHzl

SUMMER
- ^ / \
WINTER

0000 0400 0800 1200 1600 2000 2400 LOCAL TIME

Figure 6-44
Seasonal variations of the F2 layer.

The intensity of ionizing radiation that strikes the ionosphere also varies
with latitude. For example, solar radiation is considerably greater near
where the sun is more directly overhead then in the northern
the equator, ,

and southern latitudes. These variations in the ionosphere are called


geographical variations.

The 1 1-year sunspot cycle probably influences the ionosphere most. This
variationdepends upon sunspot activity, which is constantly changing
throughout an approximate 11-year cycle.
6-64 UNIT SIX

NUMBER 200-.CYCLE NUMBER 12 3 4 5 6 7 8 9 10 1 1 12 13 14 15 16 1 7 18 19 20

Scots' ;
j^yvAA/yWy^
1700 1750 '1800 '
' '
isV
1850 ' '
\oTn^ '
^ ', '

[950
' '

YEAR

Figure 6-45
Sunspots greatly affect the ionosphere.
(A)shows the surface of the sun with a number of sunspots.
(B) shows the cyclical variation
in the sunspot count.
Antennas 6-65

A photograph of the surface of the sun that shows a number of sunspots is


shown in Figure 6-45A. A direct relationship exists between these
sunspots and the intensity of ultraviolet energy radiated by the sun: the
number of sunspots, the greater the intensity. Therefore, since
greater the
ultraviolet radiation is responsible for ionizing the various layers, it

follows that the sunspot count directly affects the condition of the iono-
sphere. This effect is dramatically illustrated in Figure 6-46. It shows a
comparison between critical frequencies for the F2 layer during a sunspot
maximum and a sunspot minimum. Notice that the critical frequency
during a sunspot maximum is over twice that of a sunspot minimum. The
result is that communications range and reliability is greatly increased
during years of high sunspot activity.

IB

16

14

12
CRITICAL
FREQUENCY
(MHz)
10

MAXIMUM
6 1200 SUNSPOTS)

4 MINIMUM
(11 SUNSPOTSI
?

10 12 14 16 18 20 22

LOCAL TIME

Figure 6-46
Comparison of the critical frequency of the F2 layer versus
the number ot sunspots.
6-66 UNIT SIX

The ionosphere is also subject to abnormal variations which can cause


unpredictable changes in wave propagation. Generally, ionospheric dis-
turbances on the HF bands weaken signal levels, either suddenly or
gradually, sometimes to the point where signals disappear completely.
On the other hand, in the VHF bands, these ionospheric disturbances can
produce very long distance communications "openings". One of these
disturbances is known as "sporadic E-layer ionizations". These are errat-
ic areas of high density ionization that occur at E-layer heights. This
usually results in ionospheric refraction of signals high up into the VHF
region, and it allows extremely distant contacts on these normally line-
of-sight frequencies.

The other two main types of ionospheric disturbances are the "iono-
spheric storm" and the sudden ionospheric disturbance. In both cases,
they are believed to have their origin in solar flares which occur on the
surface of the sun. An ionospheric storm may develop either gradually or
suddenly, and may continue from one or two days to almost a week. The
sudden ionospheric disturbance, on the other hand, usually commences
suddenly and lasts from about twenty minutes to an hour or two. At the
height of these disturbances, the absorption is high enough to cause a
radio blackout, particularly at the higher frequencies.
Antennas 6-67

VHF-UHF Propagation

The maximum usable frequency seldom rises above 30 MHz and only at

the sunspot maximum does it reach as high as 50 — 60 MHz. For this


reason, most communication in the VHF and UHF bands takes place via
the space wave and is, therefore, strictly limited to line-of-sight paths.
Thus, the limiting factor for VHF-UHF communications is the local
terrain and the curvature of the earth.

Actually, the radio horizon for space waves about 1-1/3 as far as the
is

optical horizon. This is due to refraction in the earth's lower atmosphere.


Refraction occurs because of the linear decrease in density of the earth's
atmosphere as height increases. What happens, in effect, is that the top of
the wave travels slightly faster than the bottom. As a result, the wave is
bent slightly downward to follow the earth's curvature beyond the opti-
cal horizon.

The radio horizon of an antenna is given by the formula:

D, =4VHT

where D, = radio horizon distance in km.

H, = transmitting antenna height in m.

You can use same formula for a receiving antenna. Therefore, the
the
maximum space wave communications distance is the sum of these
figures or

Dm = D, + D r = 4VHT+ 4VH7
6-68 UNIT SIX

As an example, consider antenna height of 100 m and a


a transmitting
receiving antenna height of 49 m. What is the maximum space wave
communications distance?

D w =4VhT+4Vh7

= 4Vl00m + 4V49m

= 4(10) + 4(7)

= 40 + 28

= 68 km

In this case, the maximum communications distance is 68 km. Of course,


thisdoes not account for any path obstructions, such as mountains, hills,
buildings, etc. All these factors must be considered when setting up a
communications link.

Extending VHF-UHF Communications


There are many ways of extending the range of distance of VHF-UHF and
higher frequency communications. However, only two techniques have
proven reliable enough to merit widespread use. These are tropospheric
scatter communications and satellite communications.

TROPOSPHERIC SCATTER PROPAGATION

Figure 6-47 illustrates a typical troposcatter communications link. Here,


a very high power UHF transmitter uses a high gain directional antenna
to transmit energy at the horizon. This energy is then "scattered," with
some minute portion being scattered in the direction of the receiver. The
reasons for the scattering are not entirely understood. However, one
theory suggests that the signal is reflected from irregularities in the
atmosphere, another suggests reflection from atmospheric layers. At any
rate, the phenomena does exist and is a quite reliable method of extend-
ing UHF communications beyond the horizon.
Antennas 6-69
J

SCATTER xj

AREA ' s\

Figure 6-47
Tropospheric scatter propagation.

