Communication Systems

CHAPTER ONE

1.1 Communication
This is defined as the transfer of meaningful information (intelligence) from a source to a destination.

Source Flow of Information Destination

Transmitter Receiver

Figure 1: Transfer of information

In communication engineering one is concerned with the transmission of various signals from one point
(source) to another (destination). The source originates an input or message, such as a human voice, a
television picture, a code, a teletype message or data. This information (message) flows through a
channel before arriving at its destination, which is the unit to which the message is communicated.

1.2 Methods of Communication

There are two basic methods of communication. These are aural and visual.
a) Aural Communication:
This is a communication in which the transfer of information is in the form of sound waves,
generated by the sender. The reception of information is by the listener’s ears which convert
those sound waves to hearing. These include:
Speech, telegraph, telephone, wireless telegraph, radio etc.

b) Visual Communication: These include sign language, flag signals, light signals, pictorial
language, printing press, photography, electronic visual communication

Having defined these, it can still be said that, today, the commonest and most important forms of
communication are speech, music, picture, and computer data. Speech is the most common, followed
by music. The third source or form of communication is picture, and then computer data, such as e-mail,
exchange of software, and sharing of resources.

1.3 Communication System

A communication system is a collection of individual communications networks. These consist of
intelligence origination, transmission systems, receiving systems and intelligence reproduction.
In the most fundamental sense, communication involves implicitly the transmission of information
through a succession of processes as described below:
i. The generation of a thought pattern or image in the mind of an originator
ii. The description of that image, with a certain measure of precision, by a set of aural or visual
symbols.
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iii. The encoding of these symbols in a form that is suitable for transmission over a physical
medium of interest.
iv. The transmission of the encoded symbols to the desired destination.
v. The decoding and reproduction of the original symbols.
vi. The recreation of the original thought pattern or image, with a definable degradation in
quality, in the mind of a recipient. The degradation is caused by imperfections in the system.

1.4 Elements of a Digital Communication System

The whole digital communication system is divided as per Figure 1.2. These are the basic elements of
any digital communication system and it gives a basic understanding of communication systems. These
basic elements will be discussed.

INTELLIGENCE TRANSMISSION SYSTEM

ORIGINATION

INPUT ENCODER/ TRANSMITTER

TRANSDUCER MODULATOR

COMMUNICATION CHANNEL
NOISE SOURCE

RECEIVER DECODER/ OUTPUT

DEMODULATOR TRANSDUCER

RECEIVING SYSTEM INTELLIGENCE

REPRODUCTION

Figure 1.2: Electronic Communication System

1.4.1 Information Source and Input Transducer:

The communication systems communicate messages. The message comes from the information sources.
The source of information can be analog or digital. If the message or data is non-electrical, e.g. human
voice, picture etc. it must be converted by an input transducer into an electrical waveform. In digital
communication the signal produced by this source is converted into digital signal which consists of 1′s
and 0′s. For this there is the need for a source encoder.

1.4.2 Source Encoder

In digital communication the signal is converted from source into digital signal as already mentioned.
The objective is to use as few binary digits as possible to represent the signal. In this sense, this efficient
representation of the source output results in little or no redundancy. This sequence of binary digits is
called information sequence.
Source Encoding or Data Compression: the process of efficiently converting the output of either analog
or digital source into a sequence of binary digits is known as source encoding.
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1.4.3 Digital Modulator:

The binary sequence is passed to a digital modulator which converts the sequence into electric signals
so that this could be transmitted through a channel. The digital modulator maps the binary sequences
into signal wave forms, for example if we represent 1 by sin x and 0 by cos x then we will transmit sin
x for 1 and cos x for 0.

1.4.4 Transmitter:
The transmitter (TX) is a collection of electronic circuits designed to convert the information into a
signal suitable for transmission over a given communication medium. That is, the transmitter processes
the input signal to produce information. For example in radio broadcasting, the electrical signal obtained
from sound signal, is processed to restrict its range of audio frequencies and is often amplified. Most of
the transmitters have built-in amplifier circuits. These circuits amplify the incoming signals
(information) before transmission which help in faithful reception of the transmitted information at the
receiver end. All these are done so that the transmitted signal could suit the characteristic of the
transmission channel.

1.4.5 Channel:
The communication channel is the physical medium that is used for transmitting signals from transmitter
to receiver. It acts partly as a filter to attenuate the signal and its waveform. The signal attenuation
increases with the length of the channel, varying from a few percent for short distances to orders of very
high magnitude. The waveform may be distorted because of different amounts of attenuation and phase
shift suffered by different frequency components of the signal.
Depending on the type of communication medium, the communication system may be classified as:
i. Wire or Line communication (channels based on guided propagation).
ii. Wireless or Radio communication (channels based on free propagation).

Channels

Wired Wireless

Telephone Coaxial Optical Wireless Satellite Mobile

channels cable fibre broadcast channels radio
channels channels

Figure 1.3: Classification of Communication Channels

Thus the communication channel can be a channel based on guided propagation such as a pair of
conducting wire, a coaxial cable or optical fibre or it can be that of channel based on free propagation
such as free space or radio link. For traditional telephony, this channel is wired. In wireless system, this
channel is mainly the atmosphere. There are optical channels, under water acoustic channels etc.
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1.4.6 Noise
Noise is a random, undesirable (unwanted) energy or signal that enters the communication system
through the communication channel or medium and interferes with the transmitted message or signal.
The noise signal is random in character and may interfere with signal at any point in a communication
system. For example some noise are produced in the receiver, however it has its greatest effect on the
signal in the channel. They can be natural or man-made.
Noise is one of the serious problems of electronic communication, however, it cannot be completely
eliminated. Fortunately, there are ways to deal with noise, and reduce the possibility of degradation of
signal due to noise.

1.4.7 Receiver
A receiver (RX) is a collection of electronic circuits and components designed to convert the signal back
to the original information. It reprocesses the signal received from the transmitter through the channel,
by undoing the signal modifications made at the transmitter and channel. In other words the receiver
accepts the transmitted message from the channel and coverts it back into a form understandable by
human. The receiver operation includes demodulation, decoding and amplification for compensating
transmission losses, attenuation etc. Demodulation and decoding are the reverse of the signal processing
performed at the transmitter. Thus the receiver consists of amplifier, detector, mixer, oscillator,
transducer, and so on.

1.4.8 Digital Demodulator

The digital demodulator processes the channel corrupted transmitted waveform and reduces the
waveform to the sequence of numbers that represents estimates of the transmitted data symbols.

1.4.9 Source Decoder

At the end, if an analog signal is desired then source decoder tries to decode the sequence from the
knowledge of the encoding algorithm. And which results in the approximate replica of the input at the
transmitter end.

1.4.10 Output Transducer

Finally we get the desired signal in desired format - analog or digital.
Note: The output transducer converts the electrical signal to its original form – the message.

Human message Message for

Input (voice, code, Transmitter Communication Receiver human
pictures, data etc.) channel or medium
application

Noise

Figure 1.4: Flow of Information disturbed by noise

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1.5 Types of Electronic Communication

They can be categorized using the types of intelligence signals transmitted. These are:
i. Analog ii. Digital signals

1.5.1 Analog Signals

This is a smooth, continuously varying voltage or current. Examples are voice and video voltages. Thus,
the output voltage from an audio amplifier might be any one of the infinite values between -10 V and +
10 V at any particular instant of time. A typical analog signal is a sine wave tone. Typical examples of
analog devices are radio frequency transmitters and receivers.

1.5.2 Digital Signals

This is a type of signal that used some form of digital codes. They have one of a limited number of
values. In most applications, the applications have two (2) values which in crude terms could be
explained as ON and OFF. These values are usually described as 1 and 0, being the presence and
absence of the supply voltage or current.
For example, the earliest forms of the wire and radio communications used a kind of on/off digital code,
similar to the Morse code invented for the telegraph.
In computers, binary codes/digits are used to represent data. That is, the letters, numbers and special
symbols like comma, apostrophe, question mark etc. are transmitted by wire or radio in the form of
binary codes.
It must be noted however that both analog and digital signals may be transmitted in digital form. This is
because the analog signals could be converted using the analog-to-digital converter.

1.6 Baseband Signal

This is the other name for the original information or signal, regardless of whether they are analog or
digital. In modulation, they are also termed as the modulating or intelligence signal. Often times, in
communications system, the baseband signals may be transmitted over a medium either by themselves
or by modulation.
When the transmission is by themselves, where the original voice, video or digital signals are put directly
into the medium, then baseband transmission is taking place. A typical example is a transmission over
intercom and some telephone lines where the voice itself is placed on the wires and transmitted. This is
similar to computer networking, where the digital signals are applied directly to co-axial cables for
transmission to another computer.

1.7 Bandwidth
It is that portion of the electromagnetic spectrum occupied by a signal. It is also the frequency range
over which an information signal is transmitted or over which a receiver or other electronic circuit
operates. More specifically, the bandwidth is the difference between the upper and lower frequency
limits of the signal or the equipment operation range. Knowing the different types of passband signals
such as voice signal, music signal, TV signal etc., each of these signals will have its own frequency
range.
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Bandwidth

𝑓1 = 300 𝐻𝑧 𝑓2 = 3400 𝐻𝑧

Figure 1.5: Determining Bandwidth

The range of frequencies that is needed for a good enough quality of voice, so that the speaker can be
recognized, was defined to be the range from 300 to 3,400 Hz.
From figure 1.5, the frequency spectrum ranges from lower frequency, 𝑓1 = 300 𝐻𝑧 to upper frequency,
𝑓2 = 3400 𝐻𝑧. Thus the bandwidth, BW is given by
𝐵𝑊 = 𝑓2 − 𝑓1
= 3400 – 300 = 3100 𝐻𝑧
This means that the bandwidth of the telephone channel through the network is 3.1 kHz, as shown in
Figure 1.5. A human voice contains much higher frequencies, but this bandwidth was defined as a
compromise between quality and cost. It is wide enough to recognise the speaker, which was one
requirement for telephone channel.

Table 1.1: Types of Signal and Available Bandwidth

S/No. Type of the signal Range of Frequency (Hz) Bandwidth (Hz)
1. Voice signal (speech) for telephony 300 – 3400 3100
2. Music signal 20 – 15000 14,980
3. TV signals (picture) 0 – 5 MHz 5 MHz
300 – 3400
4. Digital data (when using the telephone line 3100
for its transmission)

1.8 Simplex, Half-Duplex, and Full-Duplex Communication

In telecommunications systems the transmission of information may be unidirectional or bidirectional.
The unidirectional systems transmit in one direction only and are called simplex, whilst bidirectional
systems are able to transmit in both directions and are called duplex systems. Bidirectional information
transfer can be implemented with half- or full-duplex transmission.

1.8.1 Simplex

Source Destination

Figure 1.6: Simplex

For simplex, the signal is transmitted in only one direction. An example of this principle is broadcast
television, where TV signals are sent from a transmitter to TV sets only and not in the other direction.
Another example is radio broadcasts.

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1.8.2 Half-Duplex

Source Destination

Figure 1.7: Half-duplex

In half-duplex operation the signal is transmitted in both directions but only in one direction at a time.
An example of this is a mobile radio system (e.g. Walkie-Talkie) where the person speaking must
indicate by saying the word over that she is done transmitting and the other person is allowed to transmit.
LANs use a high-speed, half-duplex transmission over the cable even though users may feel that the
communication is continuously bidirectional, that is, full duplex.

1.8.3 Full-Duplex

Source Destination

Figure 1.8: Full-duplex

In full-duplex operation signals are transmitted in both directions at the same time. An example of this
is an ordinary telephone conversation where it is possible for both people to speak simultaneously. Most
modern telecommunications systems use the full-duplex principle, which we call duplex operation for
short.

1.9 Modulation
It is the process of putting information in the form of a low frequency signal (baseband, voice, video or
digital) called the modulating signal on to a high-frequency (shorter wavelength) signal called the
carrier. The carrier which is usually a sine wave is of a higher frequency than the highest intelligence
signal frequency. In the modulation, some characteristics of the carrier is varied in accordance with the
modulating signal. Its voltage-time variation is represented by the equation:
𝑉𝑐 (𝑡) = 𝐴 sin(𝜔𝑐 𝑡 + 𝜑) (1.1)
The device that is used to undertake the modulation process is termed as a modulator. It forms some
kind of function
𝑠(𝑡) = 𝑓 [𝑚(𝑡), 𝑐(𝑡)] (1.2)
where 𝑠(𝑡) = modulated signal, 𝑚(𝑡) = information signal, 𝑐(𝑡) = the carrier

𝑚 (𝑡) 𝑠(𝑡)
Information Signal Modulator Modulated Signal

Carrier 𝑐(𝑡)
Figure 1.9: The Modulation Process
The modulation process simply involves performing an operation on 𝑚(𝑡) to translate its frequency
spectrum to a frequency band suitable for propagation over the available physical channel, and at the
specified frequency.
The opposite process of modulation is called Demodulation, which is the process of removing the carrier
from the modulated signal. This is done so as to recover the original information.

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1.9.1 The Need for Modulation/ Benefits of Modulation

There is the need for intelligence/information signals to be modulated before transmission due to the
following:
i. difficulty in transmitting a baseband signal over long distance
ii. height of antenna
iii. multiplexing of signals
iv. mixing of signals
v. quality of reception etc.

(a) Difficulty in Transmitting Baseband Signal over Long Distance

The frequency of the human voice ranges from about 100 to 3000 Hz. If those frequencies are
transmitted directly as radio waves, then interference would cause them to be ineffective, since
radiation is poor and signal gets highly attenuated. This means baseband signals cannot be
transmitted directly over long distance. Modulation tends to increase the frequency of the signal
about to be transmitted and hence tends to increase the distance over which signals can be
transmitted.

(b) Height of Antenna

Also, for efficient radiation, antenna dimensions had to be of the same order as the wavelength of
the signal being radiated.
Given the frequency, f, and the wavelength, λ of a wave, the velocity, V is given by
𝑉 = 𝑓𝜆 (1.3)
But for an electromagnetic wave (EM), travelling at the speed of light, then its wavelength is given
by
𝐶
𝜆=𝑓 (1.4)
Thus for an EM wave of audio frequency of say, 3000 Hz, then its wavelength is given by
3×108
𝜆= = 1.0 × 105 = 100 𝑘𝑚
3000
𝜆
If the antenna is a half-wave antenna where 𝑙 = , then height of antenna, 𝑙, should be
2
approximately,
𝜆 100
𝑙= = = 50 𝑘𝑚
2 2
From the result, it is obvious that it is virtually impossible to transmit low frequencies since the
required antennas for efficient propagation would be miles (kilometres) in length.
On the other hand, considering a modulated signal with a frequency of 10 MHz in the broadcast
band,
𝐶 3×108
Wavelength will be 𝜆 = 𝑓 = = 30 𝑚
10×106
Hence the height, 𝑙, of the half-wave antenna will be
𝜆 30
𝑙= = = 15 𝑚
2 2
This is a realistic height and the antenna can be installed.
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(c) Multiplexing of Signals

The transmission of one signal at a time on a channel is a highly wasteful situation. By the use of
modulation, several signals can be transmitted at a time on a channel. This is referred to as
multiplexing, which is a means which allows two or more signals to be transmitted simultaneously
over a channel. This increases the efficiency of the channel.

1.9.2 Classification of Modulation

The Modulation process is classified into:
a) Continuous-wave modulation and
b) Pulse modulation

Types of Modulation

Continuous wave modulation Pulse Modulation

Amplitude Angle Digital Analog

modulation modulation modulation modulation

Phase Pulse Code PAM PDM PPM

Frequency
modulation modulation
modulation

Figure 1.10: Classification of Modulation

1.9.3 Continuous-Wave (CW)

In CW modulation, a sinusoidal wave is used as the carrier.

+V

0 t

-V

Figure 1.11: A sinusoidal

Let the carrier wave bewave
represented by Process
Modulation
𝑉𝑐 (𝑡) = 𝐴 sin(𝜔𝑐 𝑡 + ∅) ............................................................................................ (1.4)
Where,

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A is for Amplitude Modulation (AM)

𝑓(𝑜𝑟 𝜔𝑐 = 2𝜋𝑓𝑐 ) is for Frequency Modulation (FM)
∅ is for Phase Modulation (PM)
When the amplitude of the carrier is varied in accordance with the intelligence (message) signal,
amplitude modulation (AM) is obtained, and when the angle of the carrier is varied, angle modulation
is obtained.
The angle modulation form of the CW modulation may be further subdivided into frequency modulation
(FM) and phase modulation (PM), in which the instantaneous frequency and phase of the carrier,
respectively, are varied in accordance with the message signal.
1.9.4 Pulse Modulation
In Pulse Modulation, the carrier consists of a periodic sequence of rectangular pulses. The pulse
modulation can be of an analog or digital type.
In Analog Pulse Modulation, the amplitude, duration or position of a pulse is varied in accordance with
sample values of the message signal. Thus the modulation could be Pulse-Amplitude Modulation
(PAM), Pulse-Duration Modulation (PDM) or Pulse-Position Modulation (PPM).
The standard digital form of Pulse Modulation is known as Pulse-Code Modulation (PCM), and this has
no CW counterpart. It starts out as PAM, but with an important modification. The amplitude of each
modulated pulse (that is, sample of the original message signal) is quantized or rounded off to the nearest
value in a prescribed set of discrete amplitude levels and then coded into a corresponding sequence of
binary.
In digital communications, the modulating signal consists of binary data or an M-ary encoded version
of it. This data is used to modulate a carrier wave (usually sinusoidal) with fixed frequency. The
modulation process involves switching or keying the amplitude, frequency or phase of the carrier in
accordance with the input data. Thus, there are three basic modulation techniques for the transmission
of digital data. They are known as amplitude-shift keying (ASK), frequency-shift keying (FSK) and
phase-shift keying (PSK).

