4/25/24, 3:39 PM Analog Communication - Quick Guide

Analog Communication - Quick Guide

Analog Communication - Introduction

The word communication arises from the Latin word commūnicāre, which means “to
share”. Communication is the basic step for exchange of information.

For example, a baby in a cradle, communicates with a cry when she needs her
mother. A cow moos loudly when it is in danger. A person communicates with the
help of a language. Communication is the bridge to share.

Communication can be defined as the process of exchange of information through

means such as words, actions, signs, etc., between two or more individuals.

Parts of a Communication System

Any system, which provides communication consists of the three important and basic
parts as shown in the following figure.

Sender is the person who sends a message. It could be a transmitting station

from where the signal is transmitted.

Channel is the medium through which the message signals travel to reach
the destination.
Receiver is the person who receives the message. It could be a receiving
station where the transmitted signal is being received.

Types of Signals

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Conveying an information by some means such as gestures, sounds, actions, etc.,

can be termed as signaling. Hence, a signal can be a source of energy which
transmits some information. This signal helps to establish a communication between
the sender and the receiver.

An electrical impulse or an electromagnetic wave which travels a distance to convey

a message, can be termed as a signal in communication systems.

Depending on their characteristics, signals are mainly classified into two types:
Analog and Digital. Analog and Digital signals are further classified, as shown in the
following figure.

Analog Signal
A continuous time varying signal, which represents a time varying quantity can be
termed as an Analog Signal. This signal keeps on varying with respect to time,
according to the instantaneous values of the quantity, which represents it.

Example

Let us consider a tap that fills a tank of 100 liters capacity in an hour (6 AM to 7
AM). The portion of filling the tank is varied by the varying time. Which means, after
15 minutes (6:15 AM) the quarter portion of the tank gets filled, whereas at 6:45
AM, 3/4th of the tank is filled.

If we try to plot the varying portions of water in the tank according to the varying
time, it would look like the following figure.

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As the result shown in this image varies (increases) according to time, this time
varying quantity can be understood as Analog quantity. The signal which
represents this condition with an inclined line in the figure, is an Analog Signal. The
communication based on analog signals and analog values is called as Analog
Communication.

Digital Signal
A signal which is discrete in nature or which is non-continuous in form can be termed
as a Digital signal. This signal has individual values, denoted separately, which are
not based on the previous values, as if they are derived at that particular instant of
time.

Example

Let us consider a classroom having 20 students. If their attendance in a week is

plotted, it would look like the following figure.

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In this figure, the values are stated separately. For instance, the attendance of the
class on Wednesday is 20 whereas on Saturday is 15. These values can be
considered individually and separately or discretely, hence they are called as
discrete values.

The binary digits which has only 1s and 0s are mostly termed as digital values.
Hence, the signals which represent 1s and 0s are also called as digital signals. The
communication based on digital signals and digital values is called as Digital
Communication.

Periodic Signal
Any analog or digital signal, that repeats its pattern over a period of time, is called
as a Periodic Signal. This signal has its pattern continued repeatedly and is easy to
be assumed or to be calculated.

Example

If we consider a machinery in an industry, the process that takes place one after the
other is a continuous procedure. For example, procuring and grading the raw
material, processing the material in batches, packing a load of products one after the
other, etc., follows a certain procedure repeatedly.

Such a process whether considered analog or digital, can be graphically represented

as follows.

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Aperiodic Signal
Any analog or digital signal, that doesn’t repeat its pattern over a period of time is
called as Aperiodic Signal. This signal has its pattern continued but the pattern is
not repeated. It is also not so easy to be assumed or to be calculated.

Example

The daily routine of a person, if considered, consists of various types of work which
take different time intervals for different tasks. The time interval or the work doesn’t
continuously repeat. For example, a person will not continuously brush his teeth
from morning to night, that too with the same time period.

Such a process whether considered analog or digital, can be graphically represented

as follows.

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In general, the signals which are used in communication systems are analog in
nature, which are transmitted in analog or converted to digital and then transmitted,
depending upon the requirement.

Analog Communication - Modulation

For a signal to be transmitted to a distance, without the effect of any external
interferences or noise addition and without getting faded away, it has to undergo a
process called as Modulation. It improves the strength of the signal without
disturbing the parameters of the original signal.

What is Modulation?
A message carrying a signal has to get transmitted over a distance and for it to
establish a reliable communication, it needs to take the help of a high frequency
signal which should not affect the original characteristics of the message signal.

The characteristics of the message signal, if changed, the message contained in it

also alters. Hence, it is a must to take care of the message signal. A high frequency
signal can travel up to a longer distance, without getting affected by external
disturbances. We take the help of such high frequency signal which is called as a
carrier signal to transmit our message signal. Such a process is simply called as
Modulation.

Modulation is the process of changing the parameters of the carrier signal, in

accordance with the instantaneous values of the modulating signal.

Need for Modulation

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Baseband signals are incompatible for direct transmission. For such a signal, to
travel longer distances, its strength has to be increased by modulating with a high
frequency carrier wave, which doesn’t affect the parameters of the modulating
signal.

Advantages of Modulation
The antenna used for transmission, had to be very large, if modulation was not
introduced. The range of communication gets limited as the wave cannot travel a
distance without getting distorted.

Following are some of the advantages for implementing modulation in the

communication systems.

Reduction of antenna size

No signal mixing

Increased communication range

Multiplexing of signals
Possibility of bandwidth adjustments

Improved reception quality

Signals in the Modulation Process

Following are the three types of signals in the modulation process.

Message or Modulating Signal

The signal which contains a message to be transmitted, is called as a message

signal. It is a baseband signal, which has to undergo the process of modulation, to
get transmitted. Hence, it is also called as the modulating signal.

Carrier Signal

The high frequency signal, which has a certain amplitude, frequency and phase but
contains no information is called as a carrier signal. It is an empty signal and is
used to carry the signal to the receiver after modulation.

Modulated Signal

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The resultant signal after the process of modulation is called as a modulated

signal. This signal is a combination of modulating signal and carrier signal.

Types of Modulation
There are many types of modulations. Depending upon the modulation techniques
used, they are classified as shown in the following figure.

The types of modulations are broadly classified into continuous-wave modulation and
pulse modulation.

Continuous-wave Modulation

In continuous-wave modulation, a high frequency sine wave is used as a carrier

wave. This is further divided into amplitude and angle modulation.

If the amplitude of the high frequency carrier wave is varied in accordance

with the instantaneous amplitude of the modulating signal, then such a
technique is called as Amplitude Modulation.

If the angle of the carrier wave is varied, in accordance with the

instantaneous value of the modulating signal, then such a technique is called
as Angle Modulation. Angle modulation is further divided into frequency
modulation and phase modulation.

If the frequency of the carrier wave is varied, in accordance with the

instantaneous value of the modulating signal, then such a technique is
called as Frequency Modulation.

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If the phase of the high frequency carrier wave is varied in accordance

with the instantaneous value of the modulating signal, then such a
technique is called as Phase Modulation.

Pulse Modulation

In Pulse modulation, a periodic sequence of rectangular pulses, is used as a carrier

wave. This is further divided into analog and digital modulation.

In analog modulation technique, if the amplitude or duration or position of a pulse is

varied in accordance with the instantaneous values of the baseband modulating
signal, then such a technique is called as Pulse Amplitude Modulation (PAM) or Pulse
Duration/Width Modulation (PDM/PWM), or Pulse Position Modulation (PPM).

In digital modulation, the modulation technique used is Pulse Code Modulation (PCM)
where the analog signal is converted into digital form of 1s and 0s. As the resultant
is a coded pulse train, this is called as PCM. This is further developed as Delta
Modulation (DM). These digital modulation techniques are discussed in our Digital
Communications tutorial

Amplitude Modulation
A continuous-wave goes on continuously without any intervals and it is the baseband
message signal, which contains the information. This wave has to be modulated.

According to the standard definition, “The amplitude of the carrier signal varies in
accordance with the instantaneous amplitude of the modulating signal.” Which
means, the amplitude of the carrier signal containing no information varies as per
the amplitude of the signal containing information, at each instant. This can be well
explained by the following figures.

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The first figure shows the modulating wave, which is the message signal. The next
one is the carrier wave, which is a high frequency signal and contains no

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information. While, the last one is the resultant modulated wave.

It can be observed that the positive and negative peaks of the carrier wave, are
interconnected with an imaginary line. This line helps recreating the exact shape of
the modulating signal. This imaginary line on the carrier wave is called as Envelope.
It is the same as that of the message signal.

Mathematical Expressions
Following are the mathematical expressions for these waves.

Time-domain Representation of the Waves

Let the modulating signal be,

m (t) = Am cos(2πfm t)

and the carrier signal be,

c (t) = Ac cos(2πfc t)

Where,

Am and Ac are the amplitude of the modulating signal and the carrier signal
respectively.

fm and fc are the frequency of the modulating signal and the carrier signal
respectively.

Then, the equation of Amplitude Modulated wave will be

s(t) = [Ac + Am cos(2πfm t)] cos(2πfc t) (Equation 1)

Modulation Index
A carrier wave, after being modulated, if the modulated level is calculated, then such
an attempt is called as Modulation Index or Modulation Depth. It states the level
of modulation that a carrier wave undergoes.

Rearrange the Equation 1 as below.

Am
s(t) = Ac [1 + ( ) cos(2πfm t)] cos(2πfc t)
Ac

⇒ s (t) = Ac [1 + μ cos(2πfm t)] cos(2πfc t) (Equation 2)

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Where, μ is Modulation index and it is equal to the ratio of Am and Ac .

Mathematically, we can write it as

Am
μ = (Equation 3)
Ac

Hence, we can calculate the value of modulation index by using the above formula,
when the amplitudes of the message and carrier signals are known.

Now, let us derive one more formula for Modulation index by considering Equation 1.
We can use this formula for calculating modulation index value, when the maximum
and minimum amplitudes of the modulated wave are known.

Let Amax and Amin be the maximum and minimum amplitudes of the modulated
wave.

We will get the maximum amplitude of the modulated wave, when cos(2πfm t) is 1.

⇒ Amax = Ac + Am (Equation 4)

We will get the minimum amplitude of the modulated wave, when cos(2πfm t) is -1.

⇒ Amin = Ac − Am (Equation 5)

Add Equation 4 and Equation 5.

Amax + Amin = Ac + Am + Ac − Am = 2Ac

A max +A min
⇒ Ac = (Equation 6)
2

Subtract Equation 5 from Equation 4.

Amax − Amin = Ac + Am − (Ac − Am ) = 2Am

A max −A min
⇒ Am = (Equation 7)
2

The ratio of Equation 7 and Equation 6 will be as follows.

Am (Amax − Amin ) /2
=
Ac (Amax + Amin ) /2

A max −A min
⇒ μ = (Equation 8)
A max +A min

Therefore, Equation 3 and Equation 8 are the two formulas for Modulation index. The
modulation index or modulation depth is often denoted in percentage called as
Percentage of Modulation. We will get the percentage of modulation, just by
multiplying the modulation index value with 100.

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For a perfect modulation, the value of modulation index should be 1, which implies
the percentage of modulation should be 100%.

For instance, if this value is less than 1, i.e., the modulation index is 0.5, then the
modulated output would look like the following figure. It is called as Under-
modulation. Such a wave is called as an under-modulated wave.

If the value of the modulation index is greater than 1, i.e., 1.5 or so, then the wave
will be an over-modulated wave. It would look like the following figure.

As the value of the modulation index increases, the carrier experiences a 180o phase
reversal, which causes additional sidebands and hence, the wave gets distorted.
Such an over-modulated wave causes interference, which cannot be eliminated.

