Introduction To Communication System

Introduction to Communication systems (ECEg-3152)
Chapter One
Introduction to Electronic communication
A communication system can be broadly defined as the transfer of information signals from a
source, located at one point in space, to a user destination, located at another point. It could be
over short or long distances, out of which communication at a distance is known as tele-
communication. In today’s world, there are number of modern communication systems in use,
which may include radio telephony and telegraph, broadcasting (both radio and television), point
to point and mobile communication, radio telemetry, internet and so on. In order to become
familiar with these communication systems, it is necessary to understand the basic building blocks
of communication systems, the concept of noise, modulation, multiplexing, and various other
systems. The following section introduces the basic building blocks of a communication system
and important concepts related to the communication system.
1.1.Elements of Communication Systems
As shown in Fig. 1.1 below, any communication system consists of basic elements such as,
information source, input transducer, transmitter block, communication channel/medium, receiver
block, output transducer and information sink/destination.
The source originates a message, such as a human voice, a television picture, a computer text
message, or data. If the data is non-electrical (human voice, teletype message, television picture),
it must be converted by an input transducer into an electrical waveform referred to as the
baseband signal or message signal. For example, a microphone is used to translate the sound into
an electronic audio signal. For TV, a camera converts the light information in the scene to a video
signal. In computer systems, the message is typed on a keyboard and converted to binary codes
that can be stored in memory or transmitted serially. In general, input transducers convert physical
characteristics (temperature, pressure, light intensity, and so on) into electrical signals.
Fig.1.1: Basic parts of communication system
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Transmitter: -The main purpose of the transmitter is to modify the message signal into a form
suitable for transmission over the channel. It may involve signal processing techniques such as,
multiplexing, modulation, encoding and amplification. Modulation is a process of mixing the
message signal with a very high frequency carrier, which is suitable for propagation. Multiplexing
is also another process that allows two or more signals to share the same medium or channel.
Communication Channel: - is the medium by which the electronic signal is sent from one place
to another. Many different types of media are used in communication systems, including wire
conductors, fiber-optic cable, and free space.
 Electrical Conductors: - In its simplest form, the medium may simply be a pair of wires
that carry a voice signal from a microphone to a headset. It may be a coaxial cable such as
the one used to carry cable TV signals. Or it may also be a twisted-pair cable used in a
local-area network (LAN).
 Optical Media: - The communication medium may also be a fiber-optic cable or “light
pipe” that carries the message on a light wave. These are widely used today for long
distance communications due to their high bandwidth and relatively small signal loss
(0.2dB/km).
 Free Space: - When free space is the medium, the resulting system is known as radio. Also
known as wireless, radio is the broad general term applied to any form of wireless
communication from one point to another. Radio makes use of the electromagnetic
spectrum. Information signals are converted to electric and magnetic fields that propagate
nearly instantaneously through space over long distances. Communication by visible or
infrared light also occurs in free space.
Various unwanted and undesirable effects crop up in the course of signal transmission.
Attenuation is undesirable since it reduces signal strength at the receiver. More serious, however,
are distortion, interference, and noise, which appear as alterations of the signal shape. Although
such contaminations may occur at any point, the standard convention is to blame them entirely on
the channel, treating the transmitter and receiver as being ideal.
 Noise refers to random and unpredictable electrical signals produced by natural processes
both internal and external to the system. When such random variations are superimposed
on an information-bearing signal, the message may be partially corrupted or totally
obliterated. Filtering reduces noise contamination, but there inevitably remains some
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amount of noise that cannot be eliminated. This noise constitutes one of the fundamental
system limitations.
 Interference is contamination by extraneous signals from human sources, other
transmitters, power lines and machinery, switching circuits, and so on. Interference occurs
most often in radio systems whose receiving antennas usually intercept several signals at
the same time. Appropriate filtering removes interference to the extent that the interfering
signals occupy different frequency bands than the desired signal.
 Distortion is waveform perturbation caused by imperfect response of the system to the
desired signal itself. Unlike noise and interference, distortion disappears when the signal is
turned off. If the channel has a linear but distorting response, then distortion may be
corrected, or at least reduced, with the help of special filters called equalizers.
Receiver: - The main purpose of the receiver is to reproduce the original message signal from the
degraded version of the transmitted signal after propagation through the channel. This is
accomplished by using a process of demodulation and amplification. Demodulation is a reverse of
modulation. It is a process of extracting the original message from the received signal.
Output Transducer. This device converts the electric signal at its input into the form desired by
the system user. Perhaps the most common output transducers may include a loudspeaker or ear
phone, monitor, etc.
The destination is the unit to which the message is communicated.
1.2.Classification of Communication Systems
There are three ways in which communication systems are classified: analog or digital systems,
one-way (simplex) or two-way (half & full duplex) systems, and base band or modulated systems.
According to the message signal communicated, communication systems can be classified in to
two types: Analog and Digital. Analog communication system is the one in which message signal
is transmitted and received in analog form. Digital communication systems are systems in which
message signal is transmitted and received in digital form. Analog systems were the first to be
developed, however in recent years digital systems have become more popular due to its superior
performance.
There are also two basic types of communication systems. The simplest is one-way
communication, normally referred to as simplex where information travels in one direction only.
For example radio and TV broadcasting are simplex. The bulk of communication systems,
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however, are two way communication systems which are referred as duplex. We could have two
types of duplex communication systems. The first one is known as half duplex communication in
which only one party is able to transmit at a time. The communication is two-way, but the direction
alternates: the communicating parties take turns transmitting and receiving. Most radio
transmissions, such as those used in the military, police, aircraft, marine, and other services, are
examples of half duplex communication. The second one is known as full duplex communication
in which both parties are able to transmit and receive at a time. For example, people
communicating with one another over the telephone can talk and listen simultaneously.
In a communication system, the information signal may be transmitted by itself over the medium
or may be used to modulate a carrier for transmission over a long distance. The former is a base
band communication while the later is a band pass (modulated signal) communication.
1.3.Typical Communication Systems
Analog Communication System
The block diagram of a typical analog communication system is shown in Fig 1.2 below. The
analog signal to be transmitted can be a voice waveform, television signal, or any other
information-bearing signal. Typically, this message signal must first be filtered to eliminate
undesired components and amplified to a suitable level, depending on the source.
Analog Modulation Power Amp

Message IN
Carrier Osc. Channel
Analog Filter Demodulation LNA

Message Out
Fig. 1.2: Typical Block diagram of Analog communication Systems

The message signal often modulated onto a carrier, which can be a sinusoidal signal, pulse train,
or a light wave. In the modulation process, the signal affects some parameter of the carrier in a
predetermined way. The modulated signal is then amplified and radiated from an antenna. Various
things may happen to the signal in transmission through the channel.
The corrupted signal received from the channel are amplified to a suitable level and filter to
eliminate noise and interfering signals that are all outside the frequency range of the desired signal.
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The amplified signal is then demodulated to recover the original message. After it is filtered the
original signal will be reproduced.
Digital Communication System
Nowadays digital communication systems are found better in performance than analog
communication systems, and therefore communication systems are becoming fully digital. Digital
systems demand efficient and reliable digital data transmission and storage systems. This demand
has been accelerated by the emergence of large scale, high speed data networks for the exchange,
processing and storage of digital information in military, government and private spheres. A
merging of communication and computer technology is also required in the design and
implementations of these systems.
A major concern of the designer in digital systems is the control of error so that reliable
reproduction of data can be obtained. From Shannon's theory, it is known that by proper encoding
of the information, errors induced by a noisy channel or storage medium can be reduced to any
desired level without sacrificing the rate of information transmission and storage. Thus the use of
coding for error control has become an integral part in the design of modern digital communication
systems and digital computers. A typical digital communication system is shown in Fig 1.3.
Fig. 1.3: Typical Block diagram of digital Communication Systems

The analog to digital converter (ADC) converts the source information in to digital form if it has
been analog. Then the source encoder further compresses the incoming digital signal for efficient
transmission over the channel. The channel encoder further encodes the source coded information
using a certain coding technique so that errors can be detected and corrected at the receiver. The
coded digital signal is then sent to the digital modulator and it would be modulated using a certain
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type of digital modulator. The amplifier will amplify and send the modulated signal through the
communication channel.
In the receiver side, the first stage receives a very weak signal and amplifies it. Then the
demodulator section separates the coded digital information from the carrier. The channel decoder
decodes the coded digital signal in its un-coded form and detects and corrects errors found in the
signal. The source decoder finally converts the digital information back to the original form.
Comparison of Digital and Analog communication Systems
Digital communication has a number of advantages
 Relatively inexpensive digital circuits can be used
 Privacy is preserved by using data encryption
 Greater dynamic range is possible
 Data from voice, video, and data sources can be merged and transmitted over a common
digital transmission system.
 In long distance systems, noise does not accumulate from repeater to repeater.
 Errors in detected data are small, even when there is a large amount of noise on the received
signal.
 Errors can often be corrected by the use of coding
Digital communication also has disadvantages
 Generally more bandwidth is required than that for analog systems.
 Synchronization between the transmitter and receiver is required.
1.4 Base Band Transmission
In a communication system, the original information signal (baseband signals) could be
transmitted over the medium. Putting the original signal directly into the medium is referred to as
baseband transmission. A common example is telephony, especially for local calls. Here the
voice signal converted into electrical form, is placed on the wires and transmitted over some
distance to the receiver. Also in some computer networks, digital signals could be applied directly
to coaxial cables for transmission to another computer, making it another example of baseband
transmission.
Limitations of Baseband Transmission
There are many instances when the baseband signals are incompatible for direct transmission over
the communication medium. Although it is theoretically possible to transmit voice signals directly
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by radio, realistically it is impractical. For example, voice signals can’t travel longer distances in
air as they get attenuated rapidly. Hence to overcome the limitations of baseband transmission,
modulation techniques has to be used.
1.5 Modulation Techniques
In modulation process, the baseband signal (such as voice, video, etc.) modifies another higher-
frequency signal called carrier which is usally a sinusoidal wave that is higher in frequency than
the highest baseband signal frequency. The baseband signal modifies the amplitude or frequency
or phase of the carrier in the modulation process.
1.5.1 Need for Modulation
We have seen that baseband signals are incompatible for direct transmission over the medium and
therfore we have to use modulation techniques for the communication of baseband signal. The
advantages of using modulation technique are as given below;
 Reduce the height of antenna
 Avoids mixing of signals
 Increase the range of communication
 Allows multiplexing of signals
 Allows adjustments in the bandwidth
1. Reduces the height of antenna
The height of the antenna required for transmission and reception of radio waves in wireless
transmission is proportional to one-forth of its wavelength, where the wavelength is itself gives as;
𝑐
𝜆=
𝑓
Where c is the speed of light and f is the frequency of the information signal
From the above equation it can be easily noticed that at low frequencies wavelength is very high
and hence the antenna height. For example, consider the baseband signal with 𝑓 = 15𝑘𝐻𝑧.
𝜆 𝑐 3 × 108 𝑚⁄𝑠
𝐻𝑒𝑖𝑔ℎ𝑡 𝑜𝑓 𝑎𝑛𝑡𝑒𝑛𝑛𝑎 = = = = 5000 𝑚𝑒𝑡𝑒𝑟𝑠
4 4 × 𝑓 15 × 103 × 4
This height of vertical antenna is unthinkable. On the other hand, if we consider a modulated signal
with 100 MHz frequency, the height of the antenna could become;
𝜆 𝑐 3 × 108 𝑚⁄𝑠
𝐻𝑒𝑖𝑔ℎ𝑡 𝑜𝑓 𝑎𝑛𝑡𝑒𝑛𝑛𝑎 = = = = 0.75 𝑚𝑒𝑡𝑒𝑟𝑠
4 4 × 𝑓 1 × 108 × 4 1⁄𝑠
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This height of antenna is practical and can easily be installed.

2. Avoids mixing of signals
All sound signals are concentrated within the range from 20Hz to 20KHz. The transmission of
baseband signals from various sources causes the mixing of signal and then it is difficult to separate
at the receiver end.
In order to separate the various signals, it is necessary to translate them to different portions of the
electromagnetic spectrum (channel); each must be given its own bandwidth commonly known as
channel bandwidth. This can be achieved by taking different carrier frequency for different signal
source as shown in Fig.1.4. Once the signals have been transmitted, a tuned circuit at the receiver
end selects the portion of the electromagnetic spectrum it is tuned for. Therefore modulating
different signal sources by different carrier frequencies avoid mixing of signals.
Fig. 1.4: Illustration of frequency translation to avoid mixing of sinal

3. Increases the range of communication
At low frequencies, radiation is poor and signal gets highly attenuated. Therefore besaband
signals can’t be transmitted directly over long distance. Modulation effectively increase the
frequency of the signal to be radiated and thus increase the distance over which signals can
be transmitted faithfully.
4. Allows multiplexing of signals
Modulation permits multiplexing to be used. Multiplexing means transmission of two or more
signals simultaneously over the same channel. The common examples of multiplexing are the
number of Television channels operating simultanously or number of radio stations broadecasting
the signal in MW and SW band, simultaneously.
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The different signals from different stations can be separated in the receiver since the carrier
frequencies for these signals are different. It is commonly known as tuning the same the receiver
to the desired station. By tuning process, the desired signal is selected and at the same time, other
unwanted signals are rejected.
5. Allows adjustments in the bandwidth
Bandwidth of a modulated signal may be made smaller or larger than the origional signal. Signal
to noise ratio in the reciever which is a function of the signal bandwidth can thus be improved by
proper control of the bandwidth at the modulating stage.
6. Improve quality of reception
The signal communication using modulation techniques such as frequency modulation and pulse
code modulation reduce the effect of noise to great extent. Reduction in noise improves the quality
of reception.
Modulation Types
There are many modulation and demodulation techniques;
 Continuous wave modulation- DSB, DSB-SC, SSB, VSB, FM, PM
 Pulse modulation- PAM, PWM, PPM, PCM, DPCM, DM
 Digital modulation- ASK, FSK, PSK, QAM
1.6.Frequency Allocations
The frequency spectrum is divided into segments for the purpose of classifying the various portions
the frequency band. This allocation helps to provide some semblance of order and to minimize
interference, specify the modulation types, bandwidth, power, and type of information that a user
can transmit over designated frequency bands. These frequency assignments and technical
standards are set internationally by the International Telecommunications Union (ITU), which is
a specialized agency of the United Nations.
The following table gives a general listing of frequency bands, their common designations, and
typical services assigned to these bands.
Band Name Frequency Wavelength Applications
Extremely Low frequency 30-300Hz 107-106m Ac line frequency, low end of human
(ELF) hearing
Voice frequency (VF) 300-3000Hz 106-105m Normal range of human speech.
Very Low Frequency (VLF) 3-30KHz 105-104m Higher end of human hearing, sounds
from musical instruments
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Low frequency (VF) 30-300KHz 104-103m Used as sub carrier, and also in marine
navigation.
Medium frequency (MF) 300- 103-102m MW AM radio broadcasting
3000KHz
High frequency (HF) 3-30MHz 102-101m SW AM radio, Two way
communications
Very High frequency (VHF) 30-300MHz 101-1m FM radio, VHF TV channels, marine
and aeronautical Communication,
Mobile
Ultra High frequency (UHF) 300-3GHz 1-10-1m UHF TV channels, Cellular Mobile,
Super High frequency (SHF) 3-30GHz 10-1-10-2m Microwave frequency used in satellite,
radar and long distance communication.
Extremely High frequency 30-300GHz 10-2-10-3m Limited activities so far
(EHF)
Infrared 43-430TH z 0.7 – 10 m In astronomy to detect stars, TV remote
control, To guide weapons
The Visible Spectrum 430-750THz 0.4 –0.8 m Optical communication
1.7.Fundamental Limitations of Communication Systems

The goal of a communication system engineer is to design systems that provide high quality service
for the maximum number of user with the smallest cost and least usage of limited resources. The
resources to be conserved include hardware for generating, transmitting, and receiving information
signal, the channel bandwidth, and the transmitter power. In other words, engineers attempt to
design communication systems that transmit information at a high rate, with high performance,
using the minimum amount of transmitted power and bandwidth.
Given these requirements, communication systems face fundamental limitations such as noise,
distortion and bandwidth which determine the performance each system. Usually, the transmitter
and the receiver are carefully designed so as to minimize the effects of these limitations on the
quality of reception.
Bandwidth of a communication system is the range of frequencies that the signal can pass through.
On the other hand, information capacity of a communication system is a measure of how much
information can be carried through the system in a given period of time. It is a function of system
bandwidth.
The main question related to information capacity and system limitations could be, is it possible
to invent a system with no bit error at the output even when we have noise introduced in to the
channel? The answer to this question is stated by Shannon-Hartely capacity theorem, according
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to which communication systems that attain as close to zero error probability as described are
theoretically possible, provided that the rate of information transmitted is less than the capacity of
the channel C. In other words, Shannon showed that if the rate of information R (in b/s) is less
than C, the probability of error would approach zero. The channel capacity C (in b/s) could be
calculated by using the equation given below which is referred to as Shannon equation. Here, B is
bandwidth in Hz, SNR is signal- to-noise power ratio at the receiver input and as already mentioned
C is the channel capacity. Signal-to-noise power ratio indicates the measure of noise power relative
to information signal power.
𝐶 = 𝐵 log 2 (1 + 𝑆𝑁𝑅)
In analog systems the optimum system might be defined as the one that achieves the largest signal
to noise ratio at the receiver output subject to design constraints such as channel bandwidth and
transmitted power. Another question could be, is it possible to design a system with infinite signal
to noise ratio at the output when noise is introduced by the channel? The answer is of course no.
1.8. Signal Distortion in Transmission
A signal transmission medium is the electrical channel between an information source and
destination. These systems range in complexity from a simple pair of wires to a sophisticated laser-
optics links. But all transmission systems have two physical attributes of particular concern in
communication: internal power dissipation that reduces the size of the output signal, and energy
storage that alters the shape of the output (distortion). This section deals with signal distortion in
transmission.
Distortion-less Transmission
Distortion-less transmission means that the output signal has the same shape as the input. More
precisely, given an input signal 𝑥(𝑡), we say that the output is undistorted if it differs from the
input by a multiplying constant and a finite time delay.
Analytically, we have distortion-less transmission if its output is related to its input as;
𝑦(𝑡) = 𝐾𝑥(𝑡 − 𝑡𝑑 )
Where K and td are constants.
The properties of a distortion-less system are easily found by examining the output spectrum.
𝑌(𝑓) = 𝐾𝑒 −𝑗𝜔𝑡𝑑 𝑋(𝑓)
Now by definition of transfer function, 𝑌(𝑓) = 𝐻(𝑓)𝑋(𝑓), so
𝐻(𝑓) = 𝐾𝑒 −𝑗𝜔𝑡𝑑
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In word, a system giving distortion-less transmission must have constant amplitude response and
negative linear phase shift, so
|𝐻(𝑓)| = |𝐾| 𝑎𝑛𝑑 arg 𝐻(𝑓) = −2𝜋𝑡𝑑 𝑓 ± 𝑚 × 180𝑜
The stringent demands of distortion-less transmission can only be satisfied approximately in
practice, so transmission systems always produce some amount of signal distortion. For the
purpose of studying distortion effects on various signals, we'll define three major types of
distortion:
1. Amplitude distortion, which occurs when
|𝐻(𝑓)| ≠ |𝐾|
2. Delay distortion, which occurs when
arg 𝐻(𝑓) ≠ −2𝜋𝑡𝑑 𝑓 ± 𝑚 × 180𝑜
3. Nonlinear distortion, which occurs when the system includes nonlinear elements.
The first two types can be grouped under the general designation of linear distortion, described in
terms of the transfer function of a linear system. For the third type, the nonlinearity prevents the
existence of a transfer function.
Linear Distortion
Linear distortion includes any amplitude or delay distortion associated with a linear transmission
system. Amplitude distortion is easily described in the frequency domain; it means simply that the
output frequency components are not in correct proportion. Since this is caused by |𝑯(𝒇)| not
being constant with frequency, amplitude distortion is sometimes called frequency distortion.
The most common forms of amplitude distortion are excess attenuation or enhancement of extreme
high or low frequencies in the signal spectrum. Less common but equally bothersome is
disproportionate response to a band of frequencies within the spectrum. For illustration, a suitably
1 1
simple test signal is 𝑥(𝑡) = 𝑐𝑜𝑠(𝑤0 𝑡) − 𝑐𝑜𝑠(3𝑤0 𝑡) + 𝑐𝑜𝑠(5𝑤0 𝑡), a rough approximation
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to a square wave sketched in Fig. 1.6. If the low-frequency or high-frequency component is

attenuated by one-half, the resulting outputs are as shown in Fig. 1.7. As expected, loss of the high-
frequency term reduces the "sharpness" of the waveform.
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1 1
Fig. 1.6: Test signal 𝑥(𝑡) = 𝑐𝑜𝑠(𝑤0 𝑡) − 𝑐𝑜𝑠(3𝑤0 𝑡) + 𝑐𝑜𝑠(5𝑤0 𝑡)
3 5
Fig. 1.7: (a) Low frequency attenuated; (b) high Frequency attenuated
A common area of confusion is constant time delay versus constant phase shift. The former is desirable
and is required for distortion-less transmission. The latter, in general, causes distortion. Suppose a
system has the constant phase shift not equal to 0° or + m180°. Then each signal frequency component
will be delayed by 𝜃⁄2𝜋 cycles of its own frequency; this is the meaning of constant phase shift. But
the time delays will be different, the frequency components will be scrambled in time, and distortion
will result.
The constant phase shift that gives distortion could be simply illustrated by returning to the test signal
of Fig. 1.6 and shifting each component by one-fourth cycle  = −90°. Whereas the input was roughly
a square wave, the output will look like the triangular wave in Fig. 1.8.
Fig. 1.8: Test signal with constant phase shift  = −90°.
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Nonlinear Distortion
A system having nonlinear elements cannot be described by a transfer function. Instead, the
instantaneous values of input and output are related by a curve or function 𝑦(𝑡) = 𝑇[𝑥(𝑡)], commonly
called the transfer characteristic. Fig. 1.9 shows a representative transfer characteristic; the
flattening out of the output for large input excursions is the familiar saturation-and cut-off effect of
transistor amplifiers. We'll consider only memory-less devices, for which the transfer characteristic is
a complete description.
Under small-signal input conditions, it may be possible to linearize the transfer characteristic in a
piecewise fashion, as shown by the thin lines in the figure. The more general approach is a
polynomial approximation to the curve, of the form.
𝑦(𝑡) = 𝑎1 𝑥(𝑡) + 𝑎2 𝑥 2 (𝑡) + 𝑎3 𝑥 3 (𝑡) + 𝑎4 𝑥 4 (𝑡) + ⋯
And the higher powers of 𝑥(𝑡) in this equation give rise to the nonlinear distortion. Even though
we have no transfer function, the output spectrum can be found, at least in a formal way, by
transforming the above equation. Specifically, invoking the convolution theorem,
𝑌(𝑓) = 𝑎1 𝑋(𝑓) + 𝑎2 𝑋 ∗ 𝑋(𝑓) + 𝑎3 𝑋 ∗ 𝑋 ∗ 𝑋(𝑓) + 𝑎4 𝑋 ∗ 𝑋 ∗ 𝑋 ∗ 𝑋(𝑓) + ⋯
Fig. 1.9: Transfer characteristic of a nonlinear device
Equalization
Linear distortion-both amplitude and delay-is theoretically curable through the use of equalization
networks.
Channel Equalizer
𝑥(𝑡) 𝐻𝐶 (𝑓) 𝐻𝑒𝑞 (𝑓) 𝑦(𝑡)
Fig. 1.10 Channel with equalization for linear distortion
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Fig. 1.10 shows an equalizer 𝐻𝑒𝑞 (𝑓) in cascade with a distorting transmission channel 𝐻𝐶 (𝑓).
Since the overall transfer function is 𝐻(𝑓) = 𝐻𝑒𝑞 (𝑓)𝐻𝐶 (𝑓), the final output will be distortion-less
if 𝐻𝑒𝑞 (𝑓)𝐻𝐶 (𝑓) = 𝐾𝑒 −𝑗𝜔𝑡𝑑 , where K and td are more or less arbitrary constants.
Companding
Although nonlinear distortion has no perfect cure, it can be minimized by careful design. The basic
idea is to make sure that the signal does not exceed the linear operating range of the channel's
transfer characteristic. Ironically, one strategy along this line utilizes two nonlinear signal
processors, a compressor at the input and an expander at the output, as shown in Fig. 1.9.
𝑥(𝑡) Compressor Channel Expander 𝑦(𝑡)
Fig.1.11: Companding System

A compressor has greater amplification at low signal levels than at high signal levels, and thereby
compresses the range of the input signal. Ideally, then, the expander has a characteristic that
perfectly complements the compressor so the expanded output is proportional to the input, as
desired. The joint use of compressing and expanding is called companding.
1.9 Electromagnetic Wave Propagation Models
Radio communication involves transmission, emission, or reception of signs, signals, writing,
images, sounds or intelligence of any nature by means of EM waves of frequencies lower than 300
GHz propagated in space without artificial guide.
The propagation of EM waves between a transmitting and a receiving antenna is usually influenced
by many different phenomena as the transmission paths between them may vary from simple line-
of-sight to severely obstructed (buildings, mountains, foliage, etc.). The “propagation channel”
places fundamental limitations on the performance of wireless communication systems. Its
modeling is a very complex task. The analysis is typically based on a combination of simplified
(statistical) physical models and empirical knowledge. Depending on the distance involved and
the frequency of the radiated waveform, a terrestrial communication link may depend on line-of-
sight, ground-wave, or sky-wave propagation. The next section deals with the introduction of these
three dominant propagation characteristics.
Ground wave propagation (illustrated in Fig. 1.11) is the dominant mode of propagation for
frequencies below 2 MHz. Here, the electromagnetic wave tends to follow the contour of the Earth.
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That is, diffraction of the wave causes it to propagate along the surface of the Earth. This is the
propagation mode used in AM broadcasting, where the local coverage follows the Earth’s contour
and the signal propagates over the visual horizon.
Fig. 1.12 Ground-Wave Propagation (Below 2 MHz)

Sky-wave propagation is illustrated in Fig. 1.13. It is the dominant mode of propagation in 2- 30
MHz frequency range. Here, long-distance coverage is obtained by reflecting the wave at the
ionosphere, and at the Earth’s boundaries. Actually, in the ionosphere the waves are refracted (i.e.,
bent) gradually, because the index of refraction varies with altitude as the ionization density
changes.
Fig. 1.13: Sky-Wave Propagation (2 to 30 MHz)

LOS propagation (illustrated in Fig. 1.14) is the dominant mode for frequencies above 30 MHz.
Here, the electromagnetic wave propagates in a straight line and there is very little refraction by
the ionosphere. In fact, the signal will propagate through the ionosphere. This propagation
mechanism is used for satellite communications.
16 | P a g e
Fig. 1.14: Line-of-Sight (LOS) Propagation (Above 30 MHz)