1.10 Electromagnetic Spectrum

Before information can be transmitted, it must be converted into electronic signals compatible with the
medium. This makes it possible for a microphone to change voice into a voltage of varying frequency
and amplitude. This baseband signal is then passed over wires to a receiver or headphone.
Instead of using wires, free space can be used. The information is converted into Electronic signals
which radiate into space. Such signals consist of both Electric and Magnetic fields. These so-called
Electromagnetic (EM) signals are generated through the generation of electromagnetic fields and travel
through space for long distances.
The EM waves vary sinusoidally, and their frequency is measured in cycles per second (CPS) or hertz
(Hz). These oscillations may occur at a very low frequency or at extremely high frequency. This entire
range of frequencies is referred to as the Electromagnetic spectrum, and this comprises of the range of
all possible EM radiation. In effect, the spectrum of an object is the characteristic distribution of EM
radiation from that object.
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The behaviour of the EM radiation largely depends on its wavelength.

1.10.1 Classification
The EM spectrum by virtue of wavelength comprises of a number of parts called bands. Bands, however,
do not exist naturally. They are used in order to explain the different properties of various spectrum
parts. As a result, there is not a clear distinction between some bands of the EM spectrum. The
classification is according to electrical energy, radio, microwave, infrared, the visible region which is
perceived as light, ultraviolet, x-rays and gamma rays.
The classification scheme of radiation is generally accurate, but in reality, there is often some overlap
between neighbouring types of EM energy.

X-rays
Radio waves Microwaves Infrared Gamma rays
U-V

106 107 108 109 1010 1011 1012 1013 1014 1015 1016 1017 1018 1019 1020 1021

Figure 1.12: The Electromagnetic

Frequency (Hz)
Spectrum
The EM spectrum extends from below the frequencies used for modern radio (at the long-wavelength
end) through Gamma radiation (at the short-wavelength end). This covers EM wave energy having
wavelengths from thousands of metres down to a fraction, the size of an atom.
For example, light is an EM wave at about 5 × 1014 𝐻𝑧 (600 μm) while the usable radio waves extend
from about 1.5 × 104 𝐻𝑧 (20 km) to 3 × 1010 𝐻𝑧 (10 mm). The human eye is able to perceive to the
very narrow range of light frequencies and consequently blind to the radio-waves, though it could be
said to be a good thing since the great number of radio waves surrounding the Earth would otherwise
paint a very chaotic picture.

1.10.2 Electromagnetic Energy

EM energy at a particular wavelength, λ (in vacuum), has an associated frequency, 𝑓 and photon energy,
E. The EM spectrum may be expressed equally well in terms of any of these quantities:
𝐸 = ℎ𝑓 (1.5)
ℎ𝑐
𝐸= (1.6)
𝜆
Where h = Planck’s constant
C = 299,792,458 m/s or 3.0 × 108 m/s
ℎ ≈ 6.62069 × 10−34 𝐽𝑠 or ℎ ≈ 4.13567𝜇𝑒𝑉/𝐺𝐻𝑧

From equations (1.5) and (1.6), it could be observed that:

i. High-frequency EM waves have a short wavelength and high energy whilst
ii. Low-frequency waves have a long wavelength and low energy.
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Assignment
Determine the relation between Joules (J) and electron-volt (eV).

1.11 Classification of the EM Spectrum

The EM spectrum as used in electronic communications is classified largely as observed in Figure 1.12
but further broken down as seen in Table 1.2. Table 1.2 also gives the wavelength, frequency and energy
range of each class of spectrum within the radio wave and microwaves.

Table 1.2: EM spectrum used in Electronic Communications

CLASS WAVELENGTH FREQUENCY ENERGY
Extremely Low Frequency 10 – 100 Mm 3 – 30 Hz 12.4 – 124 feV
Super Low frequency 1 – 10 Mm 30 – 300 Hz 124 feV – 1.24 peV
Voice Freq. /Ultra Low Freq. 100 km – 1 Mm 300 Hz – 3 kHz 1.24 – 124 peV
Very Low frequency 10 – 100 km 3 – 30 kHz 12.4 – 124 peV
Low Frequency 1 – 10 km 30 – 300 kHz 124 peV – 1.24 neV
Medium Frequency 100 m – 1 km 300 kHz – 3 MHz 1.24 neV – 12.4 neV
High Frequency 10 – 100 m 3 – 30 MHz 12.4 neV – 124 neV
Very High Frequency 1 – 10 m 30 – 300 MHz 124 neV – 1.24 μeV
Ultra High Frequency 1 dm – 1 m 300 MHz – 3 GHz 1.24 – 12.4 μeV
Super High Frequency 1 cm – 1 dm 3 – 30 GHz 12.4 – 124 μeV
Extremely High Frequency 1 mm – 1 cm 30 – 300 GHz 124 μeV – 1.24 meV
Infrared 1μm – 1 mm 300 GHz – 300THz 1.24 meV – 1.24 eV
Visible Radiation (Light) 0.7 – 0.4μm 429 THz – 750 THz 1.77 eV – 3.1 eV

The rest are Ultraviolet, X-ray and Gamma rays.

1.12 Radio Waves

Radio waves occupy the lowest part of the electromagnetic spectrum. They were the first to be applied
for Telecommunications (both wired and wireless). Lower frequency radio bands have lower bandwidth
than higher frequency bands. They are generally utilized by antennas of appropriate size (according to
the principle of resonance), with wavelengths ranging from hundreds of meters to about one millimetre.
However, more modern communications systems use high frequency radio bands (short wavelengths)
for fast data services, while lower frequency radio bands (long wavelengths) are limited to TV and radio
broadcasting. TV, mobile phones, wireless networking and amateur radio all use radio waves.
The radio waves are broken down further into:

1.12.1 Extremely and Super Low Frequencies (ELF/SLF)

They form the lowest end of the Radio frequency (wave) and as well have the longest wavelength. The
frequencies range between 3 and 300 Hz. These include ac power line frequencies (50 and 60 Hz) and
also those frequencies in the low end of the human hearing range.

1.12.2 Voice Frequencies or Ultra Low Frequencies (VF/ULF)

Voice frequencies are those in the 300 to 3000 Hz range, and this is the normal range of human speech.
Even though the human hearing extends from approximately 20 Hz to 20 kilohertz (kHz), most
intelligible sound occurs in the voice frequency range.
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1.12.3 Very Low Frequencies (VLF)

VLFs include the higher end of human hearing range up to about 15 to 20 kHz. Many musical
instruments also make sounds in this range as well as in the ELF and the VF ranges. The VLF range is
also used in some government and military communications e.g. in the Navy.

1.12.4 Low Frequencies (LF)

LFs are those in the 30 to 300 kHz range. The primary communication services in this range are those
used in aeronautical and marine navigation. Frequencies in this range are also used as subcarriers.
Subcarriers are signals that carry the baseband modulating information but which, in turn modulate
another higher frequency carrier.

1.12.5 Medium Frequencies (MF)

These are in the 300 to 3000 kHz (3 MHz) range. The major application of frequencies in this range is
AM radio broadcasting (535 to 1605 kHz). Other services in this range include various marine and
aeronautical communications applications.

1.12.6 High Frequencies (HF)

HFs cover the range of 3 to 30 MHz. They are known generally as short waves (SWs). All kinds of two-
way radio communications take place in this band as well as some short wave radio broadcasting. These
include BBC, GBC, VOA, etc.

1.12.7 Very High Frequencies (VHF)

VHFs fall within the range of 30 – 300 MHz, which is an extremely popular one. It is used by many
services, including mobile radio, marine and aeronautical communications, FM radio broadcasting (88
– 108 MHz) and television channels 2 through 13.

1.12.8 Ultra High Frequencies (UHF)

UHFs are those in the 300 – 3000 MHz range. Just like the VHF, the UHF is also a very popular band,
since it is widely used. The range takes care of the UHF television channels 14 through 67, as well as
land mobile communications and cellular telephones. The military services use these frequencies, and
in addition some radar and navigation services occupy this portion of the frequency spectrum.
A part of the UHF fall within the microwave range.

1.13 Microwaves
They are EM radiation which has a smaller wavelength (1mm and 30cm) than radio waves.
Microwaves obtain their name from the fact that they have small wavelengths compared to other radio
waves. They are frequencies above 1 GHz range. SHF and EHF specifically fall fully within this range.
SHFs cover the 3 – 30 GHz range. These are microwave frequencies that are widely used for satellite
communications and radar. The EHFs extend from 30 to 300 GHz. Presently there is only a limited
amount of activity in this range, but it does include satellite and some specialized radar.
The Electromagnetic signals whose frequencies are higher than 300 GHz are given special names such
as Infrared, Visible light, Ultraviolet rays, X-rays, Gamma rays, etc.

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1.14 Infrared Radiation

The IR part of the EM spectrum covers the range from approximately 300 GHz (1 mm) to 300 THz
(1μm). The region is sandwiched between the highest radio frequencies and the visible portion of the
electromagnetic spectrum. Infrared refers to radiation generally associated with heat. Such rays are
emitted by very hot objects and the frequency depends on the temperature of the emitting body. IR is
divided arguably into 3 areas:

i. Far-Infrared: The lower part of this range may also be called microwaves.
ii. Mid-Infrared: Hot objects (black-body radiators) can radiate strongly in this range.
iii. Near-Infrared: Physical processes that are relevant for this range are similar to those for
visible light.

IR signals are used for various special kinds of communications. They are used in astronomy to detect
stars and other physical bodies in the heavens. IR is also used for guidance in weapons systems where
the heat radiated from airplanes or missiles can be picked up by IR detectors and used to guide missiles
towards these targets.
IR is also used in most remote-control units where special coded signals are transmitted by infrared to
receivers for the purpose of changing channels, setting volumes, and other functions.

1.15 Visible Radiation (Light)

This is the range in which the sun and stars similar to it emit most of their radiation. EM radiation with
a wavelength between approximately 400 nm (Violet) and 700 nm (Red) is detected by the human eye
and perceived as visible light. Other wavelengths, especially nearby infrared (longer than 700 nm) and
ultraviolet (shorter than 400 nm) are also sometimes referred to as light. The wavelengths of light are
often expressed in Angstrom (1Å = 10−10 𝑚 ).
Light is widely used for various kinds of communications, including light waves which can be
modulated and transmitted through glass fibres just as electrical signals can be transmitted over wires.
Light signals can also be transmitted through free space.

1.16 Ultraviolet (UV) Rays

This is radiation which is very energetic and can break chemical bonds, making molecules unusually
reactive or ionizing them in general changing their mutual behaviour.
The sun emits a large amount of UV radiation, which could quickly turn Earth into a barren desert.
However, most of it is absorbed by the atmosphere’s ozone layer, before reaching the surface. The rays
can also be produced by the ultraviolet lamps. Generally, UV radiation is dangerous to humans.

1.17 X-Rays
X-rays, also known as Rontgen rays are used for seeing through some things and not others, as well as
for high energy physics and astronomy. They are also useful in medicine and industry because of their
ability to pass through most substances.

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 Communication Systems

There are two types of X-rays: soft and hard x-ray. Hard x-rays have shorter wavelengths than soft x-
rays. X-rays are also dangerous to human health as they can easily penetrate body cells.

1.18 Gamma Rays

These are the most energetic photons having no defined lower limit to their wavelength. They typically
have frequencies above 1019 𝐻𝑧 and therefore have energies above 100 keV. They are very dangerous
to human life since they can easily penetrate the human body and its cells. They are more useful to
astronomers in the study of high-energy objects or regions and also physicists.

1.19 Applications
Table 1.3: Applications of the Frequency Bands

CLASSIFICATION FREQUENCY APPLICATIONS

ELF 3 – 30 Hz Detection of buried metal objects
Communication with submerged submarine,
SLF 30 – 300 Hz
Electric power
VF/ULF 300 Hz – 3 kHz Telephone audio range
Musical Instruments, Long-range navigation,
VLF 3 – 30 kHz
sonar, submarine communication
Long range navigation, Aeronautical and Marine
LF also (LW) 30 – 300 kHz
communication, radio beacon, AM radio
MF 300 kHz – 3 MHz AM broadcast, maritime radio, direction finding
International broadcasting (SW radio, e.g. BBC,
HF 3 – 30 MHz GBC, VOA), long distance aircraft and ship
communication, Telephone, Telegraph, fax
VHF 30 – 300 MHz TV, FM, Police, mobile radio, air traffic control
UHF 300 MHz – 3 GHz TV, Cellular telephony
SHF 3 – 30 GHz Satellites, radars
EHF 30 – 300 GHz Satellites, radars
Used to detect stars and other celestial bodies, in
Infrared 300 GHz – 300THz
weapons guidance systems, remote-control units

Table 1.4: Channel Bandwidth

DESIGNATION FREQUENCY RANGE CHANNELS BANDWIDTH

AM Broadcast 535 – 1605 kHz 106 10 kHz
Citizens Band (Family radio) 26.965 – 27.405 MHz 40 10 kHz
FM Broadcast 87.5 – 108 MHz 100 200 kHz
TV Broadcast 54 – 806 MHz 67 6 MHz

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 Communication Systems

For the TV broadcast, there are two different frequency bands

i. The VHF band, which is between 54 and 216 MHz and
ii. The UHF band, 470 – 806 MHz (channels 14 through 69)

For VHF
The channels are from 2 to 13, where the range 72 – 76 MHz is skipped between channels 4 and 5 and
the range 88 – 174 MHz is skipped between channels 6 and 7.
Digital audio broadcasting (DAB) takes place between 223 – 230 MHz and 1452 – 1472 MHz, whilst
digital TV is currently being installed (470 – 862), reusing some of the old frequencies for analog TV.

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CHAPTER 2

FILTERS, SIGNALS AND NOISE

2.0 Introduction
A filter is a circuit that passes certain frequencies or frequency ranges and attenuates or rejects other
frequencies. In other words, filters are capable of passing input signals with certain selected
frequencies through to the output while rejecting signals with other frequencies. This property is
called selectivity. A filter circuit, therefore, possesses at least one passband – a band of frequencies in
which the output is approximately equal to the input (i.e. attenuation is zero) and an attenuation band
in which the output is zero (i.e. attenuation is infinite).
As frequency selected devices, they are used in radio and TV receivers to allow one to select one
desired signal out of a multitude of broadcast signals in the environment.

2.1 Naming of Filters

Filters may be of any type such as electrical, mechanical, pneumatic, hydraulic, acoustical etc., but the
most commonly used filters are of the electric type. They are variously named by

• What they pass or suppress e.g. low-pass, band-pass, high-pass, band reject (stop) or notch.
• Part of frequency spectrum where they are used e.g. audio, microwave, infrared, UV, x-ray, etc.
• Bandwidths e.g. wideband, narrowband.
• many other ways

2.2 Electrical Filters

They are used in practically all circuits which require separation of signals according to their
frequencies. Almost all communication systems use filters. Applications include (but are not limited
to) noise rejection and signal separation in industrial and measurement circuits, smoothing of digitally
generated analog signals, audio-signal shaping and sound enhancement, channel separation, and signal
enhancement in communication circuits.

Some application of electrical filters

• Selecting one or more desired radio signals from one or more undesired signals and noise
• Obtaining upper- or lower-sidebands from double-sideband (DSB) signals
• carrier-waves or sidebands from amplitude-modulated (AM) signals
• long pulses or short pulses from mixtures of long and short pulses

Electrical filters are divided into two major categories.

i. Passive Filters
ii. Active Filters
Other groups of filters include tuned and wave filters.

2.2.1 Passive filter

A passive filter consists of only passive elements Resistors, R, Inductors, L and Capacitors, C (that is,
RC, RL, LC or RLC). They are also called passive filters because they do not depend on an external
power supply and/or do not contain active components such as transistors. They are most responsive
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to frequencies between around 100 Hz and 300 MHz. The lower frequency limit results from the fact
that at low frequencies the capacitance and inductance values become exceedingly large, meaning
prohibitively large components are needed.

When designing passive filters with very steep attenuation/falloff responses, the number of inductor
and capacitor sections must increase.