The best frequencies for this mode are 900 MHz, 2 GHz, and 5 GHz. The
typical troposcatter path length is 300 1000 km. Compared to the
to
maximum practical space wave distance of 100 km, troposcatter appears
quite attractive. The only problem is that a high power transmitter, a high
gain receiver, and elaborate antenna arrays must be used. Therefore, the
cost of a troposcatter system is relatively high compared to using several
space wave repeater stations or a coaxial cable to cover the same distance.
Even so, troposcatter systems are used when the communications path
must cross rough or inaccessible terrain. As an example, they are used
almost exclusively to provide communications to offshore oil drilling
platforms in the North Sea and elsewhere.
6-70 UNIT SIX

SATELLITE COMMUNICATIONS

A communications satellite is basically a microwave repeater station. It


receives an "uplink" signal from earth, amplifies it, and retransmits it on

a separate "downlink" frequency. By virtue of its position above the


earth, it allows any station in its coverage area enormously increased
communications range. Figure 6-48 shows a typical satellite communica-
tions system. Most systems presently use a 6 GHz uplink frequency and a
4 GHz downlink frequency. In addition, there are several separate com-
munications channels in each satellite. In fact, presently operational
satellites can provide up to 11,000 telephone circuits each!

COMMUNICATIONS

Figure 6-48
A satellite communications system.
Antennas 6-71

The vast majority of communications satellites now in use are "parked"


in a synchronous orbit. That is, they appear to hover over a certain spot
above the earth. To do this, they must be precisely 35,800 km above the
earth's equator. Once this is accomplished they will complete one orbit
every 24 hours, just as the earth completes one revolution. Therefore,
they will have the same angular velocity as the earth and will remain
above the same spot on earth. Thus, the satellite is geostationary, or
stationary with respect to the earth.

Once a satellite is parked in its synchronous orbit, it is a relatively simple


matter to aim the ground station antennas. This is because the satellite is
always at a precise point above the horizon. This eliminates costly satel-
lite tracking equipment that is required with satellites in other types of
orbits.
6-72 UNIT SIX

Figure 6-49 shows the coverage pattern of a geostationary satellite using


an omnidirectional antenna. The satellite is parked at approximately 15°
west longitude directly above the equator. From this map, you can see
that it would be a relatively simple matter to cover the entire earth,
excluding the polar regions, using just three equally spaced satellites. In
practice,many more are used. Some are used simply to provide more
channels or communication capacity. Others provide more specialized
applications or coverage. For example, the ANIK series of satellites is
used to provide communications within Canada. As a result, they can use
directional antennas on the satellite to provide better coverage of Canada
and, in particular, its far northern regions.

EAST^

Figure 6-49
Coverage of a geostationary satellite parked at 15° W.
Antennas 6-73

An example of a domestic satellite is shown in Figure 6-50. This satellite


uses a highly directional antenna to provide coverage of the continental
United States. The satellite consists of the directional antenna (which is

used for both transmit and receive) and a receiver/transmitter which is

called a transponder. The receiver does not demodulate the signal, in-
stead, it amplifies the 6 GHz uplink signal and heterodynes it down to 4
GHz. The transmitter then amplifies this 4 GHz downlink signal and
applies it to the antenna. The approximately 130
overall amplification is

dB and the total power consumption about 150 W. The power is provided
by batteries which are recharged by solar cells.

Since the power consumption must be kept low, the transmit-


satellite's

ter's output power also kept low. This means that the earth stations
is

must use highly directional antennas, high transmitter power, and very
high gain receivers. However, this is more than offset by the increased
communications range and reliability.

Figure 6-50
A domestic satellite and its coverage area.
6-74 UNIT SIX

Self-Review Questions

26. A radio wave that travels along the surface of the earth is called a
Propagation by this means is predominant
in the and bands.

27. When the transmitting and receiving antennas are in line-of-sight


of each other, the mode of propagation is
This type of propagation is used primarily in the ,

and the
, frequency bands.

28. Radio waves that are refracted by the ionosphere and returned to
earth are called This type of propa-
gation occurs primarily in the band.

29. Name the ionospheric layers that are present during the daytime.

What layers are present at night?

30 What is the rritiral frequency?

31. What is the maximum usable frequency?

32. What is the optimum usable frequency?

33. What is the primary pause nf fading?


J 6-75
Antennas

34. List the four variations of the ionosphere and their causes.

1.

2.

4.

35. What are the two main types of ionospheric disturbances?

36. A UHF communications path of 64 km must be established over


relatively smooth terrain. The transmitting antenna height is 36 m.
What is the minimum acceptable height of the receiving antenna
for reliable space wave communications?

37. List the two common methods used to extend VHF, UHF, and
higher frequency communications range.
6-76 UNIT SIX

Self-Review Answers

26. A radio wave that travels along the surface of the earth is called a
ground wave. Propagation by this means is predominant in the LF
and MF bands.

27. When the transmitting and receiving antennas are in line-of-sight


of each other, the mode of propagation is space wave or direct
wave. This type of propagation is used primarily in the VHF, UHF,
and the higher frequency bands.

28. Radio waves that are refracted by the ionosphere and returned to
earth are called sky waves. This type of propagation occurs primar-
ily in the HF band.

29. The ionospheric layers present in the daytime are the D, E, F, , and
F2 layers. The layer present at night is the F layer.

30. The critical frequency is the highest frequency that is returned to


earth when transmitted in a vertical direction.

31. The maximum usable frequency (MUF) is the highest frequency


that can be used for transmission between two locations.

32. The optimum usable frequency is slightly below the MUF. It pro-
vides reliable communications and is not affected by minute-to-
minute solar fluctuations.

33. Fading is caused by multiple path reception and changing iono-


spheric conditions.

34. The four variations of the ionosphere are —


1. Diurnal or hour-to-hour variations — caused by the earth's
rotation around its axis.

2. Seasonal variations — caused by the earth's changing position


relative to the sun.

3. Geographical variations — caused by different levels of solar


radiation reaching different latitudes. For example, maximum
radiation reaches the equatorial regions, with less reaching the
northern or southern latitudes.

4. Cyclical variations — caused by the 11-year sunspot cycle.


Antennas 6-77

35. The two main types of ionospheric disturbances are the iono-
spheric storm and sudden ionospheric disturbance.

36. The maximum space wave distance equation must first be rear-

ranged.