The LOS mode has the disadvantage that, for communication between two terrestrial (Earth)
stations, the signal path has to be above the horizon. Otherwise, the Earth will block the LOS path.
Thus, antennas need to be placed on tall towers so that the receiver antenna can “see” the
transmitting antenna.
17 | P a g e
Introduction to Communication Systems (ECEg-3152)
Chapter Two
Amplitude Modulation and Demodulation
Before an information-bearing signal is transmitted through a communication channel, some type
of modulation process is typically utilized to produce a signal that can easily be accommodated by
the channel. Modulation is defined as the process by which some characteristic of a carrier wave
is varied in accordance with an information-bearing signal. The carrier is needed to facilitate the
transportation of the modulated signal across a band-pass channel from the transmitter to the
receiver. A commonly used carrier is a sinusoidal wave, the source of which is physically
independent of the source of the information-bearing signal.
The modulation process is important for the following reasons:
 To radiate from an antenna of reasonable size.
 To enable more than one user to communicate over a channel at one time by
selecting different carrier frequency (FDM).
 To increase range of communication
 For effective radiation.
There are three different types of modulation techniques:
1. Analog Modulation: - is a process of changing amplitude, frequency or phase of an analog
carrier in accordance with analog message signal. It has three different forms: Amplitude
Modulation (AM), Frequency Modulation (FM), Phase Modulation (PM).
2. Pulse Modulation: - is method of converting message signal in to pulse forms for
transferring pulses from a source to a destination. The predominant methods are; Pulse
Amplitude Modulation (PAM), Pulse Position Modulation (PPM), Pulse Width
Modulation (PWM) and Pulse Code Modulation (PCM), Differential PCM (DPCM), Delta
Modulation (DM).
3. Digital Modulation: - is the same as analog modulations but the modulating signals are
digital signals and thus the modulation type is different. The predominant methods in
digital modulation are Amplitude Shift Keying (ASK), Frequency Shift Keying (FSK),
Phase Shift Keying (PSK), and Quadrature Amplitude Modulation (QAM).
Amplitude Modulation (AM)
AM is analog modulation, which is the process of changing the amplitude of a relatively high
frequency carrier signal in accordance with the amplitude of the modulating message signal. AM
1|Page
is a relatively inexpensive, low quality form of modulation that is used for broadcasting of both
audio and video signals.
There are four types of AM:
1. Double Side Band-with carrier (DSB-with carrier) AM
2. Double Side Band-Suppressed Carrier (DSB-SC) AM
3. Single Side Band (SSB) AM
4. Vestigial Side Band (VSB) AM
These four types of modulation differ from each other by virtue of their spectral characteristics.
2.1 DSB-with Carrier Amplitude Modulation - Standard AM
This is the form of modulation used for commercial AM broadcasting. It has the advantage that
the receiver is extremely simple (good for commercial applications, since radio receivers can be
made very cheaply). However, we will see the power efficiency at the transmitter is very poor.
Let the carrier be 𝑐(𝑡) = 𝐴𝑐 cos 2𝜋𝑓𝑐 𝑡 and message signal be 𝑚(𝑡) = 𝐴𝑚 cos 2𝜋𝑓𝑚 𝑡. The block
diagram of an AM modulation system is shown in Fig. 2.1.
AM Modulator 𝑠(𝑡)
𝑐(𝑡)
𝑚(𝑡)
Fig. 2.1: Double-sideband modulation

Mathematically, standard AM wave 𝑠(𝑡) is described by
s(t )  Ac [1  ka m(t )]cos 2f ct (2.1)
   
envelop carrier
Where 𝐴𝑐 is un-modulated carrier amplitude, 𝑚(𝑡) is the message signal (voice, music, data,
etc.), 𝑓𝑐 is carrier frequency and 𝑘𝑎 is a constant called amplitude sensitivity of the modulator
responsible for the generation of the modulated signal.
2|Page
Fig 2.2: amplitude modulation

The amplitude of the envelope, 𝑎(𝑡) is given as
𝑎(𝑡) = 𝐴𝑐 [1 + 𝑘𝑎 𝑚(𝑡)] (2.2)
Which as we see varies in accordance with the message signal 𝑚(𝑡). Here, we assume
k a m(t )  1 for all t. (2.3)
Fig 2.2 shows a typical baseband message waveform 𝑚(𝑡) (part a), a high frequency carrier
waveform 𝑐(𝑡) (part b) and the resulting AM wave (part c), in which sine wave is used to represent
both the message and carrier signals. Note the envelope (amplitude) of the resulting AM wave
varies in accordance with 𝑚(𝑡), hence, 𝑚(𝑡) can be recovered from the envelope of 𝑠(𝑡). Note
also that the envelope function from Eq. (2.2) may be generated in the following way:
1. Multiply the message waveform 𝑚(𝑡) by a suitably small constant 𝑘𝑎 so that the value
𝑘𝑎 𝑚(𝑡) is small in comparison to unity for all time.
2. Add a DC value of 1 Volt to 𝑘𝑎 𝑚(𝑡).
3. Multiply (scale) the resulting signal [1 + 𝑘𝑎 𝑚(𝑡)] by a content Ac to bring the signal up
to a desired level.
4. The envelope from part 3 is then multiplied by a carrier (cosine wave) at the desired
frequency.
3|Page
Fig 2.3 shows the case where Eq. (2.3) is violated; i.e. 𝑘𝑎 𝑚(𝑡) > 1, when 𝑚(𝑡) < 0. This causes
the amplitude of 𝑠(𝑡) to go negative during this interval, which results in a 1800 phase reversal in
the carrier waveform. Note that this condition results in distortion of the envelope of 𝑠(𝑡).
Fig. 2.3: Phase reversal of AM modulated wave

The envelop of the modulated wave has the same shape as the baseband signal 𝑚(𝑡) provided two
requirements are satisfied:
i. |𝑘𝑎 𝑚(𝑡)| < 1 for all t. This assure that [1 + 𝑘𝑎 𝑚(𝑡)] > 1, avoiding phase reversal
of 𝑐(𝑡).
ii. 𝑓𝑐 >> 𝑤, where 𝑤 is the highest frequency component of 𝑚(𝑡). Otherwise, the
envelope cannot be visualized and hence, cannot be detected satisfactorily.
Modulation Index and Percentage of Modulation
The maximum absolute value of the quantity 𝑘𝑎 𝑚(𝑡) is called modulation index. If it is multiplied
by 100, the result is referred to as the percentage modulation. Modulation index is a factor that
shows the degree of modulation. If modulation index is greater than 1, the message signal is said
to be over-modulated and the process is called over-modulation. If it is 1 the message signal is said
to be 100% modulated and the process is called 100% modulation. Else if the modulation index is
less than 1, the message signal is said to be under-modulated and the process is called under-
modulation. Fig. 2.4 shows examples of under-modulated (part a), 100% modulated (part b), and
overmodulated (part c) waveforms.
The ideal condition for AM is at 100 percent modulation which results in the greatest output power
at the transmitter and the greatest output voltage at the receiver, with no distortion. Over
modulation however, causes distortion and affects the performance of a communication system.
4|Page
Fig. 2.4: examples of under modulated, 100% modulated and over modulated waveforms
Frequency Domain Description
Let 𝑠(𝑡) ⇔ 𝑆(𝑓) and 𝑚(𝑡) ⇔ 𝑀(𝑓) be Fourier transform pairs and also 𝑚(𝑡) be a band limited
signal, what does the spectrum 𝑆(𝑓) look like for a specific message spectrum 𝑀(𝑓)?
Rewriting the definition of 𝑠(𝑡), we have;
𝑠(𝑡) = 𝐴𝑐 [1 + 𝑘𝑎 𝑚(𝑡)]𝑐𝑜𝑠 2𝑓𝑐 𝑡 = A𝑐 cos2f𝑐 t + A𝑐 k 𝑎 m(t)cos 2f𝑐 t (2.4)
𝐴𝑐
The frequency domain representation of the first term is a set of -functions of amplitude at
2
frequencies 𝑓𝑐 . Using the frequency-shifting property of the Fourier transform for the second
term, we have;
𝐴𝑐 (2.5)
𝐴𝑐 𝑘𝑎 𝑚(𝑡) cos 2𝜋𝑓𝑐 𝑡 ⇔ 𝑘 [𝑀(𝑓 − 𝑓𝑐 ) + 𝑀(𝑓 + 𝑓𝑐 )]
2 𝑎
This is very important result: multiplication of 𝑚(𝑡) in the time domain by 𝑐𝑜𝑠2𝑓𝑐 𝑡 shifts 𝑀(𝑓)
upwards and downwards by 𝑓𝑐 Hz. Combining these two terms together, we have:
𝐴𝑐 𝐴𝑐 𝑘𝑎 (2.6)
𝑆(𝑓) = [𝛿(𝑓 − 𝑓𝑐 ) + 𝛿(𝑓 + 𝑓𝑐 )] + [𝑀(𝑓 − 𝑓𝑐 ) + 𝑀(𝑓 + 𝑓𝑐 )]
2 2
This spectrum is shown in Fig 2.5 for a generic-type spectrum 𝑀(𝑓). This spectrum contains the
𝐴𝑐 𝑘𝑎
message spectrum shifted upwards and downwards by 𝑓𝑐 , weighted by the factor . It also
2
𝐴𝑐
contains two delta-functions of weight at frequencies  𝑓𝑐 . These -functions are the most
2
predominant components present, yet they carry no information. Thus, we see that an AM
modulation is wasteful in terms of the power of the overall modulated signal to power in the
message component, i.e. 𝑀(𝑓) only.
5|Page
Fig 2.5 (a) Spectrum of message signal (b) Spectrum of AM wave.

As shown in Fig. 2.5(b), the portion of the spectrum of an AM wave above 𝑓𝑐 for positive frequency
and below −𝑓𝑐 for negative frequencies is referred to as upper side band (USB) and the portion
of the spectrum of an AM wave below 𝑓𝑐 for positive frequency and above −𝑓𝑐 for negative
frequencies is referred to as lower side band (LSB).
For positive frequencies, the highest frequency component of the AM wave is 𝑓𝑐 + 𝑊 and the
lowest frequency component is 𝑓𝑐 − 𝑊. The difference between these two frequencies define the
transmission bandwidth of the AM wave and it is exactly equal to twice the highest frequency of
the message signal i.e., 𝐵𝑇 = 2𝑊.
Single Tone Modulation
To visualize the AM process further, we examine the time and frequency domain representations
of 𝑠(𝑡) when the message signal 𝑚(𝑡) is a sinusoid. Let 𝑚(𝑡) = 𝐴𝑚 𝑐𝑜𝑠(2𝑓𝑚 𝑡), where 𝑓𝑐 >>
𝑓𝑚 . The resulting AM waveform is then,
𝑠(𝑡) = 𝐴𝑐 [1 + 𝐴𝑚 𝑘𝑎 cos 2𝜋𝑓𝑚 𝑡] cos 2𝜋𝑓𝑐 𝑡 (2.7)
Using the standard trigonometric expansion for the cos (A) cos (B) term in Eq. (2.7), we have
𝑠(𝑡) = 𝐴𝑐 cos 2𝜋𝑓𝑐 𝑡 + 𝐴𝑐 𝐴𝑚 𝑘𝑎 cos(2𝜋𝑓𝑚 𝑡) cos(2𝜋𝑓𝑐 𝑡)
6|Page
𝐴𝑐 𝐴𝑚 𝑘𝑎 𝐴𝑐 𝐴𝑚 𝑘𝑎 (2.8)
𝑠(𝑡) = 𝐴𝑐 cos 2𝜋𝑓𝑐 𝑡 + cos 2𝜋(𝑓𝑐 − 𝑓𝑚 )𝑡 + cos 2𝜋(𝑓𝑐 + 𝑓𝑚 )𝑡
2 2
Thus, we see that an AM waveform, modulated by a single-tone message 𝑚(𝑡), consists of three
components:
i. The carrier component (sinusoidal in time domain) at 𝑓𝑐 of weight 𝐴𝑐 (or two -
𝐴𝑐
functions in frequency domain at  𝑓 each with weight .
2
𝐴𝑐 𝐴𝑚 𝑘𝑎
ii. A message component at 𝑓𝑐 − 𝑓𝑚 with weight in time domain (LSB).
2
𝐴𝑐 𝐴𝑚 𝑘𝑎
iii. A message component at 𝑓𝑐 + 𝑓𝑚 with weight in time domain (USB).
2
Taking the Fourier transform Eq. (2.8) gives;

1 1 (2.9)
𝑆(𝑓) = 𝐴𝑐 [𝛿(𝑓 − 𝑓𝑐 ) + 𝛿(𝑓 + 𝑓𝑐 )] + 𝐴𝑐 𝐴𝑚 𝑘𝑎 [𝛿(𝑓 − 𝑓𝑐 − 𝑓𝑚 ) + 𝛿(𝑓 + 𝑓𝑐 + 𝑓𝑚 )]
2 4
1
+ 𝐴𝑐 𝐴𝑚 𝑘𝑎 [𝛿(𝑓 − 𝑓𝑐 + 𝑓𝑚 ) + 𝛿(𝑓 + 𝑓𝑐 − 𝑓𝑚 )]
4
Thus, the spectrum of an AM wave, for the special case of sinusoidal modulation, consists of delta
functions at ±𝑓𝑐 , 𝑓𝑐 ± 𝑓𝑚 , and −𝑓𝑐 ± 𝑓𝑚 as shown in Fig. 2.6 (c). The signals 𝑠(𝑡), 𝑚(𝑡) and the
carrier are also shown in Fig. 2.6 (a) - (c) both in time-domain and frequency-domain.
Fig. 2.6: time-domain and frequency-domain characteristics of AM waves produced by a

single tone modulation.
7|Page
For single-tone modulation, the modulation index can be obtained by definition,

1
𝑚 = max(|𝑘𝑎 𝑚(𝑡)|). Assuming 𝑘𝑎 = 𝐴 , and taking the single-tone message signal 𝑚(𝑡).
𝑐
𝐴𝑚 cos 2𝜋𝑓𝑚 𝑡 𝐴𝑚 (2.10)

𝑚 = max (| |) =
𝐴𝑐 𝐴𝑐
Fig. 2.7: AM wave

It is also possible to compute the modulation index from measurements taken on the modulated
wave itself. Whenever the message signal is displayed on an oscilloscope, the modulation index
can be computed from 𝑉𝑚𝑎𝑥 and 𝑉𝑚𝑖𝑛 of the modulated wave as shown in Fig. 2.7. The peak value
of the modulating signal 𝐴𝑚 is one-half the difference of the peak and trough values:
𝑉𝑚𝑎𝑥 − 𝑉𝑚𝑖𝑛
𝐴𝑚 =
2
The peak value of the carrier signal 𝐴𝑐 is also the average of the 𝑉𝑚𝑎𝑥 and 𝑉𝑚𝑖𝑛 values:
𝑉𝑚𝑎𝑥 + 𝑉𝑚𝑖𝑛
𝐴𝑐 =
2
Hence, the modulation index in Eq. (2.10) could be modified as;
𝑉𝑚𝑎𝑥 − 𝑉𝑚𝑖𝑛 (2.11)
𝑚=
𝑉𝑚𝑎𝑥 + 𝑉𝑚𝑖𝑛
Two-sided frequency spectrum of a single tone modulated wave is shown in Fig. 2.8 with
corresponding weight for each delta-function.
8|Page
Fig. 2.8: Frequency spectrum of single tone AM wave

AM Power Distribution of a carrier modulated by a single-tone message signal
In practice, the AM wave 𝑠(𝑡) is a voltage or current wave. In either case, the total power in the
amplitude-modulated wave consists of the sum of three components i.e. the power in the carrier,
in the USB and in the LSB. Considering the spectrum depicted in Fig. 2.8,
2
(𝐴𝑐 ⁄√2) 𝐴2𝑐
𝑃𝑐 = =
𝑅 2𝑅
2
(𝑚𝐴𝑐 ⁄2√2) 𝑚2 𝐴2𝑐 𝑚2 𝑃𝑐
𝑃𝑈𝑆𝐵 = 𝑃𝐿𝑆𝐵 = = =
𝑅 8𝑅 4 (2.12)
2
𝑚
𝑃𝑡 = 𝑃𝑐 + 𝑃𝐿𝑆𝐵 + 𝑃𝑈𝑆𝐵 = (1 + ) 𝑃𝑐
2
The power in the side bands depends upon the value of the modulation index. The greater the
modulation index, the higher the sideband power.
AM current Relations
In many cases the power output from an AM transmitter is not measured directly. Instead the
output current in the antenna is measured and the power is calculated. The total modulated current
𝐼𝑡 is determined by the un-modulated carrier current 𝐼𝑐 , and the modulation index m.
𝑃𝑡 𝐼𝑡2 𝑅 𝐼𝑡2 𝑚2 (2.13)
= = = (1 + ) ⇒ 𝐼𝑡 = 𝐼𝑐 √1 + 𝑚2 ⁄2
𝑃𝑐 𝐼𝑐2 𝑅 𝐼𝑐2 2
Efficiency of AM Transmitter
The carrier doesn’t contain message. The message is in the side bands and both the upper and
lower side bands contain the same information. Therefore the efficiency of an AM signal is power
in sidebands divided by the total power.
PUSB  PLSB m2 (2.14)

  2
PT m 2
9|Page
This has a maximum value of 33.3%, i.e., DSB-with carrier modulation has a very low efficiency.
This is one of the disadvantages of DSB-with carrier AM. Another disadvantage is that it requires
wide bandwidth ( BT  2W ). However, it is inherently easy to modulate and demodulate.
Example: - An audio frequency signal 10 sin 2𝜋 × 500𝑡 is used to amplitude modulate a carrier
of 50 sin 2𝜋 × 105 𝑡.
i) Sketch the waveforms of the message, carrier and amplitude modulated waves
ii) Calculate the modulation index
iii) Find the side band frequencies
iv) Find the amplitude of each sideband frequencies
v) What is the required bandwidth to transmit this AM signal?
vi) Calculate the total power delivered to a load of 600Ω
vii) Find the efficiency of the modulator
Solution: i) Waveforms of the message, carrier and AM signal are left for the student
ii) The given modulating signal is 𝑚(𝑡) = 10 sin 2𝜋 × 500𝑡. Hence, 𝐴𝑚 = 10. The given carrier
signal is 𝑐(𝑡) = 50 sin 2𝜋 × 105 𝑡, hence 𝐴𝑐 = 50. Therefore the modulation index will be,
𝐴𝑚 10
𝑚= = = 0.2 𝑜𝑟 20%
𝐴𝑐 50
iii) From the given equations,
𝜔𝑚 = 2𝜋 × 500, ℎ𝑒𝑛𝑐𝑒, 𝑓𝑚 = 500𝐻𝑧 𝑎𝑛𝑑
𝜔𝑐 = 2𝜋 × 105 , ℎ𝑒𝑛𝑐𝑒, 𝑓𝑐 = 100𝐾𝐻𝑧
We know that 𝑓𝑈𝑆𝐵 = 𝑓𝑐 + 𝑓𝑚 = 100𝐾𝐻𝑧 + 500𝐻𝑧 = 100.5𝐾𝐻𝑧
𝑓𝐿𝑆𝐵 = 𝑓𝑐 − 𝑓𝑚 = 100𝐾𝐻𝑧 − 500𝐻𝑧 = 99.5𝐾𝐻𝑧
𝑚𝐴𝑐 0.2×50
iv) Amplitude of upper and lower side bands is; = = = 5𝑉
2 2
v) Bandwidth requirement of the AM signal,

𝐵𝑊 𝑜𝑓 𝐴𝑀 = 2 × 𝑓𝑚 = 2 × 500𝐻𝑧 = 1𝐾𝐻𝑧
vi) Total power delivered by to the load is;
𝐴2𝑐 𝑚2 502 0.22
𝑃𝑡𝑜𝑡 = (1 + )= (1 + ) = 2.125 𝑊
2𝑅 2 2 × 600 2
vii) Efficiency of the modulator is given by;
𝑚2 0.22
𝜂= 2 = = 1.9%
𝑚 + 2 0.22 + 2
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Generation of AM waves
So far we have studied the basic concepts of amplitude modulation. In this section, we study the
techniques to generate amplitude modulated wave. These techniques can be classified as
Low level modulation and
High level modulation
In low-level modulation, the modulation takes place at low power level prior to the output element
of the final stage of the transmitter and the modulated signal passes through series of linear
amplifiers. Whereas, in high-level modulators, the modulation takes place in the final element of
the final stage at high power levels. Despite their complex design due-to the high power involved,
relatively highly efficient class C amplifiers are used in high level modulators.
When we look at the expression for 𝑠(𝑡) in Eq. (2.1), it is clear that we need a circuit that can
multiply the carrier by the modulating signal and then add the carrier. A block diagram of such a
circuit is shown in Fig. 2.9.
Fig. 2.9: Block diagram of a circuit to produce AM.

There are two basic ways to produce amplitude modulation. The first is to multiply the carrier by
a gain or attenuation factor that varies with the modulating signal. The second is to linearly mix or
add the carrier and the modulating signals and then apply the composite signal to a nonlinear device
or circuit.
1. A square-law modulator (Switching modulator)
To produce AM, the carrier and modulating signals are added and applied to the nonlinear device.
A simple way to do this is to connect the carrier and modulating sources in series and apply them
to the diode circuit, as in Fig. 2.10.
11 | P a g e
Fig. 2.10: Square law modulator circuit

The voltage applied to the diode is then;
𝑉 = 𝑉𝑐 + 𝑉𝑚 (2.15)
The diode current in the resistor is
𝑖 = 𝑎(𝑉𝑐 + 𝑉𝑚 ) + 𝑏(𝑉𝑐 + 𝑉𝑚 )2 (2.16)
Expanding Eq. (2.16), we get
𝑖 = 𝑎(𝑉𝑐 + 𝑉𝑚 ) + 𝑏(𝑉𝑐2 + 2𝑉𝑐 𝑉𝑚 + 𝑉𝑚2 ) (2.17)
If we let 𝑉𝑐 = 𝑣𝑐 sin 𝜔𝑐 𝑡 and 𝑉𝑚 = 𝑣𝑚 sin 𝜔𝑚 𝑡, substituting the these expressions in Eq. (2.17) for
the carrier and modulating signals will result in;
𝑖 = 𝑎𝑣𝑐 sin 𝜔𝑐 𝑡 + 𝑎𝑣𝑚 sin 𝜔𝑚 𝑡 + 𝑏𝑣𝑐2 sin2 𝜔𝑐 𝑡 (2.18)
2
+ 2𝑏𝑣𝑐 𝑣𝑚 sin 𝜔𝑐 𝑡 sin 𝜔𝑚 𝑡 + 𝑏𝑣𝑚 sin2 𝜔𝑚 𝑡
The first term is the carrier sine wave, which is a key part of the AM wave; the second term is the
modulating signal sine wave. Normally, this is not part of the AM wave. It is substantially lower
in frequency than the carrier, so it could be easily filtered out. The fourth term, which is the product
of the carrier and modulating signal sine waves, defines the AM wave. The third term sin2 𝜔𝑐 𝑡 is
a sine wave at two times the frequency of the carrier, i.e., the second harmonic of the carrier. The
last term sin2 𝜔𝑚 𝑡 is also the second harmonic of the modulating sine wave. These components
are undesirable, but are relatively easy to be filtered out.
Hence, the current expression could be modified as,
𝑖 = 𝑎𝑣𝑐 sin 𝜔𝑐 𝑡 + 2𝑏𝑣𝑐 𝑣𝑚 sin 𝜔𝑐 𝑡 sin 𝜔𝑚 𝑡 + 𝑢𝑛𝑤𝑎𝑛𝑡𝑒𝑑 𝑡𝑒𝑟𝑚𝑠
2𝑏𝑣𝑚 sin 𝜔𝑚 𝑡 (2.19)
𝑖 = 𝑎𝑣𝑐 (1 + ) sin 𝜔𝑐 𝑡 + 𝑢𝑛𝑤𝑎𝑛𝑡𝑒𝑑 𝑡𝑒𝑟𝑚𝑠
𝑎
Comparing Eq. (2.19) with the expression of 𝑠(𝑡) in Eq. (2.1) which describes AM wave, we can
observe that the two expressions are identical. This indicates that Eq. (2.19) also describes an AM
waveform.
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2. Diode Modulator
Another AM modulator implementation using diode is shown in Fig. 2.11 which consists of a
resistive mixing network, a diode rectifier, and an LC tuned circuit. The carrier (Fig. 2.11(b)) is
applied to one input resistor and the modulating signal (Fig. 2.11 (a)) to the other. The mixed
signals appear across 𝑅3 . This network causes the two signals to be linearly mixed, i.e.,
algebraically added. If both the carrier and the modulating signal are sine waves, the waveform
resulting at the junction of the two resistors will be like that shown in Fig. 2.11(c), where the carrier
wave is riding on the modulating signal.
The composite waveform is applied to a diode rectifier. The diode is connected so that it is forward-
biased by the positive-going half-cycles of the input wave. During the negative portions of the
wave, the diode is cut off and no signal passes. The current through the diode is a series of positive-
going pulses whose amplitude varies in proportion to the amplitude of the modulating signal [see
Fig. 2.11(d)]. These positive-going pulses are applied to the parallel-tuned circuit made up of L
and C, which are resonant at the carrier frequency. Each time the diode conducts, a pulse of current
flows through the tuned circuit. The coil and capacitor repeatedly exchange energy, causing an
oscillation, or “ringing,” at the resonant frequency. The oscillation of the tuned circuit creates one
negative half-cycle for every positive input pulse. High amplitude positive pulses cause the tuned
circuit to produce high- amplitude negative pulses. Low-amplitude positive pulses produce
corresponding low-amplitude negative pulses. The resulting waveform across the tuned circuit is
an AM signal, as Fig. 2.11(e) illustrates.
Fig. 2.11: Resistive mixing network and resulting waveforms
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3. Op amp as AM modulator
Another simple amplitude modulator is shown in Fig. 2.12. It consists of an operational amplifier
(op amp) and a field-effect transistor (FET) used as a variable resistor. The op amp is connected
as a non-inverting amplifier for the carrier signal. The gain A of the circuit for the oscillator signal
𝑅𝑓
is given by the expression Gain 𝐴 = 1 + .
𝑅𝑖
Fig. 2.12: Using a JFET to vary the gain of an op amp: Op-amp modulator
In the absence of modulating signal FET provides a fixed resistance and therefore the gain of the
non-inverting amplifier is constant, giving steady carrier output. On the application of modulating
signal the resistance of FET will vary. A positive going modulating input signal will cause the FET
resistance to decrease, where as a negative going modulating signal will cause it to increase.
Increasing the FET resistance causes the op-amp circuit gain to decrease and vice versa. This
results in an AM signal at the output of op-amp.
4. Emitter modulator
A small signal class-A amplifier such as the one shown in Fig. 2.13 can also be used to perform
amplitude modulation. In this case, the carrier signal is given to the base of the amplifier and
modulating signal is given to the emitter. In absence of modulating signal, the circuit simply
operates as linear class A amplifier. When modulating signal is applied to an emitter, the gain of
the amplifier varies according to voltage of modulating signal. Depending upon the gain variations,
carrier signal is amplified. Thus amplitude of carrier signal is modulated by the modulating signal.
The voltage gain of emitter modulator is given as;
𝐴𝑣 = 𝐴𝑞 [1 + 𝑚 sin(2𝜋𝑓𝑚 𝑡)] (2.20)
Where 𝐴𝑣 is voltage gain with modulation, and 𝐴𝑞 is quiescent voltage gain
14 | P a g e
Fig. 2.13: Class A amplifier as AM

The primary disadvantage of emitter modulation is the amplifier operates in class A, which is
extremely inefficient. Emitter modulators are also incapable of producing high-power output
waveforms.
5.4 Collector Modulator - Medium and High Power AM Modulator
One example of a high-level modulator circuit is the collector modulator shown in Fig. 2.14. The
output stage of the transmitter is a high-power class C amplifier. Class C amplifiers conduct for
only a portion of the positive half-cycle of their input signal. The collector current pulses cause
the tuned circuit to oscillate (ring) at the desired output frequency. The tuned circuit, therefore,
reproduces the negative portion of the carrier signal.
The modulator is a linear power amplifier that takes the low-level modulating signal and amplifies
it to a high-power level. The modulating output signal is coupled through modulation transformer
𝑇1 to the class C amplifier. The secondary winding of the modulation transformer is connected in
series with the collector supply voltage 𝑉𝑐𝑐 of the class C amplifier. With a zero-modulation input
signal, there is zero-modulation voltage across the secondary of 𝑇1 , the collector supply voltage is
applied directly to the class C amplifier, and the output carrier is a steady sine wave.
When the modulating signal occurs, the AC voltage of the modulating signal across the secondary
of the modulation transformer is added to and subtracted from the dc collector supply voltage. This
varying supply voltage is then applied to the class C amplifier, causing the amplitude of the current
pulses through transistor 𝑄1 to vary. As a result, the amplitude of the carrier sine wave varies in
accordance with the modulated signal. When the modulation signal goes positive, it adds to the
collector supply voltage, thereby increasing its value and causing higher current pulses and a
higher-amplitude carrier. When the modulating signal goes negative, it subtracts from the collector
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supply voltage, decreasing it. For that reason, the class C amplifier current pulses are smaller,
resulting in a lower-amplitude carrier output.
Fig. 2.14: A high-level collector modulator

5. Linear IC AM Modulators
Linear IC AM modulators are also available and offer excellent frequency stability, symmetrical
modulation characteristics, circuit miniaturization, fewer components, temperature immunity, and
simplicity of design and troubleshooting. They are low level type modulators.
Reading assignment: Please investigate and read about linear IC AM modulators and differential
amplifier modulator.
Demodulation of AM wave
The function of AM detector or demodulator is to recover or reproduce modulating signal or the
original source information/message signal from the modulated wave at the receiver.
1. Envelope Detector (Peak detector or Diode Detector)
The simplest and most widely used amplitude demodulator is the diode detector shown in Fig.
2.15. In order to recover 𝑚(𝑡) from an AM wave 𝑠(𝑡) with envelope detector,
i. M ( f )  0, f  w ⇒ 𝑚(𝑡) must be band limited.
ii. f c  w ⇒ 𝑚(𝑡) must vary very little over one period of the carrier.
For proper operation the time constant also matters:

1
i. Charging time of 𝐶1 must be small compared to
fc
ii. 𝐶1 must discharge very little between periods of the carrier for voltage across it to follow
the envelope, but not so long that the capacitor will not discharge at maximum rate of 𝑠(𝑡);
1 1
i.e.,  R1C1  where w is the highest frequency component of 𝑚(𝑡).
fc w
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The resulting output usually has small ripple at 𝑓𝑐 which could be easily removed by low pass
filter.
Fig.2.15: Envelope detector (a), and corresponding waveforms (b)

2.2 DSB-SC Modulation and Demodulation
Conventional AM have an inherent disadvantage that carrier power constitutes two-third or more
power of the total power. This is a major drawback because the carrier contains no information.
To overcome this shortcoming of AM we may suppress the carrier component from the modulated
wave resulting in DSB-SC modulation. Thus by suppressing the carrier one will get a DSB-SC
wave, which is given by
𝑆𝐷𝑆𝐵−𝑆𝐶 (𝑡) = 𝑘𝑎 𝐴𝑐 𝐴𝑚 cos 2𝜋𝑓𝑐 𝑡 cos 2𝜋𝑓𝑚 𝑡
𝑘𝑎 𝐴𝑐 𝐴𝑚 𝑘𝑎 𝐴𝑐 𝐴𝑚 (2.21)
𝑆𝐷𝑆𝐵−𝑆𝐶 (𝑡) = cos(2𝜋(𝑓𝑐 + 𝑓𝑚 )𝑡) + cos(2𝜋(𝑓𝑐 − 𝑓𝑚 )𝑡)
2 2
This modulated wave undergoes phase reversal whenever the baseband signal 𝑚(𝑡) crosses zero.
Therefore unlike AM, the envelope of DSB-SC wave is different from the base band signal. The
main advantage of DSB-SC over conventional AM is that it has a higher power efficiency. Yet its
bandwidth is the same as that of the conventional AM.
DSB-SC Modulators
The DSB-SC consists of simply the product of the baseband and the carrier wave. A device
performing the multiplication is called product modulator. This can be implemented using
balanced or ring modulator.
Balanced modulator (Ring Modulator)
A balanced modulator is a circuit that generates a DSB signal, suppressing the carrier and leaving
only the sum and difference frequencies at the output. One of the most popular and widely used
17 | P a g e
balanced modulators is the diode ring or lattice modulator as shown in Fig. 2.16, consisting of an
input transformer 𝑇1 , an output transformer 𝑇2 , and four diodes connected in a bridge circuit. The
circuit can be in one of the two similar connections shown in Fig. 2.16 (a) and (b).
Fig. 2.16: Lattice-type balanced modulator.