Advantages
i. no, amplifying elements (transistors, op-amps, etc)
ii. no signal gain
iii. 1st order - design is simple (just use standard equations to find resonant frequency of the
circuit)
iv. 2nd order - complex equations
v. require no power supplies
vi. not restricted by the bandwidth limitations of the op-amps
vii. can be used at very high frequencies
viii. can handle larger current or voltage levels than active devices

Disadvantages
i. high accuracy (1% or 2%), small physical size, or large inductance values are required
ii. standard values of inductors are not very closely spaced
iii. difficult to find an off-the-shelf inductor within 10 percent of any arbitrary value
iv. adjustable inductors are used
v. tuning such inductors to the required values is time-consuming and expensive for larger
quantities of filters
vi. inductors are often prohibitively expensive

2.2.2 Active Filter

A filter is said to be an active filter if it consists of active elements such as transistors and op-amps,
combined with passive elements. Active filters are capable of handling very low frequency signals
(approaching 0 Hz), and they can provide voltage gain if needed (unlike passive filters). Active filters
can be designed to offer comparable performance to LC filters, and they are typically easier to make,
less finicky, and can be designed without the need for large-sized components
One major drawback with active filters is a relatively limited high-frequency range. Above 100 kHz or
so, active filters can become unreliable (a result of the op amp’s bandwidth and slew-rate
requirements). At radio frequencies, it is best to use a passive filter.
• no inductors
• made up of op-amps, resistors and capacitors
• provides virtually any arbitrary gain
• generally easier to design
• high input impedance prevents excessive loading of the driving source
• low output impedance prevents the filter from being affected by the load
• at high frequencies is limited by the gain-bandwidth of the op-amps
• easy to adjust over a wide frequency range without altering the desired response

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 Communication Systems

Advantages of Active Filters over Passive Filters

i. Active filters can be designed to provide amplification (or gain) in addition to filtering.
ii. There is no loading problem, because of high input resistance and low output resistance of the
op-amp.
iii. Active filters are cost effective as a wide variety of economical op-amps are available.
iv. No inductors are required – these are often large components that need to be designed separately.

2.2.3 Tuned Filters

Tuned filters are widely used to separate signals in relatively-narrow bands of frequencies from
signals in wider spectrums.

Tuned filters utilize one or more tuned circuits (resonant or anti-resonant circuits) to separate signals
in relatively-narrow bands of frequencies from signals wider frequency spectrums. Most transmitters
and receivers incorporate many tuned filters. External tuned filters also are sometimes used as pre-
selectors to greatly attenuate strong undesired signals received by antennas before they reach
receivers, and between transmitters and antennas to greatly attenuate harmonics and other emissions
that could cause interference.

2.3 Some Terminologies Associated with Filters

2.3.1 –3 dB Frequency (f3dB).

This represents the input frequency that causes the output signal to drop to –3 dB relative to the input
signal. The –3 dB frequency is equivalent to the cutoff frequency; the point where the input-to-output
1
power is reduced by one-half (50%) or the point where the input-to-output voltage is reduced by
√2
(70.7%). For low-pass and high-pass filters, there is only one –3-dB frequency. However, for band-pass
and notch filters, there are two –3-dB frequencies, typically referred to as f1 and f2.

0 𝑑𝐵
−3 𝑑𝐵
{

Passband Stopband

𝑓𝑐 𝑓
Figure 2.1: A Filter Depicting – 3-dB Frequency, Passband and Cut-off Frequency

2.3.2 Passband
The pass band of a filter is the region of frequencies that are allowed to pass through the filter with
minimum attenuation, usually defined as less than – 3 decibels (dB) of attenuation. Ideally, the
attenuation is zero for the band of frequencies. Every filter has to have at least one pass band and at
least one attenuation band as could be seen in Figure 2.1. For the attenuation band, ideally the
attenuation is infinite for the band of frequencies.

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2.3.3 Critical Frequency, 𝑓𝑐

This is often called the cut-off frequency, 𝑓𝑐 defines the end of the passband and is normally specified
at the point where the response drops – 3 dB from the passband response.

2.3.4 Stop-band frequency (fs).

This is a specific frequency where the attenuation reaches a specified value set by the designer. For
lowpass and high-pass filters, the frequencies beyond the stop-band frequency are referred to as the
stopband. For band-pass and notch filters, there are two stop-band frequencies, and the frequencies
between the stop-bands are also collectively called the stop-band frequencies.

2.3.5 Bandwidth
The bandwidth of a filter is a measure of its passband and is defined as the difference between the
upper and lower 3-dB cut-off frequencies of the passband.
𝐵𝑤 = 𝑓𝑐2 − 𝑓𝑐1

2.3.6 Roll-off Rate

This is the rate at which a gain drops, e.g., - 20 dB/decade. This means that at a frequency of 10𝑓𝑐, the
output will be – 20 dB (10%) of the input.

2.3.7 Decade
To describe the rate of rise or fall in attenuation as the frequency changes, description such as “20
dB/decade” is used.
𝜔
|𝐻(𝑗𝜔)| ≃ −20 log10
𝜔𝑜
𝜔 10
A decade is a ten times change in frequency i.e. 𝜔 =
𝑜 1
⟹ |𝐻(𝑗𝜔)| ≃ −20 𝑑𝐵
20 dB/decade means that, if the frequency changes by a factor of 10, the attenuation increases or
decreases by 20 dB.
1
It must be noted that when |𝐻(𝑗𝜔)| ≃ −20 log10 , the half-power condition, then |𝐻(𝑗𝜔)| ≃ −3𝑑𝐵
√2
This is called the –3-dB point. This point corresponds to the half-power point or cut-off frequency for
a filter.

2.3.8 Pole and Order

A pole is nothing more than an RC circuit as could be observed in Figure 2.2.
+V
R1
+
v in
C1 v out
- R f1
-V

R f2
Figure 2.2: An RC - Circuit
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An n - pole filter contains n – RC circuits, typically between 1 and 6 for most applications.
Order refers to the number of poles. That is,
1st order – 1 pole (single pole) 2nd order – 2 poles nth order – n poles
NB: The number of poles determines the roll-off rate of the filter. For example, a Butterworth response
produces -20 dB/decade/pole. So a first order (one-pole) filter has a roll-off - 20 dB/decade, a second order
(two-pole) filter – 40 dB/decade.

2.3.9 Critical Frequency and Roll-off Rate

The critical frequency is determined by the values of the resistors and capacitors in the RC circuit. For
1
a single – pole (first –order) filter, the critical frequency 𝑓𝑐 is 𝑓𝑐 =
2𝜋𝑅𝐶

2.4 Types of Filters

Ideally, a filter would completely eliminate signals above a cut-off frequency and perfectly pass signals
below it (in the pass-band). There is a wide range of filter circuits, each with its own set of advantages
and disadvantages. All filters introduce phase shift, and (almost all) filters change the frequency
response. In real filters, various trade-offs are made in an attempt to approximate the ideal. Some filter
types are optimized for gain flatness in the pass-band (Butterworth), some trade off gain variation
(ripple) in the pass-band for steeper roll-off (Chebyshev), and still others trade off both flatness and rate
of roll-off in favour of pulse-response fidelity (Bessel).
When describing how a filter behaves, a response curve is used, which is simply the attenuation or
gain (Vout/Vin) versus frequency graph (refer figure 2.3).
𝐺𝑎𝑖𝑛

0 𝑑𝐵

𝑓𝑐
Figure 2.3: An Ideal Low-Pass (Brick Wall) Filter

Figure 2.4: Action of Filters on Complex Signal

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The types of filters to be discussed briefly in this chapter are:

a) low-pass
b) high-pass
c) band-pass
d) notch, or band-reject or band-stop

2.4.1 Low Pass Filter

This passes low frequencies and stops (attenuates) high frequencies. The low frequencies are from dc
(0 Hz) up to the upper cut-off frequency (𝑓𝐻). Thus frequencies higher than the cut-off frequency are
attenuated. A low-pass filter is sometimes called a high-cut filter, or treble cut filter when used in
audio applications.
Av(dB)

-3dB {

f
f2

Figure 2.5: Actual response of Low-pass Filter

The response drops gradually to zero at frequencies beyond the passband as depicted in figure 2.5.
This ideal response is sometimes referred to as “brick wall” because nothing gets through beyond the
wall. In the case of a low pass-pass filter the lower critical frequency is 0 Hz, so the bandwidth is
equal to 𝑓𝐻.
1
Bandwidth 𝐵𝑊 = 𝑓𝐻 ; 𝑓𝐻 occurs when 𝑋𝑐 = 𝑅 where 𝑓𝐻 = 2𝜋𝑅𝐶
+V
R1
+
v in
C1 v out
- R f1
-V

R f2
Figure 2.6: Single-Pole Active Low Pass Filter

The most basic low-pass filter is a simple RC circuit consisting of just one resistor and one capacitor.
This basic RC filter as observed in Figure 2.6 has a single pole and it decreases at - 20 dB/decade
beyond the critical frequency.
Figure 2.7 depicts a two-pole low pass filter.
C2

+V
R2 R1
+
vin
C1 vout
- Rf1
-V

Rf2
Figure 2.7: Two-Pole Low Pass Filter
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2.4.2 High - Pass Filter

It is the filter that significantly attenuates or rejects all frequencies below the cut-off frequency, 𝑓𝐿 and
passes all frequencies above it. By expansion the 𝑓𝐿 is the frequency at which the output is 70.7% of
the input (or – 3 dB). It is sometimes called a low-cut filter; the terms bass-cut filter or rumble filter
are also used in audio applications.
A v (dB )

-
3dB {
Stopband Passband

𝑓𝐿 f
f 𝑓𝐿
( a ) Ideal filter ( b ) Practical filter
Figure 2 .8: High- Pass Filter Response

Ideally, the passband of a high-pass filter is all frequencies above 𝑓𝐿. The high-frequency response of
practical circuits is limited by the op-amp or other components that make up the filter.
A simple RC circuit consisting of a single resistor and capacitor can be configured as a high-pass filter
by taking the output across the resistor (figure 2.9(a)). As in the case of the low-pass filter, the basic
RC circuit has a roll-off rate of – 20 dB/decade. Also the critical frequency for the basic high-pass
1
filter occurs when 𝑋𝑐 = 𝑅 where 𝑓𝐿 = 2𝜋𝑅𝐶
+V
C1 +V
+ C C
v in R1 . 2 1
+
v out v in
- R f1 R 1 v out
- R f1
-V
-V
R f2
R f2

(a) Single-pole (b) Two-pole

Figure 2.9: Active High Pass Filters

2.4.3 Band - Pass Filter

A v(dB)

-3 dB {

f
f1 f2
fo
Figure 2.10: Frequency Response of Band-pass filter response

A band-pass filter passes all signals lying within a band between a lower-frequency limit and an
upper frequency limit and essentially rejects all other frequencies that are outside this specific band. A
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 Communication Systems

generalized band-pass response curve is shown in Figure 2.10 whilst Figure 2.11 depicts a two-stage
band-pass filter.

Figure 2.11: Two-stage Band - Pass Filter

The type of filter configuration shown in figure 2.12, is a multiple-feedback band-pass filter. This
filter has an advantage over the cascaded band-pass filter in figure 2.11 because it is simpler and can
achieve narrower bandwidths.
C2
Rf
+V
R1 C1
-
v in R2
v out
+
-V

Figure 2.12: Multiple-Feedback Band - Pass Filter

There are two feedback paths in this configuration, one through 𝐶2 and the other through 𝑅𝑓.

Centre frequency (f0)

On a standard log plot, band-pass filters are geometrically symmetrical around the filter’s resonant
frequency or center frequency, provided the response is plotted on linear-log graph paper (the
logarithmic axis representing the frequency). On standard-log paper, the centre frequency, 𝑓𝑜 is related
to the –3-dB frequencies by the following expression:
𝑓𝑜 = √𝑓1 × 𝑓2 i.e. the geometric mean of the cut-off frequencies.
For narrow-band band-pass filters, where the ratio of f2 to f1 is less than 1.1, the response shape
approaches arithmetic symmetry. Here, we can approximate f0 by taking the average of –3-dB
𝑓1 +𝑓2
frequencies: 𝑓𝑜 = 2

Quality Factor (Q)

There are basically two types of band-pass filters, that is, wide band-pass and narrow band-pass filters.
Unfortunately, there is no set dividing line between the two. However, a band-pass filter is defined as
a wide band-pass if its figure of merit or quality, Q is less than 10 while the band-pass filters with 𝑄 >
10 are called the narrow band-pass filters.
The quality factor (Q) of a band-pass filter is the ratio of the centre frequency to the bandwidth.
𝑓𝑜
𝑄= 𝐵
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 Communication Systems

The value of Q is an indication of the selectivity of a band-pass filter. The higher the value of Q, the
narrower the bandwidth and the better the selectivity for a given value of 𝑓𝑜.

2.4.4 Band - Stop Filter

This is also known as the notch, band-reject, or band-elimination filter. The band-stop filter attenuates
a single band of frequencies and allows those on either side to pass through. The bandwidth is the
band of frequencies between the 3 – dB points just as in the case of the band-pass filter response.
Thus, the band-stop filter can be thought of as the opposite to that of the band-pass filter since the
frequencies within a certain bandwidth are rejected, and frequencies outside the bandwidth are passed.
The notch filter is designed to block all frequencies that fall within its bandwidth. The circuit is made
up of a high pass filter, a low-pass filter and a summing amplifier. The summing amplifier will have
an output that is equal to the sum of the filter output voltages.
Av(dB)

-
3dB {

f
f1 f2

Figure 2.13 Frequency Response of Band-stop Filter

A general response curve for the band-stop filter is shown in Figure 2.13.

2.5 Filter Response Characteristics

Each type of filter (low-pass, high-pass, band-pass or band-stop) can be tailored by circuit component
values to have either, a Butterworth, Chebyshev, or Bessel characteristic. Each of these characteristics
is identified by the shape of the response curve, and each has an advantage in certain applications. In
this section, you will learn the three basic filter response characteristics and other filter parameters.
Av
Chebyshev

Bessel
Butterworth
Bessel Butterworth
Chebyshev

f
Figure 2.14: Comparative Plots of Three Types of Filter Response
characteristics
Butterworth, Chebyshev, or Bessel response characteristics can be realized with most active filter
circuit configurations by proper selection of certain component values. A general comparison of
the three response characteristics for a low-pass filter response curve is shown in figure 2.14. The
other filters can also be designed to have any one of the characteristics.
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 Communication Systems

2.5.1 The Butterworth Characteristic

The Butterworth characteristic provides a very flat amplitude response in the passband and a roll-off
rate of – 20 dB/decade/pole. The phase response is not linear, however, and the phase shift (thus, time
delay) of signals passing through the filter varies non-linearly with frequency.
Av

Figure 2.15: Frequency Response of the Butterworth characteristic

Filters with the Butterworth response are normally used when all frequencies in the passband must
have the same gain. The Butterworth response is often referred to as a maximally flat response.

2.5.2 The Chebyshev Characteristic

Filters with the Chebyshev response characteristic are useful when a rapid roll-off is required because
it provides a roll-off rate greater than – 20 dB/decade/pole.
This is a greater rate than that of the Butterworth, so filters can be implemented with the Chebyshev
response with fewer poles and less complex circuitry for a given roll-off rate.
Av

Figure 2.16: Frequency response of the Chebyshev characteristic

This type of filter response is characterized by overshoot or ripples in the passband (depending on the
number of poles) and an even less linear phase response than the Butterworth.

2.5.3 The Bessel Characteristic

Av

Figure 2.17: Frequency response of the Bessel characteristic

The Bessel response exhibits a linear phase characteristic, meaning that the phase shift increases
linearly with frequency. The result is almost no overshoot on the output with a pulse input. For this
reason, filters with the Bessel response are used for filtering pulse waveforms without distorting the
shape of the waveform.

In summary
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 Communication Systems

For Bessel
• Flat response in the passband.
• Role-off rate less than 20dB/decade/pole.
• Phase response is linear.
• Used for filtering pulse waveforms without distorting the shape of the waveform.

For Butterworth
• Very flat amplitude, Av(dB), response in the passband.
• Role-off rate is 20dB/decade/pole.
• Phase response is not linear.
• Used when all frequencies in the passband must have the same gain.
Often referred to as a maximally flat response

For Chebyshev
• Overshoot or ripples in the passband.
• Role-off rate greater than 20dB/decade/pole.
• Phase response is not linear - worse than Butterworth.
• Used when a rapid roll-off is required.

2.6 Signal
A signal may be defined as the single valued function of time. Time plays the role of an independent
variable. This means that at every instant of time, the signal has a unique value. Thus, a signal is a
function of one or more independent variables which contains some information.
In electrical sense, the signal can be voltage or current. The voltage or current is the function of time
as an independent variable.

2.7 Classification of Signals

The signals could be classified into different forms.

2.7.1 Dimension
Signal could be classified under dimension as observed in Figure 2.18.
Signals

One Dimensional Two Dimensional Three Dimensional Four Dimensional

Signals Signals Signals Signals

Speech
Pictures Video data Volume data
Music
Computer data over time
Figure 2.18: Classification of Signals

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 Communication Systems

2.7.2 Continuous-time and Discrete-time Signals

A signal 𝑥(𝑡) is a continuous-time signal if 𝑡 is a continuous variable. This means that a continuous
time signal is defined continuously in the time-domain. On the other hand, if time 𝑡, is a discrete
variable, i.e. 𝑥(𝑡) is defined at discrete times (𝑡 = 𝑛𝑇), then 𝑥(𝑡) is a discrete time-signal. Since a
discrete-time signal is defined at discrete times, it is often identified as a sequence of numbers and is
denoted by
𝑥(𝑛), where 𝑛 = integer.