Bandwidth of AM Wave
Bandwidth (BW) is the difference between the highest and lowest frequencies of
the signal. Mathematically, we can write it as

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BW = fmax − fmin

Consider the following equation of amplitude modulated wave.

s (t) = Ac [1 + μ cos(2πfm t)] cos(2πfc t)

⇒ s (t) = Ac cos(2πfc t) + Ac μ cos(2πfc t) cos(2πfm t)

Ac μ Ac μ
⇒ s (t) = Ac cos(2πfc t) + cos[2π (fc + fm ) t] + cos
2 2

[2π (fc − fm ) t]

Hence, the amplitude modulated wave has three frequencies. Those are carrier
frequency fc , upper sideband frequency fc + fm and lower sideband frequency
fc − fm

Here,

fmax = fc + fm and fmin = fc − fm

Substitute, fmax and fmin values in bandwidth formula.

BW = fc + fm − (fc − fm )

⇒ BW = 2fm

Thus, it can be said that the bandwidth required for amplitude modulated wave is
twice the frequency of the modulating signal.

Power Calculations of AM Wave

Consider the following equation of amplitude modulated wave.

Ac μ Ac μ
s (t) = Ac cos(2πfc t) + cos[2π (fc + fm ) t] + cos
2 2

[2π (fc − fm ) t]

Power of AM wave is equal to the sum of powers of carrier, upper sideband, and
lower sideband frequency components.

Pt = Pc + PU SB + PLSB

We know that the standard formula for power of cos signal is

– 2
2
vrms (vm /√2)
P = =
R 2

Where,

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vrms is the rms value of cos signal.

vm is the peak value of cos signal.

First, let us find the powers of the carrier, the upper and lower sideband one by one.

Carrier power

– 2 2
(Ac /√2) Ac
Pc = =
R 2R

Upper sideband power

– 2 2 2
(Ac μ/2√2) Ac μ
PU SB = =
R 8R

Similarly, we will get the lower sideband power same as that of the upper side band
power.

2 2
Ac μ
PLSB =
8R

Now, let us add these three powers in order to get the power of AM wave.

2 2 2 2 2
Ac Ac μ Ac μ
Pt = + +
2R 8R 8R

2 2 2
Ac μ μ
⇒ Pt = ( ) (1 + + )
2R 4 4

2
μ
⇒ Pt = Pc (1 + )
2

We can use the above formula to calculate the power of AM wave, when the carrier
power and the modulation index are known.

If the modulation index μ = 1 then the power of AM wave is equal to 1.5 times the
carrier power. So, the power required for transmitting an AM wave is 1.5 times the
carrier power for a perfect modulation.

Numerical Problems 1
In the previous chapter, we have discussed the parameters used in Amplitude
Modulation. Each parameter has its own formula. By using those formulas, we can

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find the respective parameter values. In this chapter, let us solve a few problems
based on the concept of amplitude modulation.

Problem 1
A modulating signal m (t) = 10 cos(2π × 10 t)
3
is amplitude modulated with a
carrier signal c (t) = 50 cos(2π × 10 t) .
5
Find the modulation index, the carrier
power, and the power required for transmitting AM wave.

Solution

Given, the equation of modulating signal as

3
m (t) = 10 cos(2π × 10 t)

We know the standard equation of modulating signal as

m (t) = Am cos(2πfm t)

By comparing the above two equations, we will get

Amplitude of modulating signal as Am = 10volts

and Frequency of modulating signal as

3
fm = 10 H z = 1K H z

Given, the equation of carrier signal is

5
c (t) = 50 cos(2π × 10 t)

The standard equation of carrier signal is

c (t) = Ac cos(2πfc t)

By comparing these two equations, we will get

Amplitude of carrier signal as Ac = 50volts

and Frequency of carrier signal as

5
fc = 10 H z = 100K H z

We know the formula for modulation index as

Am
μ =
Ac

Substitute, Am and Ac values in the above formula.

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10
μ = = 0.2
50

Therefore, the value of modulation index is 0.2 and percentage of modulation is

20%.

The formula for Carrier power, Pc = is

2
Ac
Pc =
2R

Assume R = 1Ω and substitute Ac value in the above formula.

2
(50)
Pc = = 1250W
2 (1)

Therefore, the Carrier power, Pc is 1250 watts.

We know the formula for power required for transmitting AM wave is

2
μ
⇒ Pt = Pc (1 + )
2

Substitute Pc and μ values in the above formula.

2
(0.2)
Pt = 1250 (1 + ) = 1275W
2

Therefore, the power required for transmitting AM wave is 1275 watts.

Problem 2
The equation of amplitude wave is given by
s (t) = 20 [1 + 0.8 cos(2π × 10 t)] cos(4π × 10 t). Find the carrier power, the total
3 5

sideband power, and the band width of AM wave.

Solution

Given, the equation of Amplitude modulated wave is

3 5
s (t) = 20 [1 + 0.8 cos(2π × 10 t)] cos(4π × 10 t)

Re-write the above equation as

3 5
s (t) = 20 [1 + 0.8 cos(2π × 10 t)] cos(2π × 2 × 10 t)

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We know the equation of Amplitude modulated wave is

s (t) = Ac [1 + μ cos(2πfm t)] cos(2πfc t)

By comparing the above two equations, we will get

Amplitude of carrier signal as Ac = 20volts

Modulation index as μ = 0.8

Frequency of modulating signal as

3
fm = 10 H z = 1K H z

Frequency of carrier signal as

5
fc = 2 × 10 H z = 200K H z

The formula for Carrier power, Pc is

2
Ae
Pc =
2R

Assume R = 1Ω and substitute Ac value in the above formula.

2
(20)
Pc = = 200W
2 (1)

Therefore, the Carrier power, Pc is 200watts.

We know the formula for total side band power is

2
Pc μ
PSB =
2

Substitute Pc and μ values in the above formula.

2
200 × (0.8)
PSB = = 64W
2

Therefore, the total side band power is 64 watts.

We know the formula for bandwidth of AM wave is

BW = 2fm

Substitute fm value in the above formula.

BW = 2 (1K ) = 2K H z

Therefore, the bandwidth of AM wave is 2 KHz.

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Analog Communication - AM Modulators

In this chapter, let us discuss about the modulators, which generate amplitude
modulated wave. The following two modulators generate AM wave.

Square law modulator

Switching modulator

Square Law Modulator

Following is the block diagram of the square law modulator

Let the modulating and carrier signals be denoted as m (t) and A cos(2πfc t)

respectively. These two signals are applied as inputs to the summer (adder) block.
This summer block produces an output, which is the addition of the modulating and
the carrier signal. Mathematically, we can write it as

V1 t = m (t) + Ac cos(2πfc t)

This signal V1 t is applied as an input to a nonlinear device like diode. The

characteristics of the diode are closely related to square law.

V2 t = k 1 V1 (t) + k 2 V
1
2
(t) (Equation 1)

Where, k1 and k2 are constants.

Substitute V1 (t) in Equation 1

2
V2 (t) = k 1 [m (t) + Ac cos(2πfc t)] + k 2 [m (t) + Ac cos(2πfc t)]

2
⇒ V2 (t) = k 1 m (t) + k 1 Ac cos(2πfc t) + k 2 m (t) +

2 2
k 2 Ac cos (2πfc t) + 2k 2 m (t) Ac cos(2πfc t)

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2 2 2
⇒ V2 (t) = k 1 m (t) + k 2 m (t) + k 2 Ac cos (2πfc t)+

2k2
k 1 Ac [1 + ( ) m (t)] cos(2πfc t)
k1

The last term of the above equation represents the desired AM wave and the first
three terms of the above equation are unwanted. So, with the help of band pass
filter, we can pass only AM wave and eliminate the first three terms.

Therefore, the output of square law modulator is

2k 2
s (t) = k 1 Ac [1 + ( ) m (t)] cos(2πfc t)
k1

The standard equation of AM wave is

s (t) = Ac [1 + k a m (t)] cos(2πfc t)

Where, Ka is the amplitude sensitivity

By comparing the output of the square law modulator with the standard equation of
AM wave, we will get the scaling factor as and the amplitude sensitivity as .
2k2
k1 ka
k1

Switching Modulator
Following is the block diagram of switching modulator.

Switching modulator is similar to the square law modulator. The only difference is
that in the square law modulator, the diode is operated in a non-linear mode,
whereas, in the switching modulator, the diode has to operate as an ideal switch.

Let the modulating and carrier signals be denoted as m (t) and

c (t) = Ac cos(2πfc t) respectively. These two signals are applied as inputs to the

summer (adder) block. Summer block produces an output, which is the addition of
modulating and carrier signals. Mathematically, we can write it as

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V1 (t) = m (t) + c (t) = m (t) + Ac cos(2πfc t)

This signal V1 (t) is applied as an input of diode. Assume, the magnitude of the
modulating signal is very small when compared to the amplitude of carrier signal Ac .
So, the diode’s ON and OFF action is controlled by carrier signal c (t). This means,
the diode will be forward biased when c (t) > 0 and it will be reverse biased when
c (t) < 0 .

Therefore, the output of the diode is

V1 (t) if c (t) > 0

V2 (t) = {
0 if c (t) < 0

We can approximate this as

V2 (t) = V1 (t) x (t) (Equation 2)

Where, is a periodic pulse train with time period

1
x (t) T =
fc

The Fourier series representation of this periodic pulse train is

∞ n
1 2 (−1) − 1
x (t) = + ∑ cos(2π (2n − 1) fc t)
2 π 2n − 1
n=1

1 2 2
⇒ x (t) = + cos(2πfc t) − cos(6πfc t)+. . . .
2 π 3π

Substitute, V1 (t) and x (t) values in Equation 2.

V2 (t)

1 2 2
= [m (t) + Ac cos(2πfc t)] [ + cos(2πfc t) − cos(6πfc t)+. . . .
2 π 3π

.]

m(t) Ac 2m(t) 2A c 2
V2 (t) = + cos(2πfc t) + cos(2πfc t) + cos (2πfc t)−
2 2 π π

2m(t) 2A c
cos(6πfc t) − cos(2πfc t) cos(6πfc t)+. .
3π 3π

...

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Ac m(t) 2A c
4 2
V2 (t) = (1 + ( ) m (t)) cos(2πfc t) + + cos (2πfc t)
2 πA c 2 π

2m(t) 2A c
cos(6πfc t) − cos(2πfc t) cos(6πfc t)+. .
3π 3π

...

The 1st term of the above equation represents the desired AM wave and the
remaining terms are unwanted terms. Thus, with the help of band pass filter, we can
pass only AM wave and eliminate the remaining terms.

Therefore, the output of switching modulator is

Ac 4
s (t) = (1 + ( ) m (t)) cos(2πfc t)
2 πAc

We know the standard equation of AM wave is

s (t) = Ac [1 + k a m (t)] cos(2πfc t)

Where, ka is the amplitude sensitivity.

By comparing the output of the switching modulator with the standard equation of
AM wave, we will get the scaling factor as 0.5 and amplitude sensitivity ka as 4
.
πA c

Analog Communication - AM Demodulators

The process of extracting an original message signal from the modulated wave is
known as detection or demodulation. The circuit, which demodulates the
modulated wave is known as the demodulator. The following demodulators
(detectors) are used for demodulating AM wave.

Square Law Demodulator

Envelope Detector

Square Law Demodulator

Square law demodulator is used to demodulate low level AM wave. Following is the
block diagram of thesquare law demodulator.

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This demodulator contains a square law device and low pass filter. The AM wave
V1 (t) is applied as an input to this demodulator.