The carrier signal is applied to the center taps of the input and output transformers, and the
modulating signal is applied to the input transformer 𝑇1 . The DSB output is collected at the
secondary of transformer 𝑇2 . Let us consider that modulating input is zero. In the positive half
cycle of the carrier signal diodes 𝐷1 and 𝐷2 are forward biased, and diodes 𝐷3 and 𝐷4 are reverse
biased as illustrated in Fig. 2.17 (a).
Fig. 2.17: Operation of the lattice modulator.

It is clear from Fig. 2.17 (a) that the current divides equally in the upper and lower portions of the
primary winding of 𝑇2 . The current in the upper part of the winding produces a magnetic field that
that is equal and opposite to the magnetic field produced by the current in the lower half of the
secondary. As magnetic fields are equal and opposite, they cancel each other, producing no output
at the secondary of 𝑇2 . Thus, the carrier is suppressed. In the negative half cycle, 𝐷1 and 𝐷2 are
reverse biased, and diodes 𝐷3 and 𝐷4 are forward biased as shown in Fig. 2.15 (b). Similar to
18 | P a g e
positive half cycle, here also magnetic fields in primary winding of 𝑇2 are equal and opposite
cancelling each other. Therefore there is no output produced at the secondary of 𝑇2 .
Consider that a sinusoidal modulating signal is applied to the primary of 𝑇1 . This signal will also
appear across the 𝑇1 secondary. In the positive half cycle of the carrier, the diodes 𝐷1 and 𝐷2 are
forward biased and they will connect the secondary of 𝑇1 to the primary of 𝑇2 . As a result, the
modulating signal at the secondary of 𝑇1 is applied to primary of 𝑇2 through diodes 𝐷1 and 𝐷2 . In
the negative half cycle of the carrier, diodes 𝐷3 and 𝐷4 are forward biased and they will connect
the secondary of 𝑇1 to the primary of 𝑇2 with reverse connections. This inverts the polarity of
modulating signal when it is applied to primary of 𝑇2 . Fig. 2.18 (c) shows DSB signal at the
primary of 𝑇2 . Thus when 𝐷3 and 𝐷4 conduct, the polarity of the signal is opposite to that of
modulating signal. Fig. 2.18 (d) shows the DSB output at the secondary of 𝑇2 . Thus carrier is
totally suppressed in this signal.
Fig. 2.18: Waveforms in the lattice-type balanced modulator. (a) Carrier. (b) Modulating
signal. (c) DSB signal primary 𝑇2 . (d) DSB output.
1.3 SSB Modulation and Demodulation
Conventional AM and DSB-SC are wasteful of bandwidth because they both require transmission
bandwidth equal to twice the message bandwidth. As far as the transmission of information is
concerned, only one sideband is necessary. Thus, it is possible to transmit only one of the side
bands because the lower side band and upper sideband carries the same information. When only
19 | P a g e
one sideband is transmitted, the modulation system is referred to as single sideband system (SSB).
It provides the following advantages as compared to conventional AM and DSB-SC.
1. It conserves frequency spectrum since only one of the side bands is transmitted.
2. It requires relatively low power as compared to conventional AM.
3. Noise decrease since the BW has decreased by half.
The benefit of using SSB is therefore derived from the reduced bandwidth requirement and the
elimination of the high power carrier wave. The principal disadvantage of the SSB system is its
cost and complexity.
Mathematically- SSB wave is given by
1 1 (2.22)
𝑆𝑆𝑆𝐵 (𝑡) = 𝑘𝑎 𝐴𝑐 𝐴𝑚 cos(2𝜋𝑓𝑚 𝑡) cos(2𝜋𝑓𝑐 𝑡) ± 𝑘𝑎 𝐴𝑐 𝐴𝑚 sin(2𝜋𝑓𝑚 𝑡) sin(2𝜋𝑓𝑐 𝑡)
2 2
where the plus sign applies to lower SSB and the minus sign applies to upper SSB.
SSB Modulators
The are two methods of generating SSB
1. Frequency discrimination method (filter method)
2. Phase discrimination method
 Frequency Discrimination Method (Filter Method)
An SSB modulator based on frequency discrimination consists basically a ring modulator and a
filter, which is designed to pass the desired sideband of the DSB-SC wave.
Fig. 2.19: An SSB transmitter using the filter method.
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In designing the band-pass filter in the SSB generator system in Fig.2.19, the filter must have a
pass band at the same frequency range as the spectrum of the desired sideband. This type of
frequency discrimination can be satisfied only by using highly selective filter, which can be
realized using crystal and ceramic filters.
 Phase Discriminator Method
This method uses two balanced modulators, which effectively eliminate the carrier. The carrier
oscillator is applied directly to the upper balanced modulator along with the audio modulating
signal as shown in Fig. 2.20. The carrier and modulating signal are then both shifted in phase by
90° and applied to the second, lower, balanced modulator. The phase-shifting action causes one
sideband to be canceled out when the two balanced modulator outputs are added to produce the
output. If the carrier signal is 𝐴𝑐 sin 2𝜋𝑓𝑐 𝑡 and the modulating signal is 𝐴𝑚 sin 2𝜋𝑓𝑚 𝑡, balanced
modulator 1 produces the product of these two signals:
1 (2.23)
𝐴𝑐 sin 2𝜋𝑓𝑐 𝑡 𝐴𝑚 sin 2𝜋𝑓𝑚 𝑡 = 𝐴𝑚 𝐴𝑐 [cos 2𝜋(𝑓𝑐 − 𝑓𝑚 )𝑡 − cos 2𝜋(𝑓𝑐 + 𝑓𝑚 )𝑡]
2
The 90° phase shifters in Fig. 2.20 create cosine waves of the carrier and modulating signals that
are multiplied in balanced modulator 2 to produce
1 (2.24)
(𝐴𝑐 cos 2𝜋𝑓𝑐 𝑡) (𝐴𝑚 cos 2𝜋𝑓𝑚 𝑡) = 𝐴 𝐴 [cos 2𝜋(𝑓𝑐 − 𝑓𝑚 )𝑡 + cos 2𝜋(𝑓𝑐 + 𝑓𝑚 )𝑡]
2 𝑚 𝑐
Adding the sine expression in Eq. (2.23) to the cosine expression in (2.24), the sum frequencies
cancel and the difference frequencies add, producing only the lower sideband cos 2𝜋(𝑓𝑐 − 𝑓𝑚 )𝑡.
Fig. 2.20: Phase discriminator method of generating SSB signal

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Demodulation of DSB-SC and SSB signals (coherent detection)

To recover the baseband signal in a DSB or SSB AM, the carrier that was suppressed at the
modulator must be reinserted at the receiver. This is called coherent detection and it involves
applying the DSB/SSB wave, together with a locally generated sinusoidal carrier wave to a
balanced modulator as shown in Fig 2.21 below. The balanced modulator is called a product
detector because it is used to recover the modulating signal. In this modulator, the carrier is
suppressed, but the sum and difference signals are generated. As an example the sum and
difference frequencies produced by the product detector for the case of DSB-SC AM could be
given as;
Sum for USB: 𝑓𝑐 + 𝑓𝑚 + 𝑓𝑐 = 2𝑓𝑐 + 𝑓𝑚 , and sum for LSB: 𝑓𝑐 − 𝑓𝑚 + 𝑓𝑐 = 2𝑓𝑐 − 𝑓𝑚
Difference for USB: 𝑓𝑐 + 𝑓𝑚 − 𝑓𝑐 = 𝑓𝑚 and difference for LSB: 𝑓𝑐 − 𝑓𝑚 − 𝑓𝑐 = −𝑓𝑚
The difference is, of course, the original modulating signal, while the sum signals have no
importance. Since these frequencies are so far apart, the higher undesired frequency components
can be easily filtered out by a low-pass filter that keeps the required baseband signal but suppresses
everything above it.
Fig. 2.21: Demodulation of DSB/SSB signal using product detector

1.4 Vestigial (VSB) AM
When the information signal contains significant components at extremely low frequencies as in
TV signals, the SSB modulation is inappropriate in transmitting such baseband signals. This is due
to the difficulty of isolating one side band as sideband filters are only approximately realizable.
Despite its simpler generation, DSB-SC has also a disadvantage bandwidth inefficiency since it
requires twice the signal bandwidth. These difficulties suggest another scheme known as VSB
modulation, which is a compromise between SSB and DSB-SC modulation techniques. It inherits
the advantages of DSB-SC and SSB techniques avoiding their disadvantages at the small cost.
22 | P a g e
VSB signals are relatively easy to generate and at the same time their bandwidth is only a little
greater than that of SSB signals.
VSB modulation distinguishes itself from SSB modulation in two practical respects:
1. Instead of completely removing a sideband, a trace or vestige of that sideband is
transmitted; hence, the name “vestigial sideband.”
2. Instead of transmitting the other sideband in full, almost the whole of this second band
is also transmitted.
Accordingly, the transmission bandwidth of a VSB modulated signal is defined by

𝐵𝑇 = 𝑓𝑣 + 𝑓𝑚
where 𝑓𝑣 is the vestige bandwidth and 𝑓𝑚 is the message bandwidth. Typically, 𝑓𝑣 is 25% of 𝑓𝑚 ,
which means that the VSB bandwidth lies between the SSB bandwidth 𝑓𝑚 , and DSB-SC
bandwidth, 2𝑓𝑚 .
VSB wave Generation
The block diagram of VSB modulator is shown in Fig 2.23. The modulating signal 𝑚(𝑡) and the
carrier 𝑐(𝑡) are applied to the product modulator.
Fig. 2.23: Generation of VSB AM signal

The output of the product modulator in time domain is given by
𝑠𝑐 (𝑡) = 𝑚(𝑡)𝑐(𝑡) = 𝑘𝑎 𝐴𝑐 𝑚(𝑡) cos 2𝜋𝑓𝑐 𝑡 (2.25)
This represents a DSB-SC modulated wave. This signal is then applied to a sideband shaping
filter 𝐻(𝑓) whose design depends on the desired spectrum of the VSB modulated signal. It will
pass the required sideband (say USB) and the vestige of the other (LSB) sideband. If the impulse
response of the filter is ℎ(𝑡), then the output of the filter will be given by;
𝑆𝑉𝑆𝐵 (𝑡) = 𝑠𝑐 (𝑡) ∗ ℎ(𝑡) = 𝑘𝑎 𝐴𝑐 𝑚(𝑡) cos 2𝜋𝑓𝑐 𝑡 ∗ ℎ(𝑡) (2.26)
Then the spectrum of VSB modulated signal is given by
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𝐴𝑐 𝑘𝑎 (2.27)
𝑆𝑉𝑆𝐵 (𝑓) = [𝑀(𝑓𝑐 + 𝑓𝑚 ) + 𝑀(𝑓𝑐 − 𝑓𝑚 )]𝐻(𝑓)
2
The only requirement that the sideband shaping performed by 𝐻(𝑓) must satisfy is that the
transmitted vestige compensates for the spectral portion missing from the other sideband. This
requirement ensures that coherent detection of the VSB modulated wave recovers a replica of the
message signal, except for amplitude scaling.
By imposing this requirement on the VSB demodulation process, it turns out that the sideband
shaping filter must itself satisfy the following condition:
𝐻(𝑓 + 𝑓𝑐 ) + 𝐻(𝑓 − 𝑓𝑐 ) = 1 𝑓𝑜𝑟 − 𝑓𝑚 ≤ 𝑓 ≤ 𝑓𝑚 (2.28)
The term 𝐻(𝑓 + 𝑓𝑐 ) is the positive-frequency part of the band-pass transfer function shifted to the
left by 𝑓𝑐 and 𝐻(𝑓 − 𝑓𝑐 ) is the negative frequency part shifted to the right by 𝑓𝑐 .
For the case of sinusoidal VSB modulation produced by the sinusoidal modulating wave, let
𝑚(𝑡) = 𝐴𝑚 cos(2𝜋𝑓𝑚 𝑡) and carrier wave 𝑐(𝑡) = 𝐴𝑐 cos(2𝜋𝑓𝑐 𝑡). If the upper side-frequency
at 𝑓𝑐 + 𝑓𝑚 as well as its image at −(𝑓𝑐 + 𝑓𝑚 ) be attenuated by the factor 𝑘. To satisfy the condition
of Eq. (2.28), the lower side-frequency at 𝑓𝑐 − 𝑓𝑚 and its image −(𝑓𝑐 − 𝑓𝑚 ) must be attenuated by
the factor 1 − 𝑘. The VSB spectrum is therefore;
1 1 (2.29)
𝑠(𝑡) = 𝑘𝐴𝑐 𝐴𝑚 cos(2𝜋(𝑓𝑐 + 𝑓𝑚 )𝑡) + (1 − 𝑘)𝐴𝑐 𝐴𝑚 cos(2𝜋(𝑓𝑐 − 𝑓𝑚 )𝑡)
2 2
Using well-known trigonometric identities to expand the cosine terms;
1 1 (2.30)
𝑠(𝑡) = 𝐴𝑐 𝐴𝑚 cos(2𝜋𝑓𝑐 𝑡) cos(2𝜋𝑓𝑚 𝑡) + (1 − 2𝑘)𝐴𝑐 𝐴𝑚 sin(2𝜋𝑓𝑐 𝑡) sin(2𝜋𝑓𝑚 𝑡)
2 2
Depending on how the attenuation factor 𝑘 in Eq. (2.30) is defined in the interval [0, 1], we may
identify three modulated waves.
1
1. When 𝑘 = 2 , 𝑠(𝑡) reduces to DSB-SC.
2. When 𝑘 = 0, 𝑠(𝑡) reduces to lower SSB, and When 𝑘 = 1, 𝑠(𝑡) reduces to upper SSB
1
3. For 0 < 𝑘 < 2, the attenuated version of the upper side-frequency defines the vestige
1
of 𝑠(𝑡), and for 2 < 𝑘 < 1, attenuated version of the lower side-frequency defines the
vestige of 𝑠(𝑡).
Demodulation of VSB AM
Similar to DSB-SC and SSB demodulations studied previously, the demodulation of VSB consists
of multiplying 𝑠(𝑡) with a locally generated sinusoid and then low-pass filtering the resulting
24 | P a g e
product signal 𝑣(𝑡). It is assumed that the local sinusoid in the coherent detector of Fig. 2.24 is in
perfect synchronism with the carrier in the modulator responsible for generating the VSB-
modulated wave.
Fig. 2.24: Coherent detection of VSB AM

Multiplying 𝑠(𝑡) in Eq. (2.30) by 𝐴′𝑐 cos(2𝜋𝑓𝑐 𝑡) in accordanc+e with perfect coherent detection
yields the product signal
1 (2.31)
𝑣(𝑡) = 𝑠(𝑡)𝐴′𝑐 cos(2𝜋𝑓𝑐 𝑡) = 𝐴 𝐴 𝐴′ cos2 (2𝜋𝑓𝑐 𝑡) cos(2𝜋𝑓𝑚 𝑡) +
2 𝑐 𝑚 𝑐
1
(1 − 2𝑘)𝐴𝑐 𝐴𝑚 𝐴′𝑐 sin(2𝜋𝑓𝑐 𝑡) sin(2𝜋𝑓𝑚 𝑡) cos(2𝜋𝑓𝑐 𝑡)
2
Next, using the trigonometric identities,
1 1
cos2 (2𝜋𝑓𝑐 𝑡) = 2 [1 + cos(4𝜋𝑓𝑐 𝑡)] and sin(2𝜋𝑓𝑐 𝑡) cos(2𝜋𝑓𝑐 𝑡) = 2 sin(4𝜋𝑓𝑐 𝑡)
We may redefine as;

1 1 (2.32)
𝑣(𝑡) = 𝐴𝑐 𝐴𝑚 𝐴′𝑐 [cos(2𝜋𝑓𝑚 𝑡) + 𝐴𝑐 𝐴𝑚 𝐴′𝑐 [cos(2𝜋𝑓𝑚 𝑡) cos(4𝜋𝑓𝑐 𝑡)
4 4
1
+ (1 − 2𝑘) sin(2𝜋𝑓𝑚 𝑡) sin(4𝜋𝑓𝑐 𝑡)]
2
The first term on the right-hand side of Eq. (2.32) is a scaled version of the message signal 𝑚(𝑡),
while the second term is a new sinusoid which represents the high-frequency components. This
high-frequency component can be removed by the low-pass filter in the detector of Fig. 2.24,
provided that the cutoff frequency of the filter is just slightly greater than the message frequency.
2.5 AM Transmitters
The transmitter is a part of communication system that accepts the message signal to be transmitted
and converts it into an RF signal capable of being transmitted over long distances. Every
transmitter has three basic functions. First, the transmitter must generate a signal of the correct
frequency at a desired point in the spectrum. Second, it must provide some form of modulation
that causes the information signal to modify the carrier signal. Third, it must provide sufficient
25 | P a g e
power amplification to ensure that the signal level is high enough so that it will carry over the
desired distance.
An AM transmitter is shown in Fig. 2.25 below. An oscillator generates the final carrier frequency.
In most applications, this will be a crystal oscillator due to high frequency stability of the crystal.
The carrier signal is then fed to a buffer amplifier whose primary purpose is to isolate the oscillator
from the remaining power amplifier stages. The signal from the buffer is applied to the driver
amplifier which is a class C amplifier designed to provide an intermediate level of power
amplification. The purpose of this circuit is to generate sufficient output power to drive the final
power amplifier stage. The final power amplifier, also operates in class C at very high power. The
actual amount of power depends upon the intended application. Assuming it as a voice transmitter,
the voice from the microphone is amplified and processed by the speech processor. The speech
processor is basically used for two main purposes: filtering (frequency control) and amplitude
control to avoid out of band radiation and distortion due-to over-modulation respectively.
Fig. 2.25: An AM transmitter using high-level collector modulation

To design an AM transmitter one should know about the main blocks of the transmitter. These
include RF oscillators, Buffer amplifiers, Driver amplifiers, power amplifiers, filters and
impedance matching.
2.6 AM receiver
A sensitive and selective receiver can be made using only amplifiers, selective filters, and a
demodulator. This is called a tuned radio frequency receiver. Early radios used this design.
26 | P a g e
However, such a receiver does not usually deliver the kind of performance expected in modern
communications applications. One type of receiver that can provide that performance is the
superheterodyne receiver. It converts all incoming signals to a lower frequency, known as the
intermediate frequency (IF), at which a single set of amplifiers and filters is used to provide a fixed
level of sensitivity and selectivity. Most of the gain and selectivity in a superheterodyne receiver
are obtained in the IF amplifiers. The key circuit is the mixer, which acts as a simple amplitude
modulator to produce sum and difference frequencies. The incoming signal is mixed with a local
oscillator signal to produce this conversion. Fig. 2.26 below shows a general block diagram of a
superhetrodyne receiver.
Fig. 2.26: Block diagram of a superheterodyne receiver.

RF amplifier: the antenna picks up weak radio signal and feeds it to the RF amplifier, also called
a low-noise amplifier (LNA). Because RF amplifiers provide some initial gain and selectivity, they
are sometimes referred to as pre-selectors. The RF amplifier between the mixer and the antenna
also isolates the two, significantly reducing any local oscillator radiation.
Mixers and Local Oscillators: output of the RF amplifier is applied to the input of the mixer,
which may also receive an input from a local oscillator. Its output is the input signal, the local
oscillator signal, and the sum and difference frequencies of these signals. Usually a tuned circuit
at the output of the mixer selects the difference frequency, or IF. The sum frequency may also be
selected as the IF in some applications.
IF Amplifiers: output of the mixer is amplified by one or more IF amplifier stages, and most of
the receiver gain is obtained in these stages.
27 | P a g e
Demodulators: the highly amplified IF signal is applied to the demodulator, or detector, which
recovers the original modulating information. Its output is then usually fed to an audio amplifier
with sufficient voltage and power gain to operate a speaker. For non-voice signals, the detector
output may be sent elsewhere, to a TV, tablet, cell phone screen, computer, or some other device.
Automatic Gain Control (AGC): with AGC, the overall gain of the receiver is automatically
adjusted depending on the input signal level.
Noise in AM receivers
Noise can broadly be defined as any unknown signal that affects the recovery of the desired signal.
It is an electronic signal which is a mixture of many random frequencies at many amplitudes that
gets added to a radio or information signal as it is transmitted from one place to another. There
may be many sources of noise in a communication system, but often the major sources are the
communication devices themselves or interference encountered during the course of transmission.
There are also several ways that noise can affect the desired signal, and for proper analysis of its
effect on the performance of the receiver, we need appropriate receiver models. The customary
practice is to model the receiver noise (channel noise) as additive, white, and Gaussian.
Fig. 2.26: receiver model

As shown in Fig. 2.26 above, the received signal is modeled as
𝑟(𝑡) = 𝑠(𝑡) + 𝑤(𝑡) (2.33)
Where 𝑠(𝑡) denotes the incoming modulated signal and 𝑤(𝑡) denotes front-end receiver noise.
The power spectral density of the noise 𝑤(𝑡) is denoted by 𝑁0 /2, defined for both positive and
negative frequencies. 𝑁0 is the average noise power per unit bandwidth measured at the front end
of the receiver. The bandwidth of the band-pass filter used in the model is just wide enough to pass
the modulated signal without distortion. We assume the band-pass filter is ideal, having a
bandwidth equal to the transmission bandwidth 𝐵𝑇 of the modulated signal 𝑠(𝑡), and a mid-band
frequency equal to the carrier frequency 𝑓𝑐 , 𝑓𝑐 >> 𝐵𝑇 .
28 | P a g e
Fig. 2.27: Power spectral density of band-pass noise.

The filtered noise 𝑛(𝑡) may be treated as a narrow band noise represented in the canonical form:
𝑛(𝑡) = 𝑛𝐼 (𝑡) cos(2𝜋𝑓𝑐 𝑡) − 𝑛𝑄 (𝑡) sin(2𝜋𝑓𝑐 𝑡) (2.34)
where 𝑛𝐼 (𝑡) is the in-phase noise component and 𝑛𝑄 (𝑡) is the quadrature noise component, both
measured with respect to the carrier wave 𝐴𝑐 cos(2𝜋𝑓𝑐 𝑡). The filtered signal 𝑥(𝑡) available for
demodulation can be defined by
𝑥(𝑡) = 𝑠(𝑡) + 𝑛(𝑡) (2.35)
The average noise power at the demodulator input is equal to the total area under the curve of the
power spectral density 𝑆𝑁 (𝑓):
𝑁0 (2.36)
𝑃𝑎𝑣𝑔−𝑛𝑜𝑖𝑠𝑒 = 2 × 𝐵𝑇 × = 𝐵𝑇 𝑁0
2
Signal-To-Noise Ratios (SNR)
Both of the terms on the right-hand side of Eq. (2.35) are random. It is due to the unpredictability
of both its information content and the noise. For partially describing a random variable mean and
variance parameters are used where both information and noise signals have zero mean and
constant variance. Consequently, for zero-mean processes, a simple measure of the signal quality
is the ratio of the variances of the desired and undesired signals. On this basis, the signal-to-noise
ratio is formally defined by
𝐸[𝑠 2 (𝑡)] (2.37)
𝑆𝑁𝑅 =
𝐸[𝑛2 (𝑡)]
where E is the expectation operator and for a communication signal, a squared signal level is
usually proportional to power. Consequently, the SNR is often considered to be a ratio of the
average signal power to the average noise power. SNR is measured at the receiver, but there are
several points in the receiver where the measurement can be carried out. In fact, measurements at
particular points in the receiver have their own particular importance and value.
For instance:
29 | P a g e
If the SNR is measured at the front-end of the receiver, then it is usually a measure of the
quality of the transmission link and the receiver front-end.
If it is measured at the output of the receiver, it is a measure of the quality of the recovered
information-bearing signal.
If we consider the block diagram of a typical analog communication receiver in Fig. 2.26, the SNR
measured at the input to the demodulator is referred to as the pre-detection SNR. Of equal or
greater importance is the SNR of the recovered message at the output of the demodulator. This
metric defines the quality of the signal that is delivered to the end user. We refer to this output
SNR as the post-detection SNR. It should be noted that the signal and noise characteristics may
differ significantly between the pre-detection and post-detection calculations.
In order to compare different analog modulation-demodulation schemes, we introduce the idea of
a reference transmission model as depicted in Fig. 2.28. This reference model is equivalent to
transmitting the message at baseband. In this model, two assumptions are made:
The message power is the same as the modulated signal power of the modulation scheme
under study.
The baseband low-pass filter passes the message signal and rejects out-of-band noise.
Accordingly, we may define the reference SNR, as
average power of the modulated mesage signal (2.38)
𝑆𝑁𝑅𝑟𝑒𝑓 =
average power of noise in the message bandwidth
Fig. 2.28: Reference transmission model for analog communications.