Figures 2.19 and 2.20 show respectively the graphs of continuous-time and discrete-time signal.
Audio and video recordings are examples of continuous-time signals whilst monthly sales of a
company is an example of discrete-time signal.

-2 1 5
0 t -5 -4 -3 -1 0 2 3 4 n

Figure 2.19 Continuous - Time Signal Figure 2.20 Discrete - Time Signal

2.7.3 Periodic and Aperiodic Signals

A signal is said to be periodic if it has a definite pattern and repeats over and over with a repetition
period of T. It thus exhibits periodicity i.e.
𝑥(𝑡 + 𝑇) = 𝑥(𝑡), −∞ < 𝑡 < ∞ (2.1)
Where, T is the period of the signal.
The smallest value of T which satisfies equation (2.1) is termed as the fundamental period, 𝑇𝑜. Figure
2.21 is a periodic signal.
𝑥 (𝑡)

0 t

Figure 2.21 Periodic Signal

An aperiodic signal on the other hand, does not repeat. An aperiodic signal is supposed to have a
period equal to infinity. For example, considering equation (2.2):
𝑥(𝑡) = 𝑒−𝑎𝑡 (2.2)
Thus for 𝑥(𝑡 + 𝑇𝑜) = 𝑒−𝑎(𝑡+𝑇𝑜) = 𝑒−𝑎(𝑡+∞)
= 𝑒−𝑎𝑡 × 𝑒−∞)
= 𝑒−𝑎𝑡 × 0 ⟹ 𝑥(𝑡 + 𝑇0) = 0
Thus 𝑥(𝑡 + 𝑇0) ≠ 𝑥(𝑡)

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This indicates that the signal having period, 𝑇𝑜 = ∞ is an aperiodic signal. Figure 2.22 depicts
aperiodic signal.
𝑥 (𝑡 )
1 .00
𝑥 ( 𝑡 ) = 𝑒 −𝑎𝑡

0 t
Figure 2.22: Aperiodic Signal

2.7.4 Real and Complex Signals

A signal 𝑥(𝑡) is a real signal if its value is a real number. Similarly, a signal 𝑥(𝑡) is a complex signal if
its value is a complex number.

2.7.5 Deterministic and Random Signals

Deterministic signals are signals which can be completely specified in time. The signal has a regular
pattern and can be characterised mathematically. Its nature and amplitude at any time can be
predicted.
An example is a linear or ramp signal, i.e.
𝑥(𝑡) = 𝑎𝑡 (2.3)
Its amplitude increases linearly with time and the slope is a.

𝑥(𝑡) = 𝐴 sin 𝜔𝑡 (2.4)

For this signal, the amplitude varies sinusoidally with time and its maximum amplitude is A.
A random signal is one whose occurrence is always random in nature. The pattern of such a signal is
quite irregular. Random signals are also called non-deterministic signals. An example of a random
signal is thermal noise, generated in an electric circuit. Such a noise signal has probabilistic behaviour.

2.7.6 Even and Odd Signals

An even signal is that type of signal which exhibits symmetry in the time domain. This type of signal
is identical about the origin. Mathematically, an even signal must satisfy the following condition:
𝑥(𝑡) = 𝑥(−𝑡) (2.5)

Figure 2.23: Even Signal Figure 2.24: Odd Signal

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On the other hand, an odd signal is that type of signal which exhibits anti-symmetry. It is not identical
about the origin, but rather identical to its negative. Mathematically, an odd signal must satisfy the
condition:
𝑥(𝑡) = −𝑥(𝑡) (2.6)

2.8 Singularity Functions

These are functions which play very important roles in communications systems. These singularity
functions serve as foundations for the construction of more complex signals and may be used to model
a large number of physical signals which occur in nature. These singularity functions are:
i. Unit step function
ii. Unit impulse function
iii. Ramp function

2.8.1 Unit Step Function

It is a type of singularity function which exists only for positive side and is zero for negative side. The
unit step function is discontinuous at 𝑡 = 0. The unit step function, denoted by 𝑢(𝑡), is defined
mathematically by
0, 𝑡 < 0
𝑢(𝑡) = { (2.7)
1, 𝑡 ≥ 0

Here unit step means that the amplitude of u(t) is equal to 1 for t ≥ 0.

NB: We are following the convention that u(0) = 1. From a strict mathematical standpoint, u(t) is not
defined at t = 0. Nevertheless, we usually take u(0) = 1. Figure 2.25 shows the unit-step function (𝑡).
𝑢(𝑡)

0 1 3 4 𝑡
2
Figure 2.25: Unit-Step Function

Any signal (voltage or current) that is switched on or off at some instant of time, to, is easily described
mathematically by using the unit step function.

2.8.2 Unit Impulse Function

The unit-impulse function, (t), also called the delta function or the Dirac distribution, is defined by

 (t ) = 0, t0

   ( )d ( ) = 1,
−
for any real number   0

The first condition states that (t) is zero for all nonzero values of t, while the second condition states
that the area under the impulse is 1, so (t) has unit area. It is important to point out that the value (0)

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 Communication Systems

of (t) at t = 0 is not defined; in particular, (0) is not equal to infinity. For any real number K, K(t) is
the impulse with area K. It is defined by

K (t ) = 0, t0

  K ( )d ( ) = K,
−
for any real number   0

The graphical representation of K(t) is shown in Figure 2.26. The notation K in the figure refers to the
area of the impulse K(t).

Figure 2.26: Unit-Impulse Function

2.9 Noise
Noise is defined as an unwanted signal/wave that corrupts and distorts the desired signal (s). It can
cause the original signal to be distorted in shape, increased or decreased in amplitude, delayed slightly
in time, or otherwise corrupted and modified. Noise tends to disturb the transmission and processing
of signals in communication systems and over which one has incomplete control. The corruption can
take many forms. Noise is always present in electronic systems, and its effects tend to be devastating
to the performance of the system. It is generated by random electron motion in all electronic devices
and components whether passive or active. The higher operating temperatures, the greater the amount
of noise produced.

2.9 Categories of Noise

There are two (2) broad categories of noise. These are: external and internal noise

2.9.1 External Noise

This is a type that is generated outside a receiver. It is termed as external to the system. If it is present
in a received radio signal then it was introduced in the transmitting medium. They include:

i. Man-made
ii. Atmospheric and
iii. Space, solar and cosmic noise

(i) Man-made: It is any form of electromagnetic interference that can be traced to non-natural
causes. They are the most troublesome form of external noise. They occur randomly at
frequencies up to approximately 500 MHz. They are usually produced by mechanisms such as
engine ignition systems, fluorescent lights and power lines that supply energy for most electronic
systems.
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 Communication Systems

(ii) Atmospheric noise: As the name implies it is caused by naturally occurring disturbances in
earth’s atmosphere, and manifests itself in what is commonly termed as static. The most
prominent contributor is the lightening discharges. This type of noise is more worrying at lower
frequencies, hence it is not significant at frequencies exceeding about 20 MHz.

(iii) Space, solar and cosmic noise.

2.9.2 Internal Noise

This is introduced by the receiver itself. This type of noise can be divided into two broad categories:

(i) Thermal noise (ii) Shot noise

Thermal noise: it is caused by thermal agitation of electrons in a conductor. It is generated by the rapid
and random motion of the atoms and electrons of which any resistance is constructed. The power of
this random noise is related directly to the bandwidth. The power is also related directly to the
absolute temperature of the resistance.

Thus noise power, 𝑃𝑛, is given by

𝑃𝑛 = 𝑘𝑇∆𝑓 (2.7)
−23
Where 𝑘 =Boltzmann’a constant proportionality constant = 1.38 × 10 𝐽/𝐾
𝑇 = absolute resistance temperature in degrees Kelvin (K)
∆𝑓 = bandwidth of frequencies to be measured (in Hz)
Thus mean square value of thermal noise voltage, 𝑉𝑇𝑁,
𝑉𝑇𝑁2 = 4𝑘𝑇∆𝑓𝑅

Or rms noise voltage, 𝑉𝑇𝑁 is

Shot Noise: This is caused by random fluctuations in the motion of charge carriers in a conductor. It is
present in all amplifying devices.

Mean square value of shot noise current, 𝐼𝑁2, is given by

𝐼𝑁2 = 2𝑞𝑒𝐼𝐵

Where 𝑞𝑒 = electronic charge

𝐼 = average current
∆𝑓 = bandwidth (in Hz)

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CHAPTER THREE
AMPLITUDE MODULATION
3.0 INTRODUCTION
In Amplitude Modulation (AM), the information signal varies the amplitude of the carrier sine wave. In
other words, the instantaneous value of the carrier amplitude changes in accordance with the amplitude
and frequency variations of the modulating signal.
In AM, the carrier frequency remains constant during the modulation process but its amplitude varies in
accordance with the modulating signal.

3.1 AM Waveforms
Ac

Vc

Am
Vm

(a) (b) (c)

Figure 3.1: Three Different AM waveforms

All the diagrams depict the AM waveform when the Intelligence or modulating signal is subjected to
different conditions.

For diagram 3.1a: The AM waveform obtained is a signal at the carrier frequency whose amplitude is
changing at the same rate as the intelligence frequency. The AM waveform attains maximum
amplitude when the modulating amplitude reaches a maximum positive value.
On the other hand, as the modulating (intelligence/information) amplitude reaches a maximum
negative value, the AM waveform attains minimum value.

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For diagram 3.1b: The modulating amplitude is increased while letting its frequency remains the
same. The resultant AM waveform ends up obtaining a larger maximum value and relatively smaller
minimum value.

For diagram 3.1c: The modulating frequency is increased whilst its corresponding amplitude is
decreased. The resultant AM waveform has reduced maximums and minimums, and its frequency
matches that of the modulating frequency.

3.2 Derivation of AM Equation

Let 𝑓𝑚 = modulating frequency, 𝐴𝑚 = modulating Amplitude, 𝑓𝑐 = carrier frequency
𝐴𝑐 = carrier (peak) amplitude
Now,
The modulating signal can be written as
𝑉𝑚 (𝑡) = 𝐴𝑚 sin 𝜔𝑚 𝑡 (at phase zero) (3.1)

Considering the obtained AM waveform (modulated AM signal)

If A is the amplitude of the AM waveform then its equation can be given as the sum of the carrier peak
amplitude and modulating signal.
i.e. 𝐴 = 𝐴𝑐 + 𝑉𝑚 (𝑡) (3.2)
hence from (3.1) 𝐴 = 𝐴𝑐 + 𝐴𝑚 sin 𝜔𝑚 𝑡 (3.3)
But
The AM signal is given by
𝑉(𝑡) = 𝐴 sin 𝜔𝑐 𝑡 (3.4)
Thus
𝑣(𝑡) = [𝐴𝑐 + 𝐴𝑚 sin 𝜔𝑚 𝑡] sin 𝜔𝑐 𝑡 (3.5)
𝐴𝑚
𝑣(𝑡) = 𝐴𝑐 [1 + sin 𝜔𝑚 𝑡] sin 𝜔𝑐 𝑡
𝐴𝑐

𝑉(𝑡) = 𝐴𝑐 [1 + 𝑚 sin 𝜔𝑚 𝑡] sin 𝜔𝑐 𝑡 (3.6)

Where, m which is called modulation index is given by
𝐴𝑚
𝑚 = , (3.7)
𝐴𝑐

From (3.7)
𝑉(𝑡) = 𝐴𝑐 sin 𝜔𝑐 𝑡 + 𝑚𝐴𝑐 sin 𝜔𝑚 𝑡 sin 𝜔𝑐 𝑡 (3.8)

Now using the Trigonometry identity

1
sin 𝐴 sin 𝐵 = 2 [cos(𝐴 − 𝐵) − cos(𝐴 + 𝐵)] (3.9)

Thus
𝑚𝐴𝑐
𝑉(𝑡) = 𝐴𝑐 sin 𝜔𝑐 𝑡 + [cos(𝜔𝑐 − 𝜔𝑚 )𝑡 − cos(𝜔𝑐 + 𝜔𝑚 )𝑡]
2
𝑚𝐴𝑐 𝑚𝐴𝑐
𝑉(𝑡) = 𝐴𝑐 sin 𝜔𝑐 𝑡 + cos(𝜔𝑐 − 𝜔𝑚 )𝑡 − cos(𝜔𝑐 + 𝜔𝑚 )𝑡 (3.10)
2 2
𝑚𝐴𝑐 𝑚𝐴𝑐
𝑉(𝑡) = 𝐴𝑐 sin 𝜔𝑐 𝑡 + cos 2π(𝑓𝑐 − 𝑓𝑚 )𝑡 − cos 2𝜋(𝑓𝑐 + 𝑓𝑚 )𝑡 (3.11)
2 2

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3.3 Frequency Spectrum, Sidebands and Bandwidth

The sidebands are produced when a carrier is modulated in any manner for conveying intelligence
(information) of any sort. The sidebands, but sometimes called side frequencies, have signals that are
symmetrical about the carrier frequency, occurring above and below it.
The signals are called side frequencies if the modulating signal is a single-frequency sine wave.
On the other hand, if the modulating signal contains multiple frequencies such as voice, video or digital
signals, the result is a range of multiple side frequencies, and they are referred to as sidebands.
The sidebands occur at frequencies that are the sum and difference of the carrier and modulating
frequencies. Thus from equation (3.11) with carrier and modulating frequencies of 𝑓𝑐 and 𝑓𝑚
respectively, the upper sideband, 𝑓𝑈𝑆𝐵 is given by: 𝑓𝑈𝑆𝐵 = 𝑓𝑐 + 𝑓𝑚
whilst that of the lower sideband 𝑓𝐿𝑆𝐵 is given by: 𝑓𝐿𝑆𝐵 = 𝑓𝑐 − 𝑓𝑚

From equation 3.11, for a single tone signal, the amplitude spectrum of the modulated carrier consists
of three components:
i. The carrier frequency 𝑓𝑐 with relative amplitude of 1
𝑚
ii. the lower frequency 𝑓𝑐 − 𝑓𝑚 with relative amplitude of 2
𝑚
iii. the upper frequency𝑓𝑐 + 𝑓𝑚 , also with relative amplitude of 2

Figure 3.2 depicts the frequency spectrum of the AM signal.

1
𝑚 𝑚
2 2

fc - fm fc fc + fm
ωc – ωm ωc ωc + ωm
Figure 3.2: Frequency Spectrum of a single tone AM

NB: The higher the frequency components of the modulating signal, the farther the sidebands
will appear from the carrier.

3.3.1 Bandwidth, 𝐵𝑤
Bw = Upper frequency – Lower frequency
𝐵𝑤 = (𝑓𝑐 + 𝑓𝑚 ) − (𝑓𝑐 − 𝑓𝑚 ) = 2𝑓𝑚 (3.12)
Hence the bandwidth of an AM signal is twice its modulating (intelligence) frequency.

3.4 Modulation Index (Depth/Factor/Coefficient) – m

This is the depth of modulation and is defined as,
modulating signal amplitude Am
𝑚= = (3.13)
unmodulated carrier amplitude Ac

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In order for proper AM to occur, the amplitude of the modulating signal, Am should be less than the
amplitude of the carrier, Ac. Thus the modulation index should be a number between 0 and 1. This is
to avoid distortion. The ideal condition for AM is where Am = Ac or m = 1.
Example 3.1
A carrier wave of frequency 20MHz and peak value 20V is amplitude modulated by a 5 kHz sine
wave of amplitude 10V. Determine the modulation index, and draw the spectrum.

Solution 3.1
Carrier frequency, fc = 20 MHz carrier amplitude, Ac = 20 V
Modulating frequency, fm = 5 kHz modulating amplitude, Am = 10V
𝐴𝑚 10
Modulation index, 𝑚 = = = 0.5
𝐴𝑐 20

Side frequencies
𝑓𝑐 − 𝑓𝑚 = (20,000– 5) = 19,995 kHz; 𝑓𝑐 + 𝑓𝑚 = (20,000 + 5) = 20,005 kHz

Side frequency amplitude

1 1
= 𝑚𝐴𝑐 = 2 × 0.5 × 20 = 5𝑉
2

Amplitude

20V

5V 5V

19,995 20,000 20,005 kHz

Figure 3.3: Resulting Frequency Spectrum for Example 3.1

3.5 Modulation Envelope

An amplitude modulated waveform has an envelope which is a signal at the carrier frequency whose
amplitude is changing at the same rate as the modulating frequency. The waveform attains maximum
amplitude when the modulating amplitude reaches a maximum value. On the other hand, as the
modulating amplitude reaches a maximum negative value, the AM waveform reaches a minimum
value.
V(t)

t m = 60%

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 Communication Systems

V(t)

m = 80%

V(t)

m = 100%
t
This is the ideal condition for
AM, where Am = Ac. This will
produce the greatest output at
the receiver with no distortion.