The standard form of AM wave is

V1 (t) = Ac [1 + k a m (t)] cos(2πfc t)

We know that the mathematical relationship between the input and the output of
square law device is

V2 (t) = k 1 V1 (t) + k 2 V
2
1
(t) (Equation 1)

Where,

V1 (t) is the input of the square law device, which is nothing but the AM wave

V2 (t) is the output of the square law device

k1 and k2 are constants

Substitute V1 (t) in Equation 1

V2 (t) = k 1 (Ac [1 + k a m (t)] cos(2πfc t))

2
+ k 2 (Ac [1 + k a m (t)] cos(2πfc t))

⇒ V2 (t) = k 1 Ac cos(2πfc t) + k 1 Ac k a m (t) cos(2πfc t)+

2 1+cos(4πf t)
2 2 c
k 2 Ac [1 + Ka m (t) + 2k a m (t)] (
2

2
K2 A c
⇒ V2 (t) = k 1 Ac cos(2πfc t) + k 1 Ac k a m (t) cos(2πfc t) + +
2

2 2 2 2 2 2 2
K2 A c k2 A c ka m (t) k2 A c ka m (t)
cos(4πfc t) + + cos
2 2 2

(4πfc t)+

2 2
k 2 Ac k a m (t) + k 2 Ac k a m (t) cos(4πfc t)

In the above equation, the term k 2 Ac k a m (t) is the scaled version of the message
2

signal. It can be extracted by passing the above signal through a low pass filter and

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2
k2 A c
the DC component 2
can be eliminated with the help of a coupling capacitor.

Envelope Detector
Envelope detector is used to detect (demodulate) high level AM wave. Following is
the block diagram of the envelope detector.

This envelope detector consists of a diode and low pass filter. Here, the diode is the
main detecting element. Hence, the envelope detector is also called as the diode
detector. The low pass filter contains a parallel combination of the resistor and the
capacitor.

The AM wave s (t) is applied as an input to this detector.

We know the standard form of AM wave is

s (t) = Ac [1 + k a m (t)] cos(2πfc t)

In the positive half cycle of AM wave, the diode conducts and the capacitor charges
to the peak value of AM wave. When the value of AM wave is less than this value,
the diode will be reverse biased. Thus, the capacitor will discharge through resistor R
till the next positive half cycle of AM wave. When the value of AM wave is greater
than the capacitor voltage, the diode conducts and the process will be repeated.

We should select the component values in such a way that the capacitor charges
very quickly and discharges very slowly. As a result, we will get the capacitor voltage
waveform same as that of the envelope of AM wave, which is almost similar to the
modulating signal.

Analog Communication - DSBSC Modulation

In the process of Amplitude Modulation, the modulated wave consists of the carrier
wave and two sidebands. The modulated wave has the information only in the
sidebands. Sideband is nothing but a band of frequencies, containing power, which
are the lower and higher frequencies of the carrier frequency.

The transmission of a signal, which contains a carrier along with two sidebands can
be termed as Double Sideband Full Carrier system or simply DSBFC. It is plotted

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as shown in the following figure.

However, such a transmission is inefficient. Because, two-thirds of the power is being

wasted in the carrier, which carries no information.

If this carrier is suppressed and the saved power is distributed to the two sidebands,
then such a process is called as Double Sideband Suppressed Carrier system or
simply DSBSC. It is plotted as shown in the following figure.

Mathematical Expressions
Let us consider the same mathematical expressions for modulating and carrier
signals as we have considered in the earlier chapters.

i.e., Modulating signal

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m (t) = Am cos(2πfm t)

Carrier signal

c (t) = Ac cos(2πfc t)

Mathematically, we can represent the equation of DSBSC wave as the product of

modulating and carrier signals.

s (t) = m (t) c (t)

⇒ s (t) = Am Ac cos(2πfm t) cos(2πfc t)

Bandwidth of DSBSC Wave

We know the formula for bandwidth (BW) is

BW = fmax − fmin

Consider the equation of DSBSC modulated wave.

s (t) = Am Ac cos(2πfm t) cos(2πfc t)

Am Ac Am Ac
⇒ s (t) = cos[2π (fc + fm ) t] + cos[2π (fc − fm ) t]
2 2

The DSBSC modulated wave has only two frequencies. So, the maximum and
minimum frequencies are fc + fm and fc − fm respectively.

i.e.,

fmax = fc + fm and fmin = fc − fm

Substitute, fmax and fmin values in the bandwidth formula.

BW = fc + fm − (fc − fm )

⇒ BW = 2fm

Thus, the bandwidth of DSBSC wave is same as that of AM wave and it is equal to
twice the frequency of the modulating signal.

Power Calculations of DSBSC Wave

Consider the following equation of DSBSC modulated wave.

Am Ac Am Ac
s (t) = cos[2π (fc + fm ) t] + cos[2π (fc − fm ) t]
2 2

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Power of DSBSC wave is equal to the sum of powers of upper sideband and lower
sideband frequency components.

Pt = PU SB + PLSB

We know the standard formula for power of cos signal is

– 2
2
vrms (vm √2)
P = =
R R

First, let us find the powers of upper sideband and lower sideband one by one.

Upper sideband power

– 2 2 2
(Am Ac /2√2) Am Ac
PU SB = =
R 8R

Similarly, we will get the lower sideband power same as that of upper sideband
power.
2 2
Am Ac
PU SB =
8R

Now, let us add these two sideband powers in order to get the power of DSBSC
wave.

2 2 2 2
Am Ac Am Ac
Pt = +
8R 8R

2 2
Am Ac
⇒ Pt =
4R

Therefore, the power required for transmitting DSBSC wave is equal to the power of
both the sidebands.

Analog Communication - DSBSC Modulators

In this chapter, let us discuss about the modulators, which generate DSBSC wave.
The following two modulators generate DSBSC wave.

Balanced modulator
Ring modulator

Balanced Modulator
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Following is the block diagram of the Balanced modulator.

Balanced modulator consists of two identical AM modulators. These two

modulators are arranged in a balanced configuration in order to suppress the carrier
signal. Hence, it is called as Balanced modulator.

The same carrier signal c (t) = Ac cos(2πfc t) is applied as one of the inputs to
these two AM modulators. The modulating signal m (t) is applied as another input to
the upper AM modulator. Whereas, the modulating signal m (t) with opposite
polarity, i.e., −m (t) is applied as another input to the lower AM modulator.

Output of the upper AM modulator is

s1 (t) = Ac [1 + k a m (t)] cos(2πfc t)

Output of the lower AM modulator is

s2 (t) = Ac [1 − k a m (t)] cos(2πfc t)

We get the DSBSC wave s (t) by subtracting s2 (t) from s1 (t) . The summer block is
used to perform this operation. s1 (t) with positive sign and s2 (t) with negative sign
are applied as inputs to summer block. Thus, the summer block produces an output
s (t) which is the difference of s1 (t) and s2 (t) .

⇒ s (t) = Ac [1 + k a m (t)] cos(2πfc t) − Ac [1 − k a m (t)] cos(2πfc t)

⇒ s (t) = Ac cos(2πfc t) + Ac k a m (t) cos(2πfc t) − Ac cos(2πfc t)+

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Ac k a m (t) cos(2πfc t)

⇒ s (t) = 2Ac k a m (t) cos(2πfc t)

We know the standard equation of DSBSC wave is

s (t) = Ac m (t) cos(2πfc t)

By comparing the output of summer block with the standard equation of DSBSC
wave, we will get the scaling factor as 2k a

Ring Modulator
Following is the block diagram of the Ring modulator.

In this diagram, the four diodes D1 ,D2 ,D3 and D4 are connected in the ring
structure. Hence, this modulator is called as the ring modulator. Two center tapped
transformers are used in this diagram. The message signal m (t) is applied to the
input transformer. Whereas, the carrier signals c (t) is applied between the two
center tapped transformers.

For positive half cycle of the carrier signal, the diodes D1 and D3 are switched ON
and the other two diodes D2 and D4 are switched OFF. In this case, the message
signal is multiplied by +1.

For negative half cycle of the carrier signal, the diodes D2 and D4 are switched ON
and the other two diodes D1 and D3 are switched OFF. In this case, the message

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signal is multiplied by -1. This results in phase shift in the resulting DSBSC
0
180

wave.

From the above analysis, we can say that the four diodes D1 , D2 , D3 and D4 are
controlled by the carrier signal. If the carrier is a square wave, then the Fourier
series representation of c (t) is represented as

∞ n−1
4 (−1)
c (t) = ∑ cos[2πfc t (2n − 1)]
π 2n − 1
n=1

We will get DSBSC wave s (t), which is just the product of the carrier signal c (t) and
the message signal m (t) i.e.,

∞ n−1
4 (−1)
s (t) = ∑ cos[2πfc t (2n − 1)]m (t)
π 2n − 1
n=1

The above equation represents DSBSC wave, which is obtained at the output
transformer of the ring modulator.

DSBSC modulators are also called as product modulators as they produce the
output, which is the product of two input signals.

DSBSC Demodulators
The process of extracting an original message signal from DSBSC wave is known as
detection or demodulation of DSBSC. The following demodulators (detectors) are
used for demodulating DSBSC wave.

Coherent Detector
Costas Loop

Coherent Detector
Here, the same carrier signal (which is used for generating DSBSC signal) is used to
detect the message signal. Hence, this process of detection is called as coherent or
synchronous detection. Following is the block diagram of the coherent detector.

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In this process, the message signal can be extracted from DSBSC wave by
multiplying it with a carrier, having the same frequency and the phase of the carrier
used in DSBSC modulation. The resulting signal is then passed through a Low Pass
Filter. Output of this filter is the desired message signal.

Let the DSBSC wave be

s (t) = Ac cos(2πfc t)m (t)

The output of the local oscillator is

c (t) = Ac cos(2πfc t + ϕ)

Where, ϕ is the phase difference between the local oscillator signal and the carrier
signal, which is used for DSBSC modulation.

From the figure, we can write the output of product modulator as

v (t) = s (t) c (t)

Substitute, s (t) and c (t) values in the above equation.

⇒ v (t) = Ac cos(2πfc t)m (t) Ac cos(2πfc t + ϕ)

2
= Ac cos(2πfc t) cos(2πfc t + ϕ)m (t)

2
Ac
= [cos(4πfc t + ϕ) + cos ϕ] m (t)
2

2 2
Ac Ac
v (t) = cos ϕm (t) + cos(4πfc t + ϕ)m (t)
2 2

In the above equation, the first term is the scaled version of the message signal. It
can be extracted by passing the above signal through a low pass filter.

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Therefore, the output of low pass filter is

2
Ac
v0 t = cos ϕm (t)
2

The demodulated signal amplitude will be maximum, when ϕ = 0 . That’s why the
0

local oscillator signal and the carrier signal should be in phase, i.e., there should not
be any phase difference between these two signals.

The demodulated signal amplitude will be zero, when . This effect is called
0
ϕ = ±90

as quadrature null effect.

Costas Loop
Costas loop is used to make both the carrier signal (used for DSBSC modulation) and
the locally generated signal in phase. Following is the block diagram of Costas loop.

Costas loop consists of two product modulators with common input s (t), which is
DSBSC wave. The other input for both product modulators is taken from Voltage
Controlled Oscillator (VCO) with phase shift to one of the product modulator
0
−90

as shown in figure.

We know that the equation of DSBSC wave is

s (t) = Ac cos(2πfc t)m (t)

Let the output of VCO be

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c 1 (t) = cos(2πfc t + ϕ)

This output of VCO is applied as the carrier input of the upper product modulator.

Hence, the output of the upper product modulator is

v1 (t) = s (t) c 1 (t)

Substitute, s (t) and c 1 (t) values in the above equation.

⇒ v1 (t) = Ac cos(2πfc t)m (t) cos(2πfc t + ϕ)

After simplifying, we will get v1 (t) as

Ac Ac
v1 (t) = cos ϕm (t) + cos(4πfc t + ϕ)m (t)
2 2

This signal is applied as an input of the upper low pass filter. The output of this low
pass filter is

Ac
v01 (t) = cos ϕm (t)
2

Therefore, the output of this low pass filter is the scaled version of the modulating
signal.