Figure of merit: for the purpose of comparing different continuous-wave (CW) modulation
systems, we normalize the receiver performance by dividing the post detection SNR by the
reference SNR. That is called figure of merit for it is defined as follows:
𝑃𝑜𝑠𝑡 − 𝑑𝑒𝑡𝑒𝑐𝑡𝑖𝑜𝑛 𝑆𝑁𝑅 (2.39)
𝐹𝑖𝑔𝑢𝑟𝑒 𝑜𝑓 𝑚𝑒𝑟𝑖𝑡 =
𝑅𝑒𝑓𝑒𝑟𝑒𝑛𝑐𝑒 𝑆𝑁𝑅
Clearly, the higher the value of the figure of merit, the better will the noise performance of the
receiver be. It is dimensionless metric which can have a value equal to 1, less than 1, or greater
than 1, depending on the type of modulation used.
30 | P a g e
Noise in DSB-SC Receiver

The model of a DSB-SC receiver using a coherent detector can be given as shown in Fig. 2.29.
Fig. 2.29: model of DSB-SC receiver using coherent demodulation.

For the demodulation scheme to operate satisfactorily, it is necessary that the local oscillator be
synchronized both in phase and in frequency with the oscillator generating the carrier wave in the
transmitter. We assume that this synchronization has been achieved. The DSB-SC component of
the modulated signal 𝑠(𝑡) is expressed as
𝑠(𝑡) = 𝐴𝑐 𝑚(𝑡) cos(2𝜋𝑓𝑐 𝑡) (2.40)
Pre-detection SNR
For the signal 𝑠(𝑡) of Eq. (2.40), the average power of the signal component is given by expected
value of the squared magnitude. Since the carrier and modulating signal are independent, this can
be broken down into two components as follows:
𝐸[𝑠 2 (𝑡)] = 𝐸[(𝐴𝑐 cos 2𝜋𝑓𝑐 𝑡)2 ]𝐸[𝑚2 (𝑡)] (2.41)
If we let, 𝑃 = 𝐸[𝑚2 (𝑡)]
be the average signal (message) power, we have
2
𝐴2𝑐 𝑃 (2.42)
𝐸[𝑠 (𝑡)] =
2
𝐴2𝑐 𝑃
That is, the average received signal power due to the modulated component is . If the band-pass
2
filter has a noise bandwidth 𝐵𝑇 , then the noise power passed by this filter is by definition 𝑁0 𝐵𝑇 .
Consequently, the SNR of the signal is
𝐷𝑆𝐵−𝑆𝐶
𝐴2𝑐 𝑃 (2.43)
𝑆𝑁𝑅𝑝𝑟𝑒 =
2𝑁0 𝐵𝑇
Post-detection SNR
Next, we wish to determine the post-detection SNR of the DSB-SC system. Using the narrowband
representation of the band-pass noise, the signal at the input to the coherent detector of Fig. 2.29
may be represented as
31 | P a g e
𝑥(𝑡) = 𝑠(𝑡) + 𝑛𝐼 (𝑡) cos 2𝜋𝑓𝑐 𝑡 − 𝑛𝑄 (𝑡) sin 2𝜋𝑓𝑐 𝑡 (2.44)

Where 𝑛𝐼 (𝑡) and 𝑛𝑄 (𝑡) are the in-phase and quadrature components of 𝑛(𝑡) with respect to the
carrier. The output of the product modulator in Fig. 2.29 is given by
1 1 (2.45)
𝐴𝑐 𝑚(𝑡) + 𝑛𝐼 (𝑡)
𝑣(𝑡) = 𝑥(𝑡) cos(2𝜋𝑓𝑐 𝑡) =
2 2
1 1
+ [𝐴𝑐 𝑚(𝑡) + 𝑛𝐼 (𝑡)] cos(4𝜋𝑓𝑐 𝑡) − 𝑛𝑄 (𝑡) sin(4𝜋𝑓𝑐 𝑡)
2 2
Where we have used the double-angle formula
1 + cos 2𝜃 sin 2𝜃
cos 𝜃 cos 𝜃 = 𝑎𝑛𝑑 cos 𝜃 sin 𝜃 =
2 2
After the low-pass filter, the output 𝑦(𝑡) will become;
1 1 (2.46)
𝐴𝑐 𝑚(𝑡) + 𝑛𝐼 (𝑡)
𝑦(𝑡) =
2 2
This equation indicates the following points:
The message signal 𝑚(𝑡) and in-phase noise component 𝑛𝐼 (𝑡) of the filtered noise 𝑛(𝑡)
appear additively at the receiver output.
The quadrature component 𝑛𝑄 (𝑡) of the noise 𝑛(𝑡) is completely rejected by the coherent
detector.
From Eq. (2.46), we may compute the post-detection SNR by noting the following:
1
The message component is 2 𝐴𝑐 𝑚(𝑡), so analogous to the computation of the pre-detection
1
signal power, the post-detection signal power is 4 𝐴2𝑐 𝑃, where 𝑃 is the average message
power.
1
The noise component is 𝑛𝐼 (𝑡) after low-pass filtering and this in-phase component has a
2
noise spectral density of 𝑁0 over the bandwidth from −𝐵𝑇 ⁄2 to 𝐵𝑇 ⁄2. If the low-pass filter
has a noise bandwidth 𝑊, corresponding to the message bandwidth, which is less than or
equal to 𝐵𝑇 ⁄2, then the output noise power is
𝑊 (2.47)
𝐸[𝑛𝐼2 (𝑡)] = ∫ 𝑁0 𝑑𝑓 = 2𝑁0 𝑊
−𝑊
Combining these observations, we obtain the post-detection SNR of
𝐴2𝑐 𝑃 ⁄4 𝐴2𝑐 𝑃 (2.48)
𝑆𝑁𝑅𝑝𝑜𝑠𝑡 = =
2𝑊𝑁0 ⁄4 2𝑊𝑁0
32 | P a g e
Consequently, if 𝑊 ≈ 𝐵𝑇 ⁄2, the post-detection SNR is twice the pre-detection SNR. This is due
to the fact that the quadrature component of the noise has been discarded by the synchronous
demodulator.
Figure of merit
For DSB-SC modulation the average modulated message power is given by Eq. (2.42), and the
average noise power for a message of bandwidth W is 𝑁0 𝑊. Consequently the reference SNR
for this transmission scheme is 𝑆𝑁𝑅𝑟𝑒𝑓 = 𝐴2𝑐 𝑃⁄2𝑁0 𝑊 . The corresponding figure of merit for this
receiver is;
𝑆𝑁𝑅𝑝𝑜𝑠𝑡 (2.49)
𝐹𝑖𝑔𝑢𝑟𝑒 𝑜𝑓 𝑚𝑒𝑟𝑖𝑡 = =1
𝑆𝑁𝑅𝑟𝑒𝑓
This illustrates that we lose nothing in performance by using a band-pass modulation scheme
compared to the baseband scheme, even though the bandwidth of the former is twice as wide.
Noise in AM receivers using envelop detection
Recall that the envelope-modulated signal is represented by
𝑠(𝑡) = 𝐴𝑐 (1 + 𝑘𝑎 𝑚(𝑡)) cos 2𝜋𝑓𝑐 𝑡 (2.50)
Now we would like to perform noise analysis for an AM system using an envelope detector. The
receiver model is depicted in Fig. 2.30;
Fig. 2.30: Model of AM receiver using envelope detection.

Pre-detection SNR
In expression of 𝑠(𝑡) for standard AM, the average power of the carrier component is 𝐴2𝑐 ⁄2 due to
the sinusoidal nature of the carrier. The power in the modulated part of the signal is
𝐸[(1 + 𝑘𝑎 𝑚(𝑡))2 ] = 𝐸[1 + 2𝑘𝑎 𝑚(𝑡) + 𝑘𝑎2 𝑚2 (𝑡)] (2.51)
1 + 2𝑘𝑎 𝐸[𝑚(𝑡)] + 𝑘𝑎2 𝐸[𝑚2 (𝑡)] = 1 + 𝑘𝑎2 𝑃
where we assume the message signal 𝑚(𝑡) has zero mean, and the message power 𝑃 = 𝐸[𝑚2 (𝑡)].
Consequently, the received signal power is 𝐴2𝑐 (1 + 𝑘𝑎2 𝑃)⁄2.
The pre-detection signal-to-noise ratio is given by
33 | P a g e
𝐴𝑀
𝐴2𝑐 (1 + 𝑘𝑎2 𝑃) (2.52)
2𝑁0 𝐵𝑇
Where 𝐵𝑇 is the noise bandwidth of the band-pass filter.
Post-detection SNR
We can represent the noise in terms of its in-phase and quadrature components, and consequently
model the input to the envelope detector as
𝑥(𝑡) = 𝑠(𝑡) + 𝑛(𝑡) = [𝐴𝑐 + 𝐴𝑐 𝑘𝑎 𝑚(𝑡) + 𝑛𝐼 (𝑡)] cos 2𝜋𝑓𝑐 𝑡 − 𝑛𝑄 (𝑡) sin 2𝜋𝑓𝑐 𝑡 (2.53)
Conceptually, this can be represented in a phasor diagram as shown in Fig. 2.31, where the signal
component of the phasor is 𝐴𝑐 (1 + 𝑘𝑎 𝑚(𝑡)), and the noise has two orthogonal phasor components,
𝑛𝐼 (𝑡) and 𝑛𝑄 (𝑡).
Fig. 2.31: Phasor diagram for AM wave plus narrowband noise.

From Fig. 2.31, the output of the envelope detector is the amplitude of the phasor representing
𝑥(𝑡) and it is given by
1⁄2 (2.54)
𝑦(𝑡) = 𝑒𝑛𝑣𝑒𝑙𝑜𝑝 𝑜𝑓 𝑥(𝑡) = {[𝐴𝑐 (1 + 𝑘𝑎 𝑚(𝑡) + 𝑛𝐼 (𝑡))]2 + 𝑛𝑄2 (𝑡)}
If we assume that the signal is much larger than the noise, then using the approximation when we
may write √𝐴2 + 𝐵 2 ≈ 𝐴 when 𝐴 ≫ 𝐵, hence;
𝑦(𝑡) = 𝐴𝑐 + 𝐴𝑐 𝑘𝑎 𝑚(𝑡) + 𝑛𝐼 (𝑡) (2.55)
The dc component (𝐴𝑐 ), could be easily removed by dc-blocking capacitor. Accordingly, the
post-detection SNR for the envelope detection of AM, using a message bandwidth W, is given
by
𝐴𝑀
𝐴2𝑐 𝑘𝑎2 𝑃 (2.56)
𝑆𝑁𝑅𝑝𝑜𝑠𝑡 =
2𝑁0 𝑊
Figure of merit
For AM modulation, the average transmitted power is given by 𝐴2𝑐 (1 + 𝑘𝑎2 𝑃)⁄2, consequently the
𝐴2𝑐 (1+𝑘𝑎
2 𝑃)
reference SNR is . Combining this result with Eq. (2.56), the figure of merit for this AM
2𝑁0 𝑊
modulation-demodulation scheme becomes;
34 | P a g e
𝐴𝑀
𝑆𝑁𝑅𝑝𝑜𝑠𝑡 𝑘𝑎2 𝑃 (2.57)
𝐹𝑖𝑔𝑢𝑟𝑒 𝑜𝑓 𝑚𝑒𝑟𝑖𝑡 = =
𝑆𝑁𝑅𝑟𝑒𝑓 1 + 𝑘𝑎2 𝑃
2
Since the product 𝑘𝑎 𝑃 is always less than unity (otherwise the signal would be over modulated),
the figure of merit for this system is always less than 0.5. Hence, the noise performance of an
envelope-detector receiver is always inferior to a DSB-SC receiver, the reason is that part of the
power is wasted transmitting the carrier as a component of the modulated (transmitted) signal.
Noise in SSB Receivers
Using the definitions of SSB modulation, we assume that only one sideband is transmitted, so that
we may express the modulated wave as
1 1 (2.58)
𝑆𝑆𝑆𝐵 (𝑡) = 𝐴𝑐 𝑚(𝑡) cos(2𝜋𝑓𝑐 𝑡) + 𝐴𝑐 𝑚
̂ (𝑡) sin(2𝜋𝑓𝑐 𝑡)
2 2
̂ (𝑡) is the Hilbert transform of the message signal. Proceeding in a manner similar to that
where 𝑚
for the DSB-SC receiver, we find that the in-phase and quadrature components of the SSB
modulated wave contribute an average power of 𝐴2𝑐 𝑃 ⁄8 each. The average power of 𝑠(𝑡) is
therefore 𝐴2𝑐 𝑃⁄4. This result is half that of the DSB-SC case.
Pre-detection SNR
For SSB signal, the transmission bandwidth 𝐵𝑇 is approximately equal to the message
bandwidth 𝑊. Consequently, using the signal power calculation of the previous section, the pre-
detection SNR of a coherent receiver with SSB modulation is
𝑆𝑆𝐵
𝐴2𝑐 𝑃 (2.59)
4𝑁0 𝑊
Post-detection SNR
Using the same receiver of Fig. 2.29, the band-pass signal after multiplication with the
synchronous oscillator output is
1 𝐴𝑐 1 𝐴𝑐 (2.60)
𝑣(𝑡) = 𝑥(𝑡) cos(2𝜋𝑓𝑐 𝑡) = ( 𝑚(𝑡) + 𝑛𝐼 (𝑡)) + ( 𝑚(𝑡) + 𝑛𝐼 (𝑡)) cos(4𝜋𝑓𝑐 𝑡)
2 2 2 2
1 𝐴𝑐
− ( 𝑚 ̂ (𝑡) + 𝑛𝑄 (𝑡)) cos(4𝜋𝑓𝑐 𝑡)
2 2
After low-pass filtering 𝑣(𝑡), we are left with
1 𝐴𝑐 (2.61)
𝑦(𝑡) = ( 𝑚(𝑡) + 𝑛𝐼 (𝑡))
2 2
35 | P a g e
The message component in the receiver output is 𝐴𝑐 𝑚(𝑡)⁄4 so that the average power of the
1
recovered message is 𝐴2𝑐 𝑃 ⁄16 and the corresponding noise power is 4 𝑁0 𝑊. Accordingly, the post-
detection SNR of a system using SSB modulation in the transmitter and coherent detection in the
receiver is the ratio of these two powers; namely,
𝑆𝑆𝐵
𝐴2𝑐 𝑃 (2.62)
4𝑁0 𝑊
Figure of merit
𝐴2𝑐 𝑃
The average signal power for the SSB system, as discussed above, is . Consequently, the
4
𝐴2 𝑃
reference SNR is 4𝑁𝑐 𝑊. The figure of merit for the SSB system is the ratio of Eq. (2.62) to the
0
reference SNR becomes;

𝑆𝑆𝐵
𝑆𝑁𝑅𝑝𝑜𝑠𝑡 (2.63)
𝐹𝑖𝑔𝑢𝑟𝑒 𝑜𝑓 𝑚𝑒𝑟𝑖𝑡 = =1
𝑆𝑁𝑅𝑟𝑒𝑓
Consequently, SSB transmission has the same figure of merit as DSB-SC. The performance of
vestigial sideband with coherent detection is also similar to that of SSB.
Comparing the results for the different AM schemes, we find that there are a number of design
tradeoffs. DSB-SC provides the same SNR performance as the baseband reference model but
requires synchronization circuitry to perform coherent detection while standard AM simplifies the
receiver design significantly as it is implemented with an envelope detector. However, standard
AM requires significantly more transmitter power to obtain the same SNR performance as the
baseband reference model. SSB achieves the same SNR performance as the baseband reference
model but only requires half the transmission bandwidth of the DSC-SC system. On the other
hand, it requires more transmitter processing. These observations show that communication system
design involves a tradeoff between power, bandwidth, and processing complexity.
36 | P a g e
Chapter Three
Angle Modulation
Previously, with amplitude modulation systems, the amplitude of the carrier is varied in some way
with the message, keeping the phase and frequency of the carrier fixed. There is also another way of
modulating a sinusoidal carrier wave namely, angle modulation, in which the angle of the carrier
wave is varied according to the message signal. In this second family of modulation techniques, the
amplitude of the carrier wave is maintained constant.
Let 𝜃𝑖 (𝑡) denote the angle of a modulated sinusoidal carrier at time 𝑡; it is assumed to be a function
of the message signal. We express the resulting angle-modulated wave as;
𝑠(𝑡) = 𝐴𝑐 cos[𝜃𝑖 (𝑡)] (3.1)
where 𝐴𝑐 is the carrier amplitude. A complete oscillation occurs whenever the angle 𝜃𝑖 (𝑡) changes
by 2𝜋 radians. If 𝜃𝑖 (𝑡) increases monotonically with time, then the average frequency in hertz, over
a small interval from 𝑡 to 𝑡 + ∆𝑡 is given by
𝜃𝑡 (𝑡 + ∆𝑡) − 𝜃𝑖 (𝑡)
𝑓∆𝑡 (𝑡) =
2𝜋∆𝑡
Allowing the time interval ∆𝑡 to approach zero leads to the following definition for the instantaneous
frequency of the angle-modulated signal 𝑠(𝑡).
𝜃𝑡 (𝑡 + ∆𝑡) − 𝜃𝑖 (𝑡) 1 𝑑𝜃𝑖 (𝑡)
𝑓𝑖 (𝑡) = lim 𝑓∆𝑡 (𝑡) = lim [ ]=
∆𝑡→0 ∆𝑡→0 2𝜋∆𝑡 2𝜋 𝑑𝑡
Thus according to Eq. (3.1), we may interpret the angle-modulated signal as a rotating phasor of
𝑑𝜃𝑖 (𝑡)
length 𝐴𝑐 and angle 𝜃𝑖 (𝑡). The angular velocity of such a phasor is measured in radians per
𝑑𝑡
second. In the simplest case of an unmodulated carrier, the angle 𝜃𝑖 (𝑡) can be;
𝜃𝑖 (𝑡) = 2𝜋𝑓𝑐 𝑡 + ∅𝑐 𝑓𝑜𝑟 𝑚(𝑡) = 0 (3.2)
There are two common ways in which a message 𝑚(𝑡) may be embedded onto the angle of a carrier.
1. A frequency modulator (FM) accepts the input as a frequency.
𝑚(𝑡) Frequency 𝑆𝐹𝑀 (𝑡)

Modulator
𝑐(𝑡) = 𝐴𝑐 cos(2𝜋𝑓𝑐 𝑡)
Fig 3.1 Block diagram of frequency modulator
1|Page
FM is that form of angle modulation in which the instantaneous frequency is varied linearly with the
message signal as shown by
𝑓𝑖 (𝑡) = 𝑓𝑐 + 𝑘𝑓 𝑚(𝑡) (3.3)
The constant term 𝑓𝑐 represents the frequency of the unmodulated carrier; the constant 𝑘𝑓 represents
the frequency-sensitivity factor of the modulator, expressed in hertz per volt on the assumption
that 𝑚(𝑡) is a voltage waveform. Integrating Eq. (3.3) with respect to time and multiplying the result
by 2𝜋, we get
𝑡 𝑡 (3.4)
𝜃𝑖 (𝑡) = 2𝜋 ∫ 𝑓𝑖 (𝜏) 𝑑𝜏 = 2𝜋𝑓𝑐 𝑡 + 2𝜋𝑘𝑓 ∫ 𝑚(𝜏)𝑑𝜏
0 0
where the second term accounts for the increase or decrease in the instantaneous phase 𝜃𝑖 (𝑡) due to
the message signal 𝑚(𝑡). The frequency-modulated wave is therefore
𝑡 (3.5)
𝑆𝐹𝑀 (𝑡) = 𝐴𝑐 cos [2𝜋𝑓𝑐 𝑡 + 2𝜋𝑘𝑓 ∫ 𝑚(𝜏)𝑑𝜏]
0
2. A phase modulator interprets the input as a phase.
Phase
𝑚(𝑡) 𝑆𝑃𝑀 (𝑡)
Modulator
Fig 3.2 Block diagram of phase modulator
PM is another form of angle modulation in which the instantaneous angle 𝜃𝑖 (𝑡) is varied linearly with
the message signal 𝑚(𝑡) as shown by
𝜃𝑖 (𝑡) = 2𝜋𝑓𝑐 𝑡 + 𝑘𝑝 𝑚(𝑡) (3.6)
The term 2𝜋𝑓𝑐 𝑡 represents the angle of the unmodulated carrier with the constant ∅𝑐 set equal to zero
for convenience of presentation; the constant 𝑘𝑝 represents the phase sensitivity factor of the
modulator, expressed in radians per volt on the assumption that 𝑚(𝑡) is a voltage waveform. The
phase-modulated wave 𝑠(𝑡) is correspondingly described in the time domain by
𝑆𝑃𝑀 (𝑡) = 𝐴𝑐 cos[2𝜋𝑓𝑐 𝑡 + 𝑘𝑝 𝑚(𝑡)] (3.7)
2|Page
Basic Principles of Frequency Modulation

In FM, the carrier amplitude remains constant and the carrier frequency is changed by the modulating
signal. As the amplitude of the information signal varies, the carrier frequency shifts proportionately.
As the modulating signal amplitude increases, the carrier frequency increases. If the amplitude of the
modulating signal decreases, the carrier frequency decreases. The reverse relationship can also be
implemented. A decreasing modulating signal increases the carrier frequency above its center value,
whereas an increasing modulating signal decreases the carrier frequency below its center value. As
the modulating signal amplitude varies, the carrier frequency varies above and below its normal
center, or resting, frequency with no modulation. The amount of change in carrier frequency produced
by the modulating signal is known as the frequency deviation 𝑓𝑑 . Maximum frequency deviation
occurs at the maximum amplitude of the modulating signal. The frequency of the modulating signal
determines the frequency deviation rate, or how many times per second the carrier frequency deviates
above and below its center frequency.
An FM signal is illustrated in Fig. 3.1 (c). Normally the carrier in Fig. 3.1 (a) is a sine wave, but it is
shown as a triangular wave here to simplify the illustration. With no modulating signal applied, the
carrier frequency is a constant-amplitude sine wave at its normal resting frequency. The modulating
information signal Fig. 3.1 (b)] is a low-frequency sine wave. As the sine wave goes positive, the
frequency of the carrier increases proportionately. The highest frequency occurs at the peak amplitude
of the modulating signal. As the modulating signal amplitude decreases, the carrier frequency
decreases. When the modulating signal is at zero amplitude, the carrier is at its center frequency point.
When the modulating signal goes negative, the carrier frequency decreases. It continues to decrease
until the peak of the negative half-cycle of the modulating sine wave is reached. Then as the
modulating signal increases toward zero, the carrier frequency again increases. Note that the
frequency of the modulating signal has no effect on the amount of deviation, which is strictly a
function of the amplitude of the modulating signal.
3|Page
Fig. 3.3: FM and PM signals. (a) Carrier. (b) Modulating signal. (c) FM signal. (d) PM signal.
Principles of Phase Modulation

When the amount of phase shift of a constant-frequency carrier is varied in accordance with a
modulating signal, the resulting output is a phase modulation (PM) signal [see Fig. 3.1 (d)]. Imagine
a modulator circuit whose basic function is to produce a phase shift, i.e., a time separation between
two sine waves of the same frequency. Assume that a phase shifter can be built that will cause the
amount of phase shift to vary with the amplitude of the modulating signal. The greater the amplitude
of the modulating signal, the greater the phase shift. Assume further that positive alternations of the
modulating signal produce a lagging phase shift and negative signals produce a leading phase shift.
4|Page
If a constant-amplitude, constant-frequency carrier sine wave is applied to the phase shifter whose
phase shift is varied by the intelligence signal, the output of the phase shifter is a PM wave. As the
modulating signal goes positive, the amount of phase lag, and thus the delay of the carrier output,
increases with the amplitude of the modulating signal. The result at the output is the same as if the
constant-frequency carrier signal had been stretched out, or had its frequency lowered. When the
modulating signal goes negative, the phase shift becomes leading. This causes the carrier sine wave
to be effectively speeded up, or compressed. The result is the same as if the carrier frequency had
been increased.
Interrelationships between PM and FM
If we differentiate 𝑚(𝑡) before applying to a frequency modulator, the resulting 𝜃𝑖 (𝑡) from (3.4) is
𝑡
𝑑𝑚(𝑡) (3.8)
𝜃𝑖 (𝑡) = 2π𝑓𝑐 (𝑡) + 2π𝑘𝑓 ∫ 𝑑𝑡 = 2π𝑓𝑐 (𝑡) + 2π𝑘𝑓 𝑚(𝑡)
0 𝑑𝑡
This phase term has exactly the same form as the PM version of 𝜃𝑖 (𝑡) in Eq. (3.6), except for the
presence of a constant 2. The presence of this constant can be compensated by setting 𝑘𝑓 as
that 2𝑘𝑓 = 𝑘𝑝 . Thus, the following scheme is equivalent to PM:
PM signal
Differentiator Frequency
m(t) SPM (t )
Modulator
Fig. 3.4: PM signal can be obtained using frequency modulation
Further, suppose we integrate 𝑚(𝑡) before applying it to a phase modulator. The result from (3.6) is
𝑡 (3.9)
𝜃𝑖 (𝑡) = 2𝜋𝑓𝑐 𝑡 + 𝑘𝑝 ∫ 𝑚(𝑡)𝑑𝑡
0
This version of 𝜃𝑖 (𝑡) now has the same form as 𝜃𝑖 (𝑡) for the case of FM in Eq. (3.4), except for the
presence of the 2 term which can be compensated by adjusting 𝑘𝑝 . Thus, the following scheme is
equivalent to FM:
𝑚(𝑡) Integrator Phase 𝑆𝐹𝑀 (𝑡)
Modulator
c(t)  A c cos 2f c t

Fig. 3.5 FM signal can be obtained using phase modulation
5|Page
It follows therefore that PM and FM are uniquely related to each other. This relationship, in turn,
means that we may deduce the properties of PM from those of FM and vice versa.
Properties of Angle-Modulated Waves
Angle-modulated waves are characterized by some important properties, which distinguish them from
the family of amplitude modulated waves.
1. Constancy of transmitted power
The amplitude of PM and FM waves is maintained at a constant value equal to the carrier amplitude
𝐴𝑐 for all time 𝑡, irrespective of the sensitivity factors 𝑘𝑓 and 𝑘𝑝 . Consequently, the average
transmitted power of angle-modulated waves is a constant, as shown by
1 2
𝑃𝑎𝑣 = 𝐴
2 𝑐
Where it is assumed that the load resistor is 1 ohm.
2. Nonlinearity of the modulation process
Another distinctive property of angle modulation is its nonlinear character. We say so because both
PM and FM waves violate the principle of superposition. Suppose, for example, that the message
signal 𝑚(𝑡) is made up of two different components 𝑚1 (𝑡) and 𝑚2 (𝑡) as shown by
𝑚(𝑡) = 𝑚1 (𝑡) + 𝑚2 (𝑡)
Let 𝑠(𝑡), 𝑠1 (𝑡) and 𝑠2 (𝑡) denote the PM waves produced by 𝑚(𝑡), 𝑚1 (𝑡) and 𝑚2 (𝑡), respectively. In
light of this equation, we may express these PM waves as follows:
𝑠(𝑡) = 𝐴𝑐 [2𝜋𝑓𝑐 𝑡 + 𝑘𝑝 (𝑚1 (𝑡) + 𝑚2 (𝑡))]
𝑠1 (𝑡) = 𝐴𝑐 [2𝜋𝑓𝑐 𝑡 + 𝑘𝑝 𝑚1 (𝑡)]
And 𝑠2 (𝑡) = 𝐴𝑐 [2𝜋𝑓𝑐 𝑡 + 𝑘𝑝 𝑚2 (𝑡)]
From these expressions, despite the fact that 𝑚(𝑡) = 𝑚1 (𝑡) + 𝑚2 (𝑡), we readily see that the principle
of superposition is violated because
𝑠(𝑡) ≠ 𝑠1 (𝑡) + 𝑠2 (𝑡)
The fact that the angle-modulation process is nonlinear complicates the spectral analysis and noise
analysis of PM and FM waves, compared to amplitude modulation. By the same token, the angle-
modulation process has practical benefits of its own. For example, frequency modulation offers a
superior noise performance compared to amplitude modulation, which is attributed to the nonlinear
character of frequency modulation.
6|Page
3. Visualization difficulty of message waveform