D m =4VHT+4VHT

D wi -4VhT=4 VHT
Dm -4VHr
=VH,

f D w - 4VH7 2
j = H r

Therefore

I 10 J*
= 100 m
37. The two common methods of extending VHF, UHF, and higher
frequency communications range are tropospheric scatter propaga-
tion, and satellite communications.
6-78 UNIT SIX
Unit 7

COMMUNICATIONS SYSTEMS
7-2 UNIT SEVEN
Communications Systems 7-3

INTRODUCTION
This unit discusses three basic communications systems. These are the
television broadcasting system, the FM stereo broadcasting system, and
several methods of data communication.

The "Unit Objectives" on the next page state, in a concise manner, the
goals of this unit. Review this list now and be sure you can satisfactorily
complete all the objectives before taking the unit exam.
7-4 UNIT SEVEN

.
Communications Systems 7-5

UNIT OBJECTIVES
When you have completed this unit, you should be able to:

1. Identify the signals used in the FM stereo multiplex system.

2. List the characteristics of the FM stereo multiplex system and


typical transmitters and receivers.

3. State the purpose of pre-emphasis and de-emphasis as used in FM


broadcasting.

4. State the requirements of television transmission and reception.

5. Identify the frequency components of a television broadcast signal.

6. Identify synchronizing pulses used in a composite video wave.

7. Define frame, field, interlaced scanning, vestigial sideband, sync


pulse, blanking, equalizing pulse, and diplexer.

8. State the purpose of a data communications system, and list the


basic system requirements.

9. Define on-line, off-line, real-time, modem, and FSK.

10. List the three modes of operation for data transmission.

11. Construct an FSK modem.


7-6 UNIT SEVEN

FM STEREO BROADCASTING
Stereo (or stereoscope) originally referred to a special photographic
technique used to give the viewer the impression of observing a scene in
three dimensions. This was accomplished using two photographs of the
same object taken from slightly different angles. When one photo was
viewed by one eye and the other photo by the other eye, the result was the
appearance of a three-dimensional image. Almost exactly the same
technique is used in stereophonic audio. Here, the same sound source
(speaker, orchestra, etc) is recorded from two different angles, in this
case, the left and right sides. When the recording is "played back"
through left and right loudspeakers to the left and right ears, the result is
the appearance of a "three-dimensional" sound source. It lends realism to
the recorded sound just as the stereoscope brought the two-dimensional
photograph to life.

Adapting stereo audio to broadcasting presented two problems. First any ,

system used to transmit stereo signals had to be compatible with the


existing monophonic receivers. And second, the stereo signal had to be
transmitted within the 200 kHz bandwidth allotted to each FM station.
Eventually, a system was approved that used a frequency division multi-
plex scheme.

The Stereo Multiplex System

To provide the necessary compatibility with monophonic receivers, the


stereo multiplex system transmits both the right and left channels at their
normal audio frequencies from 50 Hz to 15 kHz. This is called the L + R
signal. The system also transmits an L — R signal. It is transmitted as a
double-sideband suppressed-carrier signal on a 38 kHz subcarrier. As
you learned earlier, a double-sideband suppressed-carrier signal is ex-
tremely difficult to demodulate without a synchronizing signal. For this
reason, a special pilot carrier is transmitted at 19 kHz, exactly one-half
the frequency of the 38 kHz sub-carrier. The resulting stereo multiplex
signal spectrum is shown in Figure 7-1. This multiplex signal is then
used to modulate the FM broadcast transmitter.
Communications Systems
J7-7

«*
£
£
S

L + R L-R L-R

% ^ % % \ %
FREQUENCY

Figure 7-1
The Stereo Multiplex Spectrum.

Once the stereo multiplexed FM signal is received and demodulated, its


spectrum again appears just as shown in Figure 7-1. A monophonic
receiver, whose audio response is limited to 16 or 17 kHz, utilizes only
the L+R signal. Therefore, it reproduces both audio channels, but with-
out any stereo separation. Nonetheless, the multiplex system does pro-
duce a compatible monophonic signal containing both left and right
channels. On the other hand, a stereophonic receiver uses special demul-
tiplexing circuits to obtain separate left and right audio channels. Before
examining the stereo receiver in detail, let's look at the stereo transmitter.
7-8 UNIT SEVEN

The Stereo Transmitter

The block diagram of a typical stereo multiplex transmitter is shown in


Figure There is a microphone and a preamplifier for both the left and
7-2.
Both the left and right signals are immediately applied to
right channels.
summing amplifier "A." Its output is a linear mix of the two channels and
is, therefore, the L+R signal.

LEFT

RIGHT

Figure 7-2
A Stereo Multiplex Transmitter.

To form the L-R signal, the left channel is applied directly to summing
amplifier "B". However, the right channel passes through an inverter to
become the -R signal before being applied to the summing amplifier.
Therefore, the output of summing amplifier "B" is a linear mix of the left
channel and an inverted right channel, or the L— R signal.

i
Communications Systems 7-9

The L-R signal then applied to a balanced modulator. A 38 kHz carrier


is

is also applied. This carrier is derived from a 19 kHz master oscillator

using a frequency doubler. The output of the balanced modulator is the


double-sideband suppressed-carrier L-R signal. This signal is then
coupled to the FM transmitter through an amplifier. It is joined by the
L+R signal which passed through a delay network to compensate for
balanced modulator delay of the L-R signal. The 19 kHz pilot is also
applied to the transmitter.

In FM broadcasting, the maximum allowable carrier deviation is ±75


kHz. This is monaural or stereo programming. Therefore,
true for either
when the modulating signal is stereo multiplex, deviation by the pro-
gram audio is slightly reduced. This is due to the 19 kHz pilot carrier. It is
always present and does not vary in amplitude. It deviates the carrier 10%
of its maximum deviation or ±75 kHz x 10% = ±7.5 kHz. Thus, the
maximum deviation for the audio signals is 90% of ±75 kHz or ±67.5
kHz. The difference between 90% and 100% deviation is minimal and not
readily discernible to the listener. It may also seem that this remaining
90% deviation must be divided equally between the L + R and L-R
signals. However, in practice, a process called interleaving takes place.
When the L+R signal is at its peak amplitude, the L-R signal is at its
minimum amplitude, and vice versa. Therefore, due to interleaving, each
modulating signal can independently produce almost full deviation
without over deviating the carrier.
7-10 UNIT SEVEN

The Stereo Receiver

One example of a stereo multiplex receiver is shown in Figure 7-3. A


normal FM tuner and demodulator are used to recover the actual stereo
multiplex signal. Then, filters are used to separate the basic components
A low pass filter is used to obtain the L+R signal, which is
of the signal.
50 Hz to 15 kHz. A bandpass filter separates the 23-53 kHz double-
sideband L-R signal. And finally, a 19 kHz bandpass filter separates the
pilot carrier.

w
LOW PASS RIGHT
FM L +R
DEMODULATOR •—• MATRIX
TUNER FILTER 'LEFT

L-R

DSB
(AND PASS L-R BALANCED
FILTER
23-53kHz
MODULATOR

38kHz

19kHz
19kHz
!AND PASS DOUBLER
FILTER

Figure 7-3
A Stereo Multiplex Receiver.