The output of phase shifter is

0
−90

0
c 2 (t) = cos (2πfc t + ϕ − 90 ) = sin(2πfc t + ϕ)

This signal is applied as the carrier input of the lower product modulator.

The output of the lower product modulator is

v2 (t) = s (t) c 2 (t)

Substitute, s (t) and c 2 (t) values in the above equation.

⇒ v2 (t) = Ac cos(2πfc t)m (t) sin(2πfc t + ϕ)

After simplifying, we will get v2 (t) as

Ac Ac
v2 (t) = sin ϕm (t) + sin(4πfc t + ϕ)m (t)
2 2

This signal is applied as an input of the lower low pass filter. The output of this low
pass filter is

Ac
v02 (t) = sin ϕm (t)
2

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The output of this Low pass filter has phase difference with the output of the
0
−90

upper low pass filter.

The outputs of these two low pass filters are applied as inputs of the phase
discriminator. Based on the phase difference between these two signals, the phase
discriminator produces a DC control signal.

This signal is applied as an input of VCO to correct the phase error in VCO output.
Therefore, the carrier signal (used for DSBSC modulation) and the locally generated
signal (VCO output) are in phase.

Analog Communication - SSBSC Modulation

In the previous chapters, we have discussed DSBSC modulation and demodulation.
The DSBSC modulated signal has two sidebands. Since, the two sidebands carry the
same information, there is no need to transmit both sidebands. We can eliminate one
sideband.

The process of suppressing one of the sidebands along with the carrier and
transmitting a single sideband is called as Single Sideband Suppressed Carrier
system or simply SSBSC. It is plotted as shown in the following figure.

In the above figure, the carrier and the lower sideband are suppressed. Hence, the
upper sideband is used for transmission. Similarly, we can suppress the carrier and
the upper sideband while transmitting the lower sideband.

This SSBSC system, which transmits a single sideband has high power, as the power
allotted for both the carrier and the other sideband is utilized in transmitting this
Single Sideband.

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Mathematical Expressions
Let us consider the same mathematical expressions for the modulating and the
carrier signals as we have considered in the earlier chapters.

i.e., Modulating signal

m (t) = Am cos(2πfm t)

Carrier signal

c (t) = Ac cos(2πfc t)

Mathematically, we can represent the equation of SSBSC wave as

Am Ac
s (t) = cos[2π (fc + fm ) t] for the upper sideband
2

Or

Am Ac
s (t) = cos[2π (fc − fm ) t] for the lower sideband
2

Bandwidth of SSBSC Wave

We know that the DSBSC modulated wave contains two sidebands and its bandwidth
is 2fm . Since the SSBSC modulated wave contains only one sideband, its bandwidth
is half of the bandwidth of DSBSC modulated wave.
2fm
i.e., Bandwidth of SSBSC modulated wave = 2
= fm

Therefore, the bandwidth of SSBSC modulated wave is fm and it is equal to the

frequency of the modulating signal.

Power Calculations of SSBSC Wave

Consider the following equation of SSBSC modulated wave.

for the upper sideband

Am Ac
s (t) = cos[2π (fc + fm ) t]
2

Or

Am Ac
s (t) = cos[2π (fc − fm ) t] for the lower sideband
2

Power of SSBSC wave is equal to the power of any one sideband frequency
components.

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Pt = PU SB = PLSB

We know that the standard formula for power of cos signal is

– 2
2
vrms (vm /√2)
P = =
R R

In this case, the power of the upper sideband is

– 2 2 2
(Am Ac /2√2) Am Ac
PU SB = =
R 8R

Similarly, we will get the lower sideband power same as that of the upper side band
power.
2 2
Am Ac
PLSB =
8R

Therefore, the power of SSBSC wave is

2 2
Am Ac
Pt = PU SB = PLSB =
8R

Advantages

Bandwidth or spectrum space occupied is lesser than AM and DSBSC waves.

Transmission of more number of signals is allowed.
Power is saved.
High power signal can be transmitted.
Less amount of noise is present.
Signal fading is less likely to occur.

Disadvantages

The generation and detection of SSBSC wave is a complex process.

The quality of the signal gets affected unless the SSB transmitter and receiver
have an excellent frequency stability.

Applications

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For power saving requirements and low bandwidth requirements.

In land, air, and maritime mobile communications.
In point-to-point communications.
In radio communications.

In television, telemetry, and radar communications.

In military communications, such as amateur radio, etc.

Analog Communication - SSBSC Modulators

In this chapter, let us discuss about the modulators, which generate SSBSC wave.
We can generate SSBSC wave using the following two methods.

Frequency discrimination method

Phase discrimination method

Frequency Discrimination Method

The following figure shows the block diagram of SSBSC modulator using frequency
discrimination method.

In this method, first we will generate DSBSC wave with the help of the product
modulator. Then, apply this DSBSC wave as an input of band pass filter. This band
pass filter produces an output, which is SSBSC wave.

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Select the frequency range of band pass filter as the spectrum of the desired SSBSC
wave. This means the band pass filter can be tuned to either upper sideband or
lower sideband frequencies to get the respective SSBSC wave having upper sideband
or lower sideband.

Phase Discrimination Method

The following figure shows the block diagram of SSBSC modulator using phase
discrimination method.

This block diagram consists of two product modulators, two −90

0
phase shifters,
one local oscillator and one summer block. The product modulator produces an
output, which is the product of two inputs. The phase shifter produces an
0
−90

output, which has a phase lag of with respect to the input.

0
−90

The local oscillator is used to generate the carrier signal. Summer block produces an
output, which is either the sum of two inputs or the difference of two inputs based
on the polarity of inputs.

The modulating signal Am cos(2πfm t) and the carrier signal Ac cos(2πfc t) are
directly applied as inputs to the upper product modulator. So, the upper product
modulator produces an output, which is the product of these two inputs.

The output of upper product modulator is

s1 (t) = Am Ac cos(2πfm t) cos(2πfc t)

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Am Ac
⇒ s1 (t) = {cos[2π (fc + fm ) t] + cos[2π (fc − fm ) t]}
2

The modulating signal Am cos(2πfm t) and the carrier signal Ac cos(2πfc t) are
phase shifted by −90 before applying as inputs to the lower product modulator. So,
0

the lower product modulator produces an output, which is the product of these two
inputs.

The output of lower product modulator is

0 0
s2 (t) = Am Ac cos(2πfm t − 90 ) cos(2πfc t − 90 )

⇒ s2 (t) = Am Ac sin(2πfm t) sin(2πfc t)

Am Ac
⇒ s2 (t) = {cos[2π (fc − fm ) t] − cos[2π (fc + fm ) t]}
2

Add s1 (t) and s2 (t) in order to get the SSBSC modulated wave s (t) having a lower
sideband.

Am Ac
s (t) = {cos[2π (fc + fm ) t] + cos[2π (fc − fm ) t]} +
2

Am Ac
{cos[2π (fc − fm ) t] − cos
2

[2π (fc + fm ) t]}

⇒ s (t) = Am Ac cos[2π (fc − fm ) t]

Subtract s2 (t) from s1 (t) in order to get the SSBSC modulated wave s (t) having a
upper sideband.

Am Ac
s (t) = {cos[2π (fc + fm ) t] + cos[2π (fc − fm ) t]} −
2

Am Ac
{cos[2π (fc − fm ) t] − cos[2π (fc + fm ) t]}
2

⇒ s (t) = Am Ac cos[2π (fc + fm ) t]

Hence, by properly choosing the polarities of inputs at summer block, we will get
SSBSC wave having a upper sideband or a lower sideband.

SSBSC Demodulator
The process of extracting an original message signal from SSBSC wave is known as
detection or demodulation of SSBSC. Coherent detector is used for demodulating
SSBSC wave.

Coherent Detector

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Here, the same carrier signal (which is used for generating SSBSC wave) is used to
detect the message signal. Hence, this process of detection is called as coherent or
synchronous detection. Following is the block diagram of coherent detector.

In this process, the message signal can be extracted from SSBSC wave by
multiplying it with a carrier, having the same frequency and the phase of the carrier
used in SSBSC modulation. The resulting signal is then passed through a Low Pass
Filter. The output of this filter is the desired message signal.

Consider the following SSBSC wave having a lower sideband.

Am Ac
s (t) = cos[2π (fc − fm ) t]
2

The output of the local oscillator is

c (t) = Ac cos(2πfc t)

From the figure, we can write the output of product modulator as

v (t) = s (t) c (t)

Substitute s (t) and c (t) values in the above equation.

Am Ac
v (t) = cos[2π (fc − fm ) t]Ac cos(2πfc t)
2

2
Am Ac
= cos[2π (fc − fm ) t] cos(2πfc t)
2

2
Am Ac
= {cos[2π (2fc − f m)] + cos(2πfm )t}
4

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2 2
Am Ac Am Ac
v (t) = cos(2πfm t) + cos[2π (2fc − fm ) t]
4 4

In the above equation, the first term is the scaled version of the message signal. It
can be extracted by passing the above signal through a low pass filter.

Therefore, the output of low pass filter is

2
Am Ac
v0 (t) = cos(2πfm t)
4

Here, the scaling factor is .

Ac

We can use the same block diagram for demodulating SSBSC wave having an upper
sideband. Consider the following SSBSC wave having an upper sideband.

Am Ac
s (t) = cos[2π (fc + fm ) t]
2

The output of the local oscillator is

c (t) = Ac cos(2πfc t)

We can write the output of the product modulator as

v (t) = s (t) c (t)

Substitute s (t) and c (t) values in the above equation.

Am Ac
⇒ v (t) = cos[2π (fc + fm ) t]Ac cos(2πfc t)
2

2
Am Ac
= cos[2π (fc + fm ) t] cos(2πfc t)
2

2
Am Ac
= {cos[2π (2fc + fm ) t] + cos(2πfm t)}
4

2 2
Am Ac Am Ac
v (t) = cos(2πfm t) + cos[2π (2fc + fm ) t]
4 4

In the above equation, the first term is the scaled version of the message signal. It
can be extracted by passing the above signal through a low pass filter.

Therefore, the output of the low pass filter is

2
Am Ac
v0 (t) = cos(2πfm t)
4

Here too the scaling factor is .

Ac

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Therefore, we get the same demodulated output in both the cases by using coherent
detector.

Analog Communication - VSBSC Modulation

In the previous chapters, we have discussed SSBSC modulation and demodulation.
SSBSC modulated signal has only one sideband frequency. Theoretically, we can get
one sideband frequency component completely by using an ideal band pass filter.
However, practically we may not get the entire sideband frequency component. Due
to this, some information gets lost.

To avoid this loss, a technique is chosen, which is a compromise between DSBSC and
SSBSC. This technique is known as Vestigial Side Band Suppressed Carrier
(VSBSC) technique. The word “vestige” means “a part” from which, the name is
derived.

VSBSC Modulation is the process, where a part of the signal called as vestige is
modulated along with one sideband. The frequency spectrum of VSBSC wave is
shown in the following figure.

Along with the upper sideband, a part of the lower sideband is also being transmitted
in this technique. Similarly, we can transmit the lower sideband along with a part of
the upper sideband. A guard band of very small width is laid on either side of VSB in
order to avoid the interferences. VSB modulation is mostly used in television
transmissions.

Bandwidth of VSBSC Modulation

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We know that the bandwidth of SSBSC modulated wave is fm . Since the VSBSC
modulated wave contains the frequency components of one side band along with the
vestige of other sideband, the bandwidth of it will be the sum of the bandwidth of
SSBSC modulated wave and vestige frequency fv .

i.e., Bandwidth of VSBSC Modulated Wave = fm + fv

Advantages

Following are the advantages of VSBSC modulation.

Highly efficient.
Reduction in bandwidth when compared to AM and DSBSC waves.

Filter design is easy, since high accuracy is not needed.