In AM, we see the message waveform as the envelope of the modulated wave. This is not so in angle
modulation, as illustrated by the corresponding waveforms of Figs. 3.3(c) and 3.3(d) for PM and FM,
respectively. In general, the difficulty in visualizing the message waveform in angle-modulated waves
is also attributed to the nonlinear character of angle-modulated waves.
4. Trade-off of increased transmission bandwidth for improved noise performance
The use of angle modulation offers the possibility of exchanging an increase in transmission
bandwidth for an improvement in noise performance. Such a tradeoff is not possible with amplitude
modulation schemes.
Narrow band FM (NBFM)
Consider a sinusoidal modulating wave defined by
𝑚(𝑡) = 𝐴𝑚 cos(2𝜋𝑓𝑚 𝑡)
The instantaneous frequency of the resulting FM wave is
𝑓𝑖 (𝑡) = 𝑓𝑐 + 𝑘𝑓 𝐴𝑚 cos(2𝜋𝑓𝑚 𝑡) = 𝑓𝑐 + ∆𝑓 cos(2𝜋𝑓𝑚 𝑡) (3.10)
Where ∆𝑓 = 𝑘𝑓 𝐴𝑚 and it is called the frequency deviation, representing the maximum departure of
the instantaneous frequency of the FM wave from the carrier frequency 𝑓𝑐 . A fundamental
characteristic of sinusoidal frequency modulation is that the frequency deviation is proportional to
the amplitude of the modulating signal and is independent of the modulating frequency. Using Eq.
(3.10) in Eq. (3.4) the angle 𝜃𝑖 (𝑡) of the FM wave is obtained as
∆𝑓 (3.11)
𝜃𝑖 (𝑡) = 2𝜋𝑓𝑐 (𝑡) + sin(2𝜋𝑓𝑚 𝑡)
𝑓𝑚
The ratio of the frequency deviation ∆𝑓 to the modulation frequency 𝑓𝑚 is commonly called the
modulation index of the FM wave. We denote this new parameter by 𝛽 so we write
∆𝑓 (3.12)
𝛽=
𝑓𝑚
And 𝜃𝑖 (𝑡) = 2𝜋𝑓𝑐 (𝑡) + 𝛽 sin(2𝜋𝑓𝑚 𝑡) (3.13)
From Eq. (3.13) we see that, in a physical sense, the parameter 𝛽 represents the phase deviation of
the FM wave that is, the maximum departure of the angle 𝜃𝑖 (𝑡) from the angle 2𝜋𝑓𝑐 (𝑡) of the
unmodulated carrier. Hence, 𝛽 is measured in radians.
The FM wave itself is given by
𝑠𝐹𝑀 (𝑡) = 𝐴𝑐 cos(2𝜋𝑓𝑐 (𝑡) + 𝛽 sin(2𝜋𝑓𝑚 𝑡)) (3.14)
7|Page
For the FM wave of Eq. (3.14) to be narrow-band the modulation index must be small compared to
one radian. To proceed further, we use the trigonometric identity cos(𝐴 + 𝐵) = cos 𝐴 cos 𝐵 −
sin 𝐴 sin 𝐵 to expand Eq. (3.14) as
𝑠𝐹𝑀 (𝑡) = 𝐴𝑐 cos 2𝜋𝑓𝑐 (𝑡) cos[𝛽 sin(2𝜋𝑓𝑚 𝑡)] − 𝐴𝑐 sin 2𝜋𝑓𝑐 (𝑡) sin[𝛽 sin(2𝜋𝑓𝑚 𝑡)] (3.15)
Then under the condition that the modulation index 𝛽 is small compared to one radian, we may use
the following two approximations for all times t:
cos[𝛽 sin(2𝜋𝑓𝑚 𝑡)] ≈ 1 𝑎𝑛𝑑 sin[𝛽 sin(2𝜋𝑓𝑚 𝑡)] ≈ 𝛽 sin(2𝜋𝑓𝑚 𝑡)
Accordingly, Eq. (3.15) simplifies to
𝑠𝐹𝑀 (𝑡) ≈ 𝐴𝑐 cos 2𝜋𝑓𝑐 (𝑡) − 𝛽𝐴𝑐 sin 2𝜋𝑓𝑐 (𝑡) sin(2𝜋𝑓𝑚 𝑡) (3.16)
Equation (3.16) defines the approximate form of a narrow-band FM wave produced by the sinusoidal
modulating wave 𝐴𝑚 cos(2𝜋𝑓𝑚 𝑡). From this approximate representation, we deduce the modulator
shown in block diagram form in Fig. 3.4. This modulator involves splitting the carrier wave into two
paths. One path is direct; the other path contains a −90 degree phase-shifting network and a product
modulator, the combination of which generates a DSB-SC modulated wave. The difference between
these two signals produces a narrow-band FM wave, but with some amplitude distortion.
We may expand the modulated wave in Eq. (3.16) further into three frequency components:
1 (3.17)
𝑠𝐹𝑀 (𝑡) ≈ 𝐴𝑐 cos 2𝜋𝑓𝑐 (𝑡) + 𝛽𝐴𝑐 {cos[2𝜋(𝑓𝑐 + 𝑓𝑚 )𝑡] − cos[2𝜋(𝑓𝑐 − 𝑓𝑚 )𝑡]}
2
This expression is somewhat similar to the corresponding one defining an AM wave with basic
difference between an AM wave and a narrow-band FM wave is that the algebraic sign of the lower
side-frequency in the narrow-band FM is reversed. Nevertheless, a narrow-band FM wave requires
essentially the same transmission bandwidth (i.e., 2𝑓𝑚 for sinusoidal modulation) as the AM wave.
Wide-Band Frequency Modulation
We next wish to determine the spectrum of the single-tone FM wave defined by the exact formula in
Eq. (3.14) for an arbitrary value of the modulation index 𝛽. Using the complex baseband
representation of a modulated signal;
𝑠𝐹𝑀 (𝑡) = 𝑹𝒆[𝐴𝑐 𝑒𝑥𝑝(𝑗2𝜋𝑓𝑐 (𝑡) + 𝑗𝛽 sin(2𝜋𝑓𝑚 𝑡))] = 𝑹𝒆[𝑠̃𝐹𝑀 (𝑡)𝑒𝑥𝑝(𝑗2𝜋𝑓𝑐 (𝑡))] (3.18)
Where 𝑠̃𝐹𝑀 (𝑡) = 𝐴𝑐 𝑒𝑥𝑝(𝑗𝛽 sin(2𝜋𝑓𝑚 𝑡)) is the complex envelope of the FM wave 𝑠𝐹𝑀 (𝑡). We may
therefore expand 𝑠̃𝐹𝑀 (𝑡) in the form of a complex Fourier series as follows:
∞
(3.19)
𝑠̃𝐹𝑀 (𝑡) = ∑ 𝑐𝑛 exp(𝑗2𝜋𝑛𝑓𝑚 𝑡)
𝑛=−∞
8|Page
where the complex Fourier coefficient

1⁄2𝑓𝑚 (3.20)
𝑐𝑛 = 𝑓𝑚 𝐴𝑐 ∫ exp[𝑗𝛽 sin(2𝜋𝑓𝑚 𝑡) − 𝑗2𝜋𝑛𝑓𝑚 𝑡] 𝑑𝑡
−1⁄2𝑓𝑚
Define the new variable:

𝑥 = 2𝜋𝑓𝑚 𝑡 𝑎𝑛𝑑 𝑑𝑥 = 2𝜋𝑓𝑚 𝑑𝑡 (3.21)
Hence, we may redefine the complex Fourier coefficient in Eq. (3.20) in the new form
𝐴𝑐 𝜋 (3.22)
𝑐𝑛 = ∫ exp[𝑗(𝛽 sin 𝑥 − 𝑛𝑥)] 𝑑𝑥
2𝜋 −𝜋
The integral on the right-hand side of Eq. (3.22), except for the carrier amplitude 𝐴𝑐 , is referred to as
the 𝑛𝑡ℎ order Bessel function of the first kind and argument 𝛽. This function is commonly denoted
by the symbol 𝐽𝑛 (𝛽), so we may write
1 𝜋 (3.23)
𝐽𝑛 (𝛽) = ∫ exp[𝑗(𝛽 sin 𝑥 − 𝑛𝑥)] 𝑑𝑥
2𝜋 −𝜋
Accordingly, we may rewrite Eq. (3.22) in the compact form
𝑐𝑛 = 𝐴𝑐 𝐽𝑛 (𝛽) (3.24)
Substituting Eq. (3.24) into (3.19), we get, in terms of the Bessel function 𝐽𝑛 (𝛽), the following
expansion for the complex envelope of the FM wave:
∞
(3.25)
𝑠̃ 𝐹𝑀 (𝑡) = 𝐴𝑐 ∑ 𝐽𝑛 (𝛽)exp(𝑗2𝜋𝑛𝑓𝑚 𝑡)
𝑛=−∞
Next, substituting Eq. (3.25) into (3.18), we get

∞
(3.26)
𝑠𝐹𝑀 (𝑡) = 𝑹𝒆 [𝐴𝑐 ∑ 𝐽𝑛 (𝛽) 𝑒𝑥𝑝(𝑗2𝜋(𝑓𝑐 + 𝑛𝑓𝑚 )𝑡)]
𝑛=−∞
The carrier amplitude 𝐴𝑐 is a constant and may therefore be taken outside the real-time operator Re[.].
Moreover, we may interchange the order of summation and real-part operation, as they are both linear
operators. Accordingly, we may rewrite Eq. (3.26) in the simplified form
∞
(3.27)
𝑠𝐹𝑀 (𝑡) = 𝐴𝑐 ∑ 𝐽𝑛 (𝛽) cos(2𝜋(𝑓𝑐 + 𝑛𝑓𝑚 )𝑡)
𝑛=−∞
The discrete spectrum of 𝑠𝐹𝑀 (𝑡) is obtained by taking the Fourier transforms of both sides of Eq.
(3.27), which yields
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∞
𝐴𝑐 (3.28)
𝑆𝐹𝑀 (𝑓) = ∑ 𝐽𝑛 (𝛽) [𝛿(𝑓 − 𝑓𝑐 − 𝑛𝑓𝑚 ) + 𝛿(𝑓 + 𝑓𝑐 + 𝑛𝑓𝑚 )]
2
𝑛=−∞
Equation (3.25) shows that the spectrum of 𝑆𝐹𝑀 (𝑓) consists of an infinite number of delta functions
spaced at 𝑓 = 𝑓𝑐 ± 𝑛𝑓𝑚 𝑓𝑜𝑟 𝑛 = 0, 1, 2, ….
Note that the average power of an FM wave may also be determined from Eq. (3.18), as shown by
∞
1 (3.29)
𝑃 = 𝐴2𝑐 ∑ 𝐽𝑛2 (𝛽)
2
𝑛=−∞
Fig. 3.7 shows the plot of Bessel function versus the modulation index 𝛽 for different positive integer
values of 𝑛.
Fig.3.6: Plots of the Bessel function of the first kind, 𝐽𝑛 (𝛽) for varying order n.
Transmission Bandwidth
In theory an FM wave contains infinite number of side frequencies so the BW required to transmit
such a signal is similarly infinite in extent. In practice however the FM wave is effectively limited to
a finite number. Using Carson’s rule it is given by
BW  2f  2 f m  2f (1  1 /  )  2(   1) f m (3.30)
Consider next the more general case of an arbitrary modulating wave 𝑚(𝑡) with its highest frequency
component denoted by W. We now have a more difficult situation to deal with. One way of tackling
it is to seek a worst-case evaluation of the transmission bandwidth. Specifically, the bandwidth
required to transmit an FM wave generated by an arbitrary modulating wave is based on a worst-case
tone modulation analysis. We first determine the so-called deviation ratio D, defined as the ratio of
10 | P a g e
the frequency deviation which corresponds to the maximum possible amplitude of the modulation
wave to the highest modulation frequency W. These conditions represent the extreme cases possible.
We may thus formally write
∆𝑓
𝐷=
𝑊
The deviation ratio D plays the same role for non-sinusoidal modulation that the modulation index 𝛽
plays for the case of sinusoidal modulation. Hence, replacing 𝛽 by D and replacing 𝑓𝑚 with W, we
may generalize Eq. (3.30) as follows:
𝐵𝑇 = 2(∆𝑓 + 𝑊)
This expression is known as the generalized Carson rule for the transmission bandwidth of an
arbitrary FM signal.
Generation of FM Waves
According to Eq. (3.3), the instantaneous frequency of an FM wave varies linearly with the message
signal 𝑚(𝑡). For the design of a frequency modulator, we therefore need a device that produces an
output signal whose instantaneous frequency is sensitive to variations in the amplitude of an input
signal in a linear manner. There are two methods of generating an FM signal. They are
1. Indirect, and
2. Direct
The indirect method is based on the idea of NBFM, whereas direct FM is based on WBFM.
1. Direct Method
The direct method uses a sinusoidal oscillator, with one of the reactive elements (e.g., capacitive
element) in the tank circuit of the oscillator being directly controllable by the message signal. This
method is very straightforward to implement and it is also capable of providing large frequency
deviations. However, a serious limitation of the direct method is the tendency for the carrier frequency
to drift, which is usually unacceptable for commercial radio applications. To overcome this limitation,
frequency stabilization of the FM generator is required, which is realized through the use of feedback
around the oscillator. Although the oscillator may itself be simple to build, the use of frequency
stabilization adds system complexity to the design of the frequency modulator.
2. Indirect Method of FM Generation
In indirect method, on the other hand, the message signal is first used to produce a NBFM, which is
followed by frequency multiplication to increase the frequency deviation to the desired level. In this
11 | P a g e
method, the carrier-frequency stability problem is alleviated by using a highly stable oscillator (e.g.,
crystal oscillator) in the NBFM generation; this modulation scheme is called the Armstrong WBFM,
in recognition of its inventor. A simplified block diagram of this system is shown in Fig. 3.7.
Fig. 3.7: Block diagram of the indirect method of generating a wide-band FM wave.
The message signal 𝑚(𝑡) is first integrated and then used to phase-modulate a crystal-controlled
oscillator; the use of crystal control provides frequency stability. In order to minimize the distortion
inherent in the phase modulator, the maximum phase deviation or modulation index 𝛽 is purposely
kept small, thereby resulting in a NBFM wave; for the implementation of the narrow-band phase
modulator, we may use the arrangement described in Fig. 3.8. The NBFM wave is next multiplied in
frequency by means of a frequency multiplier so as to produce the desired WBFM wave.
Fig.3.7: Block diagram of an indirect method for generating a narrow-band FM wave.

A frequency multiplier consists of a memoryless nonlinear device followed by a band-pass filter, as
shown in Fig. 3.8. The implication of the nonlinear device being memoryless is that it has no energy-
storage elements. The input-output relation of such a device may be expressed in the general form
𝑣(𝑡) = 𝑎1 𝑠(𝑡) + 𝑎2 𝑠 2 (𝑡) + ⋯ + 𝑎𝑛 𝑠 𝑛 (𝑡) (3.31)
NBFM wave 𝑣(𝑡) WBFM wave

𝑛𝑡ℎ Order BPF
𝑠(𝑡) 𝑠 ′(𝑡)
Non-linearity
(Mod. Index , (Mod. Index 𝑛,
𝐶𝑎𝑟𝑟𝑖𝑒𝑟 𝑓𝑟𝑒𝑞. = 𝑓𝑐 ) Centered at 𝑛𝑓𝑐 𝐶𝑎𝑟𝑟𝑖𝑒𝑟 𝑓𝑟𝑒𝑞. = 𝑛𝑓𝑐 )
Excites nth harmonic of
𝐵𝑊 = 2𝑛𝑓𝑚
input signal
Fig.3.9: Block diagram of frequency multiplier.
12 | P a g e
where 𝑎1 , 𝑎2 , … , 𝑎𝑛 are coefficients determined by the operating point of the device, and 𝑛 is the
highest order of nonlinearity. In other words, the memoryless nonlinear device is an 𝑛𝑡ℎ power-law
device. The only problem with indirect method of FM generation, is that if a high value of  is
required, the result could be that the value 𝑓𝑐 becomes too large to be of practical use (recall both 𝑓𝑐
and  get multiplied by 𝑛). To fix this problem, we perform the multiplication in two steps, with a
frequency-mixing (heterodyning) operation in the middle. The mixer translates the FM spectrum to a
lower value of 𝑓𝑐 without affecting the value of .
Extracts lower freq. component and hence
shifts FM spectrum to a lower value
NBFM WBFM wave
𝑛1𝑡ℎ order BPF Mixer 𝑛2𝑡ℎ order BPF
wave Non-linearity Non-linearity
𝑀𝑜𝑑. 𝐼𝑛𝑑𝑒𝑥 = 𝑛1 𝑛2 ,
Multiplies freq. by 𝑛2 𝐶𝑎𝑟𝑟𝑖𝑒𝑟 𝑓𝑟𝑒𝑞.
Multiplies freq. by 𝑛1 LO
= 𝑛2 (𝑛1 𝑓𝑐  𝑓0 )
Fig 3.10 Frequency multiplication in two steps, with a frequency-mixing
A difficulty with this approach is that if there is any phase jitter in the original NBFM generator
oscillator, it gets multiplied by a factor 𝑛1 𝑛2 at the output. This can result in excessive phase jitter at
the output, which will interfere with the desired frequency modulation. Likewise, any jitter in the
mixer local oscillator will be multiplied 𝑛2 times at the output. Therefore, this approach requires very
stable crystal-controlled oscillators to function properly.
Practical Circuits to Generate FM Waves
1. Varactor Diode Modulator
A varactor diode may be used to generate FM directly. All reverse-biased diodes exhibit a junction
capacitance that varies inversely with the amount of reverse bias. A diode that is physically
constructed so as to enhance this characteristic is termed as varactor diode. Fig 2.37 shows a
schematic of a varactor diode modulator.
Fig. 3.11 Varactor diode modulator
13 | P a g e
In varactor diode modulator, the modulating signal applied to the varactor changes the capacitance of
the diode and, thus, the frequency of oscillation. With no message signal applied, the parallel
combination of C1, L1, and D1's capacitance forms the resonant carrier frequency. The diode D1 is
effectively in parallel with L1 and C1 assuming the coupling capacitors C3 and C2 that isolates the dc
levels and from mixing with intelligence signal and high-frequency carrier looks like a short to the
signals at frequencies other than dc. When the message signal is applied to the varactor diode, its
reverse bias is varied, which causes the diode's junction capacitance to vary in step with the message
signal amplitude. The oscillator frequency is subsequently varied as required for FM, and FM signal
is generated. For simplicity, the oscillator circuitry is not shown in the circuit.
Varactor diode FM modulators are extremely popular because they are simple to use and are reliable.
However they are used primarily for low index applications, such as two-way mobile radio.
2. Reactance Modulator
Fig 3.12 illustrates a typical reactance modulator circuit diagram using a BJT as an active device. This
circuit configuration is called a reactance modulator because the amplifier looks like a variable
reactance load to the LC tank circuit. The circuit consists of reactance circuit and the master oscillator
which is left out for simplicity. The reactance circuit operates on the master oscillator to cause its
resonant frequency to shift up or down depending on the modulating signal being applied. The
reactance circuit appears capacitive to the master oscillator. In this case, the reactance looks like a
variable capacitor in the oscillator's tank circuit.
Transistor Q1 makes up the reactance modulator circuit. Resistors R1 and R2 establish a voltage divider
network that biases Q1. Resistor R3 furnishes emitter feedback to thermally stabilize Q1. Capacitor
C3 is a bypass component that prevents ac input signal degeneration. Capacitor Cs interacts with
transistor Q1's inter-electrode capacitance to cause a varying capacitive reactance directly influenced
by the input modulating signal. The master oscillator can be a Colpitts oscillator built around the
second transistor. Capacitor C4 effectively couples the changes at Q1's collector to the tank circuit of
while blocking dc voltages. When a modulating signal is applied to the base of transistor Q1 via
capacitor C1 and chokes RFC1, the reactance of the transistor changes in relation to that signal. If the
modulating voltage goes up the capacitance of Q1 goes down, and if the modulating voltage goes down
the reactance of Q1 goes up. This change in reactance is felt on Q1's collector and also at the tank
circuit of the oscillator transistor. As capacitive reactance at Q1 goes up, the resonant frequency of the
oscillator decreases. Conversely, if Q1's capacitive reactance goes down the oscillator resonant
14 | P a g e
frequency increases. The reactance modulator is efficient and provides a large deviation. It is popular
and used often in FM transmitters.
Fig. 3.12 Reactance modulator

3. Linear IC Modulators
Linear IC voltage-controlled oscillators and function generators can generate a direct FM output
waveform that is relatively stable, accurate, and directly proportional to the input modulating
signal.
Demodulation of FM Signals
Frequency demodulation is the process by means of which the original message signal is
recovered from an incoming FM wave. In other words, frequency demodulation is the
inverse of frequency modulation. With the frequency modulator being a device that
produces an output signal whose instantaneous frequency varies linearly with the amplitude
of the input message signal, it follows that for frequency demodulation we need a device
whose output amplitude is sensitive to variations in the instantaneous frequency of the input
FM wave in a linear manner too.
One device, called a frequency discriminator, relies on slope detection followed by
envelope detection. The other device, called a phase-locked loop, performs frequency
demodulation in a somewhat indirect manner.
Recall that the FM signal is given by
𝑡
𝑠(𝑡) = 𝐴𝑐 cos(2𝜋𝑓𝑐 𝑡 + 2𝜋𝑘𝑓 ∫ 𝑚(𝜏) 𝑑𝜏
0
15 | P a g e
The question to be addressed is: how do we recover the message signal 𝑚(𝑡)from the
modulated signal 𝑠(𝑡). We can motivate the formulation of a receiver for doing this
recovery by noting that if we take the derivative of 𝑠(𝑡)with respect to time, then we obtain
𝑡
𝑑𝑠(𝑡)
= −2𝜋𝐴𝑐 [𝑓𝑐 + 𝑘𝑓 𝑚(𝑡)] sin(2𝜋𝑓𝑐 𝑡 + 2𝜋𝑘𝑓 ∫ 𝑚(𝜏) 𝑑𝜏
𝑑𝑡 0
Inspecting this equation, we observe that the derivative is a band-pass signal with amplitude
modulation defined by the multiplying term [𝑓𝑐 + 𝑘𝑓 𝑚(𝑡)]. Consequently, if 𝑓𝑐 is large
enough such that the carrier is not overmodulated, then we can recover the message signal
with an envelope detector in a manner similar to that described for AM signals. This idea
provides the motivation for the frequency discriminator, which is basically a demodulator
that consists of a differentiator followed by an envelope detector.
However, there are practical issues related to implementation of the discriminator as just
described, particularly, the differentiator.
𝑑
⇌ 𝑗2𝜋𝑓
𝑑𝑡
Where, as usual, ⇌ implies a Fourier-transform relationship. In practical terms, it is
difficult to construct a circuit that has a transfer function equivalent to the right-hand side
of Eq. (4.46) for all frequencies. Instead, we construct a circuit that approximates this
transfer function over the band-pass signal bandwidth, in particular, for 𝑓𝑐 − (𝐵𝑇 ⁄2) ≤
|𝑓| ≤ 𝑓𝑐 + (𝐵𝑇 ⁄2) where 𝐵𝑇 is the transmission bandwidth of the incoming FM
signal 𝑠(𝑡). A typical transfer characteristic that satisfies this requirement is described by
𝑗2𝜋[𝑓 − (𝑓𝑐 − 𝐵𝑇 ⁄2)], 𝑓𝑐 − (𝐵𝑇 ⁄2) ≤ |𝑓| ≤ 𝑓𝑐 + (𝐵𝑇 ⁄2)
𝐻1 (𝑓) = {
0, 𝑂𝑡ℎ𝑒𝑟 𝑤𝑖𝑠𝑒
The transfer characteristic of this so-called slope circuit is illustrated in Fig. 3.12 for
positive frequencies. A practical slope circuit would have a non-unity gain associated with
the slope; but, to simplify matters, we assume that it has unity gain without loss of
generality. The circuit is also not required to have zero response outside the transmission
bandwidth, provided that the circuit is preceded by a band-pass filter centered on 𝑓𝑐 with
bandwidth 𝐵𝑇 .
16 | P a g e
Fig. 3.12: Frequency response of an ideal slope circuit.

It is simplest to proceed with a complex baseband representation of the signal processing
performed by the discriminator. Specifically, following the theory of this representation
developed, we find that the complex envelope of the FM signal 𝑠(𝑡) is
𝑡
𝑠̃ (𝑡) = 𝐴𝑐 𝑒𝑥𝑝 (𝑗2𝜋𝑘𝑓 ∫ 𝑚(𝜏) 𝑑𝜏)
0
the applicability of which requires that the carrier frequency 𝑓𝑐 be large compared to
Correspondingly, we may express the complex baseband filter (i.e., slope circuit)
corresponding to Eq. (4.48) as
−𝐵𝑇 𝐵𝑇
̃1 (𝑓) = { 𝑗2𝜋[𝑓 + (𝐵𝑇 ⁄2)],
𝐻 2
≤𝑓≤
2
Let 𝑠̃1 (𝑡) denote the complex envelope of the response of the slope circuit due
to 𝑠̃ (𝑡).Then, according to the band-pass to low-pass transformation, we may express the
Fourier transform of 𝑠̃1 (𝑡) as
1 −𝐵𝑇 𝐵𝑇
𝑆̃1 (𝑓) = 𝐻̃ (𝑓)𝑆(𝑓) = {𝑗2𝜋[𝑓 + (𝐵𝑇 ⁄2)]𝑆(𝑓), 2
≤𝑓≤
2
2 1
To determine 𝑠̃ (𝑡), which is the inverse of 𝑆̃(𝑓), we invoke two pertinent properties of the
Fourier transform.
1. Multiplication of the Fourier transform 𝑆̃(𝑓) by 𝑗2𝜋𝑓 is equivalent to
differentiating the inverse Fourier transform 𝑠̃ (𝑡).
𝑑𝑠̃ (𝑡)
⇌ 𝑗2𝜋𝑓𝑆̃(𝑓)
𝑑𝑡
2. Application of the linearity property to the nonzero part of yields 𝑆̃1 (𝑓)
1 𝑑𝑠̃ (𝑡) 1
𝑠̃1 (𝑡) = + 𝑗𝜋𝐵𝑇 𝑠̃ (𝑡)
2 𝑑𝑡 2
17 | P a g e
𝑡
Substituting for 𝑠̃ (𝑡) = 𝐴𝑐 𝑒𝑥𝑝 (𝑗2𝜋𝑘𝑓 ∫0 𝑚(𝜏) 𝑑𝜏)
𝑡
1 𝑑(𝐴𝑐 𝑒𝑥𝑝 (𝑗2𝜋𝑘𝑓 ∫0 𝑚(𝜏) 𝑑𝜏))
𝑠̃1 (𝑡) =
2 𝑑𝑡
𝑡
1
+ 𝑗𝜋𝐵𝑇 (𝐴𝑐 𝑒𝑥𝑝 (𝑗2𝜋𝑘𝑓 ∫ 𝑚(𝜏) 𝑑𝜏))
2 0
2𝑘𝑓 𝑡
1
𝑠̃1 (𝑡) = 𝑗𝜋𝐴𝑐 𝐵𝑇 [1 + ( ) 𝑚(𝑡)] 𝐴𝑐 𝑒𝑥𝑝 (𝑗2𝜋𝑘𝑓 ∫ 𝑚(𝜏) 𝑑𝜏)
2 𝐵𝑇 0
Finally, the actual response of the slope circuit due to the FM wave 𝑠(𝑡)is given by
𝑠1 (𝑡) = 𝑅𝑒[ 𝑠̃1 (𝑡)exp(𝑗2𝜋𝑓𝑐 𝑡)]
2𝑘𝑓 𝑡
1 𝜋
𝑠1 (𝑡) = 𝑗𝜋𝐴𝑐 𝐵𝑇 [1 + ( ) 𝑚(𝑡)] cos (2𝜋𝑓𝑐 𝑡 + 2𝜋𝑘𝑓 ∫ 𝑚(𝜏) 𝑑𝜏 + )
2 𝐵𝑇 0 2
We can see that 𝑠̃1 (𝑡) is a hybrid modulated wave, exhibiting both amplitude modulation
and frequency modulation of the message signal Provided that we maintain the extent of
amplitude modulation, namely,
2𝑘𝑓
( ) |𝑚(𝑡)|𝑚𝑎𝑥 < 1, 𝑓𝑜𝑟 𝑎𝑙𝑙 𝑡
𝐵𝑇
Then the envelope detector recovers the message signal except for a bias. Specifically,
under ideal conditions, the output of the envelope detector is given by
1 2𝑘𝑓
𝑣1 (𝑡) = 𝑗𝜋𝐴𝑐 𝐵𝑇 [1 + ( ) 𝑚(𝑡)]
2 𝐵𝑇
1
The bias 𝑣1 (𝑡) is defined by the constant term 2 𝑗𝜋𝐴𝑐 𝐵𝑇 . To remove the bias, we may use
a second slope circuit followed by an envelope detector of its own. This time, however, we
design the slope circuit so as to have a negative slope and its output is given by
1 2𝑘𝑓
𝑣2 (𝑡) = 𝑗𝜋𝐴𝑐 𝐵𝑇 [1 − ( ) 𝑚(𝑡)]
2 𝐵𝑇
Accordingly, solving the difference between 𝑣1 (𝑡) and 𝑣2 (𝑡)
𝑣(𝑡) = 𝑣1 (𝑡) − 𝑣2 (𝑡) = 𝑐𝑚(𝑡)
where 𝑐 is a constant.
Hence, we may now construct the FM demodulator block diagram as Fig. 3.13.
18 | P a g e
Fig.3.13: Block diagram of balanced frequency discriminator.