Figure 3.4: Modulation at different modulation indices

3.5.1 Overmodulation
If the amplitude of the modulating signal is greater than the amplitude of the carrier, m will be greater
than 1. This will cause severe distortion of the modulated waveform. This is illustrated in figure 3.5
below.
Distortion causes incorrect information to be transmitted. This condition is called overmodulation,
where the waveform is flattened near the zero line. In the case of overmodulation, the information
cannot be retrieved.

V(t)

Figure 3.5: Distortion caused by overmodulation

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3.5.2 Determining Modulation Index by Measurements of Composite Modulated Wave

As was seen from section 3.4, the modulation index can be determined by measuring the actual values
of the modulating and the carrier voltages and determining the ratio using equation 3.13.
Another way of determining the modulation index is by taking measurements of the composite
modulated wave itself.

Ac Am
Vmax
Am
Vmin

Figure 3.6: An AM wave showing Vmax and Vmin

Practically, whenever the AM signal is displayed on an oscilloscope, the modulation index can be
computed from Vmax and Vmin as could be seen in figure 3.6 above.
From figure 3.6, 𝑉𝑚𝑎𝑥 = 2𝐴𝑚 + 𝑉𝑚𝑖𝑛 ,
𝑉𝑚𝑎𝑥 −𝑉𝑚𝑖𝑛
⟹ 𝐴𝑚 = (3.14)
2
Again, 𝑉𝑚𝑎𝑥 = 𝐴𝑐 + 𝐴𝑚
Thus, 𝐴𝑐 = 𝑉𝑚𝑎𝑥 − 𝐴𝑚 (3.15)

Substituting equation (3.15) into (3.16)

Then
𝑉𝑚𝑎𝑥 − 𝑉𝑚𝑖𝑛
𝐴𝑐 = 𝑉𝑚𝑎𝑥 −
2
𝑉𝑚𝑎𝑥 + 𝑉𝑚𝑖𝑛
⟹ 𝐴𝑐 = (3.16)
2
𝐴𝑚
But 𝑚 =
𝐴𝑐
𝐴𝑚 𝑉𝑚𝑎𝑥 − 𝑉𝑚𝑖𝑛
Hence, 𝑚= =
𝐴𝑐 𝑉𝑚𝑎𝑥 + 𝑉𝑚𝑖𝑛

Thus, the modulation index, m produces the result:

𝑉𝑚𝑎𝑥 − 𝑉𝑚𝑖𝑛
𝑚= (3.17)
𝑉𝑚𝑎𝑥 + 𝑉𝑚𝑖𝑛

The values for Vmax and Vmin can be read directly from an oscilloscope screen and inserted into the
formula to determine m.

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3.6 AM Signal Power Consideration

The total power transmitted in a DSB-AM signal is the sum of the carrier power and the power in the
sidebands. Now, power in a sinusoidal signal is proportional to the square of its amplitude.
Thus total power transmitted, Pt
𝑚𝐴𝑐 2 𝑚𝐴𝑐 2
𝑃𝑡 = [𝐴𝑐 2 + ( ) +( ) ] (3.18)
2 2

𝑚2 𝐴𝑐 2 𝑚2 𝐴𝑐 2
= 𝐴𝑐 2 + +
4 4
𝑚2 𝐴𝑐 2
= 𝐴𝑐 2 +
2
𝑚2
𝑃𝑡 = 𝐴𝑐 2 [1 + ] (3.19)
2

In terms of power
𝑚2 2+𝑚2
𝑃𝑡 = 𝑃𝑐 [1 + ] = 𝑃𝑐 [ ] (3.20)
2 2
2
Thus 𝑃𝑐 = 𝑃𝑡 [2+𝑚2 ] (3.21)

3.6.1 Ratio of Useful Power to Total Power (Transmission Efficiency)

The transmission efficiency of the AM wave is defined as the ratio of the useful power to the total
transmitted power. The useful power can be regarded as the power in the sidebands which contains the
information. The carrier component carries no information.

Useful power = 𝑃𝐿𝑆𝐵 + 𝑃𝑈𝑆𝐵

𝑚 2 𝐴𝑐 2 𝑚 2 𝐴𝑐 2
= + (3.22)
4 4

𝑚 2 𝐴𝑐 2
=
2
𝑚2
But from equation 3.19, total Power, 𝑃𝑡 = 𝐴𝑐 2 [1 + ]
2

Thus transmission efficiency, 𝜂 (ratio of useful power to total power) is

𝑚2 𝐴𝑐 2 𝑚2
2 2
= 𝑚2
= 𝑚2
𝐴𝑐 2 [1+ 2 ] 1+ 2

𝑚2
𝜂 =
2+𝑚2

For peak amplitude modulation, m = 1

1
2 1
𝜂 = 1 =
3
or 33.3%
1+2

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Example 3.2
A 100 W, AM transmitter is modulated firstly to 50% and then to 100%.
Determine:
a) total power transmitted
b) power in each side band, in each case

Solution 3.2
For 𝑚 = 50% = 0.5, Pc = 100 W
𝑚2 0.25
a) Total power, 𝑃𝑡 = 𝑃𝑐 [1 + ] = 100(1 + ) = 112.5 𝑊
2 2
𝑚2
b) Power in each sideband, 𝑃𝐿𝑆𝐵 = 𝑃𝑈𝑆𝐵 = 𝑃𝑐 = 6.25𝑊
4

3.6.2 Effective Voltage and Current

Let V = the effective or rms voltage of the modulated wave.
𝑉2
Then 𝑃𝑡 =
𝑅
2
𝑉2 𝑉𝑐 𝑚2
 = (1 + )
𝑅 𝑅 2

Where Vc = rms voltage of unmodulated carrier

𝑚2
Thus 𝑉 = 𝑉𝑐 √1 + 2

𝑚2
Similarly, 𝐼 = 𝐼𝑐 √1 + 2
NB:
𝐼 𝑚2
The ratio = √1 +
𝐼𝑐 2
The above equation is used as the basis for measuring the Modulation Index by measuring the Antenna
current.
Let Ir ≡ ratio of modulated to unmodulated rms Current
𝐼
or 𝐼𝑟 = , then in terms of Ir
𝐼𝑐

𝑚 = √2(𝐼𝑟 2 − 1)

Example 3.3
The rms antenna current of a radio transmitter is 5A when unmodulated, rising to 6A when the carrier
is sinusoidally modulated. Calculate the modulation index.

Solution 3.3
Modulated current, I = 6A; Unmodulated current, Ic = 5A, m=?
𝐼 6
𝐼𝑟 = = = 1.2
𝐼𝑐 5

But 𝑚 = √2(𝐼𝑟 2 − 1) = √2(1.22 − 1) = 0.94

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Example 3.4
An unmodulated carrier current is 7.5A. The current becomes 9.0A when the carrier is modulated with
a sine wave.
Determine the:
a) Index of modulation
b) Total current when the signal is modulated to a level of 80%.

Solution 3.4
I = 9.0A Ic = 7.5A
𝐼 2 9.0 2
a) 𝑚 = √2 [(𝐼 ) − 1] = √2 [(7.5) − 1] = 0.938 = 93.8%
𝑐

𝑚2 0.82
b) 𝐼 = 𝐼𝑐 √1 + = 7.5√1 + = 8.62𝐴
2 2

3.7 Modulating Signal of a Complex Wave

The AM waveform so far considered only looks at modulation of a carrier with a pure sine-wave
intelligence signal. However, in most systems the intelligence is a rather complex waveform that
contains many frequency components. For example, the human voice contains components from
roughly 200 Hz to 3 kHz and has a very erratic shape. If it were used to modulate the carrier, a whole
band of side frequencies would be generated.
This could be illustrated using a carrier of 2 MHz.
Let f1 = 200 Hz, f2 = 3000 Hz, fc = 2,000,000 Hz
 fc – f1 = 2,000,000 – 200 = 1,999,800 Hz fc + f1 = 2,000,000 + 200 = 2,000,200 Hz
fc – f2 = 2,000,000 – 3,000 = 1, 997,000 Hz fc + f2 = 2,000,000 + 3,000 = 2,003,000 Hz

LSB USB

fc – f2 fc – f1 fc fc + f1 fc + f2
1997 1999.8 2000 2000.2 2003 𝑘𝐻𝑧
Figure 3.7: Spectrum of a Complex wave ranging from 200 – 3000 Hz with a carrier frequency of 2 MHz

1
𝑚1 𝑚1
2 2
𝑚2 𝑚2
2 2
𝑚3 𝑚3
2 2

fc – 3fm fc – 2fm fc – fm fc fc + fm fc + 2fm fc + 3fm

Figure 3.8: Spectrum of a Complex wave containing several frequencies
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 Communication Systems

1
𝑚2 𝑚2
2 2
𝑚1 𝑚1
2 2
𝑚3 𝑚3
2 2

fc – f3 fc – f2 fc – f1 fc fc + f1 fc + f2 fc + f3
Figure 3.9: Spectrum Containing Several Unrelated Modulating Frequencies

3.8 Total Sideband Power

If several sine waves simultaneously modulate the carrier as in the case of the complex wave, the
carrier power is unaffected but the total sideband power will be the sum of the individual sideband
powers.
If PSBT = total sideband power, then
𝑚1 2 𝑚2 2 𝑚3 2
𝑃𝑆𝐵𝑇 = 𝑃𝑐 + 𝑃𝑐 + 𝑃𝑐 + … ..
2 2 2
𝑚𝑡 2
But 𝑃𝑆𝐵𝑇 = 𝑃𝑐 where mt = total modulation index
2
 𝑚𝑡 2 = 𝑚1 2 + 𝑚2 2 + 𝑚3 2

or 𝑚𝑡 = √𝑚1 2 + 𝑚2 2 + 𝑚3 2 + … .

Example 3.5
A 360 W carrier is simultaneously modulated by two audio waves with modulation percentages of 55
and 65, respectively. What is the total sideband power radiated?

3.9 Advantages and Disadvantages of Standard AM

A standard AM has clear advantages and disadvantages but there are more disadvantages than
advantages.

3.9.1 Disadvantages of Standard AM

- A great amount of power is wasted in the transmitted signal. This is because two-thirds (or more) of
the total power is in the carrier, which does not contribute to the transmitted intelligence.
- The standard AM is wasteful of bandwidth. The transmitted signal requires twice the bandwidth of
the transmitted intelligence.

3.9.2 Main Advantage of Standard AM

The AM system is relatively cheap to build. It uses straight forward and inexpensive transmitting and
receiving equipment.

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3.10 Single and Double Sideband Communications

From section 3.6.1 (Transmission efficiency) and section 3.9.1 (disadvantages of Standard AM), it is
obvious that AM is an inefficient and wasteful method of communications. This is because the real
information is contained within the sidebands and only about a third of the total power is within these
sidebands, whilst the major part (two-thirds or more) is taken up by the carrier wave which conveys no
intelligence. With the carrier not providing any useful information, it need not be transmitted. The
carrier can be suppressed without any detriment to information carrying capacity of the modulated
wave thereby resulting in saving more than 66% of the total power. The carrier can of course be
reconstructed at the receiver.

3.10.1 Double Sideband-Suppressed Carrier (DSB-SC)

In DSB-SC, the carrier wave is suppressed (or reduced) in order to save transmitted power. Thus the
transmitted wave consists of either only the sidebands (the upper and lower sidebands) or the
sidebands with a reduced carrier.

(b) Modulating signal

(a) AM wave

(c) Suppressed carrier wave

Figure 3.10: Waveforms for Double Sideband with Suppressed Carrier

fc - fm fc fc + fm
Figure 3.11: Frequency spectrum of a DSB-SC

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Instead of two-thirds of the power lost in the carrier, nearly all of the available power is used in
sidebands. From figure 3.11, the bandwidth, B, however, remains the same as that of the standard AM,
that is, twice the modulating frequency. i.e. 𝐵𝑤 = (𝑓𝑐 + 𝑓𝑚 ) − (𝑓𝑐 − 𝑓𝑚 ) = 2𝑓𝑚

3.10.2 Single Sideband (SSB)

SSB transmission is an attempted solution to the disadvantage of the Standard AM. This is a form of
AM in which one of the sidebands or the carrier and one of the sidebands are eliminated. This leaves
one of the sidebands and the carrier, or only one of the sidebands to be used to transmit the information
(audio) signal. If it is just one of the sidebands, the system is called SSB-TC (Single Sideband
Transmitted Carrier).
On the other hand, if the carrier and one of the sidebands is suppressed the system is called Single
Sideband Suppressed Carrier (SSB-SC). This is possible because the sidebands are symmetrical and
contain the same intelligence, and therefore eliminating one of them also cuts the required bandwidth
for transmission in half.
This system has many advantages and has been adopted by many communication systems. Besides
saving considerable amount of power, it results in bandwidth saving and can transmit good quality
communication signal at low power and narrow bandwidth.
Again, the SSB receiver is quieter due to the narrower bandwidth (receiver noise is a function of
bandwidth).

3.10.3 Advantages of SSB over DSB and AM

(a) Bandwidth conservation: The spectrum space occupied by the SSB signal is 𝑓𝑚 , which is only half
that of AM and DSB signals. That is, SSB required half the bandwidth required of AM and DSB
signals. The reduction allows more signals to transmit in the same frequency range without
interference.
(b) Power conservation: Due to suppression of carrier and one sideband, power is saved and this saved
power can be used to produce a stronger signal that will travel farther, and it will be reliably
received at greater distances.
(c) No fading: Fading does not occur in SSB transmission. Fading means that a signal alternately
increases and decreases in strength as it is picked up by the receiver. It occurs because the carrier
and sideband may reach the receiver shifted in time and phase with respect to each other. The
carrier and sideband signals have different frequencies, which are affected by the ionosphere in
different ways. The ionosphere bends the carrier and sideband signals at slightly different angles,
resulting in fading.
(d) Noise reduction: Bandwidth of the single sideband system is half of the conventional AM,
therefore the thermal noise power is reduced to half in SSB system than the conventional AM.

3.10.4 Disadvantages of SSB

(a) Complex process: The generation and reception of SSB signal is a complex process.
(b) Moderate transmission quality: Since carrier is absent, the SSB transmitter and receiver need to
have excellent frequency stability. A little change in frequency hampers the quality of signal such
as music signal. It is usually used for transmission of speech signal.
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 Communication Systems

Example 3.6
An AM broadcast station has modulation index which is 0.75 on the average. What would be its
average power saving, if it could go over to single sideband suppressed carrier transmissions, while
having to maintain the same signal strength in its reception area?

3.11 Receivers
In radio communications systems, the transmitted signal is very weak when it reaches the receiver,
particularly if it has travelled any distance. Further, the signal has had to share the free-space
transmission media with thousands of other radio signals. Various kinds of noise also get added to the
signal. The job of the radio receiver then is to have the sensitivity and selectivity to fully reproduce the
modulating signal at its output.
The receiver as found in broadcasting system is supposed to perform certain system functions. These
are:
a) Demodulating the incoming modulated signal
b) The selection of the desired signal (e.g. desired radio or T.V.), which is termed as carrier-
frequency tuning
c) Filtering, which represent a means of separating the desired signal from the other picked-up
modulated signals and
d) Amplification, the purpose of which is to compensate for the loss of signal power during the
transmission.

The receiver must be able to perform the above functions irrespective of whether they are amplitude or
frequency modulated.
The primary requirement for any communications receiver is that it has the ability to:
i. Select the desired signal from among thousands of others present and
ii. To provide sufficient amplification to recover the modulating signal.

These two requirements are generally referred to as Selectivity and Sensitivity

Selectivity
It refers to the ability of a receiver to select a signal of a desired frequency while rejecting those on
closely adjacent frequencies. A receiver with good selectivity will isolate the desired signal in the
radio-frequency (RF) spectrum and eliminate all other signals.

Sensitivity
This refers to the receiver’s ability to pick up weak signals. It is primarily a function of the overall
receiver gain, which is the factor by which an input signal is multiplied to produce an output signal. In
general, the higher the gain of a receiver, the better its sensitivity.

3.11.1 Reception Process

The overall reception process consists of:
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i. Coupling the signal from the physical medium to the receiver

ii. Channel selection
iii. Extraction of sideband content

This is illustrated in figure 3.12:

Receiving s(t) Demodulation

m(t)
Antenna
fc-fm fc fc+fm

Figure 3.12

: Channel Tuning Control

The reception process begins with the coupling antenna. It may be in the form of:

i. Parabola

ii. Monopole

iii. Dipole

Figure 3.13: Types of Antennas

• The coupled signal level in some cases may be so low that a pre-amplifier is required
preceding the radio frequency stage
• The purpose of receiver front is to reject unwanted signals
• After channel selection, the incoming signal is AM, DSB or SSB.
• The tuned RF amplifier accomplishes channel selection. The ideal bandwidth required are:

AM – 2fm; DSB – 2fm; SSB – fm

3.12 Amplitude Demodulation

Demodulation is the process of recovering the information signal m(t) from the modulated signal s(t),
that is by way of extraction of the sideband(s) from the modulated radio frequency signal.