The transmission of low frequency components is possible, without any
difficulty.
Possesses good phase characteristics.

Disadvantages

Following are the disadvantages of VSBSC modulation.

Bandwidth is more when compared to SSBSC wave.

Demodulation is complex.

Applications

The most prominent and standard application of VSBSC is for the transmission of
television signals. Also, this is the most convenient and efficient technique when
bandwidth usage is considered.

Now, let us discuss about the modulator which generates VSBSC wave and the
demodulator which demodulates VSBSC wave one by one.

Generation of VSBSC
Generation of VSBSC wave is similar to the generation of SSBSC wave. The VSBSC
modulator is shown in the following figure.

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In this method, first we will generate DSBSC wave with the help of the product
modulator. Then, apply this DSBSC wave as an input of sideband shaping filter. This
filter produces an output, which is VSBSC wave.

The modulating signal m (t) and carrier signal Ac cos(2πfc t) are applied as inputs
to the product modulator. Hence, the product modulator produces an output, which
is the product of these two inputs.

Therefore, the output of the product modulator is

p (t) = Ac cos(2πfc t)m (t)

Apply Fourier transform on both sides

Ac
P (f ) = [M (f − fc ) + M (f + fc )]
2

The above equation represents the equation of DSBSC frequency spectrum.

Let the transfer function of the sideband shaping filter be H (f ). This filter has the
input p (t)and the output is VSBSC modulated wave s (t). The Fourier transforms of
p (t) and s (t) are P (t) and S (t) respectively.

Mathematically, we can write S (f ) as

S (t) = P (f ) H (f )

Substitute P (f ) value in the above equation.

Ac
S (f ) = [M (f − fc ) + M (f + fc )] H (f )
2

The above equation represents the equation of VSBSC frequency spectrum.

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Demodulation of VSBSC
Demodulation of VSBSC wave is similar to the demodulation of SSBSC wave. Here,
the same carrier signal (which is used for generating VSBSC wave) is used to detect
the message signal. Hence, this process of detection is called as coherent or
synchronous detection. The VSBSC demodulator is shown in the following figure.

In this process, the message signal can be extracted from VSBSC wave by
multiplying it with a carrier, which is having the same frequency and the phase of the
carrier used in VSBSC modulation. The resulting signal is then passed through a Low
Pass Filter. The output of this filter is the desired message signal.

Let the VSBSC wave be s (t) and the carrier signal is Ac cos(2πfc t).

From the figure, we can write the output of the product modulator as

v (t) = Ac cos(2πfc t)s (t)

Apply Fourier transform on both sides

Ac
V (f ) = [S (f − fc ) + S (f + fc )]
2

We know that
Ac
S (f ) = [M (f − fc ) + M (f + fc )] H (f )
2

From the above equation, let us find S (f − fc ) and S (f + fc ).

Ac
S (f − fc ) = [M (f − fc − fc ) + M (f − fc + fc )] H (f − fc )
2

Ac
⇒ S (f − fc ) = [M (f − 2fc ) + M (f )] H (f − fc )
2

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Ac
S (f + fc ) = [M (f + fc − fc ) + M (f + fc + fc )] H (f + fc )
2

Ac
⇒ S (f + fc ) = [M (f ) + M (f + 2fc )] H (f + fc )
2

Substitute, S (f − fc ) and S (f + fc ) values in V (f ).

Ac Ac
V (f ) = [ [M (f − 2fc ) + M (f )]H (f − fc )+
2 2

Ac
[M (f ) + M (f + 2fc )]H (f + fc )]
2

2
Ac
⇒ V (f ) = M (f ) [H (f − fc ) + H (f + fc )]
4

2
Ac
+ [M (f − 2fc ) H (f − fc )
4

+ M (f + 2fc ) H (f + fc )]

In the above equation, the first term represents the scaled version of the desired
message signal frequency spectrum. It can be extracted by passing the above signal
through a low pass filter.

2
Ac
V0 (f ) = M (f ) [H (f − fc ) + H (f + fc )]
4

Analog Communication - Angle Modulation

The other type of modulation in continuous-wave modulation is Angle Modulation.
Angle Modulation is the process in which the frequency or the phase of the carrier
signal varies according to the message signal.

The standard equation of the angle modulated wave is

s (t) = Ac cos θi (t)

Where,

Ac is the amplitude of the modulated wave, which is the same as the amplitude of
the carrier signal

θi (t) is the angle of the modulated wave

Angle modulation is further divided into frequency modulation and phase

modulation.

Frequency Modulation is the process of varying the frequency of the carrier

signal linearly with the message signal.

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Phase Modulation is the process of varying the phase of the carrier signal
linearly with the message signal.

Now, let us discuss these in detail.

Frequency Modulation
In amplitude modulation, the amplitude of the carrier signal varies. Whereas, in
Frequency Modulation (FM), the frequency of the carrier signal varies in
accordance with the instantaneous amplitude of the modulating signal.

Hence, in frequency modulation, the amplitude and the phase of the carrier signal
remains constant. This can be better understood by observing the following figures.

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The frequency of the modulated wave increases, when the amplitude of the
modulating or message signal increases. Similarly, the frequency of the modulated
wave decreases, when the amplitude of the modulating signal decreases. Note that,
the frequency of the modulated wave remains constant and it is equal to the
frequency of the carrier signal, when the amplitude of the modulating signal is zero.

Mathematical Representation

The equation for instantaneous frequency fi in FM modulation is

fi = fc + k f m (t)

Where,

fc is the carrier frequency

kt is the frequency sensitivity

m (t) is the message signal

We know the relationship between angular frequency ωi and angle θi (t) as

dθi (t)
ωi =
dt

dθ i (t)
⇒ 2πfi =
dt

⇒ θi (t) = 2π ∫ fi dt

Substitute, fi value in the above equation.

θi (t) = 2π ∫ (fc + k f m (t)) dt

⇒ θi (t) = 2πfc t + 2πk f ∫ m (t) dt

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Substitute, θi (t) value in the standard equation of angle modulated wave.

s (t) = Ac cos(2πfc t + 2πk f ∫ m (t) dt)

This is the equation of FM wave.

If the modulating signal is m (t) = Am cos(2πfm t), then the equation of FM wave
will be

s (t) = Ac cos(2πfc t + β sin(2πfm t))

Where,

Δf kf A m
β = modulation index =
fm
=
fm

The difference between FM modulated frequency (instantaneous frequency) and

normal carrier frequency is termed as Frequency Deviation. It is denoted by Δf ,

which is equal to the product of kf and Am .

FM can be divided into Narrowband FM and Wideband FM based on the values of

modulation index β.

Narrowband FM

Following are the features of Narrowband FM.

This frequency modulation has a small bandwidth when compared to

wideband FM.
The modulation index β is small, i.e., less than 1.
Its spectrum consists of the carrier, the upper sideband and the lower
sideband.
This is used in mobile communications such as police wireless, ambulances,
taxicabs, etc.

Wideband FM

Following are the features of Wideband FM.

This frequency modulation has infinite bandwidth.

The modulation index β is large, i.e., higher than 1.

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Its spectrum consists of a carrier and infinite number of sidebands, which are
located around it.
This is used in entertainment, broadcasting applications such as FM radio, TV,
etc.

Phase Modulation
In frequency modulation, the frequency of the carrier varies. Whereas, in Phase
Modulation (PM), the phase of the carrier signal varies in accordance with the
instantaneous amplitude of the modulating signal.

So, in phase modulation, the amplitude and the frequency of the carrier signal
remains constant. This can be better understood by observing the following figures.

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The phase of the modulated wave has got infinite points, where the phase shift in a
wave can take place. The instantaneous amplitude of the modulating signal changes
the phase of the carrier signal. When the amplitude is positive, the phase changes in
one direction and if the amplitude is negative, the phase changes in the opposite
direction.

Mathematical Representation

The equation for instantaneous phase ϕi in phase modulation is

ϕi = k p m (t)

Where,

kp is the phase sensitivity

m (t) is the message signal

The standard equation of angle modulated wave is

s (t) = Ac cos(2πfc t + ϕi )

Substitute, ϕi value in the above equation.

s (t) = Ac cos(2πfc t + k p m (t))

This is the equation of PM wave.

If the modulating signal, m (t) = Am cos(2πfm t), then the equation of PM wave will
be

s (t) = Ac cos(2πfc t + β cos(2πfm t))

Where,

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β = modulation index = Δϕ = k p Am

Δϕ is phase deviation

Phase modulation is used in mobile communication systems, while frequency

modulation is used mainly for FM broadcasting.

Numerical Problems 2
In the previous chapter, we have discussed the parameters used in Angle
modulation. Each parameter has its own formula. By using those formulas, we can
find the respective parameter values. In this chapter, let us solve a few problems
based on the concept of Frequency Modulation.

Problem 1
A sinusoidal modulating waveform of amplitude 5 V and a frequency of 2 KHz is
applied to FM generator, which has a frequency sensitivity of 40 Hz/volt. Calculate
the frequency deviation, modulation index, and bandwidth.

Solution

Given, the amplitude of modulating signal, Am = 5V

Frequency of modulating signal, fm = 2K H z

Frequency sensitivity, k f = 40H z/volt

We know the formula for Frequency deviation as

Δf = k f Am

Substitute kf and Am values in the above formula.

Δf = 40 × 5 = 200H z

Therefore, frequency deviation, Δf is 200H z

The formula for modulation index is

Δf
β =
fm

Substitute Δf and fm values in the above formula.

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200
β = = 0.1
2 × 1000

Here, the value of modulation index, β is 0.1, which is less than one. Hence, it is
Narrow Band FM.

The formula for Bandwidth of Narrow Band FM is the same as that of AM wave.

BW = 2fm

Substitute fm value in the above formula.

BW = 2 × 2K = 4K H z

Therefore, the bandwidth of Narrow Band FM wave is 4K H z.

Problem 2
An FM wave is given by s (t) = 20 cos(8π × 10 t + 9 sin(2π × 10 t)) .
6 3
Calculate the
frequency deviation, bandwidth, and power of FM wave.

Solution

Given, the equation of an FM wave as

6 3
s (t) = 20 cos(8π × 10 t + 9 sin(2π × 10 t))

We know the standard equation of an FM wave as

s (t) = Ac cos(2πfc t + β sin(2πfm t))

We will get the following values by comparing the above two equations.

Amplitude of the carrier signal, Ac = 20V

Frequency of the carrier signal,

6
fc = 4 × 10 H z = 4M H z

Frequency of the message signal,

3
fm = 1 × 10 H z = 1K H z

Modulation index, β = 9

Here, the value of modulation index is greater than one. Hence, it is Wide Band FM.

We know the formula for modulation index as

Δf
β =
fm

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Rearrange the above equation as follows.

Δ = βfm

Substitute β and fm values in the above equation.

Δ = 9 × 1K = 9K H z

Therefore, frequency deviation, Δf is 9K H z.

The formula for Bandwidth of Wide Band FM wave is

BW = 2 (β + 1) fm

Substitute β and fm values in the above formula.

BW = 2 (9 + 1) 1K = 20K H z

Therefore, the bandwidth of Wide Band FM wave is 20K H z

Formula for power of FM wave is

2
Ac
Pc =
2R

Assume, R = 1Ω and substitute Ac value in the above equation.

2
(20)
P = = 200W
2 (1)

Therefore, the power of FM wave is 200 watts.

Analog Communication - FM Modulators

In this chapter, let us discuss about the modulators which generate NBFM and WBFM
waves. First, let us discuss about the generation of NBFM.

Generation of NBFM
We know that the standard equation of FM wave is

s (t) = Ac cos(2πfc t + 2πk f ∫ m (t) dt)

⇒ s (t) = Ac cos(2πfc t) cos(2πk f ∫ m (t) dt)−

Ac sin(2πfc t) sin(2πk f ∫ m (t) dt)

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For NBFM,

∣ ∣
∣ 2πk f ∫ m (t) dt∣ << 1
∣ ∣

We know that cos θ ≈ 1 and sin θ ≈ 1 when θ is very small.