This particular detection system is called a balanced frequency discriminator, where the
term “balanced” refers to the fact that the two slope circuits of the system are related to
each other in the manner described above.
Phase Locked Loop
The phase-locked loop is a feedback system whose operation is closely linked to frequency
modulation. It is commonly used for carrier synchronization, and indirect frequency
demodulation. The latter application is the subject of interest here. Basically, the phase-locked
loop consists of three major components:
Voltage-controlled oscillator (VCO), which performs frequency modulation on its own
control signal.
Multiplier, which multiplies an incoming FM wave by the output of the voltage-
controlled oscillator.
Loop filter of a low-pass kind, the function of which is to remove the high-frequency
components contained in the multiplier’s output signal and thereby shape the overall
frequency response of the system.
As shown in the block diagram of Fig. 3.14, these three components are connected together to
form a closed-loop feedback system. To demonstrate the operation of the phase-locked loop as
a frequency demodulator, we assume that the VCO has been adjusted so that when the control
signal (i.e., input) is zero, two conditions are satisfied:
1. The frequency of the VCO is set precisely at the unmodulated carrier frequency of the
incoming FM wave
2. The VCO output has a 90-degree phase-shift with respect to the unmodulated carrier
wave.
Suppose then that the incoming FM wave is defined by
𝑠(𝑡) = 𝐴𝑐 sin[2𝜋𝑓𝑐 𝑡 + ∅1 (𝑡)]
19 | P a g e
Fig. 3.14: Block diagram of the phase-locked loop.

By definition, the angle is related to the message signal by the integral
𝑡
∅1 (𝑡) = 2𝜋𝑘𝑓 ∫ 𝑚(𝜏)𝑑𝜏
0
Correspondingly, in accordance with points (1) and (2) above, we define the FM wave
produced by the VCO as
𝑟(𝑡) = 𝐴𝑣 cos(2𝜋𝑓𝑐 𝑡 + ∅2 (𝑡))
The angle is related to the control signal of the VCO by the integral
𝑡
∅2 (𝑡) = 2𝜋𝑘𝑣 ∫ 𝑣(𝜏)𝑑𝜏
0
The function of the feedback loop acting around the VCO is to adjust the angle ∅2 (𝑡) so that it
equals ∅1 (𝑡), thereby setting the stage for frequency demodulation. To deal more deeply into
this function and how it can arise, we need to develop a model for the phase-locked loop. We
first note that multiplication of the incoming FM wave by the locally generated FM wave
produces two components (except for the scaling factor 1⁄2):
1. A high-frequency component, which is defined by the double-frequency term namely,
𝑘𝑚 𝐴𝑐 𝐴𝑣 sin[4𝜋𝑓𝑐 𝑡 + ∅1 (𝑡) + ∅2 (𝑡)]
where 𝑘𝑚 is the multiplier gain.
2. A low-frequency component, which is defined by the difference-frequency term namely,
𝑘𝑚 𝐴𝑐 𝐴𝑣 sin[∅1 (𝑡) − ∅2 (𝑡)]
With the loop-filter designed to suppress the high-frequency components in the multiplier’s
output, we may henceforth discard the double-frequency term. Doing this, we may reduce the
signal applied to the loop filter to
𝑒(𝑡) = 𝑘𝑚 𝐴𝑐 𝐴𝑣 sin[∅𝑒 (𝑡)]
Where ∅𝑒 (𝑡) is the phase error defined by
𝑡
∅𝑒 (𝑡) = ∅1 (𝑡) − ∅2 (𝑡) = ∅1 (𝑡) − 2𝜋𝑘𝑣 ∫ 𝑣(𝜏)𝑑𝜏
0
20 | P a g e
When the phase error ∅𝑒 (𝑡) is zero, the phase-locked loop is said to be in phase-lock. It is said
to be near-phase-lock when the phase error ∅𝑒 (𝑡) is small compared with one radian, under
which condition we may use the approximation.
sin[∅𝑒 (𝑡)] ≈ ∅𝑒 (𝑡)
Correspondingly, we may approximate the error signal as
𝐾0
𝑒(𝑡) = 𝑘𝑚 𝐴𝑐 𝐴𝑣 ∅𝑒 (𝑡) = ∅ (𝑡), 𝑤ℎ𝑒𝑟𝑒, 𝐾0 = 𝑘𝑚 𝑘𝑣 𝐴𝑐 𝐴𝑣
𝑘𝑣 𝑒
𝐾0 is called the loop-gain parameter of the phase-lock loop.
The error signal 𝑒(𝑡) acts on the loop filter to produce the overall output 𝑣(𝑡). Let ℎ(𝑡) denote
the impulse response of the loop filter. We may then relate 𝑣(𝑡) to 𝑒(𝑡) by the convolution
integral
∞
𝑣(𝑡) = ∫ 𝑒(𝜏)ℎ(𝑡 − 𝜏)𝑑𝜏
−∞
These equations constitute a linearized feedback model of the phase-locked loop. The model is
depicted in Fig. 3.15(a) with the angle ∅1 (𝑡) of the incoming FM wave acting as input and the
loop filter’s output acting as the overall output of the phase-locked loop. From the linearized
feedback model of Fig. 3.15(a), we observe three points pertinent to the problem at hand:
1. The feedback path is defined solely by the scaled integrator described in expression
for ∅2 (𝑡), which is the VCO’s contribution to the model. Correspondingly, the inverse
of this feedback path is described in the time domain by the scaled differentiator:
1 𝑑∅2 (𝑡)
𝑣(𝑡) = ( )
2𝜋𝑘𝑣 𝑑𝑡
2. The closed-loop time-domain behavior of the phase-locked loop is described by the
overall output 𝑣(𝑡) produced in response to the angle ∅1 (𝑡) in the incoming FM
wave 𝑠(𝑡).
3. The magnitude of the open-loop transfer function of the phase-locked loop is controlled
by the loop-gain parameter of 𝐾0 .
21 | P a g e
Fig. 3.15: (a) Linearized model of the phase-locked loop. (b) Approximate form of the
model, assuming that the loop gain is large compared with unity.
We may relate the overall output to the input angle by the approximate formula
1 𝑑∅1 (𝑡)
𝑣(𝑡) ≈ ( )
2𝜋𝑘𝑣 𝑑𝑡
In light of this approximation, we may now simplify the linearized feedback model of Fig.
3.15(a) to the form shown in part (b). Hence, substituting Eq. (4.58) into (4.67), we obtain
𝑡 𝑘𝑓
1 𝑑
𝑣(𝑡) = . (2𝜋𝑘𝑓 ∫ 𝑚(𝜏)𝑑𝜏) = 𝑚(𝑡)
2𝜋𝑘𝑣 𝑑𝑡 0 𝑘𝑣
Slope Detector
The tuned circuit converts frequency variations in to amplitude variations (FM to AM
conversion). Then the peak detector converts the amplitude variations to an output voltage
whose amplitude is proportional to the frequency changes.
Fig. 3.16 Slope detector

1. Balanced slope Detector
A balanced slope detector is simply two single ended slope detectors connected in parallel and fed
180 out of phase. The circuit operation is quite simple. The output from each tuned circuit is
22 | P a g e
proportional to the input frequency, and each output is rectified by its respective peak detector.
Therefore the closer the input frequency is to the tank circuit resonant frequency, the greater the
tank circuit output voltage. The carrier frequency falls exactly halfway between the resonance
frequencies of the two tuned circuits.
Fig. 3.17 Balanced slope detector

FM Transmitters
Fig. 3.20 shows a block diagram of an FM or PM transmitter. The indirect method of FM generation
is used. A stable crystal oscillator is used to generate the carrier signal, and a buffer amplifier is
used to isolate it from the remainder of the circuitry. The carrier signal is then applied to a phase
modulator. The voice input is amplified and processed to limit the frequency range and prevent
over-deviation. The output of the modulator is the desired FM signal. Most FM transmitters are
used in the VHF and UHF range, and crystals are not available to generate those frequencies
directly. As a result, the carrier is usually generated at a frequency considerably lower than the
final output frequency. To get the desired output frequency, one or more frequency multiplier
stages are used. The frequency multiplier not only increases the carrier frequency to the desired
frequency, but also multiplies the frequency deviation produced by the modulator.
Fig. 3.20 A typical FM transmitter using indirect FM with a phase modulator
23 | P a g e
2.2.6. FM Receivers
In FM receivers, the voltage at the output of the audio detector is directly proportional to the
frequency deviation at its input while in PM receivers, the voltage at the output of the audio detector
is directly proportional to the phase deviation at its input.
The FM receiver is similar in many ways to the AM receiver. Both are usually superheterodyne
receivers. Of course the demodulation circuit for FM receivers are the inclusion of a block called
a limiter and one called a de-emphasis network in the FM case.
Fig. 3.21 Block diagram of typical FM receiver

The Limiter: its purpose is to clip all amplitude variations which may exist in the incoming
signal. This clipping by the limiter eliminates noise but doesn’t affect the information content of
the signal because the information is contained in the frequency variations, not in the amplitude
variations.
De-emphasis network: Investigators found that noise which entered the signal as a frequency
modulation occurred with greater likelihood and disturbance in the higher audio frequencies; thus,
the pre-emphasis de-emphasis system functions to reduce frequency modulated noise. The pre-
emphasis network, being located in the in the transmitter causes the higher frequency information
content of the audio signal at the transmitter to be amplified more than the lower frequency
information. The de-emphasis network on the other-hand compensates for this by reducing the gain
of the higher frequency audio signal at the receiver.
Noise in Angle modulated Waves
We now turn to the detection of a frequency-modulated carrier in noise. Recall that the
frequency-modulated signal is given by
𝑡
𝑠(𝑡) = 𝐴𝑐 cos(2𝜋𝑓𝑐 𝑡 + 2𝜋𝑘𝑓 ∫ 𝑚(𝜏) 𝑑𝜏
0
The received FM signal has a carrier frequency 𝑓𝑐 and a transmission bandwidth 𝐵𝑇 such that a
negligible amount of power lies outside the frequency band 𝑓𝑐 ± 𝐵𝑇 for positive frequencies, and
similarly for negative frequencies.
24 | P a g e
Pre-detection SNR
For FM detection, we assume a receiver model as shown in Fig. 3.22. As before, we did in Chapter
two we assume that the noise is a white zero-mean Gaussian process with power spectral
density 𝑁0 ⁄2 .
Fig. 3.22: Model for an FM receiver

The FM detector consists of a band-pass filter, a limiter, a discriminator, and a low-pass filter. The
bandpass filter has a center frequency 𝑓𝑐 and bandwidth 𝐵𝑇 so that it passes the FM signal without
distortion. Ordinarily, 𝐵𝑇 is small compared with the center frequency so that we may use the
narrowband representation 𝑛(𝑡) for the filtered version of the channel noise 𝑤(𝑡) . The pre-
detection SNR in this case is simply the carrier power 𝐴2𝑐 ⁄2 divided by the noise passed by the
bandpass filter 𝑁0 𝐵𝑇 , namely,
𝐹𝑀
𝐴2𝑐
2𝑁0 𝐵𝑇
Post-Detection SNR
The noisy FM signal after band-pass filtering may be represented as
𝑥(𝑡) = 𝑠(𝑡) + 𝑛(𝑡)
The filtered noise 𝑛(𝑡) at the band-pass filter output in Fig. 3.22 in terms of its in-phase and
quadrature components
𝑛(𝑡) = 𝑛𝐼 (𝑡) cos(2𝜋𝑓𝑐 𝑡) − 𝑛𝑄 (𝑡) sin(2𝜋𝑓𝑐 𝑡) = 𝑟(𝑡) cos(2𝜋𝑓𝑐 𝑡 + ∅𝑛 (𝑡))
1⁄2 𝑛 (𝑡)
Where the envelope is 𝑟(𝑡) = [𝑛𝐼2 (𝑡) + 𝑛𝑄2 (𝑡)] and phase ∅𝑛 (𝑡) = tan−1 ( 𝑛𝑄(𝑡) )
𝐼
One of the properties of this polar representation is that the phase ∅𝑛 (𝑡) is uniformly distributed
between 0 and 2𝜋 radians.
To proceed, we note that the phase of 𝑠(𝑡) is
𝑡
∅(𝑡) = 2𝜋𝑘𝑓 ∫ 𝑚(𝜏) 𝑑𝜏
0
Combining these terms, the noisy signal at the output of the band-pass filter may be expressed as
𝑥(𝑡) = 𝑠(𝑡) + 𝑛(𝑡) = 𝐴𝑐 cos(2𝜋𝑓𝑐 𝑡 + ∅(𝑡)) + 𝑟(𝑡) cos(2𝜋𝑓𝑐 𝑡 + ∅𝑛 (𝑡))
25 | P a g e
It is informative to represent 𝑥(𝑡) by means of a phasor diagram, as in Fig. 3.23, where we have
used the signal term 𝑠(𝑡) as the reference.
Fig. 3.23: Phasor diagram for FM signal plus narrowband noise

From the Figure, the phase difference ψ𝑛 (𝑡) = 𝜙𝑛 (𝑡) − 𝜙(𝑡) is the angle between the noise phasor
and the signal phasor. The phase 𝜃(𝑡) of the resultant is given by
𝑟(𝑡) sin(ψ𝑛 (𝑡))
𝜃(𝑡) = 𝜙(𝑡) + tan−1 ( )
𝐴𝑐 + 𝑟(𝑡) cos(ψ𝑛 (𝑡))
The envelope of 𝑥(𝑡) is of no interest to us, because the envelope variations at the bandpass filter
output are removed by the limiter.
To obtain useful results, we make some approximations regarding 𝜃(𝑡). First, we assume that the
carrier-to-noise ratio measured at the discriminator input is large. If R denotes observations of the
sample function 𝑟(𝑡) of the noise envelope, then most of the time the random variable R is small
compared to the carrier amplitude 𝐴𝑐 . Under this condition, and noting that tan−1 (𝜑) ≈ 𝜑
since 𝜑 ≪ 1, the expression for the phase simplifies tox
𝑟(𝑡)
𝜃(𝑡) = 𝜙(𝑡) + sin(ψ𝑛 (𝑡))
𝐴𝑐
We simplify this expression even further by ignoring the modulation component and
replacing ψ𝑛 (𝑡) = 𝜙𝑛 (𝑡) − 𝜙(𝑡) with 𝜙𝑛 (𝑡). This is justified because the phase 𝜙𝑛 (𝑡) is
uniformly distributed between 0 and radians 2𝜋 and, since 𝜙(𝑡) is independent of 𝜙𝑛 (𝑡) it is
reasonable to assume that the phase difference 𝜙𝑛 (𝑡) − 𝜙(𝑡) is also uniformly distributed over 2𝜋
radians. Then noting that the quadrature component 𝑛𝑄 (𝑡) = 𝑟(𝑡) sin[∅𝑛 (𝑡)] of the noise is we
may simplify 𝜃(𝑡) to
𝑡
𝑛𝑄 (𝑡) 𝑛𝑄 (𝑡)
𝜃(𝑡) = ∅(𝑡) + ≈ 2𝜋𝑘𝑓 ∫ 𝑚(𝜏) 𝑑𝜏 +
𝐴𝑐 0 𝐴𝑐
Our objective is to determine the error in the instantaneous frequency of the carrier wave caused
by the presence of the filtered noise. With an ideal discriminator, its output is proportional to the
derivative. Using the expression for 𝜃(𝑡), the ideal discriminator output, scaled by 2𝜋 is therefore
1 𝑑𝜃(𝑡)
𝑣(𝑡) = = 𝑘𝑓 𝑚(𝑡) + 𝑛𝑑 (𝑡)
2𝜋 𝑑𝑡
26 | P a g e
1 𝑛𝑄 (𝑡)
Where 𝑛𝑑 (𝑡) = 2𝜋𝐴
𝑐 𝑑𝑡
Here we can see that, the average output signal power is equal to 𝑘𝑓2 𝑃 where 𝑃 is the average power
of the message signal. To determine the average output noise power, we note that the noise at the
discriminator output 𝑛𝑑 (𝑡) is proportional to the time derivative of the quadrature noise
component 𝑛𝑄 (𝑡). Since the differentiation of a function with respect to time corresponds to
multiplication of its Fourier transform by 𝑗2𝜋𝑓, it follows that we may obtain the noise
process 𝑛𝑑 (𝑡), by passing 𝑛𝑄 (𝑡) through a linear filter with a frequency response equal to
𝑗2𝜋𝑓 𝑗𝑓
𝐺(𝑓) = =
2𝜋𝐴𝑐 𝐴𝑐
This means that the power spectral density 𝑆𝑁𝑑 (𝑓) of the noise is related to the power spectral
density 𝑆𝑁𝑄 (𝑓) of the quadrature noise component as follows:
𝑓2
𝑆𝑁𝑑 (𝑓) = [𝐺(𝑓)]2 𝑆𝑁𝑄 (𝑓) = 𝑆 (𝑡)
𝐴2𝑐 𝑁𝑄
With the band-pass filter in the receiver model of Fig. 3.22 having an ideal frequency response
characterized by bandwidth 𝐵𝑇 and mid-band frequency 𝑓𝑐 , it follows that the narrowband noise
will have a power spectral density characteristic that is similarly shaped. If the input noise is white
then, from the properties of the in-phase and quadrature components of narrowband noise, the
power spectral density of 𝑁𝑄 (𝑡) will be the low-pass equivalent of the sum of the positive and
negative frequency responses of the band-pass filter. This means that the quadrature
component 𝑁𝑄 (𝑡) of the narrowband noise will have the ideal low-pass characteristic shown in Fig.
3.24 (a). The corresponding power spectral density of the noise is shown in Fig. 2.24 (b); that is,
𝑁0 𝑓 2 ,
|𝑓| < 𝐵𝑇
𝑆𝑁𝑑 (𝑓) = { 𝐴2𝑐
0, 𝑜𝑡ℎ𝑒𝑟𝑤𝑖𝑠𝑒
Fig. 3.24: (a) Power spectral density of quadrature component of narrowband noise (b) Power
spectral density at discriminator output. (c) Power spectral density of noise at receiver output.
27 | P a g e
In the receiver model of Fig. 3.22, the discriminator output is followed by a low-pass filter with a
bandwidth equal to the message bandwidth 𝑊. For wideband FM, we usually find that 𝑊 is smaller
than 𝐵𝑇 ⁄2 where 𝐵𝑇 is the transmission bandwidth of the FM signal.
This means that the out-of-band components of noise will be rejected. Therefore, the power spectral
density of the noise appearing at the receiver output is defined by
𝑁0 𝑓 2 ,
|𝑓| < 𝑊
𝑆𝑁0 (𝑓) = { 𝐴2𝑐
0, 𝑜𝑡ℎ𝑒𝑟𝑤𝑖𝑠𝑒
The average output noise power is determined by integrating the power spectral density 𝑆𝑁0 (𝑓)
from −𝑊 to 𝑊. Doing so, we obtain the following result:
𝑁0 𝑊 2 2𝑁0 𝑊 3
𝐴𝑣𝑒𝑟𝑎𝑔𝑒 𝑝𝑜𝑠𝑡 − 𝑑𝑒𝑡𝑒𝑐𝑡𝑖𝑜𝑛 𝑛𝑜𝑖𝑠𝑒 𝑝𝑜𝑤𝑒𝑟 = ∫ 𝑓 𝑑𝑓 =
𝐴2𝑐 −𝑊 3𝐴2𝑐
As mentioned earlier, the average output signal power is 𝑘𝑓2 𝑃, and the post detection SNR becomes;
𝐹𝑀
3𝐴2𝑐 𝑘𝑓2 𝑃
2𝑁0 𝑊 3
Hence, the post-detection SNR of an FM demodulator has a nonlinear dependence on both the
frequency sensitivity and the message bandwidth.
Figure of Merit
With FM modulation, the modulated signal power is simply 𝐴2𝑐 ⁄2, hence the reference SNR
is 𝐴2𝑐 ⁄(2𝑁0 𝑊). Consequently, the figure of merit for this system is given by
𝐹𝑀
𝑆𝑁𝑅𝑝𝑜𝑠𝑡 2𝑁0 𝑊 3 𝑘𝑓2 𝑃
𝐹𝑖𝑔𝑢𝑟𝑒 𝑜𝑓 𝑚𝑒𝑟𝑖𝑡 = 𝐹𝑀 = = 3 ( 2 ) = 3𝐷2
𝑆𝑁𝑅𝑟𝑒𝑓 𝐴2𝑐 𝑊
2𝑁0 𝑊
where, in the last line, we have introduced the definition 𝐷 = 𝑘𝑓 √𝑃⁄𝑊 as the deviation ratio for
the FM system. Recall that the generalized Carson rule yields the transmission bandwidth 𝐵𝑇 =
2(𝑘𝑓 √𝑃 + 𝑊) ≈ 2(𝑘𝑓 √𝑃) for an FM signal. So, substituting 𝐵𝑇 ⁄2 for 𝑘𝑓 √𝑃 in the definition of 𝐷,
the figure of merit for an FM system is approximately given by
3 𝐵𝑇 2
𝐹𝑖𝑔𝑢𝑟𝑒 𝑜𝑓 𝑚𝑒𝑟𝑖𝑡 ≈ ( )
4 𝑊
Consequently, an increase in the transmission bandwidth 𝐵𝑇 provides a corresponding quadratic
increase in the output SNR with an FM system compared to the reference system. Thus, when the
28 | P a g e
carrier to noise level is high, unlike an amplitude modulation system an FM system allows us to
trade bandwidth for improved performance in accordance with a square law.
Example: A tone of unit amplitude and frequency 600 Hz is sent via FM. The FM receiver has
been designed for message signals with a bandwidth upto 1 kHz. The modulation index produced
by the tone is 5 rad. If the carrier signal has twice the amplitude of the message while the noise
spectral density is 2 × 10−5 𝑊 ⁄𝐻𝑧, calculate; the pre-detection SNR, post detection SNR,
reference SNR and figure of merit of the system.
Solution:
𝐹𝑀
𝐴2𝑐 𝐴2𝑐 22 𝑊
𝑆𝑁𝑅𝑝𝑟𝑒 = = = =
2𝑁0 𝐵𝑇 2𝑁0 × 2(𝛽 + 1)𝑓𝑚 2 × 2 × 10−5 × 2(5 + 1) × 600𝑊
𝐹𝑀
2𝑁0 𝑊 3
∆𝑓 𝑘𝑓 𝐴𝑚 𝛽𝑓𝑚 5×600
In which 𝛽=𝑓 = ⇒ 𝑘𝑓 = = = 3000. Substituting this value in the
𝑚 𝑓𝑚 𝐴𝑚 1
𝐹𝑀
𝑆𝑁𝑅𝑝𝑜𝑠𝑡 expression and 𝑃 = 𝐴2𝑚 ⁄2 = 1/2,
𝐹𝑀
3 ∗ 22 ∗ 30002 ∗ 1/2
𝑆𝑁𝑅𝑝𝑜𝑠𝑡 = =
2 ∗ 2 × 10−5 ∗ (103 )3
𝐹𝑀
𝐴2𝑐 22
𝑆𝑁𝑅𝑟𝑒𝑓 = = = 100
2𝑁0 𝑊 2 ∗ 2 × 10−5 ∗ 103
3 𝐵𝑇 2 3 12 × 600 2
𝐹𝑜𝑀 ≈ ( ) = ( ) =
4 𝑊 4 103
29 | P a g e
Chapter Four
Base Band Pulse Signaling
In our discussions so far, some parameter of a continuous sinusoidal carrier wave was varied continuously in
accordance with the message signal. In this chapter, we will see other families of signaling in which some
parameter of a pulse train is varied in accordance with the message signal. This is called pulse modulation, and
in this context, we may distinguish two families of pulse modulation, analog pulse modulation and digital pulse
modulation, depending on how the modulation is performed.
In analog pulse modulation, a periodic pulse train is used as the carrier wave, and some characteristic feature of
each pulse (e.g., amplitude, duration, or position) is varied in a continuous manner in accordance with the
corresponding sample value of the message signal. Thus, in analog pulse modulation, information is transmitted
basically in analog form, but the transmission takes place at discrete times. In digital pulse modulation, on the
other hand, the message signal is represented in a form that is discrete in both time and amplitude, thereby
permitting its transmission in digital form as a sequence of coded pulses. Simply put, digital pulse modulation
has no continuous wave counterpart. The last modulation technique that will be covered in this chapter is called
digital modulation which uses sinusoidal carrier signal like analog modulations but the modulating signals are
digital signals and thus the modulation type is different.
Basic Principles of Data Conversion
Translating an analog signal to a digital signal is called analog-to-digital (A/D) conversion, digitizing a signal,
or encoding. The device used to perform this translation is known as an analog-to-digital (A/D) converter or
ADC. The opposite process is called digital-to-analog (D/A) conversion. The circuit used to perform this is
called a digital-to-analog (D/A) converter (or DAC) or a decoder. The input to a D/A converter may be a serial
or parallel binary number, and the output is a proportional analog voltage level.
A/D Conversion. An analog signal is a smooth or continuous voltage or current variation. Through A/D
conversion these continuously variable signals are changed to a series of pulse trains or binary numbers. A/D
conversion is a process of sampling or measuring the analog signal at regular time intervals. At the times
indicated by the vertical dashed lines in Fig. 4.1, the instantaneous value of the analog signal is measured and a
proportional binary number is generated to represent that sample. As a result, the continuous analog signal is
translated to a series of discrete binary numbers representing samples.
1
Fig. 4.1: Sampling an analog signal.