3.12.1 The Demodulation Process

The two fundamental demodulation concepts used are:
a) Synchronous (coherent), where you should be able to generate a replica of the incoming carrier
b) Non-Synchronous (Non-Coherent), where the carrier can be generated without necessarily
generating the same thing.
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There have been two types of radio receivers that has stood the test of time, even though several of
them had been proposed. These two are:
a) The Tuned Radio Frequency (TRF) receiver
b) The Superheterodyne (Superhet) receiver.
Currently, it is only the super heterodyne receiver which is in general use, whilst the TRF could be
found in some fixed frequency applications.

3.12.2 The Superheterodyne Receiver

A block diagram of the Superhet is shown in figure 3.14. This is for use in AM broadcast band.
Basically the receiver consists of a radio-frequency (RF) stage and a mixer, which accepts two inputs,
the output of the RF amplifier and a steady sine wave from the Local Oscillator (L.O.). The others
include an Intermediate frequency (IF) section, demodulator, and Audio and Power amplifiers.
As depicted in the diagram, the incoming AM signal operating in the range of 0.535 to 1.605 MHz
(535 to 1605 kHz) is picked up by the antenna, and amplified in the RF section, which is tuned to the
carrier frequency of the incoming wave. This tuneable circuit must be able to tune over the entire
broadcast band.
Receiving 535 – 1605
Antenna kHz
455 kHz Loud speaker
RF IF Audio &
Mixer Detector
Amplifier Amplifier Power Amp

990 - 2060 kHz

Local
AGC
Oscillator

Common
Tuning
Figure 3.14: Superheterodyne Receiver

In the Mixer stage, the received RF signal and the L.O. of adjustable frequency (e.g. 990 – 2060 kHz)
are mixed. The dashed line area represents a combination for common tuning. The output of the mixer
stage is a fixed Intermediate Frequency, which is the difference in frequency between the RF signal
and the L.O. frequency. This is defined as
fIF = fLO - fRF
where fLO is the frequency of the local oscillator
and fRF is the carrier frequency of the incoming RF signal.
The standard difference frequency is 455 kHz.
The I.F. section provides most of the amplification and selectivity in the receiver. It is the key to the
superior selectivity of the Superhet. The amplified IF signal is applied to a detector (demodulator) whose
purpose is to recover the original modulating information.
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 Communication Systems

In the final stage of the receiver operation, the detected audio signal is subsequently amplified by the
Audio and Power amplifiers into the loudspeaker.
As could be seen in figure 3.14, an Automatic Gain Control (AGC) is connected by the feedback
network. The dc output of the detector is fed back to the input of the IF stage. The feedback voltage
controls the gain of the IF amplifier in such a way as to increase the gain when the received carrier is
weak and decrease it when the received carrier is too strong.

Problems
1. An antenna transmits an AM signal having a total power content of 15 kW. Determine the power
being transmitted at the carrier frequency and at each of the sidebands when the modulation is 85%.
[ 11.02 kW; 1.990 kW]
2. An amplitude-modulated wave is given by
𝑣 = 20[1 + 0.3 sin 5024𝑡] sin 6.28 × 106 𝑡 volts.
Determine the
i) Amplitude and frequency of the unmodulated carrier wave
ii) Amplitude and frequency of the modulating signal
iii) Modulation factor
iv) Sideband amplitudes and frequencies
3. The total power content of an AM signal is 3000 W. For (a) 100 percent (b) 80 percent
modulation, determine:
i. Power transmitted by carrier
ii. Power transmitted by each sideband

4. A sinusoidal carrier wave 50 sin 200000𝜋𝑡 was amplitude-modulated by an intelligence signal

given by 30 sin 5000𝜋𝑡. Determine
a) Modulation index
b) Percent modulation
c) Frequencies of signal and carrier
d) Frequency spectrum of the modulated wave
5. A bandwidth of 10 MHz is available for AM transmission. If the maximum audio signal frequency
used for modulating the carrier is not to exceed 20 kHz, how many stations can broadcast within
this band simultaneously without interfering with each other?

6. The output current of 60% modulated AM generator is 1.5 A. To what value will this current rise if
the generator is modulated additionally by another audio wave, whose modulation index is 0.7?
7. When a broadcast AM transmitter is 50% modulated, its antenna current is 12 A. What will the
current be when the modulation depth is increased to 0.9?

8. An AM transmitter radiates 9 kW of power when the carrier is unmodulated and 10.125 kW when
the carrier is sinusoidally modulated. Find the modulation index. If another sine wave,
corresponding to 40% modulation is transmitted simultaneously, calculate the total radiated power.
[ 0.50; 10.84 kW]

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CHAPTER FOUR
FREQUENCY MODULATION
4.0 INTRODUCTION
Modulation is the process of modifying a carrier wave in accordance with an information signal to be
transmitted. As was seen in Chapter three, varying the amplitude of the carrier according to the
amplitude of the intelligence (modulating) signal produces AM. It is also possible to impress an
information signal on a carrier by changing its frequency according to the amplitude of the information
signal. This leads to Frequency Modulation (FM).
Another characteristic of a carrier that can be changed is its phase shift (that is, the variation of the angle
of the signal). If the amount of phase shift that a carrier experiences is varied, information can be
impressed upon the carrier. This is known as phase modulation (PM). Both modulations are related to
time, hence they have some very similar properties. In effect, varying the phase shift of a carrier
produces FM.
FM and PM may be closely related to one another, however, they also have some important differences.
Together they are collectively referred to as types of angle modulation. They represent the alternative
to AM for impressing an information signal onto a carrier. FM and PM offer many advantages in noise
reduction, signal fidelity, and use of power, but more complex circuitry is required for the transmitter
and receiver.

4.1 FM WAVEFORM
As explained, FM is a form of angle modulation in which the amplitude of the modulated carrier is kept
constant whiles its instantaneous frequency is varied in response to the modulating (intelligence signal).
As the amplitude of the modulating signal varies, the carrier frequency shifts in response.
That is
𝑉(𝑡) = 𝑉𝑐 sin(𝜔𝑐 𝑡 + 𝜙) (4.1)
Where
V(t) = instantaneous voltage
𝜔𝑐 = angular frequency
𝜙 = phase angle
As depicted in the waveforms, figure 4.1 represents a constant unmodulated carrier or rest frequency.
In figure 4.2, up until 𝑡1 , the waveform of FM had a constant frequency and thus correspond to the
carrier or rest frequency (fc). When the modulating signal increases sinusoidally and reaches a maximum
positive amplitude value, 𝑡2 , the oscillator frequency increases accordingly and also reaches its highest
frequency (represented by the congested part of the FM waveform).
As the information signal moves from maximum positive to maximum negative, 𝑡4 ,, the resulting
frequency also falls to its minimum frequency (also represented by the sparse part of the FM waveform).

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+V

-V

Figure 1: A constant unmodulated carrier or rest frequency

3
t2
+

t1
0 t3

- No
Signal

t4
Zero
Maximum Maximum
Deviation
Negative Positive
Deviation Deviation

Figure 2: Frequency modulation

Let the frequency of carrier be raised in response to a modulating wave given by

𝑣𝑚 = 𝑉𝑚 cos 𝜔𝑚 𝑡 (4.2)
Thus the instantaneous frequency of carrier is
𝑓 = 𝑓𝑐 + 𝑘𝑓𝑐 𝑣𝑚 (4.3)
= 𝑓𝑐 + 𝑘𝑓𝑐 𝑉𝑚 cos 𝜔𝑚 𝑡
= 𝑓𝑐 (1 + 𝑘𝑉𝑚 cos 𝜔𝑚 𝑡) (4.4)
where fc = unmodulated carrier frequency and k = constant of proportionality
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4.2 Frequency Deviation

The maximum deviation in frequency away from the carrier frequency will occur when the cosine term
has its maximum value of ±1. Under this, the instantaneous frequency will be
𝑓 = 𝑓𝑐 (1 ± 𝑘𝑉𝑚 ) (4.5)
Deviation, Δf, is thus defined as
∆𝑓 = 𝑘𝑉𝑚 𝑓𝑐 (4.6)
Hence , 𝑓 = 𝑓𝑐 ± ∆𝑓

4.3 The FM equation

Under Modulation Conditions, the instantaneous amplitude of the FM signal
𝑣 = 𝑉𝑐 sin( 𝜔𝑡 + Φ) (4.7)
Where 𝜔 = 2𝜋𝑓
𝜔 = 2𝜋𝑓𝑐 (1 + 𝑘𝑉𝑚 cos 𝜔𝑚 𝑡)
𝜔 = 𝜔𝑐 (1 + 𝑘𝑉𝑚 cos 𝜔𝑚 𝑡)
Hence by deviation, the instantaneous voltage of the signal, V is given by
Δ𝑓
𝑣 = 𝑉𝑐 sin( 𝜔𝑐 𝑡 + sin 𝜔𝑚 𝑡) (4.8)
𝑓𝑚

Proof
If 𝑣 = 𝑉𝑐 sin[𝐹(𝜔𝑐 , 𝜔𝑚 ] (4.9)
Where F is an angle and defined as θ

Then 𝑣 = 𝑉𝑐 sin 𝜃 (4.10)

Also
Instantaneous angular frequency, ω is given by
𝑑𝜃
𝜔= rad/sec
𝑑𝑡
Thus, the instantaneous angle
𝜃 = ∫ 𝜔𝑑𝑡 = 𝜔𝑐 ∫(1 + 𝑘𝑉𝑚 cos 𝜔𝑚 𝑡)𝑑𝑡

𝑘𝑉𝑚 sin 𝜔𝑚 𝑡 𝜔𝑐 𝑘𝑉𝑚 sin 𝜔𝑚 𝑡

= 𝜔𝑐 (𝑡 + ) = 𝜔𝑐 𝑡 +
𝜔𝑚 𝜔𝑚
2𝜋𝑓𝑐 𝑘𝑉𝑚 sin 𝜔𝑚 𝑡 𝑘𝑉𝑚 𝑓𝑐 sin 𝜔𝑚 𝑡
= 𝜔𝑐 𝑡 + = 𝜔𝑐 𝑡 +
2𝜋𝑓𝑚 𝑓𝑚

∆f sin 𝜔𝑚 𝑡
⟹ 𝜃 = 𝜔𝑐 𝑡 + (4.11)
𝑓𝑚

Substituting eqn. (4.11) into (4.10), the instantaneous voltage of the FM signal becomes
∆𝑓
𝑣 = 𝑉𝑐 sin(𝜔𝑐 𝑡 + 𝑓 sin 𝜔𝑚 𝑡) (4.12)
𝑚

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4.4 Modulation Index (Deviation Ratio)

The modulation index for the FM wave, Mf is defined as
𝑀𝑎𝑥𝑖𝑚𝑢𝑚 𝑓𝑟𝑒𝑞𝑢𝑒𝑛𝑐𝑦 𝑑𝑒𝑣𝑖𝑎𝑡𝑖𝑜𝑛
Mf =
𝑀𝑜𝑑𝑢𝑙𝑎𝑡𝑖𝑛𝑔 𝐹𝑟𝑒𝑞𝑢𝑒𝑛𝑐𝑦
∆𝑓
𝑀𝑓 = (4.13)
𝑓𝑚

Hence substituting (4.12) into (4.11), we obtain

𝒗 = 𝑽𝒄 𝐬𝐢𝐧(𝝎𝒄 𝒕 + 𝑴𝒇 𝐬𝐢𝐧 𝝎𝒎 𝒕) (4.14)

Example 4.1
An FM signal has a resting frequency of 105 MHz and the highest frequency of 105.03 MHz when
modulated by a signal frequency of 5 kHz. Determine the:
i. frequency deviation
ii. modulation index

Solution 4.1
𝑓𝑐 = 105 MHz; 𝑓 = 105.03 MHz; 𝑓𝑚 = 5 kHz
i. But 𝑓 = 𝑓𝑐 ± ∆𝑓
⟹ ∆𝑓 = 𝑓 − 𝑓𝑐 = 0.03 MHz = 30 kHz

ii. Modulation index,

∆𝑓 30
Mf = = =6
𝑓𝑚 5

Example 4.2
A 100 MHz carrier is modulated by a 12 kHz sinewave so as to cause a frequency swing of ±50 𝑘𝐻𝑧.
Calculate the modulation index.

Solution 4.2
𝑓𝑚 = 12 𝑘𝐻𝑧; ∆𝑓 = 50 𝑘𝐻𝑧
∆𝑓 50
Mf = =
𝑓𝑚 12

Example 4.3
An FM signal is represented by the voltage equation
𝑣 = 12 sin(6 × 108 𝑡 + 5 sin 1250 𝑡)
a) Determine:
i. The carrier frequency
ii. The modulating frequency
iii. The modulation index
iv. The maximum deviation
b) What power will this FM wave dissipate in a 10 Ω resistor?
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 Communication Systems

Solution 4.3
For the FM wave
𝑣 = 𝑉𝑐 sin(𝜔𝑐 𝑡 + 𝑀𝑓 sin 𝜔𝑚 𝑡)
Comparing with 𝑣 = 12 sin(6 × 108 𝑡 + 5 sin 1250 𝑡)
a)
6×108
i. fc = = 95. 5 MHz
2𝜋
1250
ii. fm = = 199 Hz
2𝜋

iii. 𝑀𝑓 = 5
∆𝑓
iv. Mf = ⇒ ∆𝑓 = 𝑀𝑓 × 𝑓𝑚
𝑓𝑚
= 5 × 199 = 995 𝐻𝑧
𝑉 2 𝑟𝑚𝑠
b) Power, P = ,
𝑅
𝑉 = 𝑉𝑐 = 12
𝑉 12
But 𝑉𝑟𝑚𝑠 = =
√2 √2

12 2
( )
√2
∴𝑃= = 7.2 W
10

Example 4.4
In an FM system, when the audio frequency, (AF) is 500 Hz and the voltage of the AF is 2.4 V, the
deviation is 4.8 kHz.
(a) if the AF is increased to 7.2 V, what is the new deviation?
(b) If the AF voltage is raised to 10 V whilst the AF frequency is dropped to 200 Hz, what is the
deviation?
(c) Find the modulation index in each case.

Solution 4.4
∆𝑓 = 𝑘𝑉𝑚 𝑓𝑐
∆𝑓 4.8
⟹ = 𝑘𝑓𝑐 = = 2 𝑘𝐻𝑧/𝑉
𝑉𝑚 2.4

(a) when 𝑉𝑚 = 7.2 𝑉,

∆𝑓 = 𝑉𝑚 × 2 𝑘𝐻𝑧/𝑉
= 7.2 × 2 = 14.4 𝑘𝐻𝑧

(b) ∆𝑓 = 10 × 2 = 20 𝑘𝐻𝑧 (This is because f does not affect the deviation of the signal – only the
amplitude).

(c) Modulation Index

∆𝑓1 4.8
𝑀𝑓1 = = = 9.6
𝑓𝑚1 0.5

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∆𝑓2 14.4
𝑀𝑓2 = = = 28.8
𝑓𝑚2 0.5

∆𝑓3 20
𝑀𝑓3 = = = 100
𝑓𝑚3 0.2

4.5 Frequency Spectrum of FM wave (Use of Bessel Functions)

From the FM equation
𝑣 = 𝑉𝑐 sin(𝜔𝑐 𝑡 + 𝑀𝑓 sin 𝜔𝑚 𝑡)
The solution of this type of function involves the use of Bessel functions given by

𝑀𝑓 2 𝑀𝑓 4 𝑀𝑓 6
𝑀𝑓 𝑛 1 (
2
) (
2
) (
2
)
𝐽𝑛 (𝑀𝑓 ) = ( ) [𝑛! − 1!(𝑛+1)! + 2!(𝑛+2)! − 3!(𝑛+3)! + ⋯ ]
2

Where Jn (Mf) = Bessel coefficient (also designates the sideband)

Using the coefficients, the FM wave can be expanded to yield the following:
𝑣
= 𝐽𝑜 (𝑀𝑓 ) sin 𝜔𝑐 𝑡
𝑉𝑐

+ 𝐽1 (𝑀𝑓 )[sin(𝜔𝑐 + 𝜔𝑚 )𝑡 − sin(𝜔𝑐 − 𝜔𝑚 )𝑡]

+ 𝐽2 (𝑀𝑓 )[sin(𝜔𝑐 + 2𝜔𝑚 )𝑡 + sin(𝜔𝑐 − 2𝜔𝑚 )𝑡]

+ 𝐽3 (𝑀𝑓 )[sin(𝜔𝑐 + 3𝜔𝑚 )𝑡 − sin(𝜔𝑐 − 3𝜔𝑚 )𝑡]

+ .................

It is seen that the FM wave consist of a carrier and an apparently infinite number of pairs of sideband,
each preceded by a J coefficient representing the corresponding harmonic or sideband frequency. They
are Bessel functions of the first kind. The order denoted by the subscript and the argument by Mf.
The function Jn (Mf) is given in a table form or in a graphical form.