The 19 kHz pilot is then applied to a doubler to obtain the correct phase
and frequency (38 kHz) subcarrier. This is coupled to the balanced mod-
ulator along with the double-sideband L-R signal. The demodulated
L— R signal and the L+R signal are then combined in a special matrix
network to obtain the separate left and right channels.

MH i^B
Communications Systterns 7"1
I

Figure 7-4 shows the schematic diagram for a very basic matrix network.
Here, the 38 kHz subcarrier has been developed and is applied to the
junction of D,D 2 The
. FM demodulator output is also applied. The L+R
signal is separated by the low pass filter of C,L, and developed across R,.
The double-sideband L-R signal is applied to the junction of D,D 2 .

38kHz o
SUBCARRIER

DEMODULATOR
OUTPUT O

RIGHT
OUTPUT

LOW PASS
FILTER

Figure 7-4
Simplified Stereo Multiplex Matrix.

The L-R signal is demodulated by D,D 2 and developed across R2 and R3 .

However, due to the diodes' polarity, the signal across R2 is 180° out of
phase with the signal across R3 with respect to point A. Note that the L + R
,

signal is added at point A. This means that the left output "sees" a
+ (L-R) signal across R2 and the (L+R) signal across R,. These two signals
add to produce the left output, because (L + R) + (L-R) = 2L. Similarly,
the right output "sees" a - (L-R) signal across R3 and the (L+R) signal
across R,. Therefore, it produces a right output because (L+R) - (L-R) =
L + R-L + R = 2R.

when these two signals are amplified and fed


Stereophonic sound results
to loudspeakers. The examples shown were simplifications and you
should remember that many circuit variations exist. For example, the FM
tuner of schematic diagram #2, the Heathkit AJ-1219, has an integrated
circuit stereo demodulator.
7-12 UNIT SEVEN

SCA
There is one other signal that can be multiplexed onto an FM broadcast
transmission. It is an SCA or Subsidary Communication Authorization.
The SCA that most of us are familiar with is Muzak® or storecasting. This
is a special background music service that contains no advertising or

announcements. It is paid for by subscription rather than advertising


revenue; and is transmitted along with the normal FM broadcast, except
that it is multiplexed onto a special subcarrier. This subcarrier can be any
frequency from 20 kHz to 75 kHz. However, if the station is also transmit-
ting a stereo multiplex signal, the SCA subcarrier must be above the L-R
signal, or from 53 kHz to 75 kHz. The accepted standard SCA frequency is
67 kHz.

The SCA subcarrier is frequency modulated with a maximum deviation


of ±7.5 kHz. The spectrum of a stereo generator output with an SCA
subcarrier shown in Figure 7-5. This entire spectrum is then used to
is

modulate the FM broadcast transmitter. The SCA signal is only allowed


to deviate the FM transmitter 10% of its maximum of ± 75 kHz. Therefore,
since the stereo pilot also deviates the transmitter 10% of maximum, this
leaves only 80% of deviation for the L+R and L— R signals. Thus, the
maximum allowable deviation for the main channel is reduced by 10%
when adding an SCA signal.

<*

L-R
—^—

<5
L + R
/
\ % % \

Figure 7-5
Spectrum of a Stereo Multiplex Generator with an
SCA Signal.

"Trademark, Muzak Corporation.


Communications Systems 7-13

SCA program material can be any number of things. Usually it is continu-


ous background music as mentioned previously. However, it can be
special purpose programming for particular audiences such as "talking
books" for the blind or even slow scan television for the deaf. One
application is relaying telemetry signals to the studio from a remote
transmitter site.

The SCA receiver is relatively simple. After a normal FM tuner's demod-


ulator, a 67 kHz (or any other SCA frequency) bandpass filter is used to
separate the SCA signal. Then a phase-locked loop tuned to 67 kHz is
used to demodulate the FM signal. The detected signal is amplified and
fed to a loudspeaker or other output device. The listener can use the main
channel program, the SCA program, or both, simultaneously.

Pre-emphasis and De-emphasis

In FM broadcasting, a method of noise reduction is used that requires


pre-emphasis at the transmitter and de-emphasis at the receiver. Pre-
emphasis means that the higher audio frequencies are increased in
amplitude. De-emphasis means that the higher audio frequencies are
decreased in amplitude. The reason for this
is to overcome the noise level

and thereby increase the system's signal-to-noise ratio.

Noise is always present in any system. However, as the audio frequency


increases, so does the noise level. Therefore, if the audio signal is in-

creased proportionately to the noise level, the signal-to-noise ratio will


improve greatly at the higher frequencies.
1
7-14 UNIT SEVEN

In FM broadcasting, the program audio signal is fed through a special


RLC pre-emphasis network whose time constant is 75 fxs. A graph of the
output signal amplitude versus frequency is shown in Figure 7-6. This
output response curve is very similar to a high pass filter response. To
stay within FCC regulations, the transmitted audio signal must be within
the solid and dashed lines shown on the chart.

17

16

15 SI ANDARD PRE-EMPHASIS CU RVE:


-Tl CONSTANT 75 MICROSEC DNDS
VIE
14
(SOLID LINE)
13
1
:
REQUENCY RESPONSE LIMIT S
12 5H0WN BY USE OF SOLID AN D
DASHED LINES 4
11
*

10 f r

9 /
9
8
t
u 7 W
f f
4
o
UJ
6 1
i
5
i
/
4

3
/
2 9
/
1
r

-1

-2
..-'
y *

" J
--""
50 100 200 400 600 1000 2000 4000 6000 10000 15000
CYCLES PER SECOND

Figure 7-6
Standard 75 fis Pre-Emphasis Curve.