A key factor in the sampling process is the frequency of sampling 𝑓, which is the reciprocal of the sampling
interval 𝑡 shown in Fig. 4.1. To retain the high frequency information in the analog signal, a sufficient number
of samples must be taken so that the waveform is adequately represented. It has been found that the minimum
sampling frequency is twice the highest analog frequency content of the signal. For example, if the analog signal
contains a maximum frequency variation of 3000 Hz, the analog wave must be sampled at a rate of at least twice
this, or 6000 Hz. This minimum sampling frequency is known as the Nyquist frequency 𝑓𝑁 . And 𝑓𝑁 ≥ 2𝑓𝑚 ,
where 𝑓𝑚 is the frequency of the input signal. For bandwidth limited signals with upper and lower limits of 𝑓2
and 𝑓1 , the Nyquist sampling rate is just twice the bandwidth or 2(𝑓2 − 𝑓1 ).
Aliasing phenomena
The above sampling frequency requirement is based on the assumption that the signal is strictly band-limited.
In practice, however, no information-bearing signal of physical origin is strictly band-limited, with the result
that some degree of undersampling signal is always encountered. Consequently, aliasing is produced by the
sampling process which refers to the phenomenon of a high-frequency component in the spectrum of the signal
seemingly taking on the identity of a lower frequency in the spectrum of its sampled version, as illustrated in
Fig. 4.2. The aliased spectrum shown by the solid curve in Fig. 4.2(b) pertains to an “undersampled” version of
the message signal represented by the spectrum of Fig. 4.2(a).
2
Fig. 4.2: (a) Spectrum of a signal. (b) Spectrum of an undersampled version of the signal, exhibiting the
aliasing phenomenon.
To combat the effects of aliasing in practice, we may use two corrective measures:
1. Prior to sampling, a low-pass anti-alias filter is used to attenuate those high-frequency components
of a message signal that are not essential to the information being conveyed by the signal.
2. The filtered signal is sampled at a rate slightly higher than the Nyquist rate.
The use of a sampling rate higher than the Nyquist rate also has the beneficial effect of easing the design of the
synthesis filter used to recover the original signal from its sampled version.
Pulse Amplitude Modulation
In pulse-amplitude modulation (PAM), the amplitudes of regularly spaced pulses are varied in proportion to the
corresponding sample values of a continuous message signal; the pulses can be of a rectangular form or some
other appropriate shape. PAM as defined here is somewhat similar to natural sampling, where the message
signal is multiplied by a periodic train of rectangular pulses. In natural sampling, however, the top of each
modulated rectangular pulse is permitted to vary with the message signal, whereas in PAM it is maintained flat.
The waveform of a PAM signal is illustrated in Fig. 4.3. The dashed curve in this figure depicts the waveform
of the message signal 𝑚(𝑡) and the sequence of amplitude-modulated rectangular pulses shown as solid lines
represents the corresponding PAM signal 𝑠(𝑡). There are two operations involved in the generation of the PAM
signal:
1. Instantaneous sampling of the message signal every 𝑇𝑠 seconds, where the sampling rate 𝑓𝑠 = 1⁄𝑇𝑠 is
chosen in accordance with the sampling theorem.
2. Lengthening the duration of each sample, so that it occupies some finite value T.
Fig. 4.3: Flat-top sampling of a message signal.
3
In digital circuit technology, these two operations are jointly referred to as “sample and hold.” One important
reason for intentionally lengthening the duration of each sample is to avoid the use of an excessive channel
bandwidth, since bandwidth is inversely proportional to pulse duration. However, care has to be exercised in
how long we make the sample duration 𝑇, as the following analysis reveals.
Generation of PAM signal
It is very easy to generate and demodulate PAM. The signal to be converted to PAM is fed through switch which
is controlled by a pulse train. When pulse is present i.e signal is at high level, switch is closed. When pulse is
absent i.e. signal is at low level switch is open. With this control action of switch we get pulse amplitude
modulated signal waveform at the output terminal of the switch. This pulse amplitude modulated signal is passed
through a pulse shaping network, which gives them flat tops as illustrated in Fig. 4.4.
Fig. 4.4: Generation of PAM signal

Fig. 4.5 shows the waveforms related to PAM generation.
4
Fig. 4.5: PAM waveforms

Transmission Bandwidth of PAM Signal
The pulse duration ′𝜏′ is supposed to be very very small compared to time period 𝑇𝑠 between the two samples.
If the maximum frequency in the signal 𝑥(𝑡) is ′𝑊′ then by sampling theorem, 𝑓𝑠 should be higher than
Nyquist rate or,
1 1
𝑓𝑠 ≥ 2𝑊 𝑜𝑟 𝑇𝑠 ≤ 𝑠𝑖𝑛𝑐𝑒 𝑓𝑠 =
2𝑊 𝑇𝑠
1
And 𝜏 ≪ 𝑇𝑠 ≤ 2𝑊
If ON and OFF time of the pulse is same, then frequency of the PAM pulse becomes,
5
1 1
𝑓= =
𝜏 + 𝜏 2𝜏
Bandwidth required for transmission of PAM signal will be equal to maximum frequency 𝑓𝑚𝑎𝑥 given by the
above equation.
1
𝐵𝑇 ≥ 𝑓𝑚𝑎𝑥 ≥
2𝜏
1 1
Since 𝜏 ≪ 2𝑊 , 𝐵𝑇 ≥ 2𝜏 ≫ 𝑊
Reconstruction of the original signal

The original modulated signal can be detected from the natural PAM by passing the naturally modulated PAM
signal through a low-pass filter. The low-pass filter with cut-off frequency equal to 𝑓𝑚 removes high frequency
ripple and recovers the original modulating signal.
Fig. 4.6: PAM detector

Output of such a demodulator can be shown as follows.
Fig. 4.7: waveforms of PAM detection

In case of flat-top PAM to reduce the aperture effect, equalizer is used. The aperture effect is caused by the
frequency response of the low-pass filter as shown in Fig. 4.8(b). From this figure we see that by using flat-top
samples to generate a PAM signal, we have introduced amplitude distortion as well as a delay of 𝑇⁄2.
6
Fig. 4.8: (a) Rectangular pulse, (b) Spectrum of low pass filter, defined in terms of its magnitude and phase.
This distortion may be corrected by connecting an equalizer in cascade with the low-pass reconstruction filter,
as shown in Fig. 4.9. The equalizer has the effect of decreasing the in-band loss of the reconstruction filter as
the frequency increases in such a manner as to compensate for the aperture effect.
Fig. 4.9: Recovering the message signal 𝑚(𝑡) from the PAM signal 𝑠(𝑡).
Merits and Demerits of PAM
In PAM, generation and demodulation are simple process. However, PAM produces the amplitude variations.
We know that noise affects the amplitude of the waveform. Thus, like AM, PAM is also less immune to noise.
Also PAM waveform has pulses with varying amplitude and therefore power required to transmit them is not
constant. This requires that the transmitter must be able to handle the power required to transmit pulse having
maximum amplitude.
Pulse Time Modulation
In pulse time modulation (PTM), the signal to be transmitted is sampled as in PAM but amplitude of pulses is
held constant, whereas position or width of pulses is made proportional to the amplitude of signal at the sampling
instant. Then we have two types of PTM: Pulse Width Modulation (PWM) and Pulse Position Modulation
(PPM). Since in both PWM and PPM, amplitude is held constant and doesn’t carry information, amplitude
limiters can be used. The amplitude limiters, similar to those used in FM, will clip off the portion of the signal
corrupted by noise and thus provide a good degree of noise immunity. In this section we will see the generation
and demodulation of PWM and in the next one we see the generation and demodulation of PPM.
Pulse Width Modulation
7
In PWM the samples of the message signal are used to vary the width of the individual pulses. This form of
modulation is also referred to as pulse-duration modulation or pulse-length modulation. The modulating signal
may vary the time of occurrence of the leading edge, the trailing edge, or both edges of the pulse. This is
illustrated in Fig. 4.10.
Fig. 4.10: illustration of PWM, (a) trailing edge, (b) leading edge and (c) both edges.
In Fig. 4.11(c) the trailing edge of each pulse is varied in accordance with the message signal, assumed to be
sinusoidal as shown in Fig. 4.11(a). The periodic pulse carrier is shown in Fig. 4.11(b).
Fig. 4.11: Illustration of two different forms of pulse-time modulation for the case of a sinusoidal modulating
wave. (a) Modulating wave. (b) Pulse carrier. (c) PWM wave. (d) PPM wave.
Generation of PWM Signal
8
The Fig. 4.12 shows the block diagram of PWM generator. It consists of sawtooth generator and comparator.
The sawtooth generator generates a sawthooth signal which is used as a sampling signal. The comparator
compares the amplitude of modulating signal 𝑚(𝑡) and the amplitude of sampling signal, i.e. sawtooth signal.
The output of the comparator if high as long as the instantaneous amplitude of 𝑚(𝑡) is greater than that of the
sawthooth signal. Thus, the duration for which the comparator output remains high is directly proportional to
the amplitude of the modulating signal, 𝑚(𝑡). As a result, the comparator output as a PWM signal.
Fig. 4.12: Waveforms of PWM generator

Practical PWM Generator Circuits
Fig. 4.13(a) shows PWM generator which is a monostable multivibrator with a modulating input signal applied
at the control voltage input. Internally, the control voltage is adjusted to the 2⁄3 𝑉𝐶𝐶 . Externally applied
modulating signal changes the control voltage, and hence the threshold voltage level. As a result, time period
required to charge the capacitor up to threshold voltage level changes, giving pulse modulated signal at the
output, as shown in Fig. 4.13(b).
Fig. 4.13: (a) PWM generator circuit. (b) Waveforms of PWM generator
Demodulation of PWM Signal
9
Fig. 4.14(a) shows the block diagram of PWM detector in which the received PWM signal is applied to the
schmitt trigger circuit that removes the noise in the PWM waveform. The regenerated PWM is then applied to
the ramp generator and the synchronization pulse generator. The ramp generator produces ramps for the duration
of pulses such that height of ramps are proportional to the widths of PWM pulses. The maximum ramp voltage
is retaied till the next pulse. On the other hand synchronous pulse generator produces reference pulses with
constant amplitude and pulse width. These pulses are delyed by specific amount of delay as shown in Fig.
4.14(b). the delayed reference pulses and the output of ramp generator is added with the help of adder whose
output is given to the level shifter. Here, negative offset waveform is clipped by rectifier. Finally, output of the
rectifier is passed through a lowpass filter to recover the modulating signal as shown in Fig. 4.14.
Fig. 4.14: (a) PWM detector. (b) Waveforms for PWM detection circuit.
Advantages of PWM
1. Unlike PAM, noise is less, since in PWM, amplitude is held constant.
2. Signal and noise separation is very easy.
Disadvantages of PWM
1. Due to varying pulses width, power contents are variable. This requires the transmitter to be able to
handle the power contents of the pulse having maximum pulse width.
2. Large bandwidth is required for PWM communication as compared to PAM.
Pulse Position Modulation
In this system, the amplitude and width of the pulses are kept constant, while the position of each pulse, with
reference to the position of a reference pulse, is changed according to the instantaneous sampled value of the
10
modulating signal. Thus the transmitter has to send synchronizing pulses to keep the transmitter and receiver in
synchronism. As the amplitude and width of the pulses are constant, the transmitter handles constant power
output, a definite advantage over the PWM. But the disadvantage of the PPM system is the need for transmitter-
receiver synchronization. PPM is obtained from PWM. As shown in Fig. 4.11, each trailing edge of the PWM
pulse is a starting point of the pulse in the PPM.
Generation of PPM Signal
Fig. 4.15 shows the block diagram of PPM generator. It consists of PWM generator followed by the monostable
multivibrator. Since, in PPM, output remains high for fix duration from the trailing edges of the PWM signal,
the trailing edge of the PWM signal is used as a trigger input for the monostable multivibrator.
Fig. 4.15: Block diagram of PPM generator.

Practical PPM Generator
Fig. 4.16(a) shows the PPM generator. It consists of differentiator and a monostable multivibrator. The input to
the differentiator is a PWM waveform. The differentiator generates positive and negative spikes corresponding
to leading and trailing edges of the PWM waveform. Diode 𝐷1 is used to bypass the positive spikes. The negative
spikes are used to the trigger monostable multivibrator. The monostable multivibrator then generates the pulses
of same width and amplitude with reference to trigger to give PPM waveform as shown in Fig. 4.16(b).
Fig. 4.16: (a) PPM generator. (b) Waveforms of PPM generator.

Demodulation of PPM signal
In the case of PPM, it is customary to convert the received pulses that vary in position to pulses that vary in
length. One way to achieve this is illustrated in Fig. 4.17.
11
Fig. 4.17: PPM demodulator

As shown in Fig. 4.14, flip-flop circuit is set or turned ON when the reference pulse arrives. This reference
pulse is generated by reference pulse generator of the receiver with the synchronization signal from the
transmitter. The flip-flop circuit is reset or turned OFF at the leading edge of the position modulated pulse. This
repeats and we get PWM pulses at the output of the flip-flop. The PWM pulses are then demodulated by PWM
demodulator to get original modulating signal.
Fig. 4.18: Demodulation waveforms of PPM

Advantages of PPM
1. Like PWM, in PPM amplitude is held constant thus less noise interference
2. Like, PWM, signal and noise separation is very easy.
3. Due to constant pulse width and amplitude, transmission power for each pulse is the same.
Disadvantages of PPM
1. Synchronization between transmitter and receiver is required
2. Large bandwidth is required compared to PAM.
Quantization Process
Amplitude quantization is defined as the process of transforming the sample amplitude 𝑚(𝑛𝑇𝑠 ) of a baseband signal 𝑚(𝑡)
at time 𝑡 = 𝑛𝑇𝑠 into a discrete amplitude 𝑣(𝑛𝑇𝑠 ) taken from a finite set of possible levels. We confine attention to a
quantization process that is memoryless and instantaneous, which means that the transformation at time 𝑡 = 𝑛𝑇𝑠 is not
affected by earlier or later samples of the message signal. This form of quantization, though not optimal, is commonly
used in practice because of its simplicity. When dealing with a memoryless quantizer, we may simplify the notation by
12
dropping the time index. That is, we use the symbol 𝑚 in place of the sample 𝑚(𝑛𝑇𝑠 ) as indicated in Fig. 4.19(a). Then,
as shown in Fig. 4.19(b), the signal amplitude 𝑚 is specified by the index 𝑘 if it lies inside the interval
𝐼𝑘 : {𝑚𝑘 < 𝑚 ≤ 𝑚𝑘+1 }, 𝑘 = 1, 2, … , 𝐿
Where L is the total number of amplitude levels used in the quantizer, which refers to the subsystem that performs the
quantization process. The amplitudes, 𝑚𝑘 , 𝑘 = 1, 2, … , 𝐿, are called decision levels or decision thresholds. At the quantizer
output, the index 𝑘 is transformed into an amplitude that represents all amplitudes that lie inside the interval 𝐼𝑘 . The
amplitudes 𝑣𝑘 , 𝑘 = 1, 2, … , 𝐿, are called representation levels or reconstruction levels, and the spacing between two
adjacent representation levels is called a quantum or step-size. Thus, the quantizer output equals if the input signal sample
m belongs to the interval 𝐼𝑘 . The mapping
𝑣 = 𝑔(𝑚)
is the quantizer characteristic. This characteristic is described by a staircase function. Quantizers can be of a uniform or
non-uniform type. In a uniform quantizer, the representation levels are uniformly spaced; otherwise, the quantizer is non-
uniform. The quantizers considered in this section are of the uniform variety.
Fig. 4.19: Description of a memoryless quantizer.

The quantizer characteristic can also be of a midtread or midrise type. Figure 4.20(a) shows the input–output
characteristic of a uniform quantizer of the mid-tread type, which is so called because the origin lies in the
middle of a tread of the staircase-like graph. Figure 4.20(b) shows the corresponding input–output characteristic
of a uniform quantizer of the midrise type, in which the origin lies in the middle of a rising part of the staircase-
like graph. Note that both the mid-tread and midrise types of uniform quantizers, illustrated in Fig. 4.20, are
symmetric about the origin.
13
Fig. 4.20: Two types of quantization: (a) mid-tread and (b) mid-rise.
Pulse-Code Modulation
In pulse code modulation (PCM), a message signal is represented by a sequence of coded pulses, which is
accomplished by representing the signal in discrete form in both time and amplitude. The basic operations
performed in the transmitter of a PCM system are sampling, quantization, and encoding, as shown in Fig.
4.21(a); the low-pass filter prior to sampling is included merely to prevent aliasing of the message signal. The
quantizing and encoding operations are usually performed in the same circuit, which is called an analog-to-
digital converter.
The basic operations in the receiver are regeneration of impaired signals, decoding, and reconstruction of the
train of quantized samples, as shown in Fig. 4.21(c). Regeneration also occurs at intermediate points along the
transmission path as necessary, as indicated in Fig. 4.21(b).
Fig. 4.21: The basic elements of a PCM system: (a) Transmitter, (b) transmission path, connecting the
transmitter to the receiver, and (c) receiver.
Encoding
In combining the processes of sampling and quantization, the specification of a continuous message (baseband)
signal becomes limited to a discrete set of values, but not in the form best suited to transmission over a wire line
or radio path. To exploit the advantages of sampling and quantization for the purpose of making the transmitted
signal more robust to noise, interference and other channel degradations, we require the use of an encoding
process to translate the discrete set of sample values to a more appropriate form of signal. Any plan for
representing this discrete set of values as a particular arrangement of discrete events is called a code. One of the
discrete events in a code is called a code element or symbol. For example, the presence or absence of a pulse is
14
a symbol. A particular arrangement of symbols used in a code to represent a single value of the discrete set is
called a code word or character.
In a binary code, each symbol may be either of two distinct values, such as a negative pulse or positive pulse.
The two symbols of the binary code are customarily denoted as 0 and 1. In practice, a binary code is preferred
over other codes (e.g., ternary code) for two reasons:
1. The maximum advantage over the effects of noise in a transmission medium is obtained by using a
binary code, because a binary symbol withstands a relatively high level of noise.
2. The binary code is easy to generate and regenerate.
Regeneration along the transmission path
The most important feature of a PCM system lies in the ability to control the effects of distortion and noise
produced by transmitting a PCM signal over a channel. This capability is accomplished by reconstructing the
PCM signal by means of a chain of regenerative repeaters located at sufficiently close spacing along the
transmission route. As illustrated in Fig. 4.22, three basic functions are performed by a regenerative repeater:
equalization, timing, and decision making.
Fig. 4.22: Block diagram of a regenerative repeater.
Operations in the receiver

1. Decoding and Expanding
The first operation in the receiver is to regenerate (i.e., reshape and clean up) the received pulses one last time.
These clean pulses are then regrouped into code words and decoded (i.e., mapped back) into a quantized PAM
signal. The decoding process involves generating a pulse whose amplitude is the linear sum of all the pulses in
the code word; each pulse is weighted by its place value 20 , 21 , 22 , . . . ., 2𝑅−1 in the code, where 𝑅 is the number
of bits per sample.
The sequence of decoded samples represents an estimate of the sequence of compressed samples produced by
the quantizer in the transmitter. We use the term “estimate” here to emphasize the fact that there is no way for
15
the receiver to compensate for the approximation introduced into the transmitted signal by the quantizer.
Moreover, other sources of noise include bit errors and jitter produced along the transmission path. In order to
restore the sequence of decoded samples to their correct relative level, we must, of course, use a subsystem in
the receiver with a characteristic complementary to the compressor, used in the transmitter. Such a subsystem
is called an expander. Ideally, the compression and expansion laws are exactly inverse so that, except for the
effect of quantization, the expander output is equal to the compressor input if these two devices were connected
directly. The combination of a compressor and an expander is referred to as a compander.
2. Reconstruction
The final operation in the receiver is to recover the message signal. This operation is achieved by passing the
expander output through a low-pass reconstruction filter whose cutoff frequency is equal to the message
bandwidth. Recovery of the message signal is intended to signify estimation rather than exact reconstruction.
Delta Modulation
From the discussion presented in PCM, it is apparent that the design of a PCM system involves many operations,
which tend to make its practical implementation rather costly. To simplify the system design, we may use
another digital pulse modulation technique known as delta modulation.
Basic considerations
In delta modulation (DM), an incoming message signal is oversampled (i.e., at a rate much higher than the
Nyquist rate) to purposely increase the correlation between adjacent samples of the signal. The increased
correlation is done so as to permit the use of a simple quantizing strategy for constructing the encoded signal.
In its basic form, DM provides a staircase approximation to the oversampled version of the message signal.
Unlike PCM, the difference between the input signal and its approximation is quantized into only two levels—
namely, ±∆ corresponding to positive and negative differences. Thus, if the approximation falls below the input
signal at any sampling period, it is increased by ∆. If, on the other hand, the approximation lies above the signal,
it is diminished by ∆. Provided the input signal does not change too rapidly from sample to sample, we find that
the staircase approximation remains within ±∆ of the input signal.
We denote the input signal by 𝑚(𝑡) and its staircase approximation by 𝑚𝑞 (𝑡). The basic principle of delta
modulation may then be formalized in the following set of three discrete-time relations:
𝑒(𝑛𝑇𝑠 ) = 𝑚(𝑛𝑇𝑠 ) − 𝑚𝑞 (𝑛𝑇𝑠 − 𝑇𝑠 )
𝑒𝑞 (𝑛𝑇𝑠 ) = ∆𝑠𝑔𝑛[𝑒(𝑛𝑇𝑠 )]
𝑚𝑞 (𝑛𝑇𝑠 ) = 𝑚𝑞 (𝑛𝑇𝑠 − 𝑇𝑠 ) + 𝑒𝑞 (𝑛𝑇𝑠 ) … … … . .∗∗
16
where 𝑇𝑠 is the sampling period; 𝑒(𝑛𝑇𝑠 ) is an error signal representing the difference between the present sample
value 𝑚(𝑛𝑇𝑠 ) of the input signal and the latest approximation to it that is, 𝑚(𝑛𝑇𝑠 ) − 𝑚𝑞 (𝑛𝑇𝑠 − 𝑇𝑠 ) and 𝑒𝑞 (𝑛𝑇𝑠 )
is the quantized version of 𝑒(𝑛𝑇𝑠 ) and sgn[.] is the signum function, assuming the value +1 or −1. The quantizer
output 𝑒𝑞 (𝑛𝑇𝑠 ) is finally encoded to produce the desired DM data.
Fig. 4.23(a) illustrates the way in which the staircase approximately follows variations in the input signal in
accordance with the above equations and Fig. 4.23(b) displays the corresponding binary sequence at the delta
modulator output.
Fig. 4.23: Illustration of delta modulation. (a) Analog waveform 𝑚(𝑡) and its staircase approximation 𝑚𝑞 (𝑡). b) Binary
sequence at the modulator output.
System Details
The principal virtue of delta modulation is its simplicity. It may be implemented by applying a sampled
version of the incoming message signal to a transmitter that consists of a comparator, quantizer, and
accumulator connected together as shown in Fig. 4.24(a).
Fig. 4.24: DM system: (a) Transmitter and (b) receiver.
17
Eq. (**) is a difference equation of order one; the order refers to the fact the present sample 𝑚𝑞 (𝑛𝑇𝑠 ) differs
from the past sample 𝑚𝑞 (𝑛𝑇𝑠 − 𝑇𝑠 )by an amount equal to the quantization error 𝑒𝑞 (𝑛𝑇𝑠 ). Assuming that the
accumulation process starts at zero time, the solution to this equation yields the approximate result
Thus, at the sampling instant 𝑛𝑇𝑠 , the accumulator increments the approximation by the increment ∆ in a positive
or negative direction, depending on the algebraic sign of the error signal 𝑒(𝑛𝑇𝑠 ). If the input signal 𝑚(𝑛𝑇𝑠 ) is
greater than the most recent approximation 𝑚𝑞 (𝑛𝑇𝑠 ), a positive increment +∆ is applied to the approximation.
If, on the other hand, the input signal is smaller, a negative increment −∆ is applied to the approximation. In
this way, the accumulator does the best it can to track the input samples one step (of amplitude +∆ or −∆) at a
time.
In the receiver shown in Fig. 4.24(b), the staircase approximation 𝑚𝑞 (𝑡) is reconstructed by passing the
sequence of positive and negative pulses, produced at the decoder output, through an accumulator in a manner
similar to that used in the transmitter. The out-of-band quantization noise present in the high-frequency staircase
waveform 𝑚𝑞 (𝑡) is rejected by passing 𝑚𝑞 (𝑡) through a filter, as in Fig. 4.24(b). The filter is of a low-pass
kind, with a bandwidth equal to the original message bandwidth.
Quantization Errors
Delta modulation is subject to two types of quantization error: (1) slope overload distortion and (2) granular
noise. We first discuss the cause of slope overload distortion and then granular noise.
Fig. 4.25: Illustration of quantization errors, slope-overload distortion and granular noise, in delta modulation.
From this discussion we see that there is a need to have a large step size to accommodate a wide dynamic range,
whereas a small step size is required for the accurate representation of relatively low-level signals. It is therefore
clear that if we are to choose an optimum step size that minimizes the average power4 of the quantization error
18
in a delta modulator, we need to make the DM system adaptive. This requirement, in turn, means that the step
size has to vary in accordance with the incoming message signal.
Differential Pulse Code Modulation
Suppose a message signal is sampled at the rate 𝑓𝑠 = 1⁄𝑇𝑠 to produce a sequence of correlated samples 𝑇𝑠
seconds apart; this sequence is denoted by 𝑚(𝑛𝑇𝑠 ). The fact that it is possible to predict future values of the
signal 𝑚(𝑡) provides motivation for the differential quantization scheme shown in Fig. 4.26(a). In this scheme,
the input signal to the quantizer is defined by
𝑒(𝑛𝑇𝑠 ) = 𝑚(𝑛𝑇𝑠 ) − 𝑚
̂ (𝑛𝑇𝑠 )
which is the difference between the input sample 𝑚(𝑛𝑇𝑠 ) and a prediction of it, denoted by 𝑚
̂ (𝑛𝑇𝑠 ). This
predicted value is produced by using a prediction filter whose input consists of a quantized version of 𝑚(𝑛𝑇𝑠 ).
The difference signal 𝑒(𝑛𝑇𝑠 ) is called the prediction error, since it is the amount by which the prediction filter
fails to predict the incoming message signal exactly. A simple and yet effective approach to implement the
prediction filter is to use a tapped-delay-line filter or discrete-time filter, with the basic delay set equal to the
sampling period. The block diagram of this filter is shown in Fig. 4.27, according to which the
prediction 𝑚
̂ (𝑛𝑇𝑠 ) is modeled as a linear combination of 𝑝 past sample values of the quantized version
of 𝑚(𝑛𝑇𝑠 ), where 𝑝 is the prediction order.
By encoding the quantizer output in Fig. 4.26(a), we obtain a variation of PCM, which is known as differential
pulse-code modulation (DPCM). It is this encoded signal that is used for transmission.
Fig. 4.26: DPCM system: (a) Transmitter and (b) receiver.
19
Fig: 4.27: Tapped-delay line filter used as prediction filter.