By using the above relations, we will get the NBFM equation as

s (t) = Ac cos(2πfc t) − Ac sin(2πfc t)2πk f ∫ m (t) dt

The block diagram of NBFM modulator is shown in the following figure.

Here, the integrator is used to integrate the modulating signal m (t) . The carrier
signal is the phase shifted by to get with the
0
Ac cos(2πfc t) −90 Ac sin(2πfc t)

help of phase shifter. The product modulator has two inputs and
0
−90 ∫ m (t) dt

Ac sin(2πfc t). It produces an output, which is the product of these two inputs.

This is further multiplied with 2πk f by placing a block 2πk f in the forward path. The
summer block has two inputs, which are nothing but the two terms of NBFM
equation. Positive and negative signs are assigned for the carrier signal and the
other term at the input of the summer block. Finally, the summer block produces
NBFM wave.

Generation of WBFM
The following two methods generate WBFM wave.

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Direct method
Indirect method

Direct Method

This method is called as the Direct Method because we are generating a wide band
FM wave directly. In this method, Voltage Controlled Oscillator (VCO) is used to
generate WBFM. VCO produces an output signal, whose frequency is proportional to
the input signal voltage. This is similar to the definition of FM wave. The block
diagram of the generation of WBFM wave is shown in the following figure.

Here, the modulating signal m (t) is applied as an input of Voltage Controlled

Oscillator (VCO). VCO produces an output, which is nothing but the WBFM.

fi α m (t)

⇒ fi = fc + k f m (t)

Where,

fi is the instantaneous frequency of WBFM wave.

Indirect Method

This method is called as Indirect Method because we are generating a wide band FM
wave indirectly. This means, first we will generate NBFM wave and then with the help
of frequency multipliers we will get WBFM wave. The block diagram of generation of
WBFM wave is shown in the following figure.

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This block diagram contains mainly two stages. In the first stage, the NBFM wave will
be generated using NBFM modulator. We have seen the block diagram of NBFM
modulator at the beginning of this chapter. We know that the modulation index of
NBFM wave is less than one. Hence, in order to get the required modulation index
(greater than one) of FM wave, choose the frequency multiplier value properly.

Frequency multiplier is a non-linear device, which produces an output signal

whose frequency is ‘n’ times the input signal frequency. Where, ‘n’ is the
multiplication factor.

If NBFM wave whose modulation index β is less than 1 is applied as the input of
frequency multiplier, then the frequency multiplier produces an output signal, whose
modulation index is ‘n’ times β and the frequency also ‘n’ times the frequency of
WBFM wave.

Sometimes, we may require multiple stages of frequency multiplier and mixers in

order to increase the frequency deviation and modulation index of FM wave.

Analog Communication - FM Demodulators

In this chapter, let us discuss about the demodulators which demodulate the FM
wave. The following two methods demodulate FM wave.

Frequency discrimination method

Phase discrimination method

Frequency Discrimination Method

We know that the equation of FM wave is

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s (t) = Ac cos(2πfc t + 2πk f ∫ m (t) dt)

Differentiate the above equation with respect to 't'.

ds (t)
= −Ac (2πfc + 2πk f m (t)) sin(2πfc t + 2πk f ∫ m (t) dt)
dt

We can write, as sin(θ − 180 ) .

0
− sin θ

ds(t)
⇒ = Ac (2πfc + 2πk f m (t)) sin
dt

0
(2πfc t + 2πk f ∫ m (t) dt − 180 )

ds(t) kf
⇒ = Ac (2πfc ) [1 + ( ) m (t)] sin
dt kc

0
(2πfc t + 2πk f ∫ m (t) dt − 180 )

In the above equation, the amplitude term resembles the envelope of AM wave and
the angle term resembles the angle of FM wave. Here, our requirement is the
modulating signal m (t) . Hence, we can recover it from the envelope of AM wave.

The following figure shows the block diagram of FM demodulator using frequency
discrimination method.

This block diagram consists of the differentiator and the envelope detector.
Differentiator is used to convert the FM wave into a combination of AM wave and FM
wave. This means, it converts the frequency variations of FM wave into the
corresponding voltage (amplitude) variations of AM wave. We know the operation of
the envelope detector. It produces the demodulated output of AM wave, which is
nothing but the modulating signal.

Phase Discrimination Method

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The following figure shows the block diagram of FM demodulator using phase
discrimination method.

This block diagram consists of the multiplier, the low pass filter, and the Voltage
Controlled Oscillator (VCO). VCO produces an output signal v (t) , whose frequency is
proportional to the input signal voltage d (t) . Initially, when the signal d (t) is zero,
adjust the VCO to produce an output signal v (t) , having a carrier frequency and
phase shift with respect to the carrier signal.
0
−90

FM wave s (t) and the VCO output v (t) are applied as inputs of the multiplier. The
multiplier produces an output, having a high frequency component and a low
frequency component. Low pass filter eliminates the high frequency component and
produces only the low frequency component as its output.

This low frequency component contains only the term-related phase difference.
Hence, we get the modulating signal m (t) from this output of the low pass filter.

Analog Communication - Multiplexing

Multiplexing is the process of combining multiple signals into one signal, over a
shared medium. If the analog signals are multiplexed, then it is called as analog
multiplexing. Similarly, if the digital signals are multiplexed, then it is called as
digital multiplexing.

Multiplexing was first developed in telephony. A number of signals were combined to

send through a single cable. The process of multiplexing divides a communication
channel into several number of logical channels, allotting each one for a different

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message signal or a data stream to be transferred. The device that does multiplexing
can be called as Multiplexer or MUX.

The reverse process, i.e., extracting the number of channels from one, which is done
at the receiver is called as de-multiplexing. The device that does de-multiplexing
can be called as de-multiplexer or DEMUX.

The following figures illustrates the concept of MUX and DEMUX. Their primary use is
in the field of communications.

Types of Multiplexers
There are mainly two types of multiplexers, namely analog and digital. They are
further divided into Frequency Division Multiplexing (FDM), Wavelength Division
Multiplexing (WDM), and Time Division Multiplexing (TDM). The following figure gives
a detailed idea about this classification.

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There are many types of multiplexing techniques. Out of which, we have the main
types with general classification, mentioned in the above figure. Let us take a look at
them individually.

Analog Multiplexing
The signals used in analog multiplexing techniques are analog in nature. The analog
signals are multiplexed according to their frequency (FDM) or wavelength (WDM).

Frequency Division Multiplexing

In analog multiplexing, the most used technique is Frequency Division Multiplexing

(FDM). This technique uses various frequencies to combine streams of data, for
sending them on a communication medium, as a single signal.

Example − A traditional television transmitter, which sends a number of channels

through a single cable uses FDM.

Wavelength Division Multiplexing

Wavelength Division multiplexing (WDM) is an analog technique, in which many data

streams of different wavelengths are transmitted in the light spectrum. If the
wavelength increases, the frequency of the signal decreases. A prism, which can turn
different wavelengths into a single line, can be used at the output of MUX and input
of DEMUX.

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Example − Optical fiber communications use WDM technique, to merge different

wavelengths into a single light for communication.

Digital Multiplexing
The term digital represents the discrete bits of information. Hence, the available data
is in the form of frames or packets, which are discrete.

Time Division Multiplexing

In Time Division Multiplexing (TDM), the time frame is divided into slots. This
technique is used to transmit a signal over a single communication channel, by
allotting one slot for each message.

Time Division Multiplexing (TDM) can be classified into Synchronous TDM and
Asynchronous TDM.

Synchronous TDM

In Synchronous TDM, the input is connected to a frame. If there are ‘n’ number of
connections, then the frame is divided into ‘n’ time slots. One slot is allocated for
each input line.

In this technique, the sampling rate is common for all signals and hence the same
clock input is given. The MUX allocates the same slot to each device at all times.

Asynchronous TDM

In Asynchronous TDM, the sampling rate is different for each of the signals and a
common clock is not required. If the allotted device for a time slot transmits nothing
and sits idle, then that slot can be allotted to another device, unlike synchronous

This type of TDM is used in Asynchronous transfer mode networks.

De-Multiplexer
De-multiplexers are used to connect a single source to multiple destinations. This
process is the reverse process of multiplexing. As mentioned previously, it is used
mostly at the receivers. DEMUX has many applications. It is used in receivers in the
communication systems. It is used in arithmetic and logical unit in computers to
supply power and to pass on communication, etc.

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De-multiplexers are used as serial to parallel converters. The serial data is given as
input to DEMUX at regular interval and a counter is attached to it to control the
output of the de-multiplexer.

Both the multiplexers and de-multiplexers play an important role in communication

systems, both at the transmitter and the receiver sections.

Analog Communication - Noise

In any communication system, during the transmission of the signal or while
receiving the signal, some unwanted signal gets introduced into the communication,
making it unpleasant for the receiver, and questioning the quality of the
communication. Such a disturbance is called as Noise.

What is Noise?
Noise is an unwanted signal, which interferes with the original message signal and
corrupts the parameters of the message signal. This alteration in the communication
process, leads to the message getting altered. It most likely enters at the channel or
the receiver.

The noise signal can be understood by taking a look at the following figure.

Hence, it is understood that the noise is some signal which has no pattern and no
constant frequency or amplitude. It is quite random and unpredictable. Measures are
usually taken to reduce it, though it can’t be completely eliminated.

Most common examples of noise are −

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Hiss sound in radio receivers

Buzz sound amidst of telephone conversations
Flicker in television receivers, etc

Types of Noise
The classification of noise is done depending on the type of the source, the effect it
shows or the relation it has with the receiver, etc.

There are two main ways in which noise is produced. One is through some external
source while the other is created by an internal source, within the receiver
section.

External Source

This noise is produced by the external sources, which may occur in the medium or
channel of communication usually. This noise cannot be completely eliminated. The
best way is to avoid the noise from affecting the signal.

Examples

Most common examples of this type of noise are

Atmospheric noise (due to irregularities in the atmosphere).

Extra-terrestrial noise, such as solar noise and cosmic noise.
Industrial noise.

Internal Source

This noise is produced by the receiver components while functioning. The

components in the circuits, due to continuous functioning, may produce few types of
noise. This noise is quantifiable. A proper receiver design may lower the effect of this
internal noise.

Examples

Most common examples of this type of noise are

Thermal agitation noise (Johnson noise or Electrical noise)

Shot noise (due to the random movement of electrons and holes)

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Transit-time noise (during transition)

Miscellaneous noise is another type of noise which includes flicker, resistance
effect and mixer generated noise, etc.

Effects of Noise
Noise is an inconvenient feature, which affects the system performance. Following
are the effects of noise.

Noise limits the operating range of the systems

Noise indirectly places a limit on the weakest signal that can be amplified by an
amplifier. The oscillator in the mixer circuit may limit its frequency because of noise.
A system’s operation depends on the operation of its circuits. Noise limits the
smallest signal that a receiver is capable of processing.

Noise affects the sensitivity of receivers

Sensitivity is the minimum amount of input signal necessary to obtain the specified
quality output. Noise affects the sensitivity of a receiver system, which eventually
affects the output.

Analog Communication - SNR Calculations

In this chapter, let us calculate Signal to Noise Ratios and Figure of Merits of various
modulated waves, which are demodulated at the receiver.

Signal to Noise Ratio

Signal-to-Noise Ratio (SNR) is the ratio of the signal power to noise power. The
higher the value of SNR, the greater will be the quality of the received output.

Signal-to-Noise Ratio at different points can be calculated using the following

formulas.