At the receiver, an RC or RLC de-emphasis network is used at the audio


amplifier. Its response curve should be exactly opposite the 75 /xs pre-
emphasis curve. This restores the audio signal to its original frequency
-versus- amplitude response. Thus, the audio frequency response is re-
turned to normal, while the signal-to-noise ratio is greatly improved.
.

Communications Systems 7-15

Self-Review Questions

1. Is the FM stereo multiplex system time or frequency division


multiplex?

2. A monaural FM receiver receives only the signal


of a stereo multiplex transmission.

3. The L-R signal has a subcarrier of kHz and uses


modulation.

4. Label the components of the stereo multiplex signal spectrum


shown in Figure 1-1

FREQUENCY

Figure 1-1

5. A stereo multiplex receiver combines the L+R and L-R signals in a


matrix to obtain separate left and right channels.

6. The normal frequency for an SCA subcarrier is kHz. This


subcarrier is modulated.

7 . Pre-emphasis and de-emphasis are used in the FM stereo multiplex


system to improve the
1

7-16 UNIT SEVEN

Self-Review Answers

1. The FM stereo multiplex system is frequency division multiplex.

2. A monaural FM receiver receives only the L+R signal of a stereo


multiplex transmission.

3. The L— R signal has a subcarrier of 38 kHz and uses double-


sideband suppressed-carrier modulation.

4. See Figure 7-8.

<*

L-R

SCA

L + R

FREQUENCY

Figure 7-8

5. A stereo multiplex receiver combines the L+R and L— R signals in a


matrix to obtain separate left and right channels.

6. The normal frequency for an SCA subcarrier is 67 kHz. This subcar-


rier is frequency modulated.

7. Pre-emphasis and de-emphasis are used in the FM stereo multiplex


system to improve the signal-to-noise ratio.
Communications Systems J7-1T

TELEVISION BROADCASTING

Of the three basic communications systems discussed in this unit, televi-


sion broadcasting is perhaps the most widely used and by far the most
popular form of entertainment. One reason for this may be that it stimu-
lates not only hearing, as does FM broadcasting, but also sight. This
allows much greater creative freedom in the programming content.

The TV broadcasting system consists of both audio and video pickup


devices, a transmitter, and a receiver. This section covers all of these
important components.

Television Transmission

Every television broadcast consists of two separate signals: an audio


signal and a video signal. The audio signal is broadcast by an FM trans-
mitter whose maximum deviation is ±25 kHz. Other than the lower
deviation, it is virtually identical to a monaural FM broadcast transmitter.

The video signal is the output of a special camera tube, such as a vidicon
or plumbicon, that converts light into an electrical signal. A microphone
does much thesame thing for audio by coverting sound waves into
electrical signals. However, in most visual scenes, light levels vary much
more rapidly than do the sound waves in music or speech. For this
reason, the video signal's frequency can be as high as 8 MHz, although for
TV broadcasting it is limited to 4.2 MHz.

The video signal used to modulate a separate AM transmitter. How-


is

ever, since the bandwidth would be 2 x 4.2 MHz or 8.4 MHz, a com-
promise system of attenuating the lower sideband is used to reduce
bandwidth. This is called vestigial sideband transmission, since only a
vestige, or small portion, of the lower sideband is transmitted.
7-18 UNIT SEVEN

The frequency spectrum of a television transmission is shown in Figure


7-9. It shows that the vestigial sideband extends 1.25 MHz below the

carrier. The majority of visual information is contained within this 1.25


MHz band. Therefore, both sidebands transmit this important part of the
video spectrum, with the full upper sideband transmitting the finer
visual detail contained within the band from 1.25 MHz to 4.2 MHz. The
aural carrier with its ±25 kHz deviation is positioned exactly 4.5 MHz
above the visual Thus, the entire television broadcast signal
carrier.
occupies a 6 MHz bandwidth. A further bandwidth reduction could be
obtained by using a single-sideband visual signal. However, the required
receiver would be extremely complex and expensive. Thus, the com-
promise vestigial sideband system was chosen.

FREQUENCY'

Figure 7-9
Television Transmission Spectrum.

Television transmissions are separated into three frequency bands. These


are:

Channels 2 through 6 — 54 to 88 MHz


Channels 7 through 13 — 174 to 216 MHz
Channels 14 through 83 — 470 to 890 MHz
Each broadcast station is assigned a specific 6 MHz-wide channel within
these bands.
Communications Systems 7-19

The Video Signal

A television picture does not appear "instantaneously" on the screen of


the receiver's cathode ray tube. Instead, it is scanned spot-by-spot along

525 horizontal lines from the top to the bottom of the picture by the CRT's
electron beam. This process is so rapid that it appears to be instantaneous
to the eye.The actual rate is 30 complete pictures or frames every second.
This rate was determined in part by the bandwidth limitations of the TV
system. However, even at a frame rate of 30 Hz, the eye detects some
"flicker" when the picture changes. For this reason, the complete frame is
divided into 2 fields composed of 262-1/2 lines each. These fields are
then alternately scanned at a 60 Hz rate.

Figure 7-10 shows how this is accomplished. Field 1 starts in the upper
left hand corner and scans across the CRT. It is then deflected back across
the tube and starts another horizontal line. All the while, a vertical
deflection signal causes each scanned line to be slightly lower on'the
CRT. After 262-1/2 lines are scanned the vertical signal drives the elec-
tron beam back to the top of the picture tube for the start of field 2 Field 2 .

repeats this process except that its horizontal lines fall exactly in between
the lines of field 1. This is called interlaced scanning. And, even though
each frame is displayed as two separate pictures, the eye sees them as one
complete image.

Figure 7-10
Interlaced Scan.
7-20 UNIT SEVEN

Since the video image is "written" on the CRT using 525 lines, it must
also be obtained from the TV camera in the same manner. Figure 7-11
shows a simplified TV camera. The light from the scene being televised is
focused onto the light sensitive target of the vidicon camera tube. Then
the deflection coils cause an electron beam to scan the target and, thus,
the video image. The output of the vidicon is a varying voltage which
corresponds to the bright and dark portions of the picture. At the receiver,
this changing voltage causes the CRT's output to vary in intensity as the
picture is scanned. Therefore, it reproduces the camera image on the
picture tube screen.