The following observations were made using the Bessel function:

1) Unlike AM where there are only 3 frequencies (the carrier and the first two sidebands), FM has an
infinite number of sidebands as well as the carrier. They are separated from the carrier by fm, 2fm,
3fm etc. (Harmonics)
2) The sidebands (Jn) are at equal distances from fc and have equal amplitudes so that the sideband
distribution is symmetrical about the carrier frequency.
3) In AM, increased depth of modulations increases sideband power and hence total transmitted
power.
In FM, the total transmitted power is constant (since the amplitude is constant), but with increased
depth of modulation, the required bandwidth is increased.
4) In FM, unlike AM, the amplitude of the carrier component does not remain constant. Its J coefficient
is J0, which is a function of Mf.
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J0 (Mf)
J1 (Mf) J1 (Mf)
J2 (Mf) J2 (Mf)

J3 (Mf) J3 (Mf)

fc
Figure 4.3: Modulation index and significant sidebands

Note.
J coefficients decrease in value as n increases. Each J coefficient represents the amplitude of a specific
pair of sidebands.
- Some J coefficients have negative values which signify a phase shift of 180o for that pair of
sidebands.

5) It is possible for the carrier component of the FM wave to disappear completely (that is the Bessel
coefficient, 𝐽0 (𝑀𝑓 ) is zero). This happens for certain values of modulation indices called Eigen
Values. The carrier disappears for the modulation indices, 𝑀𝑓 , equal to 2.4, 5.5, 8.6, 11.8 and so on.

Figure 4.4: Graphical Representation of Bessel Functions

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 Chapter Four: Frequency Modulation Communication Systems

Table 4.1: Bessel Functions

Sidebands (Pairs)
Modulation Carrier 1st 2nd 3rd 4th 5th 6th 7th 8th 9th 10th 11th 12th 13th 14th 15th 16th
Index (Mf) J0 J1 J2 J3 J4 J5 J6 J7 J8 J9 J10 J11 J12 J13 J14 J15 J16
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

2.4 0 0.52 0.43 0.20 0.06
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
5.5 0 -0.34 -0.12 0.26 0.40 0.32 0.19 0.09 0.03 0.01
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
8.65 0 0.27 0.06 -0.24 -0.23 0.03 0.26 0.34 0.28 0.18 0.10 0.05 0.02
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

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Figure 4.5: Graphical Representation of Bessel Functions

Figure 4.6 shows the FM frequency spectrum for various levels of modulation while keeping the
modulating frequency constant. The relative amplitude of all components is obtained from Table 4.1. It
could be observed from Table 1 that between 𝑀𝑓 = 2.0 and 𝑀𝑓 = 2.5, the carrier undergoes a change
from a positive to a negative value. The negative is an indication of a phase reversal.
At 𝑀𝑓 = 2.4, the carrier component has a zero amplitude.

𝐽0 (𝑀𝑓 ) J0 (Mf)

𝑀𝑓 = 0.5
𝑀𝑓 = 0.0

J1 (Mf) J1 (Mf)
J2 (Mf) J2 (Mf)

J0 (Mf) 𝐽1 (𝑀𝑓 ) 𝐽1 (𝑀𝑓 )

𝑀𝑓 = 1.5 𝑀𝑓 = 2.4

𝐽4 (𝑀𝑓 ) 𝐽4 (𝑀𝑓 ) 𝐽4 (𝑀𝑓 ) 𝐽4 (𝑀𝑓 )

𝐽0 (𝑀𝑓 )

Figure 4.6: Frequency Spectrum for FM (Constant Modulating Frequency, Variable Deviation)

Example 4.5
Determine the relative power of the carrier wave and side frequencies when modulation index 𝑀𝑓 =
0.25 for 10 kW FM transmitter.

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4.6 Bandwidth and Required Spectra

From Table 4.1, it is possible to calculate the size of the carrier and each sideband for each specific
value of modulation index. From these values, the frequency spectrum of the FM wave for those
particular values of Mf may be plotted.
Also, the total bandwidth of the FM signal can be determined by knowing the modulation index and
using the table. It can be determined with the formula:
𝐵 = 2𝑛𝑓𝑚
Where n is the number of significant sidebands.

Example 4.6
What is the bandwidth required for an FM signal in which the modulating frequency is 2 kHz and
maximum deviation is 10 kHz?

Solution 4.6
∆𝑓 10
𝑀𝑓 = = =5
𝑓𝑚 2

Then the required bandwidth,

𝐵 = 2𝑛𝑓𝑚 , n is highest needed sideband
In this case, n = 8
𝐵 = 2 × 8 × 2 = 32 𝑘𝐻𝑧

4.7 Carson’s Rule

An alternative way to calculate the bandwidth of an FM signal is to use Carson’s rule. This rule also takes
into consideration only the power in the most significant sidebands whose amplitudes are greater than 2
percent of the carrier. Carson’s rule is given by the expression
𝐵 = 2(∆𝑓𝑚𝑎𝑥 + 𝑓𝑚 𝑚𝑎𝑥 )
Where fd max is the maximum frequency deviation
And fm max is the maximum modulating frequency

Example 4.7
Apply Carson’s rule to example 4.6
∆𝑓𝑚𝑎𝑥 = 10; 𝑓𝑚 𝑚𝑎𝑥 = 2

Thus 𝐵 = 2(10 + 2)
𝐵 = 24 𝑘𝐻𝑧
Comparing the bandwidth with that computed in the preceding example, it can be seen that Carson’s rule
gives a smaller bandwidth. It has been determined that if a circuit or system has that bandwidth (per
Carson’s rule), sufficient sideband power will be passed to ensure full intelligibility of the information
signal.

4.8 Significant Sideband Criterion

It can be seen from Example 4.6 that all the sideband frequencies corresponding to the value of Mf = 5
were used in calculating the fm bandwidth . If this rule is strictly applied, the bandwidth required for
higher values of Mf will be very large.

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In practice however, since bandwidth is a limiting factor, a rule is used to consider only “significant”
sideband components which have Bessel coefficients greater than a given value normally expressed as a
fraction of J0 (0) – the unmodulated carrier.

Mathematically expressed only the sideband component for which

𝐽𝑛 (𝑀𝑓 ) >= 𝛽𝐽0 (0)
Thus all frequency components which give Bessel values less than 𝛽𝐽0 (0), are regarded as
insignificant in the determination of the fm boundary. The above equation is referred to as the significant
sideband criterion.

Example 4.8
Evaluate the bandwidth required in Example 4.6, given that the significant sideband criterion is
𝐽𝑛 (𝑀𝑓 ) >= 0.2𝐽0 (0)

Solution 4.8
Given fm = 2 kHz, ∆𝑓 = 10 𝑘𝐻𝑧
∆𝑓 10
𝑀𝑓 = = =5
𝑓𝑚 2

From Bessel table, 𝐽0 (0) = 1

Hence it is required that 𝐽𝑛 (5) >= 0.2
From the tables
𝐽𝑛 (5) = 0.26 > 0.2
∴𝑛=5
The bandwidth thus becomes
𝐵 = 2𝑛𝑓𝑚
𝐵 = 2 × 5 × 2 = 20 𝑘𝐻𝑧

4.9 Advantages of Frequency Modulation over Amplitude Modulation

1. The primary advantage of FM over AM is its immunity to noise.
2. The transmitted power of FM wave is constant since its amplitude is constant, and independent of
modulation index. This means low level modulation can be used in FM transmitters. Whereas in
AM, modulation index controls the transmitted power.
3. Greater transmitter efficiency: In FM, all the transmitted power is useful, whereas in AM, even at
maximum modulation depth, 67% of transmitted power is in the carrier and as the carrier does not
carry any intelligence, this power is wasted.
4. Capture effect: it allows the strongest signal on a frequency to dominate without interference from
the other signal.
5. FM broadcasting operates in the VHF range. In that frequency range, there is less noise than in the
medium wave (MW) and short wave (SW) ranges allotted to the AM broadcast.
6. Since FM operates in the VHF range, the propagation is line-of-sight. Thus the radius of operation
of an FM transmitter is limited to radio horizon. This allows several independent FM transmitters
to work on the same carrier frequency with very little interference than would be possible with AM.

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4.8 Disadvantages of FM

1. For transmission of FM signal, a much larger bandwidth is required than that required to transmit
AM signal.
2. FM transmitter and receiver are quite complicated, especially, in terms of servicing and fault-finding
or repairing of FM receivers.
3. Since FM reception is limited to line-of-sight, the service area of FM transmitter is much less than
that of AM.

Problems

1. A 100-MHz carrier is deviated 50 kHz by a 4–KHz signal. Calculate the modulation index.
2. Calculate the bandwidth of an FM signal with a maximum deviation of 10 kHz and a maximum
modulating signal frequency of 4 kHz. Use the two methods given in the text and compare your
answers.
3. The carrier in an FM signal can never drop to zero amplitude. True or False.

4. A single-tone FM signal is given by 𝑣(𝑡) = 10 sin(16𝜋 × 106 𝑡 + 20 sin 2𝜋 × 103 𝑡) volts. Determine
the modulation index, modulating frequency, frequency deviation, carrier frequency and the power of
the FM signal.

5. An FM wave modulated to a depth of 8 generates a bandwidth of 180 kHz. Find the frequency
deviation.

6. Determine the carrier swing, the highest and the lowest frequencies attained and the modulation index
of the FM signal generated by frequency modulating 88.9 MHz carrier with a 10 kHz sine wave
causing a frequency deviation of 50 kHz.

7. A certain FM transmitter has modulation sensitivity, k, of 10 kHz/V. If a 5-kHz sine wave of 12 Vp-p
is applied to this transmitter, determine the frequency deviation that occurs.

8. A bandwidth of 20 MHz is available for FM transmission. If the maximum audio signal frequency
used for modulating the carrier is not to exceed 4 kHz and the maximum deviation is 24 kHz, how
many stations can broadcast within this band simultaneously without interfering with each other, given
that the significant sideband criterion is 𝐽𝑛 (𝑀𝑓 ) >= 0.2𝐽0 (0)?

9. An FM signal 2000 sin(2𝜋 × 108 𝑡 + 2 sin 𝜋 × 104 𝑡) is applied to a 50-Ω antenna. Determine: (a) the
carrier frequency
(b) the transmitted power
(c) 𝑀𝑓
(d) 𝑓𝑚
(e) bandwidth by two methods
(f) power in the largest and smallest sidebands.

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CHAPTER 5
DIGITAL MODULATION TECHNIQUES

5.0 Introduction
Modern mobile communication systems use digital modulation techniques. Advancements in very large-
scale integration (VLSI) and digital signal processing (DSP) technology have made digital modulation
more cost effective than analog transmission systems. Digital modulation offers many advantages over
analog modulation. Some advantages include greater noise immunity and robustness channel
impairments, easier multiplexing of various forms of information (e.g. voice, data and video) and greater
security.

In digital wireless communication systems, the modulating signal (e.g. the message) may be represented
as a time sequence of symbols or pulses, where each symbol has m finite states. Each symbol represents
n bits of information where 𝑛 = log 2 𝑚 bits/symbol. Many digital modulation schemes are used in
modern communication systems, and many more are sure to be introduced. Since binary data is to be
transmitted, only two signal levels are required. The signal switches (shifts) between these two levels as
the binary data signal alternates (keys) between a binary 1 and 0.

5.1 Digital-to-Analog Conversion

It is the process of changing one of the characteristics of an analog signal based on the information in
digital data.
• Digital data needs to be carried on an analog signal.
• A carrier signal (frequency fc) performs the function of transporting the digital data in an analog
waveform.
• The analog carrier signal is manipulated to uniquely identify the digital data being carried.

Figure 5.1: Digital-to-Analog Conversion

Carrier signals can be completely described by three parameters: amplitude, frequency, and phase. Any
one of these three can be manipulated to suit the data transmission requirements. Amplitude-shift keying
(ASK), frequency-shift keying (FSK), and phase-shift keying (PSK) are digital-to-analog modulation
schemes. Amplitude-shift keying, for example, modifies the amplitude of the carrier wave. Frequency-
shift keying adjusts the frequency, and phase-shift keying manipulates the phase.
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Figure 5.2: Types of Digital-to-Analog Modulation

5.2 Amplitude Shift Keying

ASK is the simplest digital modulation technique. It refers to a type of amplitude modulation that assigns
bit values to discrete amplitude levels. The carrier signal is then modulated among the members of a set
of discrete values to transmit information.
• ASK is implemented by changing the amplitude of a carrier signal to reflect amplitude levels in
the digital signal.
• Two or more signals with different amplitudes are used.
• The line encoding will determine the values of the analog waveform to reflect the digital data
being carried.

1 0 0 0 1 1 0 1
𝑉𝑑 (𝑡)

𝑉𝑐1 (𝑡)

𝑉𝑐2 (𝑡)

𝑉𝐴𝑆𝐾 (𝑡)

Figure 5.3: Formation of ASK

Signal
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Vd(t) is the message signal at the control input of a switch.

𝑉𝑐1 (𝑡) is the first carrier signal at the input of the switch.
𝑉𝑐2 (𝑡) is the second carrier signal at the input of the switch.
𝑉𝐴𝑆𝐾 (𝑡) is the output of the switch.

Although theoretically any two (or more) amplitudes could be used, the most widely implemented form
of ASK represents a one (1) as a sine wave with a specified amplitude, and a zero (0) as no signal at all
(i.e., a sine wave with zero amplitude), hence it is switched on or off. As a result, ASK is also called
OOK (ON – OFF keying).

• OOK is implemented by changing the amplitude of a carrier signal to reflect amplitude levels in
the digital signal.
• The digital “1” could not affect the signal, whereas the digital “0” would, by making it zero.

Figure 5.4: Formation of OOK

Signal
There are sharp discontinuities shown at the transition points (Figure 5.4). These result in the signal
having an unnecessarily wide bandwidth. A significant reduction can be accepted before errors at the
receiver increase unacceptably.
This can be brought about by band-limiting (pulse shaping) the message before modulation, or band-
limiting the ASK signal itself after generation.
Band-limiting is generally introduced before transmission, in which case these discontinuities would be
‘rounded off’.

Figure 5.5: ASK Generation Method

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Figure 5.6 shows the signals present in a model of Figure 5.5, where the message has been bandlimited.

Figure 5.6: Original Message (Lower), Bandlimited Message (Centre), and ASK (Top)

ASK is a simple technique, but its noise susceptibility in the electronic space limits it to low-speed
applications over telephone lines, which have become essentially obsolete. However, ASK is commonly
used over optical fibre with light-emitting diode (LED) and LASER transmitters. In simple systems, the
presence of light represents a one and the absence of light represents a zero. Amplitude-shift keying is
used extensively for commercial terrestrial applications.

5.3 Frequency Shift Keying (FSK)

FSK is also known as frequency shift modulation and frequency shift signalling. Frequency Shift Keying
is a data signal converted into a specific frequency or tone in order to transmit it over wire, cable, optical
fibre or wireless media to a destination point. It refers to a type of frequency modulation that assigns bit
values to discrete frequency levels. FSK is divided into non-coherent and coherent forms. In non-coherent
forms of FSK, the instantaneous frequency shifts between two discrete values termed the "mark" and
"space" frequencies. In coherent forms of FSK, there is no phase discontinuity in the output signal. FSK
modulation formats generate modulated waveforms that are strictly real values, and thus tend not to share
common features with quadrature modulation schemes.

• The digital data stream changes the frequency of the carrier signal, fc.
• For example, a “0” could be represented by 𝑓1 = 𝑓𝑐 − 𝑓, and a “1” could be represented by
𝑓2 = 𝑓𝑐 + 𝑓.

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Carrier
frequency 1
Carrier
frequency 2

Figure 5.7: Formation of FSK

Signal
Figure 5.7 conveys the concept using the simplest form of FSK known as binary frequency shift keying
(BFSK). In that scheme, one frequency is chosen to represent a zero and another to represent a one. There
are more complex forms of FSK that use more frequencies, but few are actively used today.
It is effective in many instances including the digital world where it is commonly used in conjunction
with computers and low speed modems. In fact, the contributions of FSK are much more far reaching.
For example, the principle of FSK has laid the path to the development of other similar techniques such
as the Audio Frequency Shift Keying (AFSK) and Minimum Frequency Shift Keying (MFSK), also
known as minimum shift keying (MSK) just to name a few.
The primary difference between AFSK and FSK is that the former is a baseband signal whereas the latter
modulates a carrier at a specific frequency. AFSK is used in North America for caller ID on an analog
POTS line. AFSK was also used in the early, low-speed modems, like the Bell 103. Today it is also used
in the U.S. Emergency Alert System.
Frequency-shift keying has some attractive characteristics, but it generally does not use bandwidth very
efficiently and is not suitable for applications where bandwidth efficiency is crucial.

5.4 Phase Shift Keying

Phase shift keying, PSK, is widely used these days within a whole raft of radio communications systems.
It is particularly well suited to the growing area of data communications. PSK enables data to be carried
on a radio communications signal in a more efficient manner than Frequency Shift Keying, and some
other forms of modulation. With more forms of communications transferring from analogue formats to
digital formats, data communications is growing in importance and along with it the various forms of
modulation that can be used to carry data.
There are several flavours of PSK that are available for use. Each form has its own advantages and
disadvantages, and a choice of the optimum format has to be made for each radio communications system
that is designed.