The quantizer output may be expressed as
𝑒𝑞 (𝑛𝑇𝑠 ) = 𝑒(𝑛𝑇𝑠 ) + 𝑞(𝑛𝑇𝑠 )
where 𝑞(𝑛𝑇𝑠 ) is the quantization error. According to Fig. 4.26(a), the quantizer output 𝑒𝑞 (𝑛𝑇𝑠 ) is added to the
predicted value 𝑚
̂ (𝑛𝑇𝑠 ) to produce the prediction-filter input
𝑚𝑞 (𝑛𝑇𝑠 ) = 𝑚
̂ (𝑛𝑇𝑠 ) + 𝑒𝑞 (𝑛𝑇𝑠 )
Substituting the expression for 𝑒𝑞 (𝑛𝑇𝑠 ), we get
𝑚𝑞 (𝑛𝑇𝑠 ) = 𝑚
̂ (𝑛𝑇𝑠 ) + 𝑒(𝑛𝑇𝑠 ) + 𝑞(𝑛𝑇𝑠 )
̂ (𝑛𝑇𝑠 ) + 𝑒(𝑛𝑇𝑠 ) is equal to the sampled message signal 𝑚(𝑛𝑇𝑠 ). Therefore, we may
However, the sum term 𝑚
rewrite 𝑚𝑞 (𝑛𝑇𝑠 ) as
𝑚𝑞 (𝑛𝑇𝑠 ) = 𝑚(𝑛𝑇𝑠 ) + 𝑞(𝑛𝑇𝑠 )
which represents a quantized version of the message sample That is, irrespective of the properties of the
prediction filter, the quantized signal 𝑚𝑞 (𝑛𝑇𝑠 ) at the prediction filter input differs from the sampled message
signal 𝑚(𝑛𝑇𝑠 ) by the quantization error 𝑞(𝑛𝑇𝑠 ). Accordingly, if the prediction is good, the average power of
the prediction error 𝑒(𝑛𝑇𝑠 ) will be smaller than the average power of 𝑚(𝑛𝑇𝑠 ) so that a quantizer with a given
number of levels can be adjusted to produce a quantization error with a smaller average power than would be
possible if 𝑚(𝑛𝑇𝑠 )were quantized directly using PCM.
The receiver for reconstructing the quantized version of the message signal is shown in Fig. 4.26(b). It consists
of a decoder to reconstruct the quantized error signal. The quantized version of the original input is reconstructed
from the decoder output using the same prediction filter in the transmitter of Fig. 4.26(a). In the absence of
channel noise, we find that the encoded signal at the receiver input is identical to the encoded signal at the
transmitter output. Accordingly, the corresponding receiver output is equal to 𝑚𝑞 (𝑛𝑇𝑠 ), which differs from the
original input 𝑚(𝑛𝑇𝑠 ) only by the quantization error 𝑞(𝑛𝑇𝑠 ) incurred as a result of quantizing the prediction
error 𝑒(𝑛𝑇𝑠 ). Finally, an estimate of the original message signal 𝑚(𝑡) is obtained by passing the sequence
through a low-pass reconstruction filter.
20
DM is the 1-bit version of DPCM. Insofar as noise is concerned, we may finally make the following two
statements
1. DPCM, like DM, is subject to slope-overload distortion whenever the input signal changes too rapidly
for the prediction filter to track it.
2. Like PCM, DPCM suffers from quantization noise.
Line Codes
In reality, PCM, DM, and DPCM represent different strategies for source encoding, whereby an analog signal
is converted into digital form. However, all three of them share a common feature: once a binary sequence of
1’s and 0’s is produced, a line code is needed for electrical representation of that binary sequence. There are
several line codes that can be used for this representation, as summarized here:
1. On–off signaling, in which symbol 1 is represented by transmitting a pulse of constant amplitude for the
duration of the symbol, and symbol 0 is represented by switching off the pulse, as in Fig. 4.28(a).
2. Non-return-to-zero (NRZ) signaling, in which symbols 1 and 0 are represented by pulses of equal
positive and negative amplitudes, as illustrated in Fig. 4.28(b).
3. Return-to-zero (RZ) signaling, in which symbol 1 is represented by a positive rectangular pulse of half-
symbol width, and symbol 0 is represented by transmitting no pulse, as illustrated in Fig. 4.28(c).
4. Bipolar return-to-zero (BRZ) signaling, which uses three amplitude levels as indicated in Fig. 4.30(d).
Specifically, positive and negative pulses of equal amplitude are used alternately for symbol 1, and no
pulse is always used for symbol 0. A useful property of BRZ signaling is that the power spectrum of the
transmitted signal has no dc component and relatively insignificant low-frequency components for the
case when symbols 1 and 0 occur with equal probability.
5. Split-phase (Manchester code), which is illustrated in Fig. 4.28(e). In this method of signaling, symbol
1 is represented by a positive pulse followed by a negative pulse, with both pulses being of equal
amplitude and half-symbol width. For symbol 0, the polarities of these two pulses are reversed. The
Manchester code suppresses the dc component and has relatively insignificant low-frequency
components, regardless of the signal statistics.
6. Differential encoding, in which the information is encoded in terms of signal transitions, as illustrated
in Fig. 4.28(f). In the example of the binary PCM signal shown in the figure, a transition is used to
designate symbol 0, whereas no transition is used to designate symbol 1. It is apparent that a
differentially encoded signal may be inverted without affecting its interpretation. The original binary
information is recovered by comparing the polarity of adjacent symbols to establish whether or not a
21
transition has occurred. Note that differential encoding requires the use of a reference bit, as indicated
in Fig. 4.28 (f).
The waveforms shown in parts (a) to (f) of Fig. 4.28 are drawn for the binary data stream 01101001. It is
important, to note that rectangular pulse-shaping is used to draw these waveforms, largely to simplify the
electrical representation.
Fig. 4.28: Line codes. (a) On–off signaling. (b) Nonreturn-to-zero signaling. (c) Return-tozero signaling. (d)
Bipolar return-to-zero signaling. (e) Split-phase or Manchester encoding. (f) Differential encoding.
22
Digital Modulation Techniques

Digital Modulation uses sinusoidal carrier signal like analog modulations but the modulating signals are digital
signals and thus the modulation type is different. The three predominant methods in digital modulation are
Amplitude Shift Keying (ASK), Frequency Shift Keying (FSK), and Phase Shift Keying (PSK). A combination
of these techniques can also be used in practical application. For example the combination of ASK and PSK
gives a rise to new modulation technique, Quadrature Amplitude Modulation (QAM). As compared to analog
modulation, digital modulation offers many advantages:
 Geater noise immunity and robustness to channel impairments
 Easier multiplexing
 Greater security (encryption)
 More flexibility
Bandpass Digital Transmission
The source information usually needs to be mapped to a high frequency bandpass signal in order to communicate
over a channel. Therefore a bandpass communication system involves modulation of a baseband on to a carrier
using some type of modulation. Bandpass digital communication system uses the different types of digital
modulation types i.e. ASK, FSK, PSK and QAM. The modulating signals in digital modulation are classified
as binary and multilevel signals.
Binary signalling: - It has two possible levels i.e. 1 and 0. Its respective digital modulations are BASK, BFSK
and BPSK.
Multilevel Signalling: - It is a type of signalling with more than two levels. Examples of multilevel modulation
include: QAM, MASK, MPSK and MFSK.
Amplitude Shift Keying (ASK)
Amplitude shift keying (ASK) is keying or shifting the magnitude of a sinusoid carrier with respect to the digital
message signal. If the message is a unipolar binary signal, the carrier will be on and off with the message as
shown in Fig. 4.29. The most basic (binary) form of ASK involves the process of switching the carrier either on
or off, in correspondence to a sequence of digital pulses that constitute the information signal. One binary digit
is represented by the presence of a carrier; the other binary digit is represented by the absence of a carrier.
Frequency remains fixed. In this particular case the ASK is known as On-off keying (OOK). The ASK signal is
represented by 𝑠(𝑡) = 𝑚(𝑡)𝑉𝑐 cos 2𝜋𝑓𝑐 (𝑡) where 𝑚(𝑡) is the digital message signal 1 or 0 in binary signaling.
23
Fig. 4.29: ASK modulation
Multilevel message signal will produce a multilevel ASK signal. As it can be observed from its expression,
ASK is identical to DSB-SC signal and therefore, the transmission bandwidth of the ASK signal is twice that
of the baseband digital signal i.e. 𝐵𝑊 = 2𝐵 where 𝐵 is baseband bandwidth. ASK (OOK) signal can be
demodulated either coherently or non-coherently with little difference in performance. OOK can be detected
using either an envelope detector or a product detector. Since it is a form of AM signaling, ASK (OOK) is
relatively low quality, low cost method of digital radio and is seldom used in high capacity/ high traffic systems.
Frequency Shift Keying (FSK)
FSK consists of shifting the frequency of a sinusoidal carrier from a mark frequency (corresponding to sending
a binary 1) to space frequency (corresponding to sending a binary 0) according to the base band digital signal.
It is identical to modulating an FM carrier with a binary digital signal. The most basic (binary) form of FSK
involves the process of varying the frequency of a carrier wave by choosing one of two frequencies (binary
FSK) in correspondence to a sequence of digital pulses that constitute the information signal. Two binary digits
are represented by two frequencies around the carrier frequency as shown in Fig. 4.30. Amplitude remains fixed.
Fig. 4.30: FSK modulation
24
There are two types of FSK signal: the discontinuous phase FSK and the continuous phase FSK. Discontinuous
phase FSK is generated by switching the output line between two different oscillators. It has a discontinuous
phase at the switching time and it is represented by
𝑉𝑐 cos(2𝜋𝑓1 𝑡 + 𝜃1 ) 𝑓𝑜𝑟 𝑏𝑖𝑛𝑎𝑟𝑦 1
𝑠(𝑡) = {
𝑉𝑐 cos(2𝜋𝑓2 𝑡 + 𝜃2 ) 𝑓𝑜𝑟 𝑏𝑖𝑛𝑎𝑟𝑦 0
Where 𝑓1 is called the mark frequency and 𝑓2 the space frequency. The continuous phase FSK signal is generated
by feeding the digital signal into a frequency modulator. The FSK signal is represented by 𝑠(𝑡) =
𝑡
𝑉𝑐 cos (2𝜋𝑓1 𝑡 + ∆𝑓 ∫−∞ 𝑚(𝜆) 𝑑𝜆) where 𝑚(𝑡) is the baseband digital signal. Although 𝑚(𝑡) is discontinuous
at the switching time, the phase function (𝑡) is continuous since it is proportional to the integral of 𝑚(𝑡).
If the digital signal is binary, the resulting FSK signal is called binary FSK (BFSK). Multilevel signaling will
produce multilevel FSK signal.
Oscillator 1
FSK 𝑚(𝑡) FM mod FSK

Oscillator 2
𝑚(𝑡) Carrier osc
Fig. 4.31: (a) Discontinuous FSK Modulator, (b) Continuous FSK Modulator
FSK demodulation is easier than FM demodulation and it can be detected using the configuration shown in Fig.
4.32.
Fig. 4.32: Non-coherent detection of FSK

Another most common circuit used for demodulating binary FSK signal is the phase-locked loop (PLL), which
is shown below.
25
Phase Comparator LPF & Amp

FSK IN
VCO
Fig. 4.33: PLL FSK demodulator

Phase Shift Keying (PSK)
PSK is similar to conventional phase modulation except that with PSK the input signal is a digital signal and a
limited number of output phases are possible. Based on the input baseband digital signal PSK can be BPSK or
multilevel PSK. BPSK consists of shifting the phase of a sinusoidal carrier by 0o or 180o with respect to a binary
signal as shown in Fig. 4.34.
Fig. 4.34: PSK waveform

The BPSK signal is represented by 𝑠(𝑡) = 𝑉𝑐 cos(2𝜋𝑓1 𝑡 + ∆𝑃𝑚(𝑡)) where 𝑚(𝑡) is a unipolar baseband binary
signal. Simplified block diagram of BPSK is shown below.
Binary Balanced BPSK Balanced
In Modulator BPF IN Modulator LPF
BPSK
Binary
out
Carrier
Carrier Osc recovery
(a) (b)
Fig. 4.35: (a) BPSK Modulator, (b) BPSK Demodulator
Multilevel Signaling
The main problem with binary signaling is that speed of data transmission is limited in a given bandwidth. One
way to increase the binary data with a fixed bandwidth is to encode more than one bit per level. This results in
multilevel signaling. With multilevel signaling digital inputs with more than two modulation levels are allowed
on the transmission input. Each level represents more than one bit. The bit rate of multilevel signaling is higher
than the bit rate of binary signaling.
26
Examples:
If 𝑀 > 2, each waveform carry log 2 (𝑀) bits. This scheme is therefore more bandwidth efficient. It is called
M-ary ASK, M-ary FSK, and M-ary PSK or M-ary QAM
Fig. 4.36: Multilevel signaling: (a) 4-ary ASK (b) 4-ary FSK (c) 8-ary PSK (d) 8-ary QAM
27
INTRODUCTION TO COMMUNICATION SYSTEMS (ECEG-3152)
28 | P a g e
Chapter 5
Multiplexing and Demultiplexing Techniques
Multiplexing is the process of simultaneously transmitting two or more individual
signals over a single communication channel. Multiplexing has the effect of increasing
the number of communications channels in which more information can be transmitted.
There are many instances in communications where it is necessary or desirable to
transmit more than one voice or data signal. The application itself may require multiple
signals, and money can be saved by using a single communication channel to send
multiple information signals. Telephone and satellite systems use multiplexing to make
the system practical and less expensive.
There are two basic types of multiplexing, namely; frequency division multiplexing
(FDM) and Time division Multiplexing (TDM). Generally speaking, FDM systems are
used to deal with analog systems, while TDM systems are used for digital systems. Of
course, TDM are also found in many analog systems. The primary difference between
these techniques is that in FDM individual signals to be transmitted are assigned a
different frequency with in a common bandwidth. In TDM, the multiple signals are
transmitted in different time slots.
Fig. 5.1: Concept of multiplexing
1|Page
Frequency Division Multiplexing (FDM)
Frequency Division Multiplexing (FDM) is a technique for transmitting multiple
messages simultaneously over a wideband channel by first modulating the message
signals onto several sub-carriers and forming a composite baseband signal that consists
of the sum of these modulated sub-carriers. This composite signal may then be
modulated on to the main carrier. Any type of modulation, such as AM, PM, FM can be
used. The composite baseband signal then modulates a main transmitter to produce the
FDM signal that is transmitted over the wide band channel.
Fig: 5.2: The transmitting end of an FDM system
The receiver portion of the system is shown below. It picks up the signal and demodulates
it into the FDM signal. This is sent to a group of bandpass filters (BPF), each centered on
one of the carrier frequencies. Each filter passes only its channel and rejects all others. A
channel demodulator then recovers each original input signal.
2|Page
Fig. 5.3: The receiving end of an FDM system
An example of a commonly used FDM system is the telephone system. Telephone
companies have been using FDM to send multiple telephone conversations over a
minimum number of cables. The original signal is voice in the 300-3000Hz range. The
voice is used to modulate a sub-carrier. Each sub-carrier is on a different frequency. These
sub-carrier are then added together to form a single group. This multiplexing process is
repeated at several levels so that very large number of users can communicate over a
single communication channel.
Example: A cable TV service uses a single coaxial cable with a bandwidth of 860 MHz to
transmit multiple TV signals to subscribers. Each TV signal is 6 MHz wide. How many
channels can be carried?
860
𝑇𝑜𝑡𝑎𝑙 𝑐ℎ𝑎𝑛𝑛𝑒𝑙𝑠 = = 143.33 𝑜𝑟 143
6
3|Page
Time division Multiplexing (TDM)
Time Division Multiplexing (TDM) is the time interleaving of samples from several
sources so that the information from these sources can be transmitted serially over a
single communication channel.
Unlike FDM, in TDM each signal can occupy the entire bandwidth of the channel.
However, each signal is transmitted for only a brief period of time. Each signal is allowed
to use the channel for fixed period of time, one after another. Once all the signals have
been transmitted, the cycle repeats again and again. One transmission of each channel
completes one cycle of operation called a frame. The cycle repeats itself at a high rate of
speed. In this way, the data bytes of the individual channels are simply interleaved.
In other words, multiple signals take turns transmitting over the single channel, as
diagrammed in Fig. 5.4. Here, each of the four signals being transmitted over a single
channel is allowed to use the channel for a fixed time, one after another. Once all four
have been transmitted, the cycle repeats. One binary word from each source creates a
frame. The frames are then repeated over and over again.
Fig. 5.4: The basic TDM concept.
Frame synchronization is needed at the TDM receiver so that the received multiplexed
data can be sorted and directed to the appropriate output channel. The frame sync can be
provided to the receiver demultiplexer circuit either by sending a frame sync signal from
4|Page
the transmitter over a separate channel or by deriving the frame sync from the TDM
signals itself. Any analog signal, be it voice or video can readily be transmitted by TDM
techniques. This is accomplished by sampling the analog signal repeatedly at a high rate.
By combining the concepts of TDM and PAM, you can see how multiple analog signals
can be transmitted over a single channel. This type multiplexer is known as PAM-TDM
system and it is shown below.
Fig. 5.5: Rotary switch and 4-channel multiplexer
Frequency Spectrum Allocation for Various Communication Technologies
Each Total Number /

Channel
Technology Frequency Band Channel Possible
Numbers
Bandwidth Channels
AM (Amplitude
540 – 1600 KHz 10 KHz 106 Channels -
Modulation) *
FM (Frequency
88 – 108 MHz 200 KHz 100 Channels 201 - 300
Modulation) *
VHFLo : 54 – 88 MHz 5 Channels 2-6

TV (Television)
VHFHi : 174 – 216 MHz 6 MHz 7 Channels 7 - 13
*
UHF : 470 – 806 MHz 56 Channels 14 - 69
5|Page
GSM -900
890 – 915 MHz (Uplink) 200 KHz

125 Channels 1 - 124
935 – 960 MHz (Downlink)
GSM (Global
Duplex spacing of 45 MHz
System for
GSM-1800
Mobile)
1710 - 1785 MHz (Uplink)
200 KHz 375 channels 512 - 885
1805 - 1880 MHz (Downlink)
Duplex spacing is 95 MHz
802.11 a – 5 GHz
5.250–5.350 GHz 20/40 MHz - -
5.470–5.725 GHz
14 Channels
802.11 b – 2.4 GHz (DSSS) (4 Non- Channels

22 MHz
2.412 – 2.472 GHz overlapping 1, 6, 11, 14
Channels)
WiFi 14 Channels
802.11 g/n – 2.4 GHz
(4 Non- Channels
(OFDM) 20 MHz
overlapping 1, 5, 9, 13
2.412 – 2.472 GHz
Channels)
14 Channels
802.11 n – 2.4 GHz (OFDM) (2 Non- Channels

40 MHz
2.412 – 2.472 GHz overlapping 3, 11
Channels)
*
OTA DTV (Over-the-Air Digital Television Broadcast Frequencies)
6|Page
Fig. 5.6: GSM Frame Format [Channel and Time Slot Assignment]
Fig. 5.7: 2.4 GHz Spectrum Channels
7|Page
Fig. 5.8: 802.11(b, g/n, n) Frequency Allocation
Multiple Access Techniques for Wireless Communication
Multiple access techniques are used to allow a large number of users to share the allocated
spectrum in the most efficient manner. Due to the spectrum limitation, the spectrum
sharing is required to increase the capacity of cell or over a geographical area by allowing
the available bandwidth to be used at the same time by different users. It must be done
in a way such that the quality of service doesn’t degrade within the existing users. There
are several different ways to allow access to the channel. These includes mainly the
following:
 Frequency division multiple-access (FDMA)
 Time division multiple-access (TDMA)
 Spread spectrum multiple-access (SSMA)
 Space division multiple-access (SDMA)
8|Page
1. Frequency Division Multiple Access (FDMA)
FDMA is the initial multiple-access technique for cellular systems in which each
individual user is assigned a pair of frequencies while making or receiving a call as shown
in Fig.1. One frequency is used for downlink and one pair for uplink. This is called
frequency division duplexing (FDD). That allocated frequency pair is not used in the
same cell or adjacent cells during the call so as to reduce the co-channel interference. Even
though the user may not be talking, the spectrum cannot be reassigned as long as a call
is in place. Different users can use the same frequency in the same cell except that they
must transmit at different times.
Fig. 5.9: The basic concept of FDMA.
The features of FDMA are as follows:
 The FDMA channel carries only one phone circuit at a time.
 If an FDMA channel is not in use, then it remains idle and it cannot be used by
other users to increase share capacity. After the assignment of the voice channel
the BS and the MS transmit simultaneously and continuously.
 The bandwidths of FDMA systems are generally narrow i.e. FDMA is usually
implemented in a narrow band system.
9|Page
 The complexity of the FDMA mobile systems is lower than that of TDMA mobile
systems. FDMA requires tight filtering to minimize the adjacent channel
interference.
Examples:
 FDMA/FDD in AMPS (Advanced Mobile Phone System).
 FDMA/TDD in CT2 (Cordless Telephone 2).
2. Time Division Multiple Access (TDMA):
In digital systems, continuous transmission is not required because users do not use the
allotted bandwidth all the time. In such cases, TDMA is a complimentary access
technique to FDMA. Global Systems for Mobile communications (GSM) uses the TDMA
technique. In TDMA, the entire bandwidth is available to the user but only for a finite
period of time.
Fig. 5.9: The basic concept of TDMA.
The available bandwidth is divided into fewer channels compared to FDMA and the
users are allotted time slots. TDMA requires careful time synchronization since users
share the bandwidth in the frequency domain. The number of channels are less, inter
channel interference is almost negligible. TDMA uses different time slots for transmission
and reception. This type of duplexing is referred to as Time division duplexing (TDD).
10 | P a g e
The features of TDMA includes the following:
TDMA shares a single carrier frequency with several users where each user makes use of
non-overlapping time slots. The number of time slots per frame depends on several
factors such as modulation technique, available bandwidth etc. Data transmission in
TDMA is not continuous but occurs in bursts. This results in low battery consumption
since the subscriber transmitter can be turned OFF when not in use. Because of a
discontinuous transmission in TDMA the handoff process is much simpler for a
subscriber unit, since it is able to listen to other base stations during idle time slots. TDMA
uses different time slots for transmission and reception thus duplexers are not required.
TDMA has an advantage that is possible to allocate different numbers of time slots per
frame to different users. Thus, bandwidth can be supplied on demand to different users
by concatenating or reassigning time slot based on priority.
Examples:
 TDMA/FDD in GSM (Global Systems for Mobile communications).
 TDMA/TDD in DECT (Digitally Enhanced Cordless Telephony).
3. Spread Spectrum Multiple Access
Spread spectrum multiple access (SSMA) uses signals which have a transmission
bandwidth whose magnitude is greater than the minimum required RF bandwidth. A
pseudo noise (PN) sequence converts a narrowband signal to a wideband noise like signal
before transmission. SSMA is not very bandwidth efficient when used by a single user.
However, since many users can share the same spread spectrum bandwidth without
interfering with one another, spread spectrum systems become bandwidth efficient in a
multiple user environment.
There are two main types of spread spectrum multiple access techniques:
 Direct Sequence Multiple Access (DSMA).
Code Division Multiple Access (CDMA).
11 | P a g e
 Frequency Hopped Multiple Access (FHMA).
Direct Sequence Spread Spectrum (DS-SS)
In DS-SS, the message signal is multiplied by a Pseudo Random Noise Code. Each user
is given his own codeword which is orthogonal to the codes of other users and in order
to detect the user, the receiver must know the codeword used by the transmitter. This is
the most commonly used technology for CDMA.
Frequency Hopped Multiple Access (FHMA)
FHMA is a digital multiple access system in which the carrier frequencies of the
individual users are varied in a pseudo random fashion within a wideband channel. The
digital data is broken into uniform sized bursts which is then transmitted on different
carrier frequencies.
Code Division Multiple Access (CDMA)
In CDMA, the same bandwidth is occupied by all the users, however they are all assigned
separate codes, which differentiates them from each other (shown in Figure 3). CDMA
utilize a spread spectrum technique in which a spreading signal (which is uncorrelated
to the signal and has a large bandwidth) is used to spread the narrow band message
signal.
Fig. 5.10: The basic concept of CDMA.
12 | P a g e
4. Space Division Multiple Access (SDMA)
SDMA utilizes the spatial separation of the users in order to optimize the use of the
frequency spectrum and, it increases the capacity of the network. Due to the RF spectrum
limitation, the same frequency is reused in different cells in a cellular wireless network.
The radiated power of each user is controlled by SDMA. SDMA serves different users by
using spot beam antenna. These areas may be served by the same frequency or different
frequencies. For avoiding the co-channel interference with neighbor cells, a region can be
divided into cells and that uses the set of frequencies. The frequency re-use factor will
define the same set of frequencies will be used in the different cells. So, the cells be
sufficiently separated in certain distance. Also, it enables frequency re-use within the cell.
In a practical cellular environment, it is improbable to have just one transmitter fall within
the receiver beam width. Therefore, it becomes imperative to use other multiple access
techniques in conjunction with SDMA. When different areas are covered by the antenna
beam, frequency can be re-used, in which case TDMA or CDMA is employed, for
different frequencies FDMA can be used.
Fig. 5.11: Space division multiple access
13 | P a g e

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Introduction To Communication System

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Introduction to Communication systems (ECEg-3152)

Fig.1.1: Basic parts of communication system

Analog Modulation Power Amp

Carrier Osc. Channel

Analog Filter Demodulation LNA

Fig. 1.2: Typical Block diagram of Analog communication Systems

Fig. 1.3: Typical Block diagram of digital Communication Systems

This height of antenna is practical and can easily be installed.

Fig. 1.4: Illustration of frequency translation to avoid mixing of sinal

1.7.Fundamental Limitations of Communication Systems

to a square wave sketched in Fig. 1.6. If the low-frequency or high-frequency component is

Fig. 1.8: Test signal with constant phase shift  = −90°.

Fig. 1.9: Transfer characteristic of a nonlinear device

𝑥(𝑡) 𝐻𝐶 (𝑓) 𝐻𝑒𝑞 (𝑓) 𝑦(𝑡)

Fig. 1.10 Channel with equalization for linear distortion

𝑥(𝑡) Compressor Channel Expander 𝑦(𝑡)

Fig.1.11: Companding System

Fig. 1.12 Ground-Wave Propagation (Below 2 MHz)

Fig. 1.13: Sky-Wave Propagation (2 to 30 MHz)

Fig. 1.14: Line-of-Sight (LOS) Propagation (Above 30 MHz)

Fig. 2.1: Double-sideband modulation

Fig 2.2: amplitude modulation

Fig. 2.3: Phase reversal of AM modulated wave

Fig 2.5 (a) Spectrum of message signal (b) Spectrum of AM wave.

Taking the Fourier transform Eq. (2.8) gives;

Fig. 2.6: time-domain and frequency-domain characteristics of AM waves produced by a

For single-tone modulation, the modulation index can be obtained by definition,

𝐴𝑚 cos 2𝜋𝑓𝑚 𝑡 𝐴𝑚 (2.10)

Fig. 2.7: AM wave

Fig. 2.8: Frequency spectrum of single tone AM wave

PUSB  PLSB m2 (2.14)

v) Bandwidth requirement of the AM signal,

Fig. 2.9: Block diagram of a circuit to produce AM.

Fig. 2.10: Square law modulator circuit

Fig. 2.11: Resistive mixing network and resulting waveforms

Fig. 2.13: Class A amplifier as AM

Fig. 2.14: A high-level collector modulator

For proper operation the time constant also matters:

Fig.2.15: Envelope detector (a), and corresponding waveforms (b)

Fig. 2.16: Lattice-type balanced modulator.

Fig. 2.17: Operation of the lattice modulator.

Fig. 2.19: An SSB transmitter using the filter method.

Fig. 2.20: Phase discriminator method of generating SSB signal

Demodulation of DSB-SC and SSB signals (coherent detection)

Fig. 2.21: Demodulation of DSB/SSB signal using product detector

Accordingly, the transmission bandwidth of a VSB modulated signal is defined by

Fig. 2.23: Generation of VSB AM signal

Fig. 2.24: Coherent detection of VSB AM

We may redefine as;

Fig. 2.25: An AM transmitter using high-level collector modulation

Fig. 2.26: Block diagram of a superheterodyne receiver.

Fig. 2.26: receiver model

Fig. 2.27: Power spectral density of band-pass noise.

Fig. 2.28: Reference transmission model for analog communications.

Noise in DSB-SC Receiver

Fig. 2.29: model of DSB-SC receiver using coherent demodulation.

𝑥(𝑡) = 𝑠(𝑡) + 𝑛𝐼 (𝑡) cos 2𝜋𝑓𝑐 𝑡 − 𝑛𝑄 (𝑡) sin 2𝜋𝑓𝑐 𝑡 (2.44)

Combining these observations, we obtain the post-detection SNR of

Fig. 2.30: Model of AM receiver using envelope detection.

Fig. 2.31: Phasor diagram for AM wave plus narrowband noise.

modulation-demodulation scheme becomes;

reference SNR becomes;

𝑚(𝑡) Frequency 𝑆𝐹𝑀 (𝑡)