DEFLECTION
COILS

LIGHT ^VIDEO OUTPUT

Figure 7-11
TV Camera-

The timing of the scanning at the camera tube and at the receiver's
cathode ray tube must be identical — not only in frequency, but also in
phase. Therefore, synchronization is necessary. This is accomplished by
using special sync pulses which trigger the camera deflection circuits.

They are also transmitted as part of the video signal. At the receiver, they
are separated and used to trigger the CRT deflection circuits. There are
both horizontal sync pulses and vertical sync pulses. The vertical sync
determines the field rate and, therefore, it is sent at a 60 Hz rate. The
horizontal sync determines the lines per field. Therefore it is sent at a ,

262-1/2 x 60 or 15,750 Hz rate. There is also a blanking pulse that cuts off
the CRT beam to prevent the sync signals from appearing on the screen.
Communications Systems 7-21

The composite or complete video signal consists of:

1. The camera video signal,

2. The blanking pulses,

3. The horizontal sync pulses, and

4. The vertical sync pulses.

A composite signal for one horizontal line is shown in Figure 7-12. It is a


negative transmission system. This means that the higher the amplitude,
the darker the picture detail. At 75% of maximum, the screen is black.
Therefore, the sync pulses, which are from 75% to 100%, are effectively
blanked out. The reference white level is at 12.5% of maximum with no
signal allowed to go lower than this level.

HORIZONTAL SYNC PULSE-

BLANKING PULSE

N 1 HORIZONTAL LINE-

TIME

Figure 7-12
A Composite Video Waveform.
7-22 UNIT SEVEN

The modulated RF wave is shown in Figure 7-13. Since this is a form of


AM, the modulation envelope corresponds to the composite video signal.

Figure 7-13
A Modulated Video Wave.

The vertical sync pulse occurs at the end of each field. It is shown in
Figure 7-14. Note that there are equalizing pulses and serrations of the
vertical sync pulse. These pulses and serrations are used to keep the
receiver's horizontal oscillator in synchronization during the long verti-
cal blanking interval. A low pass filter separates the vertical sync pulse
and then uses it to synchronize the receiver's vertical oscillator.

HORIZONTAL EQUALIZING EQUALIZING HORIZONTAL


SYNC PULSE PULSES PULSES SYNC PULSES

ONE SERRATED PICTURE SIGNALS


HORIZONAL
BLANKING PULSES VERTICAL OF ONE LINE
SYNC PULSE

Figure 7-14
The Vertical Sync Interval.
>

Communications Systems 7-23

The TV Transmitter

A greatly simplified block diagram of a TV transmitter is shown in Figure


7-15. It is modulated transmitter since the video amplitude
called an IF
modulation takes place at an intermediate frequency amplifier. Let's
examine this diagram in detail.

VISUAL TRANSMITTER

CRYSTAL POWER
OSCILLATOR
BUFFER
IF

AMPLIFIERS
MIXER
AMPLIFIER
— VSB
FILTER

M M
v M\

LIGHT VIDEO VIDEO HETERODYNE


CAMERA DIPLEXER
AMP MODULATOR OSCILLATOR
/ \ / k M

SYNC
GENERATOR

AURAL TRANSMITTER

AUDIO FM
MICROPHONE \\
V AMPl IFIER TRANSMITTER

Figure 7-15
Simplified Block Diagram of a TV Transmitter.

The visual transmitter's input comes from a camera and video amplifier.
The sync generator supplies scanning voltages and blanking pulses to the
camera. It also provides sync pulses to the video amplifier where the
composite video signal is formed. The composite video is then applied to
the video modulator, which amplitude modulates the cyrstal oscillator's
RF signal at the IF amplifier. The IF amplifier increases the amplitude of
the modulated RF and also uses a special filter to shape the output
frequency response. This is where the vestigial lower sideband is formed.
7-24 UNIT SEVEN

The mixer and heterodyne oscillator convert the IF signal to the required
output channel. The modulated RF is then amplified to the required
power level by the linear power amplifier. The VSB filter insures that the
vestigial lower sideband is sufficiently attenuated. The visual signal is
then applied to the diplexer. This device couples both the visual and
same antenna while preventing either one from
aural transmitters to the
coupling to the other transmitter. .

The aural transmitter is a straightforward FM broadcast transmitter with


a maximum deviation of ±25 kHz. However, since the aural carrier must
be spaced exactly 4.5 MHz from the visual carrier, a special automatic
frequency control is used. It effectively "locks" the aural transmitter to
the visual carrier by constantly maintaining the 4.5 MHz offset.

The TV Receiver

The simplified block diagram of a TV receiver is shown in Figure 7-16. It

is a typical superheterodyne receiver. The RF tuner is composed of the RF


amplifier, mixer, and local oscillator. It converts the incoming television
signal to the intermediate frequency. The standard IF values are 45.75
MHz for the visual carrier and 41.25 MHz for the aural carrier. The
frequency inversion occurs at the mixer. The IF amplifiers must be
broadbanded to amplify the entire television spectrum. However, their
response curve must also be tailored to equalize the lower frequency
response with the higher frequency response. The desired IF response
curve is shown in Figure 7-17. Such
compensates for having two
a curve
sidebands for video frequencies up to 1.25 MHz
and only one sideband
for the video frequencies above 1.25 MHz. In short, it eliminates any
frequency distortion caused by the vestigial sideband. The result at the
diode detector is equal amplitude response over the entire video fre-
quency range.
Communications Systems 7-25

DEFLECTION

V 45. 75MHz VISUAL


41.25 MHz AURAL
YOKE

RF IF DIODE VIDEO
TUNER AMPLIFIERS DETECTOR AMPLIFIER

SYNC VERTICAL AND


SEPARATOR HORIZONTAL
DEFLECTION

SPEAKER
4. 5 MHz
FM AUDIO
IF
DETECTOR AMPLIFIER
AMPLIFIER

Figure 7-16
Simplified Block Diagram of a TV Receiver.