5.4.1 PSK Basics

PSK in a digital transmission refers to a type of angle modulation in which the phase of the carrier is
discretely varied—either in relation to a reference phase or to the phase of the immediately preceding
signal element—to represent data being transmitted. For example, when encoding bits, the phase shift
could be 0 degree for encoding a "0," and 180 degrees for encoding a "1," or the phase shift could be –90
degrees for "0" and +90 degrees for a "1," thus making the representations for "0" and "1" a total of 180
degrees apart.
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Some PSK systems are designed so that the carrier can assume only two different phase angles, each
change of phase carries one bit of information, that is, the bit rate equals the modulation rate.
If the number of recognizable phase angles is increased to four, then 2 bits of information can be encoded
into each signal element; likewise, eight phase angles can encode 3 bits in each signal element.

That is, the method can be extended to take three input bits, choosing one of eight (or 23) phases, or four
bits, choosing one of sixteen (or 24) phases. This pattern can be extended to a generalized M phase-shift
keying signal, and all these waveforms would have the same power spectral density. But the improvement
in bandwidth efficiency comes at the cost of decreased power efficiency. The receiver must decide which
phase, out of some possible M choices, was transmitted. Those M phases are only separated by 360/M
degrees. The phase separation for binary phase-shift keying, for example, is 360/2 or 180 degrees. A
higher M value would have a correspondingly smaller phase separation. This smaller separation means it
takes less noise to corrupt the signal and cause the receiver to make an error. To combat this, more power
must be transmitted.

Another approach is to shape the waveform spectra coming out of the modulator. This can be
accomplished by filtering the waveform to narrow the main lobe and reduce side-lobes. The downside to
filtered waveforms is that the filtering process distorts the amplitude of the signal, resulting in the same
distortion through a saturated power amplifier found in amplitude-shift keying.
Another option is to use continuous phase modulation. Waveforms produced in this manner exhibit
smooth phase transitions rather than the abrupt phase transitions produced through binary or quadrature
phase-shift keying. This is important because smooth phase transitions require less bandwidth for signal
transmission. Gaussian minimum shift keying, the waveform that Aerospace proposed for the Advanced
Extremely High Frequency program, is a form of continuous phase modulation.

5.4.2 PSK Constellation Diagrams

It is often convenient to represent a phase shift keyed signal, and sometimes other types of signal using a
phasor or constellation diagram. Using this scheme, the phase of the signal is represented by the angle
around the circle, and the amplitude by the distance from the origin or centre of the circle. In this way the
signal can be resolved into quadrature components representing the sine or I for In-phase component and
the cosine for the quadrature component. In Summary, a constellation diagram
• Is a representation of a signal modulated by a digital modulation scheme;
• Is used to display the signal as a two-dimensional scatter diagram in the complex;
• Represents the possible symbols that may be selected by a given modulation

1 0

Figure 5.8: Constellation Diagrams for BPSK, QPSK and 8-PSKSignal

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Most phase shift keyed systems use a constant amplitude and therefore points appear on one circle with
constant amplitude and the changes in state being represented by movement around the circle. For binary
shift keying using phase reversals the two points appear at opposite points on the circle. Other forms of
phase shift keying may use different points on the circle and there will be more points on the circle.

5.4.3 Forms of Phase Shift Keying

Although phase modulation is used for some analogue transmissions, it is far more widely used as a digital
form of modulation where it switches between different phases. This is known as phase shift keying, PSK,
and there are many flavours of this. It is even possible to combine phase shift keying and amplitude keying
in a form of modulation known as quadrature amplitude modulation, QAM.
The list below gives some of the more commonly used forms of phase shift keying, and related forms of
modulation that are used:
• PSK - Phase Shift Keying
• BPSK - Binary Phase Shift Keying
• QPSK - Quadrature Phase Shift Keying
• O-QPSK - Offset Quadrature Phase Shift Keying
• 8 PSK - 8 Point Phase Shift Keying
• 16 PSK - 16 Point Phase Shift Keying
• QAM - Quadrature Amplitude Modulation
• 16 QAM - 16 Point Quadrature Amplitude Modulation
• 64 QAM - 64 Point Quadrature Amplitude Modulation
• MSK - Minimum Shift Keying
• GMSK - Gaussian filtered Minimum Shift Keying

5.4.4 Binary Phase Shift Keying

The basic form of phase shift keying is known as Binary Phase Shift Keying (BPSK) or it is occasionally
called Phase Reversal Keying (PRK) (Refer to Figure 5.9). BPSK takes each input bit individually and
chooses one of two possible phases to represent that bit value—so for example, to send a "0," a phase of
0 degrees might be chosen, and to send a "1," a phase of 180 degrees might be chosen.

Figure 5.9: Phase Reversal

Assuming a sinusoidal carrier is modulated by a bi-polar bit stream according to the scheme illustrated in
Figure 5.10, its polarity will be reversed every time the bit stream changes polarity. This, for a sine wave,
is equivalent to a phase reversal (shift). The multiplier output is a BPSK signal.

Figure 5.10: Generation of BPSK

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The information about the bit stream is contained in the changes of phase of the transmitted signal. A
synchronous demodulator would be sensitive to these phase reversals.

The appearance of a BPSK signal in the time domain is shown in Figure 5.11 (lower trace). The upper
trace is the binary message sequence.

Figure 5.11: BPSK Signal

The wave shape is ‘symmetrical’ at each phase transition. This is because the bit rate is a sub-multiple of
the carrier frequency w/(2π). In addition, the message transitions have been timed to occur at a zero-
crossing of the carrier.

5.4.5 Quadrature Phase Shift Keying (QPSK)

QPSK is a form of PSK in which 2 bits are modulated at once, selecting one of four possible carrier phase
shifts (0, 𝜋⁄2 , 𝜋, 𝑎𝑛𝑑 3𝜋⁄2 ).
It takes two bits at a time and chooses one of four possible phases to represent them—so, to send a "00,"
a phase of 0 degrees might be chosen; to send a "01," a phase of 90 degrees might be chosen; to send a
"11," a phase of 180 degrees might be chosen; and to send a "10," a phase of 270 degrees might be chosen.
QPSK performs by changing the phase of the In-phase (I) carrier from 𝟎𝟎 to 𝟏𝟖𝟎𝟎 and the Quadrature-
phase (Q) carrier between 𝟗𝟎𝟎 and 𝟐𝟕𝟎𝟎 . This is used to indicate the four states of the 2-bit binary code.
Each state of these carriers is referred to as a symbol.
For QPSK the speed is twice the speed of BPSK in the same bandwidth.

Figure 5.12: Quadrature Phase-shift Keying

5.4.5 MSK and GMSK Modulation - Gaussian Minimum Shift Keying
Gaussian Minimum Shift Keying, or to give it its full title Gaussian filtered Minimum Shift Keying,
GMSK, is a form of modulation used in a variety of digital radio communications systems. It has
advantages of being able to carry digital modulation while still using the spectrum efficiently. One of the
problems with other forms of phase shift keying is that the sidebands extend outwards from the main
carrier and these can cause interference to other radio communications systems using nearby channels.
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In view of the efficient use of the spectrum in this way, GMSK modulation has been used in a number of
radio communications applications. Possibly the most widely used is the GSM cellular technology which
is used worldwide and has well over 3 billion subscribers.

GMSK Basics
GMSK modulation is based on MSK, which is itself a form of continuous-phase frequency-shift keying.
One of the problems with standard forms of PSK is that sidebands extend out from the carrier. To
overcome this, MSK and its derivative GMSK can be used.

Figure 5. 13: Signal using MSK modulation

MSK and also GMSK modulation are what is known as a continuous phase scheme. Here there are no
phase discontinuities because the frequency changes occur at the carrier zero crossing points. This arises
as a result of the unique factor of MSK that the frequency difference between the logical one and logical
zero states is always equal to half the data rate. This can be expressed in terms of the modulation index,
and it is always equal to 0.5.

Figure 5.14: Spectral Density of MSK and GMSK

A plot of the spectrum of an MSK signal shows sidebands extending well beyond a bandwidth equal to
the data rate. This can be reduced by passing the modulating signal through a low pass filter prior to
applying it to the carrier. The requirements for the filter are that it should have a sharp cut-off, narrow
bandwidth and its impulse response should show no overshoot.
The ideal filter is known as a Gaussian filter which has a Gaussian shaped response to an impulse and no
ringing. In this way the basic MSK signal is converted to GMSK modulation.

Generating GMSK modulation

Figure 5.15: Generating GMSK using a Gaussian filter and VCO

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There are two main ways in which GMSK modulation can be generated. The most obvious way is to filter
the modulating signal using a Gaussian filter and then apply this to a frequency modulator where the
modulation index is set to 0.5. This method is very simple and straightforward but it has the drawback
that the modulation index must exactly equal 0.5. In practice this analogue method is not suitable because
component tolerances drift and cannot be set exactly.

Figure 5.16: Block diagram of I-Q modulator used to create GMSK

A second method is more widely used. Here what is known as a quadrature modulator is used. The
quadrature modulator uses one signal that is said to be in-phase and another that is in quadrature to this.
In view of the in-phase and quadrature elements this type of modulator is often said to be an I-Q
modulator. Using this type of modulator the modulation index can be maintained at exactly 0.5 without
the need for any settings or adjustments. This makes it much easier to use, and capable of providing the
required level of performance without the need for adjustments. For demodulation the technique can be
used in reverse.

Advantages of GMSK modulation

There are several advantages to the use of GMSK modulation for a radio communications system. One is
obviously the improved spectral efficiency when compared to other phase shift keyed modes.
A further advantage of GMSK is that it can be amplified by a non-linear amplifier and remain undistorted.
This is because there are no elements of the signal that are carried as amplitude variations. This advantage
is of particular importance when using small portable transmitters, such as those required by cellular
technology.
Non-linear amplifiers are more efficient in terms of the DC power input from the power rails that they
convert into a radio frequency signal. This means that the power consumption for a given output is much
less, and this results in lower levels of battery consumption - a very important factor for cell phones.
A further advantage of GMSK modulation again arises from the fact that none of the information is carried
as amplitude variations. This means that is immune to amplitude variations and therefore more resilient
to noise, than some other forms of modulation, because most noise is mainly amplitude based.
Generally, Phase-shift keying does not suffer the same degradation through a saturated power amplifier
as amplitude-shift keying and generally uses bandwidth more efficiently than frequency-shift keying.

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5.4.6 Quadrature Amplitude Modulation (QAM)

QAM is a form of modulation which is widely used for modulating data signals onto a carrier used for
radio communications. It is widely used because it offers advantages over other forms of data modulation
such as PSK, although many forms of data modulation operate alongside each other.
QAM is a signal in which two carriers shifted in phase by 90 degrees are modulated and the resultant
output consists of both amplitude and phase variations. In view of the fact that both amplitude and phase
variations are present it may also be considered as a mixture of amplitude and phase modulation.

Analogue and Digital QAM

QAM may exist in what may be termed either analogue or digital formats. The analogue versions of QAM
are typically used to allow multiple analogue signals to be carried on a single carrier. For example, it is
used in PAL and NTSC television systems, where the different channels provided by QAM enable it to
carry the components of chroma or colour information. In radio applications a system known as C-QUAM
is used for AM stereo radio. Here the different channels enable the two channels required for stereo to be
carried on the single carrier.
Digital formats of QAM are often referred to as "Quantised QAM" and they are being increasingly used
for data communications often within radio communications systems. Radio communications systems
ranging from cellular technology through wireless systems including WiMAX, and Wi-Fi 802.11 use a
variety of forms of QAM, and the use of QAM will only increase within the field of radio
communications.

Digital / Quantised QAM basics

Quadrature amplitude modulation, QAM, when used for digital transmission for radio communications
applications is able to carry higher data rates than ordinary amplitude modulated schemes and phase
modulated schemes. As with phase shift keying, etc., the number of points at which the signal can rest,
i.e. the number of points on the constellation is indicated in the modulation format description, e.g.
16QAM uses a 16-point constellation.
When using QAM, the constellation points are normally arranged in a square grid with equal vertical and
horizontal spacing and as a result the most common forms of QAM use a constellation with the number
of points equal to a power of 2 i.e. 2, 4, 8, 16 . . . . By using higher order modulation formats, i.e. more
points on the constellation, it is possible to transmit more bits per symbol. However the points are closer
together and they are therefore more susceptible to noise and data errors.
Phase modulation can be considered as a special form of QAM where the amplitude remains constant and
only the phase is changed. By doing this the number of possible combinations is halved.

QAM advantages and disadvantages

Although QAM appears to increase the efficiency of transmission for radio communications systems by
utilising both amplitude and phase variations, it has a number of drawbacks. The first is that it is more
susceptible to noise because the states are closer together so that a lower level of noise is needed to move
the signal to a different decision point. Receivers for use with phase or frequency modulation are both
able to use limiting amplifiers that are able to remove any amplitude noise and thereby improve the noise
reliance. This is not the case with QAM.
The second limitation is also associated with the amplitude component of the signal. When a phase or
frequency modulated signal is amplified in a radio transmitter, there is no need to use linear amplifiers,
whereas when using QAM that contains an amplitude component, linearity must be maintained.
Unfortunately, linear amplifiers are less efficient and consume more power, and this makes them less
attractive for mobile applications.
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Different Forms of QAM

A variety of forms of QAM are available and some of the more common forms include 16 QAM, 32
QAM, 64 QAM, 128 QAM, and 256 QAM. Here the figures refer to the number of points on the
constellation, i.e. the number of distinct states that can exist.
The various flavours of QAM may be used when data-rates beyond those offered by 8-PSK are required
by a radio communications system. This is because QAM achieves a greater distance between adjacent
points in the I-Q plane by distributing the points more evenly. And in this way the points on the
constellation are more distinct and data errors are reduced. While it is possible to transmit more bits per
symbol, if the energy of the constellation is to remain the same, the points on the constellation must be
closer together and the transmission becomes more susceptible to noise. This results in a higher bit error
rate than for the lower order QAM variants. In this way there is a balance between obtaining the higher
data rates and maintaining an acceptable bit error rate for any radio communications system.

QAM Applications
QAM is in many radio communications and data delivery applications. However, some specific variants
of QAM are used in some specific applications and standards.
For domestic broadcast applications for example, 64 QAM and 256 QAM are often used in digital cable
television and cable modem applications. In the UK, 16 QAM and 64 QAM are currently used for digital
terrestrial television using DVB - Digital Video Broadcasting. In the US, 64 QAM and 256 QAM are the
mandated modulation schemes for digital cable as standardised by the SCTE in the standard ANSI/SCTE
07 2000.
In addition to this, variants of QAM are also used for many wireless and cellular technology applications.

Constellation Diagrams for QAM

The constellation diagrams show the different positions for the states within different forms of QAM. As
the order of the modulation increases, and so does the number of points on the QAM constellation
diagram.
Figure 5.17, Figure 5.18 and Figure 5.19 show constellation diagrams for a variety of formats of QAM
modulation. They show the different positions, comprising of different amplitudes (distances) as against
the different angles (or phases).

Figure 5.17: Constellation Diagrams for 4-QAM and 8- QAM Signals

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Figure 5.18: Constellation Diagrams for Various 16-QAM Signals

Figure 5.19: Constellation Diagrams for 32-, 64- and 128-QAM Signals

5.5 Relationship Between Bit Rate and Baud Rate

A Bit rate is the number of bits per second whilst a baud rate is the number of signal units per second.
In the analog transmission of digital data, the signal or baud rate is less than or equal to the bit rate.

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Table 5.1: Relationship Between Baud Rate and Bit Rate

Bit
Modulation Units Bits/Baud Baud rate
Rate

ASK, FSK, 2-PSK Bit 1 N N

4-PSK, 4-QAM Dibit 2 N 2N

8-PSK, 8-QAM Tribit 3 N 3N

16-QAM Quadbit 4 N 4N

32-QAM Pentabit 5 N 5N

64-QAM Hexabit 6 N 6N

128-QAM Septabit 7 N 7N

256-QAM Octabit 8 N 8N

Example 5.1
An analog signal carries 4 bits in each signal unit. If 1000 signal units are sent per second, find the baud
rate and the bit rate.

Solution 5.1
Baud rate = 1000 bauds per second (baud/s)
Bit rate = 1000 x 4 = 4000 bps

Example 5.2
The bit rate of a signal is 3000. If each signal unit carries 6 bits, what is the baud rate?

Solution 5.2
Baud rate = 3000 / 6 = 500 baud/s

Example 5.3
If the bit rate for a 16-QAM signal is 4000 bps, what is the baud rate?

Solution 5.3
For a 16 – QAM
2𝑛 = 16 (i.e. 𝑀 = 16, 𝑤ℎ𝑒𝑟𝑒 𝑛 = log 2 𝑀)
Where n = number of bits representing 1 symbol
Thus 𝑛 = 4 (that is 4 bits = 1 symbol)
Hence 4 bit/s = 1 baud/s
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4000×1
Thus for 4000 bps = = 1000 𝑏𝑎𝑢𝑑/𝑠
4
Example 5.4
If the baud rate for a 64-QAM signal is 2000, what is the bit rate?

Solution 5.4
2𝑛 = 64
⟹𝑛=6
Thus the bit rate = 2000 × 6 = 12000

5.5.1 For an ASK

Figure 5.20: Amplitude Shift Keying

5.5.2 For a QAM

Figure 5.21: An 8-QAM

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