Average power of modulating signal

Input SNR = (S N R)
I
=
Average power of noise at input

Average power of demodulated signal

Output SNR = (S N R)
O
=
Average power of noise at output

Average power of modulated signal

Channel SNR = (S N R)
C
=
Average power of noise in message bandwidth

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Figure of Merit
The ratio of output SNR and input SNR can be termed as Figure of Merit. It is
denoted by F. It describes the performance of a device.

(S N R)
O
F =
(S N R)
I

Figure of merit of a receiver is

(S N R)
O
F =
(S N R)
C

It is so because for a receiver, the channel is the input.

SNR Calculations in AM System

Consider the following receiver model of AM system to analyze noise.

We know that the Amplitude Modulated (AM) wave is

s (t) = Ac [1 + k a m (t)] cos(2πfc t)

⇒ s (t) = Ac cos(2πfc t) + Ac k a m (t) cos(2πfc t)

Average power of AM wave is

2 2 2 2 2
Ac Ac k a m (t) Ac Ac k a P
Ps = ( –) + ( – ) = +
√2 √2 2 2

2 2
Ac (1 + k a P )
⇒ Ps =
2

Average power of noise in the message bandwidth is

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Pnc = W N0

Substitute, these values in channel SNR formula

(S N R)
C ,AM

Average P ower of AM W ave

=
Average P ower of noise in message bandwidth

2 2
Ac (1 + k a ) P
⇒ (S N R) =
C ,AM
2W N0

Where,

P is the power of the message signal=

Am

W is the message bandwidth

Assume the band pass noise is mixed with AM wave in the channel as shown in the
above figure. This combination is applied at the input of AM demodulator. Hence, the
input of AM demodulator is.

v (t) = s (t) + n (t)

⇒ v (t) = Ac [1 + k a m (t)] cos(2πfc t)+

[n1 (t) cos(2πfc t) − nQ (t) sin(2πfc t)]

⇒ v (t) = [Ac + Ac k a m (t) + n1 (t)] cos(2πfc t) − nQ (t) sin(2πfc t)

Where nI (t) and nQ (t) are in phase and quadrature phase components of noise.

The output of AM demodulator is nothing but the envelope of the above signal.
−−−−−−−−−−−−−−−−−−−−−−−−−−−−−−
2 2
d (t) = √[Ac + Ac Ka m (t) + nI (t)] + (nQ (t))

⇒ d (t) ≈ Ac + Ac k a m (t) + n1 (t)

Average power of the demodulated signal is

2 2 2
Ac k a m (t) Ac k a P
Pm = ( – ) =
√2 2

Average power of noise at the output is

Pn o = W N0

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Substitute, these values in output SNR formula.

Average P ower of demodulated signal

(S N R) =
O,AM
Average P ower of noise at Output

2 2
Ac k a P
⇒ (S N R) =
O,AM
2W N0

Substitute, the values in Figure of merit of AM receiver formula.

(S N R)
O,AM
F =
(S N R)
C ,AM

2 2
2 2
Ac k a P Ac (1 + k a ) P
⇒ F = ( )/( )
2W N0 2W N0

2
Ka P
⇒ F =
2
1 + Ka P

Therefore, the Figure of merit of AM receiver is less than one.

SNR Calculations in DSBSC System

Consider the following receiver model of DSBSC system to analyze noise.

We know that the DSBSC modulated wave is

s (t) = Ac m (t) cos(2πfc t)

Average power of DSBSC modulated wave is

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2 2
Ac m (t) Ac P
Ps = ( – ) =
√2 2

Average power of noise in the message bandwidth is

Pnc = W N0

Substitute, these values in channel SNR formula.

(S N R)
C ,DSBSC

Average P ower of DS BS C modulated wave

=
Average P ower of noise in message bandwidth

2
Ac P
⇒ (S N R) =
C ,DSBSC
2W N0

Assume the band pass noise is mixed with DSBSC modulated wave in the channel as
shown in the above figure. This combination is applied as one of the input to the
product modulator. Hence, the input of this product modulator is

v1 (t) = s (t) + n (t)

⇒ v1 (t) = Ac m (t) cos(2πfc t)

+ [nI (t) cos(2πfc t) − nQ (t) sin(2πfc t)]

⇒ v1 (t) = [Ac m (t) + nI (t)] cos(2πfc t) − nQ (t) sin(2πfc t)

Local oscillator generates the carrier signal c (t) = cos(2πfc t). This signal is applied
as another input to the product modulator. Therefore, the product modulator
produces an output, which is the product of v1 (t) and c (t).

v2 (t) = v1 (t) c (t)

Substitute, v1 (t) and c (t) values in the above equation.

⇒ v2 (t) = ([Ac m (t) + nI (t)] cos(2πfc t) − nQ (t) sin(2πfc t)) cos

(2πfc t)

2
⇒ v2 (t) = [Ac m (t) + nI (t)] cos (2πfc t) − nQ (t) sin(2πfc t) cos

(2πfc t)

1 + cos(4πfc t)
⇒ v2 (t) = [Ac m (t) + nI (t)] ( )
2

sin(4πfc t)
− nQ (t)
2

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When the above signal is applied as an input to low pass filter, we will get the output
of low pass filter as

[Ac m (t) + nI (t)]

d (t) =
2

Average power of the demodulated signal is

2 2
Ac m (t) Ac P
Pm = ( – ) =
2√ 2 8

Average power of noise at the output is

W N0
Pno =
4

Substitute, these values in output SNR formula.

Average P ower of demodulated signal

(S N R) =
O,DSBSC
Average P ower of noise at Output

2 2
Ac P W N0 Ac P
⇒ (S N R) = ( )/( ) =
O,DSBSC
8 4 2W N0

Substitute, the values in Figure of merit of DSBSC receiver formula.

(S N R)
O,DSBSC
F =
(S N R)
C ,DSBSC

2 2
Ac P Ac P
⇒ F = ( )/( )
2W N0 2W N0

⇒ F = 1

Therefore, the Figure of merit of DSBSC receiver is 1.

SNR Calculations in SSBSC System

Consider the following receiver model of SSBSC system to analyze noise.

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We know that the SSBSC modulated wave having lower sideband is

Am Ac
s (t) = cos[2π (fc − fm ) t]
2

Average power of SSBSC modulated wave is

2 2 2
Am Ac Am Ac
Ps = ( – ) =
2√ 2 8

Average power of noise in the message bandwidth is

Pnc = W N0

Substitute, these values in channel SNR formula.

(S N R)
C ,SSBSC

Average P ower of S S BS C modulated wave

=
Average P ower of noise in message bandwidth

2 2
Am Ac
⇒ (S N R) =
C ,SSBSC
8W N0

Assume the band pass noise is mixed with SSBSC modulated wave in the channel as
shown in the above figure. This combination is applied as one of the input to the
product modulator. Hence, the input of this product modulator is

v1 (t) = s (t) + n (t)

Am Ac
v1 (t) = cos[2π (fc − fm ) t] + nI (t) cos(2πfc t) − nQ (t) sin
2

(2πfc t)

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The local oscillator generates the carrier signal c (t) = cos(2πfc t). This signal is
applied as another input to the product modulator. Therefore, the product modulator
produces an output, which is the product of v1 (t) and c (t).

v2 (t) = v1 (t) c (t)

Substitute, v1 (t) and c (t) values in the above equation.

Am Ac
⇒ v2 (t) = ( cos[2π(fc − fm )t] + nI (t) cos(2πfc t)−
2

nQ (t) sin(2πfc t)) cos(2πfc t)

Am Ac
⇒ v2 (t) = cos[2π (fc − fm ) t] cos(2πfc t)+
2

2
nI (t) cos (2πfc t) − nQ (t) sin(2πfc t) cos

(2πfc t)

Am Ac
⇒ v2 (t) = {cos[2π (2fc − fm ) t] + cos(2πfm t)} +
4

1+cos(4πfc t)
nI (t) ( )
2

sin(4πf t)
c
− nQ (t)
2

When the above signal is applied as an input to low pass filter, we will get the output
of low pass filter as

Am Ac nI (t)
d (t) = cos(2πfm t) +
2 2

Average power of the demodulated signal is

2 2 2
Am Ac Am Ac
Pm = ( – )
=
4√ 2 32

Average power of noise at the output is

W N0
Pno =
4

Substitute, these values in output SNR formula

Average P ower of demodulated signal

(S N R) =
O,SSBSC
Average P ower of noise at output

2 2 2 2
Am Ac W N0 Am Ac
⇒ (S N R) = ( )/( ) =
O,SSBSC
32 4 8W N0

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Substitute, the values in Figure of merit of SSBSC receiver formula

(S N R)
O,SSBSC
F =
(S N R)
C ,SSBSC

2 2 2 2
Am Ac Am Ac
F = ( )/( )
8W N0 8W N0

F = 1

Therefore, the Figure of merit of SSBSC receiver is 1.

Analog Communication - Transmitters

The antenna present at the end of transmitter section, transmits the modulated
wave. In this chapter, let us discuss about AM and FM transmitters.

AM Transmitter
AM transmitter takes the audio signal as an input and delivers amplitude modulated
wave to the antenna as an output to be transmitted. The block diagram of AM
transmitter is shown in the following figure.

The working of AM transmitter can be explained as follows.

The audio signal from the output of the microphone is sent to the pre-
amplifier, which boosts the level of the modulating signal.

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The RF oscillator generates the carrier signal.

Both the modulating and the carrier signal is sent to AM modulator.
Power amplifier is used to increase the power levels of AM wave. This wave is
finally passed to the antenna to be transmitted.

FM Transmitter
FM transmitter is the whole unit, which takes the audio signal as an input and
delivers FM wave to the antenna as an output to be transmitted. The block diagram
of FM transmitter is shown in the following figure.

The working of FM transmitter can be explained as follows.

The audio signal from the output of the microphone is sent to the pre-
amplifier, which boosts the level of the modulating signal.
This signal is then passed to high pass filter, which acts as a pre-emphasis
network to filter out the noise and improve the signal to noise ratio.
This signal is further passed to the FM modulator circuit.
The oscillator circuit generates a high frequency carrier, which is sent to the
modulator along with the modulating signal.
Several stages of frequency multiplier are used to increase the operating
frequency. Even then, the power of the signal is not enough to transmit.
Hence, a RF power amplifier is used at the end to increase the power of the
modulated signal. This FM modulated output is finally passed to the antenna
to be transmitted.

Analog Communication - Receivers

The antenna present at the beginning of the receiver section, receives the modulated
wave. First let us discuss the requirements of a receiver.

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Requirements of a Receiver
AM receiver receives AM wave and demodulates it by using the envelope detector.
Similarly, FM receiver receives FM wave and demodulates it by using the Frequency
Discrimination method. Following are the requirements of both AM and FM receiver.

It should be cost-effective.
It should receive the corresponding modulated waves.
The receiver should be able to tune and amplify the desired station.
It should have an ability to reject the unwanted stations.
Demodulation has to be done to all the station signals, irrespective of the
carrier signal frequency.

For these requirements to be fulfilled, the tuner circuit and the mixer circuit should
be very effective. The procedure of RF mixing is an interesting phenomenon.

RF Mixing

The RF mixing unit develops an Intermediate Frequency (IF) to which any

received signal is converted, so as to process the signal effectively.

RF Mixer is an important stage in the receiver. Two signals of different frequencies

are taken where one signal level affects the level of the other signal, to produce the
resultant mixed output. The input signals and the resultant mixer output is illustrated
in the following figures.

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Let the first and second signal frequencies be f1 and f2 . If these two signals are
applied as inputs of RF mixer, then it produces an output signal, having frequencies
of f1 + f2 and f1 − f2 .

If this is observed in the frequency domain, the pattern looks like the following
figure.

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In this case, f1 is greater than f2 . So, the resultant output has frequencies f1 + f2

and f1 − f2 . Similarly, if f2 is greater than f1 , then the resultant output will have
the frequencies f1 + f2 and f1 − f2 .