FREQUENCY

Figure 7-17
IF Frequency Response.
7-26 UNIT SEVEN

The diode detector not only demodulates the visual AM signal, it also
mixes the visual carrier and aural carrier. The difference frequency is
45.75 - 41.25 or 4.5 MHz. This 4.5 MHz FM signal is amplified, detected,
and applied to the loudspeaker.

The detected composite video signal is amplified and applied to the CRT
and sync separator circuit. At the CRT, it varies the intensity of the
electron beam as it scans the screen. The sync separator removes the
horizontal sync pulses, equalizing pulses, and the vertical sync pulses
from the composite video. These signals are used to synchronize the
horizontal and vertical oscillators which, in turn, drive the deflection
circuits. The horizontal and vertical deflection circuits drive the CRT
deflection yoke and, therefore, cause the electron beam to scan the CRT
screen.
Communications Systems 7-27

Self-Review Questions

8. A television broadcast consists of an signal and a


signal.

9. In television broadcasting, the aural carrier is

modulated, while the visual carrier is mod-


ulated.

10. The maximum deviation of the aural carrier is kHz.

11 . The visual carrier uses a special form of AM called


transmission.

12. Identify the components of a television broadcast signal by filling


in the blanks of Figure 7-18.

M
/
/
\

/ \

4 MHz —
FREQUENCY
MHz
— »

Figure 7-18

13. In the scanning of a television picture, there are


horizontal lines per picture. However, there are hori-
zontal lines per field with fields per frame.

14. The field rate in television is Hz and the frame rate is


7-28 UNIT SEVEN

15. Flicker of the video image is reduced by using


scanning.

16. What are the four components of a composite video signal?

1.

2.

3.

4.

17. What is the purpose of the synchronizing pulses and the blanking
pulses?

18. In the composite video wave, what is the purpose of the equalizing
pulses and the serrations of the vertical sync pulse?

19. What is the purpose of the diplexer in the TV transmitter?

20. What is the purpose of the sync separator in a TV receiver?


Communications Systems 7-29

Self-Review Answers

8. A televison broadcast consists of an audio signal and a video


signal.

9. In television broadcasting, the aural carrier is frequency mod-


ulated, while the visual carrier is amplitude modulated.

10. The maximum deviation of the aural carrier is ±25 kHz.

11. The visual carrier uses a special form of AM called vestigial


sideband transmission.

12. See Figure 7-19.

VISUAL CARRIER

AURAL CARRIER

FREQUENCY

Figure 7-19
7-30 UNIT SEVEN

15. Flicker of the video image is reduced by using interlaced scanning.

16. The four components of a composite video signal are:

1. Camera video
2. Blanking pulses
3. Horizontal sync
4. Vertical sync

17. The blanking pulses cut off the receiver's CRT to prevent the sync
pulses from being seen. The synchronizing pulse stabilizes the
picture by insuring that the scanning at the receiver is identical to
the camera scanning.

18. The equalizing pulses and the serrations of the vertical sync pulse
maintain horizontal synchronization during the vertical sync in-
terval.

19. The diplexer couples both the visual and aural transmitters to the
same antenna while preventing either one from coupling to the
other.

20. The sync separator removes the horizontal and vertical sync pulses
from the composite video.
Communications Systems 7-31

DATA COMMUNICATIONS
A data communications system must transmit pulses, that are the output
of a data source,from one location to another. An example of this is the
transmission of pulses from a teletype unit to a computer, or vice versa.
The communication channel itself may be a cable, a switched public
telephone line, or even a radio transmission. This section is a brief
introduction to the broad and complex field of data communications.

Requirements

The basic requirement of a data communication system


is that it must be

capable of transmitting rectangular pulses from 100 pps (pulses


at rates

per second) up to 500,000 pps. However, the most commonly used speeds
are 600 to 50 ,000 pps. Also, the system may be required to transmit in one
direction only, alternately in either direction, or simultaneously in both
directions. In addition, the error rate must be as low as possible. This
requirement is relatively easy to satisfy, provided the communication
channel's signal-to-noise ratio is reasonable.

System Types

Basically, there are two types communications systems: on-line


of data
and off-line. An on-line system connects a computer directly to the
communications link. On the other hand, the off-line system uses an
intermediate storage device such as punched paper tape, magnetic tape,
or disk to store the data before transmission or upon reception.

An on-line system is means that its


generally a real-time system. This
data transmission requires an immediate response. An example of a real
time system is an airline reservation computer. Here, an inquiry from a
reservation desk anywhere world must be handled and answered
in the
rapidly by the central reservation computer. As you can imagine, on-line
real-time systems require complex computer software, elaborate com-
munications links, and therefore, are very expensive.
7-32 UNIT SEVEN

An off-line system is used to transmit data that doesn't require a real-time


response or requires no response at all. One example of an off-line system
is a national news service such as the Associated Press (AP) or United

Press International (UPI). Here, the data, which is news, is transmitted


from a central location to the subscribing newspapers, radio, and TV
stations.There is no requirement for real-time operation since no re-

sponse is expected or required.

Modes of Operation

There are basically three modes of operation for data transmission. The
first of these is simplex which is the simplest mode. It is purely a one-way

transmission channel as shown in Figure 7-20A.

The second mode is shown in Figure 7-20B. In this mode, transmission


can occur one way or the other but not simultaneously. This is called half
duplex or semiduplex operation.

The third mode, shown in Figure 7-20C, is full duplex or, simply, duplex
operation. This mode requires two separate communication channels.
However, it offers the advantage of simultaneous two-way communica-
tion.

ONE WAY ONLY


POINT A POINT B

SIMPLEX

B ONE WAY OR THE OTHER


POINT A POINT B

HALF DUPLEX

BOTH WAYS
SIMULTANEOUSLY
POINT A POINT B
4
FULL DUPLEX

Figure 7-20
Modes of Operation for Data Communication systems.
Communications Systems 7-33

Modems
Most data communications are over telephone lines, either a switched
line dialedby the user, or a special dedicated line. Since DC pulses are not
normally sent over telephone lines, a special circuit called a "modem" is
used to interface the data equipment to the line.

The term modem is a contraction of the words modulator and demod-


ulator. And, the device does just that —
it modulates an audio frequency

wave to transmit data and demodulates the incoming audio wave. Thus,
the modem consists of an audio frequency transmitter and receiver. It

may also convert parallel data to serial for transmission and vice versa on
reception.

The modulation technique generally used in modems is frequency shift


keying (FSK). In this technique, a specific audio frequency represents an
"on" condition, while another frequency represents an "off" condition.
The difference between these two frequencies is the frequency shift.
When data is being transmitted, the pulses "key" or shift the transmitter
between the two FSK frequencies just like an FM transmitter. Similarly,
the receiver uses an FM detector or phase-locked loop to distinguish
between the two frequencies and convert them to DC pulses. An example
of an FSK transmission is shown in Figure 7-21.

TRANSMIT^ RECEIVE
DATA IN DATA OUT
TELEPHONE LINE
MODEM MODEM
juuri HULL
FSK SIGNAL

Figure 7-21
An FSK Modem Simplex Transmission.

The modem audio frequencies are chosen in the 300 Hz to 3000 Hz range,
which is normal telephone line range. Many frequency combinations are
possible, however, one standard is 1270 Hz for the "on" condition and
1070 Hz for the "off" condition. This is a 200 Hz shift FSK signal.
7-34 UNIT SEVEN

AMERICAN STANDARD CODE FOR INFORMATION INTERCHANGE

COLUMN 1 2 3 4 5 6 7

ROW 000 001 010 011 100 101 110 111


765 >
4321 C
<3>
0000 NUL DEE SP @ P \
P

0001 SOH DCI ! 1 A Q a q


"
0010 STX DC 2 2 B R b r

0011 LTX DC:* = 3 C S c s

0100 EOT DC4 S 4 D I d I

0101 ENQ NAK a-


5 E u e U

0110 ACK. SYN & (3 F V I \

0111 BEL ETB / G w g w


1000 BS CAN- (
8 H X h X

1001 HT EM I
9 I Y i
y

10 1010 LF SLB *
J z J
/.

1011 VT ESC + ;
K I
k {

12 1100 FF FS .
< L \ 1
i

13 1101 CR GS - = M I
m I

14 1110 SO RS > X /~s n -

15 1111 SI LS t — o DEL

Explanation of special control functions in columns 0, 1, 2 and 7.

NUL NuJJ DLE Data Link Escape


SOH Start of Heading DCI Device Control 1
STX Start of Text DC2 Device Control 2
ETX End of Text DC3 Device Control 3
EOT End of Transmission DC4 Device Control 4
ENQ Enquiry NAK Negative Acknowledge
ACK Acknowledge SYN Synchronous Idle
BEL Bell (audible signal] ETB End of
Transmission Block
BS Backspace CAN Cancel
HT Horizontal Tabulation (punched card skip] EM End of Medium
LF Line Feed SUB Substitute
VT Vertical Tabulation ESC Escape
FF Form Feed FS File Separator
CR Carriage Return GS Group Separator
SO Shift Out RS Record Separator
SI Shift In US Unit Separator
SP Space fblankj DEL Delete

Figure 7-22
The ASCII Code
Communications Systems 7-35

Using modems and a single telephone line allows both simplex and
semiduplex operation. However, by using a form of frequency division
multiplex, full duplex operation can be achieved on a single telephone
line. This is done by transmitting in one direction at 1100 to 1300 Hz and
in the other direction at 2100 to 2300 Hz. Of course, the modems must be
adjusted for this type of operation. But nonetheless, full duplex operation
is achieved using a single line.

Data Communications Codes

The most widely used data communications code is the American Stan-
dard Code for Information Interchange, abbreviated ASCII. It is a 7-bit
binary code that used in transferring data between computers and their
is

external peripheral devices and in communicating data by radio and


telephone lines. Seven bits can represent a total of 2 7 or 128 different
states or characters. The ASCII code is used to represent the decimal
numbers through 9, the letters of the alphabet including both upper and
lower case, plus other special control symbols. The standard ASCII code
is shown in Figure 7-22.

The 7-bit ASCII code for each number, letter or control function is made
up of a 4-bit group and a 3-bit group. Figure 7-23 shows the arrangement
of these two groups and the numbering sequence. The 4-bit group is on
the right and bit 1 is the least significant bit. Note how these groups are
arranged in rows and columns in Figure 7-22.

4 BIT GROUP

7 6 5 4 3 2 1

3 BIT GROUP
7-36 UNIT SEVEN

To determine the ASCII code for a given number, letter, or control opera-
tion,you locate that item in the table. Then you use the 3- and 4-bit codes
associated with the row and column in which the item is located. For
example, the ASCII code for the letter L is 1001 100. It is located in column
4, row 12. The most significant 3-bit group is 100, while the least signific-
ant 4-bit group is 1100.

There are both 6- and 8-bit special versions of the ASCII code. In addition,
the International Business Machines Corporation (IBM) uses another
coding system called Extended Binary Coded Decimal Interchange
8-bit
Code (EBCDIC) instead of ASCII, for its peripheral and data communica-
tions operations.
Communications Systems 7-37

Self-Review Questions

21. What is the purpose of a data communications system'

22. List the basic requirements of a data communications system.

23. What are the two types of data communications systems? Define
each type. _

24. What is a real-time system?

25. List the three modes of operation for data transmission.

26. What is a modem?

27. What is FSK?

28. What is the most .widely used data communications code?


7-38 UNIT SEVEN

Self-Review Answers

21. The purpose of a data communications system is to transmit pulses,


that are the output of a data source, from one location to another.

22. The basic requirements of a data communications system are as


follows. must be able to transmit retangular pulses at 100 pps up
It

to 500,000 pps. It must transmit in either one direction, alternately


in either direction, or simultaneously in both directions. Its error
rate must be low.

23. The two types communications systems are on-line and


of data
off-line. An on-line system connects the computer directly to the
communications link. An off-line system uses an intermediate
storage device before transmission or upon reception.

24. A real-time system is usually an on-line system that requires im-


mediate response to the transmission.

25. The three modes of data transmission are: simplex, half-duplex or


semiduplex, and full duplex.

26. A modem is an interface device that connects the data equipment to


the telephone line.

27. FSK is frequency shift keying.

28. The most widely used data communications code is ASCII.


EB6106 5952692

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