AM Receiver
The AM super heterodyne receiver takes the amplitude modulated wave as an input
and produces the original audio signal as an output. Selectivity is the ability of
selecting a particular signal, while rejecting the others. Sensitivity is the capacity of
detecting RF signal and demodulating it, while at the lowest power level.

Radio amateurs are the initial radio receivers. However, they have drawbacks such as
poor sensitivity and selectivity. To overcome these drawbacks, super heterodyne
receiver was invented. The block diagram of AM receiver is shown in the following
figure.

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RF Tuner Section

The amplitude modulated wave received by the antenna is first passed to the tuner
circuit through a transformer. The tuner circuit is nothing but a LC circuit, which is
also called as resonant or tank circuit. It selects the frequency, desired by the AM
receiver. It also tunes the local oscillator and the RF filter at the same time.

RF Mixer

The signal from the tuner output is sent to the RF-IF converter, which acts as a
mixer. It has a local oscillator, which produces a constant frequency. The mixing
process is done here, having the received signal as one input and the local oscillator
frequency as the other input. The resultant output is a mixture of two frequencies
[(f1 + f2 ) , (f1 − f2 )] produced by the mixer, which is called as the Intermediate
Frequency (IF).

The production of IF helps in the demodulation of any station signal having any
carrier frequency. Hence, all signals are translated to a fixed carrier frequency for
adequate selectivity.

IF Filter

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Intermediate frequency filter is a band pass filter, which passes the desired
frequency. It eliminates all other unwanted frequency components present in it. This
is the advantage of IF filter, which allows only IF frequency.

AM Demodulator

The received AM wave is now demodulated using AM demodulator. This demodulator

uses the envelope detection process to receive the modulating signal.

Audio Amplifier

This is the power amplifier stage, which is used to amplify the detected audio signal.
The processed signal is strengthened to be effective. This signal is passed on to the
loudspeaker to get the original sound signal.

FM Receiver
The block diagram of FM receiver is shown in the following figure.

This block diagram of FM receiver is similar to the block diagram of AM receiver. The
two blocks Amplitude limiter and De-emphasis network are included before and after
FM demodulator. The operation of the remaining blocks is the same as that of AM
receiver.

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We know that in FM modulation, the amplitude of FM wave remains constant.

However, if some noise is added with FM wave in the channel, due to that the
amplitude of FM wave may vary. Thus, with the help of amplitude limiter we can
maintain the amplitude of FM wave as constant by removing the unwanted peaks of
the noise signal.

In FM transmitter, we have seen the pre-emphasis network (High pass filter), which
is present before FM modulator. This is used to improve the SNR of high frequency
audio signal. The reverse process of pre-emphasis is known as de-emphasis. Thus,
in this FM receiver, the de-emphasis network (Low pass filter) is included after FM
demodulator. This signal is passed to the audio amplifier to increase the power level.
Finally, we get the original sound signal from the loudspeaker.

Analog Communication - Sampling

So far, we have discussed about continuous-wave modulation. We will discuss about
pulse modulation in the next chapter. These pulse modulation techniques deal with
discrete signals. So, now let us see how to convert a continuous time signal into a
discrete one.

The process of converting continuous time signals into equivalent discrete time
signals, can be termed as Sampling. A certain instant of data is continually sampled
in the sampling process.

The following figure shows a continuous-time signal x(t) and the corresponding
sampled signal xs(t). When x(t) is multiplied by a periodic impulse train, the
sampled signal xs(t) is obtained.

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A sampling signal is a periodic train of pulses, having unit amplitude, sampled at

equal intervals of time Ts , which is called as sampling time. This data is
transmitted at the time instants Ts and the carrier signal is transmitted at the
remaining time.

Sampling Rate

To discretize the signals, the gap between the samples should be fixed. That gap can
be termed as the sampling period Ts . Reciprocal of the sampling period is known as
sampling frequency or sampling rate fs .

Mathematically, we can write it as

1
fs =
Ts

Where,

fs is the sampling frequency or the sampling rate

Ts is the sampling period

Sampling Theorem

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The sampling rate should be such that the data in the message signal should neither
be lost nor it should get over-lapped. The sampling theorem states that, “a signal
can be exactly reproduced if it is sampled at the rate fs , which is greater than or
equal to twice the maximum frequency of the given signal W.”

Mathematically, we can write it as

fs ≥ 2W

Where,

fs is the sampling rate

W is the highest frequency of the given signal

If the sampling rate is equal to twice the maximum frequency of the given signal W,
then it is called as Nyquist rate.

The sampling theorem, which is also called as Nyquist theorem, delivers the theory
of sufficient sample rate in terms of bandwidth for the class of functions that are
bandlimited.

For continuous-time signal x(t), which is band-limited in the frequency domain is

represented as shown in the following figure.

If the signal is sampled above Nyquist rate, then the original signal can be
recovered. The following figure explains a signal, if sampled at a higher rate than 2w
in the frequency domain.

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If the same signal is sampled at a rate less than 2w, then the sampled signal would
look like the following figure.

We can observe from the above pattern that there is over-lapping of information,
which leads to mixing up and loss of information. This unwanted phenomenon of
over-lapping is called as Aliasing.

Aliasing can be referred to as “the phenomenon of a high-frequency component in

the spectrum of a signal, taking on the identity of a low-frequency component in the
spectrum of its sampled version.”

Hence, the sampling rate of the signal is chosen to be as Nyquist rate. If the
sampling rate is equal to twice the highest frequency of the given signal W, then the
sampled signal would look like the following figure.

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In this case, the signal can be recovered without any loss. Hence, this is a good
sampling rate.

Analog Communication - Pulse Modulation

After continuous wave modulation, the next division is Pulse modulation. In this
chapter, let us discuss the following analog pulse modulation techniques.

Pulse Amplitude Modulation

Pulse Width Modulation
Pulse Position Modulation

Pulse Amplitude Modulation

In Pulse Amplitude Modulation (PAM) technique, the amplitude of the pulse
carrier varies, which is proportional to the instantaneous amplitude of the message
signal.

The pulse amplitude modulated signal will follow the amplitude of the original signal,
as the signal traces out the path of the whole wave. In natural PAM, a signal sampled
at Nyquist rate can be reconstructed, by passing it through an efficient Low Pass
Filter (LPF) with exact cutoff frequency.

The following figures explain the Pulse Amplitude Modulation.

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Though the PAM signal is passed through a LPF, it cannot recover the signal without
distortion. Hence, to avoid this noise, use flat-top sampling. The flat-top PAM signal
is shown in the following figure.

Flat-top sampling is the process in which, the sampled signal can be represented
in pulses for which the amplitude of the signal cannot be changed with respect to the
analog signal, to be sampled. The tops of amplitude remain flat. This process
simplifies the circuit design.

Pulse Width Modulation

In Pulse Width Modulation (PWM) or Pulse Duration Modulation (PDM) or Pulse
Time Modulation (PTM) technique, the width or the duration or the time of the pulse

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carrier varies, which is proportional to the instantaneous amplitude of the message

signal.

The width of the pulse varies in this method, but the amplitude of the signal remains
constant. Amplitude limiters are used to make the amplitude of the signal constant.
These circuits clip off the amplitude to a desired level, and hence the noise is limited.

The following figure explains the types of Pulse Width Modulations.

There are three types of PWM.

The leading edge of the pulse being constant, the trailing edge varies
according to the message signal. The waveform for this type of PWM is
denoted as (a) in the above figure.
The trailing edge of the pulse being constant, the leading edge varies
according to the message signal. The waveform for this type of PWM is
denoted as (b) in the above figure.
The center of the pulse being constant, the leading edge and the trailing edge
varies according to the message signal. The waveform for this type of PWM is
denoted as (c) shown in the above figure.

Pulse Position Modulation

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Pulse Position Modulation (PPM) is an analog modulation scheme in which, the

amplitude and the width of the pulses are kept constant, while the position of each
pulse, with reference to the position of a reference pulse varies according to the
instantaneous sampled value of the message signal.

The transmitter has to send synchronizing pulses (or simply sync pulses) to keep the
transmitter and the receiver in sync. These sync pulses help to maintain the position
of the pulses. The following figures explain the Pulse Position Modulation.

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Pulse position modulation is done in accordance with the pulse width modulated
signal. Each trailing edge of the pulse width modulated signal becomes the starting
point for pulses in PPM signal. Hence, the position of these pulses is proportional to
the width of the PWM pulses.

Advantage

As the amplitude and the width are constant, the power handled is also constant.

Disadvantage

The synchronization between the transmitter and the receiver is a must.

Comparison between PAM, PWM, and PPM

The following table presents the comparison between three modulation techniques.

PAM PWM PPM

Amplitude is varied Width is varied Position is varied

Bandwidth depends on Bandwidth depends on the Bandwidth depends on

the width of the pulse rise time of the pulse the rise time of the pulse

Instantaneous Instantaneous transmitter Instantaneous transmitter

transmitter power varies power varies with the power remains constant

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with the amplitude of amplitude and the width of with the width of the
the pulses the pulses pulses

System complexity is
System complexity is low System complexity is low
high

Noise interference is
Noise interference is low Noise interference is low
high

It is similar to amplitude It is similar to frequency It is similar to phase

modulation modulation modulation

Analog Communication - Transducers

Transducer is a device, which converts energy from one form to other. In this
chapter, let us discuss about the transducers used in communication systems.

Why do We Need Transducers?

In the real world, communication between any two nearby persons takes place with
the help of sound waves. But, if the persons are far away, then it is difficult to
convey the information without any loss by using sound waves in its physical form.

To overcome this difficulty, we can use modulators in the transmitter section and
demodulators in the receiver section. These modulators and demodulators operate
with electrical signals. That’s why we require a device, which has to convert the
sound waves into electrical signals or vice versa. That device is known as a
transducer.

Following is a simple block diagram of a transducer.

This transducer has a single input and a single output. It converts the energy
present at the input into its equivalent output having another energy. Basically, a
transducer converts the non-electrical form of energy into an electrical form or vice
versa.

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Types of Transducers
We can classify the transducers into following two types based on the placement
(position) of the transducer in communication systems.

Input Transducer
Output Transducer

Input Transducers

The transducer present at the input of the communication system is known as an

input transducer. Following is the block diagram of an input transducer.

This input transducer converts the non-electrical physical quantity into an electrical
signal. The physical quantities such as sound or light can be converted into electrical
quantities such as voltage or current by using this transducer. Example:
Microphone.

Microphone is used as the input transducer, which is placed between the information
source and the transmitter section. The information source produces the information
in the form of sound waves. The microphone converts these sound waves into
electrical signals with the help of a diaphragm. These electrical signals can be used
for further processing.

Output Transducers

The transducer present at the output of communication system is known as output

transducer. Following is the block diagram of an output transducer.

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This output transducer converts the electrical signal into non-electrical physical
quantity. The electrical quantities such as voltage or current can be converted into
physical quantities such as sound or light by using this transducer. Example:
Loudspeaker.

The loud speaker is used as the output transducer, which is placed between the
receiver section and the destination. The demodulator present in the receiver section
produces the demodulated output. So, the loud speaker converts the electrical
signals (demodulated output) into sound waves. Therefore, the functionality of the
loud speaker is just opposite to the functionality of the microphone.

In addition to the above transducers, there is one more transducer which is used in
communication systems. This transducer can be placed either at the end of the
transmitter section or at the starting of the receiver section. Example: Antenna.

An Antenna is a transducer, which converts electrical signals into electromagnetic

waves and vice versa. An Antenna can be used either as a transmitting antenna or
as a receiving antenna.

A transmitting antenna converts electrical signals into electromagnetic waves and

radiates them. While, a receiving antenna converts electromagnetic waves from the
received beam into electrical signals.

In this two-way communication, the same antenna can be used for both transmission
and reception.

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